May 2007 - sr-users - kamailio.org

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

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[Serusers] ser-0.9.6: modules/postgres bugreport & patch
by Alexandre Snarskii 13 Oct '08

13 Oct '08

Hi! I'm running ser-0.9.6, on FreeBSD 6.1-stable, database backend is postgresql version 8.1.3. Today I got errors in logfiles, saying: messages.2.bz2:May 29 14:35:03 <XXX> /usr/local/sbin/ser[51448]: ERROR:avpops:dbrow2avp: dbrow contains NULL fields The similar problem reported in: http://lists.iptel.org/pipermail/serusers/2005-May/019681.html with much more detailed description of error, database contents and config samples. Patch is trivial, and looks more like a fix to copy'n'paste error: in mysql/val.c function str2val states: if (!_s) { memset(_v, 0, sizeof(db_val_t)); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } VAL_NULL(_v) = 0; and the last line mentions that 'well, that's value is not NULL'. In postgresql/db_val.c, line 182, function str2valp, the same statement is the: if (!_s) { DLOG("str2valp", "got a null value"); VAL_TYPE(_v) = _t; VAL_NULL(_v) = 1; return 0; } without explicit notification that this is not-NULL value. More than, nowhere else in this function VAL_NULL(_v) not set to 0. So, if a value _v.val contained anyting but 0 at the function start, resulting value will be threated as NULL despite the fact, that _s is NOT NULL. Patch is obvious, just add VAL_NULL(_v)=0; after cited block (line 188) and everyting will be ok. At least for me it's ok for some hours :)

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[Serusers] SER Nokia CANCEL Problem
by Abdul Qadir 07 May '08

07 May '08

Hi , I tried to call from one nokia sip (E61 and other models )phone to another nokia sip phone. The call works fine. The problem comes only when I call from Phone A to Phone B and then immediately cancel the call(from Phone A). The Phone A will hangup the call as it sent CANCEL but the SER will ignore this CANCEL and still send INVITE to Phone B resulting in a ghost call situation. I tried to capture a log of message and found that Phone A "CANCEL" message is received on SER even before any provisional response from Phone B. Therefor SER doesnot relay this CANCEL request to Phone B. I even checked RFC which clearly says that UAC should not send CANCEL untill it receives any provisional response. I talked to Nokia expert and they said the 100 Trying message from your server is considered as provisional response, therefor behaviour of client is absolutely correct. Is there any way I can stop 100 Trying message and still run statefull SER, so that I can verify what nokia said. Any ideas suggestions are welcome. Thanking you all in advance. Best Regards, Abdul Qadir --------------------------------- Don't be flakey. Get Yahoo! Mail for Mobile and always stay connected to friends.

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[Users] Re: 200 ok not cought in ONREPLY
by Dan-Cristian Bogos 19 Oct '07

19 Oct '07

Guys, thanks to your support (both Bogdan and Daniel), I found out that there was nothing wrong with the server and t_onreply handling, but the INVITE was taking another routing three than the one I was checking and therefore never hit the t_on_reply section I wanted. This being said, I consider this issue resolved. Cheers, Dan

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[Users] AVP management
by Ovidiu Sas 05 Oct '07

05 Oct '07

Hi all, There are modules that are using default values for AVPs. What will happen if another module is reusing the same AVP? Or if the AVP is redefined inside the script? Is this completely orthogonal? Should I worry about this while using AVPs? Regards, Ovidiu Sas

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[Users] provisional_reply_timer
by Klaus Darilion 28 Jul '07

28 Jul '07

Hi! i wonder if there is a timer which controls the timeout for provisional replies. E.g. I want to cancel an INVITE and retry another gateway if the gateway does not respond with 100 (or 180) in a certain time. regards klaus -- Klaus Darilion nic.at

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[Users] 473 Filtered destination?
by Tobias Lindgren 23 Jul '07

23 Jul '07

Hi, recently moved from SER > OpenSER 1.2.0, but after some time when the server is running OpenSER starts to respond with: "SIP/2.0 473 Filtered destination (473/TM)." to my INVITEs. I don't have any blacklists/accesslist configured (the config is from SER which doesn't support that), why am I seeing this problem? Br, /Tobias

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[Users] openser 1.1.1 available for several embedded platforms
by Ovidiu Sas 10 Jul '07

10 Jul '07

Hi, openser 1.1.1 is now available in binary format for several embedded patforms: - Linksys NSLU - Synology DS-101(g+) - Iomega NAS 100d - Freecom FSG-3 - Qnap TurboStation ... and all routers supported by the oleg or dd-wrt distributions. Check out http://www.nslu2-linux.org Regards, Ovidiu Sas

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[Serusers] XLITEPHONE PROBLEM
by Adeyeye Michael 04 Jul '07

04 Jul '07

Kindly help me check this: root@micadeyeye:/usr/src# cd xten-xlite/ root@micadeyeye:/usr/src/xten-xlite# ls README xtensoftphone root@micadeyeye:/usr/src/xten-xlite# ./xtensoftphone I/O warning : failed to load external entity "/root/.Xscrc" Warning: /dev/dsp appears to be a valid audio device, but I cannot open it. Please ensure that no other applications are using the audio device (perhaps by trying ``lsof /dev/dsp''). -- Mike

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[Users] Malformed CANCEL request from Asterisk possibly impacting OpenSER?
by David Coulthart 27 Jun '07

27 Jun '07

We are currently experiencing problems with our production OpenSER (1.1.0) environment where the load on the server is steadily increasing and the OpenSER response time increases drastically (e.g., 30+ seconds for an internal call to ring). Capturing the network traffic when this happens we have noticed that our Asterisk (1.2.10) voicemail server is sending malformed CANCEL requests to our OpenSER proxy. We are not certain whether this is a cause or an effect of the load problems on the OpenSER server. Here is the SIP data from the bad CANCEL request: Session Initiation Protocol Request-Line: CANCEL :u43974@128.59.59.215:5069 SIP/2.0 Method: CANCEL [Resent Packet: False] Message Header Via: SIP/2.0/UDP 128.59.59.215:5069;branch=z9hG4bK6734835b;rport Transport: UDP Sent-by Address: 128.59.59.215 Sent-by port: 5069 Branch: z9hG4bK6734835b RPort: rport Route: <sip:128.59.59.199;lr=on;ftag=4FC702F4-7F812557> From: <sip:43974@sip.columbia.edu;user=phone> SIP from address: sip:43974@sip.columbia.edu To: "Jim Rios" <sip:47891@sip.columbia.edu>;tag=4FC702F4-7F812557 SIP Display info: "Jim Rios" SIP to address: sip:47891@sip.columbia.edu SIP tag: 4FC702F4-7F812557 Contact: <sip:u43974@128.59.59.215:5069> Contact Binding: <sip:u43974@128.59.59.215:5069> URI: <sip:u43974@128.59.59.215:5069> SIP contact address: sip:u43974@128.59.59.215:5069 Call-ID: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100 CSeq: 101 CANCEL Sequence Number: 101 Method: CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Accompanying this bad CANCEL request are the following error messages recorded by OpenSER to syslog when it attempts to process it: May 29 15:34:07 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:07 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:07 cocoa openser[892]: xl_get_ruri: ERROR while parsing the R-URI May 29 15:34:08 cocoa openser[892]: c9a0135b-41273fb6-f7b6cea9(a)160.39.14.100: route[0] CANCEL r-uri <<null>> from <sip:43974@sip.columbia.edu;user=phone> to <sip:47891@sip.columbia.edu> May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:08 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:08 cocoa openser[892]: loose_route: Error while parsing Request URI May 29 15:34:08 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:09 cocoa openser[892]: WARNING: do_action:error in expression May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:09 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:09 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:10 cocoa openser[892]: ERROR: parse_uri: bad uri, state 0 parsed: <:u43> (4) / <:u43974@128.59.59.215:5069> (26) May 29 15:34:10 cocoa openser[892]: ERROR: parse_sip_msg_uri: bad uri <:u43974@128.59.59.215:5069> May 29 15:34:10 cocoa openser[892]: alias_db_lookup: Error while parsing Request-URI While the malformed packet is likely the fault of the Asterisk server (we see no packets from OpenSER to Asterisk with the sip protocol missing from the URI), we are wondering if OpenSER errors out in a bad way when it tries to process this packet, possibly entering into some sort of loop that is causing the load to go through the roof. It is also unclear to us why the Asterisk voicemail server would be responding with a cancel. In our packet capture we see no record of a request initiated by the Asterisk server that it would be sending a CANCEL for. Any insight that could be offered into this problem would be greatly appreciated. Thx, David Coulthart

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[Serusers] SER Accounting.
by Robert Zorop 26 Jun '07

26 Jun '07

Hi, all. I know this is a little bit out off Topic. I'm trying to install CDRTool for CDR normalization and billing with SER. I get the things allmost working (freeradius + MySQL + SER) But i can't get the CDRs normalized with price. Can anybody please help me or post a CDRTool global.inc working file? . Thanks a lot in advance any help will be highly apreciated!

2 1

[Users] editing header multiple times
by zm23 20 Jun '07

20 Jun '07

Hi all, I am having some issues with editing remote party header. I understand that changes are saved as a diff and are sent when packet goes on wire. In my current config rpid is edited when invite is received and it seems to work fine. There is one scenario where I need to edit rpid or remove rpid altogether before relaying it. "remove_hf" did not remove the header after it was edited earlier. Is there a way to do this? Thanks in advance for your help. -- Zahid

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[Users] error: xmlrpc.h: No such file or directory
by kjcsb 18 Jun '07

18 Jun '07

I am trying to make openser-1.2.0-notls and get the following error: In file included from mi_xmlrpc.c:38: mi_xmlrpc.h:35:20: error: xmlrpc.h: No such file or directory In file included from mi_xmlrpc.c:38: mi_xmlrpc.h:37: error: syntax error before âenvâ mi_xmlrpc.h:37: warning: type defaults to âintâ in declaration of âenvâ mi_xmlrpc.h:37: warning: data definition has no type or storage class mi_xmlrpc.h:38: error: syntax error before â*â token mi_xmlrpc.h:38: warning: type defaults to âintâ in declaration of âxr_responseâ mi_xmlrpc.h:38: warning: data definition has no type or storage class In file included from mi_xmlrpc.c:39: xr_writer.h:37: error: syntax error before â*â token xr_writer.h:38: error: syntax error before â*â token In file included from mi_xmlrpc.c:40: xr_parser.h:36: error: syntax error before â*â token In file included from mi_xmlrpc.c:41: xr_server.h:35: error: syntax error before â*â token xr_server.h:35: error: syntax error before â*â token xr_server.h:36: warning: type defaults to âintâ in declaration of âdefault_methodâ xr_server.h:36: warning: data definition has no type or storage class xr_server.h:38: error: syntax error before â*â token mi_xmlrpc.c:44:26: error: xmlrpc_abyss.h: No such file or directory mi_xmlrpc.c:50: error: syntax error before âenvâ mi_xmlrpc.c:50: warning: type defaults to âintâ in declaration of âenvâ mi_xmlrpc.c:50: warning: data definition has no type or storage class mi_xmlrpc.c:51: error: syntax error before â*â token mi_xmlrpc.c:51: warning: type defaults to âintâ in declaration of âxr_responseâ mi_xmlrpc.c:51: warning: data definition has no type or storage class mi_xmlrpc.c: In function âmod_child_initâ: mi_xmlrpc.c:162: warning: implicit declaration of function âxmlrpc_server_abyss_init_registryâ mi_xmlrpc.c:172: error: âxmlrpc_server_abyss_rpc2_handlerâ undeclared (first use in this function) mi_xmlrpc.c:172: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:172: error: for each function it appears in.) mi_xmlrpc.c:178: error: âxmlrpc_server_abyss_default_handlerâ undeclared (first use in this function) mi_xmlrpc.c:193: warning: implicit declaration of function âxmlrpc_env_initâ mi_xmlrpc.c:196: warning: implicit declaration of function âxmlrpc_build_valueâ mi_xmlrpc.c:196: warning: assignment makes pointer from integer without a cast mi_xmlrpc.c:197: error: request for member âfault_occurredâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:198: error: request for member âfault_stringâ in something not a structure or union mi_xmlrpc.c:219: warning: implicit declaration of function âxmlrpc_env_cleanâ mi_xmlrpc.c:220: warning: implicit declaration of function âxmlrpc_DECREFâ make[1]: *** [mi_xmlrpc.o] Error 1 make[1]: Leaving directory `/usr/src/openser-1.2.0-notls/modules/mi_xmlrpc' make: *** [modules] Error 2 rpm -q xmlrpc-c-devel xmlrpc-c-devel-1.04-1.fc4 find -name xmlrpc.h ./usr/include/xmlrpc-c/xmlrpc.h Any suggestions would be appreciated. Regards Cameron

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[Serusers] Call waiting and accounting
by inge 15 Jun '07

15 Jun '07

Hi all, I've got a problem with call waiting function. This one allow us to hang out a call when we are already in communication. But when I press "R" on my phone to take the second call, there is one INVITE and one ACK generated by the proxy and being accounted in MySQL. It's the same when I switch between both callers. So I can have for the same callid 2 3 4...X INVITE ACK and one BYE. So in SERWeb for example, I can see for one call 3 lines of accounting. Thanks for you support. Adrien .L

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[Users] Interaction Presence Agent - Presence User Agent
by Schumann Sebastian 11 Jun '07

11 Jun '07

Dear all, I have a question for understanding: I have implemented one OpenSER as Registrar and one OpenSER as Presence Agent plus XCAP Server 1.0.0. This is all working so far with newest SVN Release (Big Thanks to Ovidiu and Anca for the hint) of OpenSER. Now I want to deploy a Web-Front-End for user interaction with the Presence Core, because not all clients used are SIMPLE capable. User of non-capable devices should be able to just login and change their presence state. They do not necessarily have to be registered at the registrar, why PUA+ pua_usrloc on registrar side is no choice. I was thinking about another OpenSER deployed as PUA plus the pua_mi module for a solution. It gets commands from Web-Frontend via PHP, forwarded to the FIFO with pua_mi module. The problem I have now is that the messages are sent just once. Means a user changes his presence state, PUA sends a PUBLISH to PA but after the expiry the user appears to be offline. Is there any possiblitiy to let the PUA send messages constantly, when the user selects online - means all the time until offline is selected? I was thinking about many possiblities (cronjob or shellscript running all the time during online time and sending data to FIFO) but they are more or less just work-arounds. Best regards Sebastian Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 910 643010 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk

2 2

[Serusers] problem with MESSAGE and un-REGISTER
by KUMAR 11 Jun '07

11 Jun '07

hi all, I've started using SER 2.0 rc1 and am having a bit of problems with UA behind NAT. I'm using Windows Messenger 5.1 as the UA. Everything works fine when both UAs are not behind NAT. But when I place one of the UAs behind NAT, then, here's what happens, in this order UA Behind Nat ----------MESSAGE------>UA outside NAT..............ok UA outside NAT-----------MESSAGE------->UA behind NAT.............OK UA behind NAT------------MESSAGE------->UA outside NAT.............message is stored by msilo module and sent after some time. The log shows: MSILO: the downstream UA doesn't support MESSAGEs Notice that this only happens once a message from outside of NAT is sent to UA inside NAT. I am using RTP Proxy for NAT solution which is on the same system as SER. Moreover, I also cannot signout using the Windows Messenger. However, when I tested using XLite, None of the above mentioned problems arose. Can anyone please suggest me as to where the problem might be. The ser.cfg that I'm using is the one from the iptel site, the one with NAT & RTPproxy. I am also attaching the ser.cfg. Thank you in advance. Kumar

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Re: [Serusers] [Serweb-users] OpenSER and SERWeb problems logging on (that old chestnut)
by flavio 07 Jun '07

07 Jun '07

May 29 11:49:19 *serweb* [error] file: /opt/*serweb*-0.9.6/html/data_layer.php:465: DB Error: connect failed - [nativecode=Access denied for user 'ser'@'localhost' (using password: YES)] ** mysql://ser:heslo@localhost/ser<http://www.voipuser.org/ship_to.php?url=mysql://ser:heslo@localhost/ser> Do you access mysql with mysql -u ser -p $>password: heslo ? If yes, try to start mysqld with --old-password option following tips at url: http://dev.mysql.com/doc/refman/5.0/en/old-client.html Bye, F. -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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[Serusers] avp_radius - ser2.1 - Access Request Contents
by tzieleniewski 07 Jun '07

07 Jun '07

hi! When I invoke the radius_load_attrs() the message send to radius has the following shape: Thu Mar 8 10:50:46 2007 Packet-Type = Access-Request User-Name = "sip:hellboy@voip.rd.touk.pl" SER-Service-Type = Get-User-Attrs NAS-Port = 0 NAS-IP-Address = 127.0.0.1 the message doesn't contain any authentication data like password. so do I understand right that radius in such case should be configured in such a way to accept any message of the SER-Service-Type = Get-User-Attrs? Are still the avp loaded from radius prefixed or they are just put into the particular track and class? Could You please give me one example of the radius one avp record and the shape in which it will appear in ser after loading. Thank you in advance Best Tomasz

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[Users] Mediaproxy Selection for Inbound DID calls
by Mike O'Connor 06 Jun '07

06 Jun '07

Hi All I would like to be able to have a difference Mediaproxy group selected per user account on inbound DID calls. Our termination provider sends the calls to our openSER with a URI of something <number>@<IP_address>. '08723451234(a)202.xxx.xxx.xxx', we use the alias_db_lookup("dbaliases") function to resolve the phone number to a user account on the DID calls. I would think that some type of rewrite of the '@202.xxx.xxx.xxx' would need to be done but I'm sure this would cause other issues. Any ideas would be great :) Thanks Mike

2 2

[Serusers] Preserve DSCP marking
by Ravindranath Gummadidala 01 Jun '07

01 Jun '07

Does SER have support for preserving the DSCP markings of (incoming) SIP messages it processes before sending them out? Or is this something one would need to change the SER source code for? I searched in the archives but could not find anything relevant. Regards, -Ravi.

3 2

[Users] changes in database setup scripts
by Henning Westerholt 01 Jun '07

01 Jun '07

Hello all, i've commited new database setup scripts for mysql and postgresql. Now are the SQL tables generated from a common XML definition, to avoid inconsistencies between the different databases. With the new system is not more necessary to update three different shell scripts for one database table change, hopefully this will lead to fewer bugs. The dbtext integration is still missing, i'll work on this and on further extension of the system in two weeks after the programming course in paris. I've tested the new system with mysql 5.0 and postgresql 8.2, but there are probably still some bugs present. So if you get an database setup script related error from trunk, don't hesitate to complain here or on the devel list. :-) You'll find further informations about this system in the wiki: http://www.openser.org/dokuwiki/doku.php/development:db-schema Best regards, Henning

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[Serusers] wrong password
by ram 01 Jun '07

01 Jun '07

Hi iam runing SER and my client is registering with Asterisk, running call center he keep getting the following error chan_sip.c: Forbidden - wrong password on authentication for INVITE to '"M0601001458000013456" <sip:user@domain.com>;tag=as5bbc2eec' and voice going choppy, when he call from single user the voice go clear when he intiates the calls from number of people voice is very choppy and some time not able to hear all are public IP, not NAT here any suggestions ram

2 1

[Users] Openser 1.2.1 with SERWeb 0.9.6-cannot open write fifo
by Lesley Liu 01 Jun '07

01 Jun '07

Hi Users, I'm setup SERWeb 0.9.6 with Openser 1.2.1. At first, when login with correct username/password, there is a "sorry -- cannot open write fifo" on the my_account.php. Referred from google, I changed the access right of openser_fifo to 666; that cause the my_account.php does not show. Please kindly advise, thanks in advance. Regards, Lesley

2 3

[Serusers] Fwd: SIP 479 Regretfully
by flavio 31 May '07

31 May '07

---------- Forwarded message ---------- From: flavio <flavio.patria(a)gmail.com> Date: 31-mag-2007 18.04 Subject: Re: [Serusers] SIP 479 Regretfully To: olivier.taylor(a)hh174.be But If I try with my BT102 GrandStream (configured as Polycom on my SER) I'm able to start a call to PSTN Number. How is it possible? Have you any suggestions about? Thanks, ps I've also asterisk running on the same machine listening on port 5062. U 2007/05/31 18:00:52.457140 10.28.19.124:5060 -> 10.28.19.202:5060 INVITE sip:0672020949@10.28.19.202 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bK5654e41ba6af4166. From: <sip:0660522016@10.28.19.202>;tag=44dcaa39db672de9. To: <sip:0672020949@10.28.19.202>. Contact: <sip:0660522016@10.28.19.124>. Supported: replaces. Proxy-Authorization: Digest username="0660522016", realm="10.28.19.202", algorithm=MD5, uri="sip:0672020949@10.28.19.202", nonce="465f0e8002ccdf57a20d382a49036bd0e5c691d0", response="554b818d1ecfc8455f1bc2e774881508". Call-ID: 19d87a9eb5a4d432(a)10.28.19.124. CSeq: 17072 INVITE. User-Agent: Grandstream BT110 1.0.8.12. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 211. . v=0. o=0660522016 8000 8001 IN IP4 10.28.19.124. s=SIP Call. c=IN IP4 10.28.19.124. t=0 0. m=audio 5004 RTP/AVP 18 8 0. a=sendrecv. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=ptime:20. 2007/5/31, olivier.taylor <olivier.taylor(a)gmail.com>: > hi, > > Regretfully, we were not able to process the URI > > probably a malformed URI, don't you have to remove the leading 0 and add the international code? > > hope it helps, > > Olivier > > > > flavio a écrit : > > Hi to all. > > I've configured my polycom ip500 IPphone to use it with ser. > > If I try a call to users registred to ser all works fine. > > If I try to call PSTN Number through my gateway I've the follow sip message: > > > > > > > > INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0. > > Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. > > From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. > > To: <sip:0672028405@10.28.19.202;user=phone>. > > CSeq: 2 INVITE. > > Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. > > Contact: <sip:0660522015@10.28.19.143>. > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, > > NOTIFY, PRACK, UPDATE, REFER. > > User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1. > > Supported: 100rel,replace. > > Allow-Events: talk,hold,conference. > > Proxy-Authorization: Digest username="0660522015", > > realm="10.28.19.202", > > nonce="465ef44bbb09c0155dc8555314519a20cac896f8", > > uri="sip:0672028405@10.28.19.202:5060;user=phone", > > response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5. > > Max-Forwards: 70. > > Content-Type: application/sdp. > > Content-Length: 237. > > . > > v=0. > > o=- 1180620549 1180620549 IN IP4 10.28.19.143. > > s=Polycom IP Phone. > > c=IN IP4 10.28.19.143. > > t=0 0. > > m=audio 2234 RTP/AVP 18 8 0 101. > > a=rtpmap:18 G729/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > > > # > > U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060 > > SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL). > > Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. > > From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. > > To: <sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7. > > CSeq: 2 INVITE. > > Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. > > Server: Sip EXpress router (0.9.6 (i386/linux)). > > Content-Length: 0. > > Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314 > > req_src_ip=10.28.19.143 req_src_port=5060 > > in_uri=sip:0672028405@10.28.19.202:5060;user=phone > > out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1". > > > > > > Have you any suggestion about? > > I use my Polycom with Asterisk and BroadSoft without any problem. > > > > Thanks for your support and great patience :D > > > > Bye, > > > > F. > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ******************************** -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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[Users] load testing presence server
by Toni Barata 31 May '07

31 May '07

Hi All, A test case with SUBSCRIBE winfo event message have reached values of hundreds of messages per second in a load test scenario. But a test case with SUBSCRIBE presence event message have only reached values of 20 messages per second. Is there any explainable reason for this big difference between the results of both test cases? Thanks, Toni

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Re: [Users] adding users
by Jeremy George 31 May '07

31 May '07

Okay, thanks. - Jeremy >-----Original Message----- >From: Ovidiu Sas [mailto:sip.nslu@gmail.com] >Sent: Thursday, May 31, 2007 10:42 AM >To: 'Jeremy George' >Cc: users(a)openser.org >Subject: Re: [Users] adding users > >Hi Jeremy, > >Have you installed the serweb tables? >If yes, you have to set HAS_SERWEB="yes" in your openserctlrc. Check >the mailing list archive, this issue was already discusses a few >times. > > >Regards, >Ovidiu Sas > >On 5/31/07, Jeremy George <jeremy(a)georgeco.us> wrote: >> >> I upgraded to 1.2.1 and completely re-created the db. Now I can add >> exactly one user to the db. Also, trying to delete that one user from >> openserctl hangs. Thoughts? >> >> Thanks much. >> >> - Jeremy >> >> Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf >> MySql password for user 'openser@localhost': >> new user 'jeremy' added >> Breathless-FC6# openserctl add helen xxxx murph(a)the.surf >> MySql password for user 'openser@localhost': >> ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 >> ERROR: introducing the new user 'helen' to the database failed >> Breathless-FC6# >> Breathless-FC6# openserctl rm jeremy >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >

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[Users] adding users
by Jeremy George 31 May '07

31 May '07

I upgraded to 1.2.1 and completely re-created the db. Now I can add exactly one user to the db. Also, trying to delete that one user from openserctl hangs. Thoughts? Thanks much. - Jeremy Breathless-FC6# openserctl add jeremy xxxx murph(a)the.surf MySql password for user 'openser@localhost': new user 'jeremy' added Breathless-FC6# openserctl add helen xxxx murph(a)the.surf MySql password for user 'openser@localhost': ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user 'helen' to the database failed Breathless-FC6# Breathless-FC6# openserctl rm jeremy

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[Serusers] SIP 479 Regretfully
by flavio 31 May '07

31 May '07

Hi to all. I've configured my polycom ip500 IPphone to use it with ser. If I try a call to users registred to ser all works fine. If I try to call PSTN Number through my gateway I've the follow sip message: INVITE sip:0672028405@10.28.19.202:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. To: <sip:0672028405@10.28.19.202;user=phone>. CSeq: 2 INVITE. Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. Contact: <sip:0660522015@10.28.19.143>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1. Supported: 100rel,replace. Allow-Events: talk,hold,conference. Proxy-Authorization: Digest username="0660522015", realm="10.28.19.202", nonce="465ef44bbb09c0155dc8555314519a20cac896f8", uri="sip:0672028405@10.28.19.202:5060;user=phone", response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 237. . v=0. o=- 1180620549 1180620549 IN IP4 10.28.19.143. s=Polycom IP Phone. c=IN IP4 10.28.19.143. t=0 0. m=audio 2234 RTP/AVP 18 8 0 101. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. # U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060 SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL). Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD. From: "0660522015" <sip:0660522015@10.28.19.202>;tag=8DDF0DB-E8BCEB84. To: <sip:0672028405@10.28.19.202;user=phone>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7. CSeq: 2 INVITE. Call-ID: 264aa927-d2a3a7c9-64f3fe3a(a)10.28.19.143. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314 req_src_ip=10.28.19.143 req_src_port=5060 in_uri=sip:0672028405@10.28.19.202:5060;user=phone out_uri=sip:0672028405@10.28.52.105:5060:5060;user=phone via_cnt==1". Have you any suggestion about? I use my Polycom with Asterisk and BroadSoft without any problem. Thanks for your support and great patience :D Bye, F.

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[Users] openserctl ping broken?
by Papadopoulos Georgios 31 May '07

31 May '07

Hello all, I am having trouble getting openserct ping to work (version 1.2). The server is running but I get [1]+ Done cat <$path | filter_fl 400 I modified openserct at the options_ping() function to do echo $RET and I get [1]+ Done cat <$path | filter_fl 400 Too few or too many arguments I tried playing around with the arguments according to the documentation for tm's function t_uac_dlg, but no luck. Can someone help? thank you George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

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[Serusers] SERVER REDIRECTION
by Cyprian Clement 31 May '07

31 May '07

Does any one know what modules(and corresponding functions if possible) that I need to use in order to make the SER do the Redirection by initiating a 302( after the reception of INVITE) that contains the contact info/uri that the caller will need to use to call the forwarded-to client? Can this be achieved without the need to use CPL language? Thanks in advance. Regards, Cyp. --------- ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

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[Users] openser: Call workflow
by Helmut Kuper 31 May '07

31 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is there a way in openser to get it work like this: 1. SIP-Request comes in 2. openser detects a special service for dialed number 3. openser sets that original sip session on hold and forwards the request to a remote service node 4. remote service node does it job and sends back a result message via network to openser 5. openser checks that result, get the session from hold and continues it's work on the session In the end I 'm looking for a way to let openser handles a sip session like a workflow, where I can inject some value added services by REMOTE service nodes (not by modules or plugins) without forwarding the complete session to a remote service node. I hope u understand what I mean. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGXrWO4tZeNddg3dwRAueLAJ48xkexHAz3GCkzKeNuRewV1hsb3ACeIr7p LST+x5cMjAuIvbzZl61PPyk= =Ita4 -----END PGP SIGNATURE-----

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[Serusers] SER Debug
by Cyprian Clement 31 May '07

31 May '07

Hi, Does any one know of a useful way of getting(turning on) SER debug report. I am trying to locate where my call is failing with the SER and currently use the "ngrep" but it does not seem to reveal much useful internal report. most of it looks external similar to ethereal. Will appreciate any share of info in this regard. Cyp. -------- ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

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[Serusers] register my asterisk on ser
by flavio 31 May '07

31 May '07

Hi, I've asterisk and ser runnig onver same machine.Is it wrong? I'm trying to register asterisk on ser as UAC, but I've the follow log error message: ERROR: forward_reply: no 2nd via found in reply sip log with ngrep are soon reported: U 2007/05/31 11:52:48.261775 10.28.19.202:5060 -> 10.28.19.202:5060 REGISTER sip:10.28.19.202 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. User-Agent: Asterisk PBX. Max-Forwards: 70. Expires: 300. Contact: <sip:2000@10.28.19.202>. Event: registration. Content-Length: 0. . # U 2007/05/31 11:52:48.262493 10.28.19.202:5060 -> 10.28.19.202:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=7153 req_src_ip=10.28.19.202 req_src_port=5060 in_uri=sip:10.28.19.202 out_uri=sip:10.28.19.202 via_cnt==1". . # U 2007/05/31 11:52:48.263017 10.28.19.202:5060 -> 10.28.19.202:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 10.28.19.202:5060;branch=z9hG4bK7bd3439f. From: <sip:0192091221@10.28.19.202>;tag=as7b047b0b. To: <sip:0192091221@10.28.19.202>;tag=979d95a734c13f6db8b9e3a72b9f44a0.020d. Call-ID: 41a53c8a015e385f4b6d951f6a0e07a5(a)127.0.0.1. CSeq: 102 REGISTER. WWW-Authenticate: Digest realm="10.28.19.202", nonce="465eb83c86f495e14144d970e6bbabb7952b6579". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=7153 req_src_ip=10.28.19.202 req_src_port=5060 in_uri=sip:10.28.19.202 out_uri=sip:10.28.19.202 via_cnt==1". Have you any suggestions? Thanks, F. -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

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Re: [Users] branch parameter in ACK set to zero?
by Bogdan-Andrei Iancu 31 May '07

31 May '07

Hi Tim, As I suspected, it is empty branch is triggered by a bogus SIP flow. The ACK generated by the caller device is bogus as the TO uri is different than the TO in INVITE. Also the caller device does use branch at all in VIA and forces openser to do pre-RFC3261 transaction matching with involves the TO uri. As the log show, the ACK does not match any transaction: DEBUG: t_lookup_request: start searching: hash=11223, isACK=1 DEBUG: proceeding to pre-RFC3261 transaction matching DEBUG: t_lookup_request: no transaction found So, the ACK is forwarded as stateless request - for this case, there is complicated alg for computing the branch param based on the branch params from the previous VIA headers. But the previous VIA is bogus (no branch at all) and openser finds itself in the impossibility to compute a branch for its VIA. regards, bogdan Tim Madorma wrote: > Hi Bogdan, > > Attached is the log and the openser.cfg file. > > thanks, > Tim > > On 5/30/07, Tim Madorma <tmadorma(a)gmail.com> wrote: >> Hi Bogdan, >> >> I have attached an html file that shows the call trace (note that the >> OpenSER is 65.185.232.62:5061in the flow). I will send the log file >> and openser.cfg in a separate email since it would put me over the 40K >> limit. >> >> thanks, >> Tim >> >> On 5/30/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> > Hi Tim, >> > >> > the VIA branch param must not be 0, indeed. Could you provide an >> network >> > trace for the entire call and the debug log for level 9? >> > >> > regards, >> > bogdan >> > >> > Tim Madorma wrote: >> > > Hi, >> > > >> > > In testing with OpenSER 1.2 I noticed that the branch parameter >> of the >> > > Via in the ACK to 200 OK has a value of zero. This does not seem >> > > right. From 16.6 of RFC-3261: >> > > >> > > The proxy MUST insert a Via header field value into the copy >> > > before the existing Via header field values. The >> construction >> > > of this value follows the same guidelines of Section >> 8.1.1.7. >> > > This implies that the proxy will compute its own branch >> > > parameter, which will be globally unique for that branch, >> and >> > > contain the requisite magic cookie. >> > > >> > > Please advise. >> > > >> > > thanks, >> > > Tim >> > > >> > > _______________________________________________ >> > > Users mailing list >> > > Users(a)openser.org >> > > http://openser.org/cgi-bin/mailman/listinfo/users >> > > >> > >> > >> >>

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[Serusers] Ser in B2B UA Mode
by Gustavo Passos Tourinho 31 May '07

31 May '07

Hi, Can ser be used like a b2b ua? I mean, can it register in others proxies? Thank you. Regards, Gustavo

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[Users] Replace destination URI (remove port 5062)
by Peter P GMX 31 May '07

31 May '07

I have set up an OpenSER to run on port 5062 instead of 5060. Attached to OpenSER I have 2 videoconferencing clients which communicate via SIP. What's working so far - internal client (registered on OpenSER) to another internal client (registered on OpenSER) - internal client (registered on OpenSER) to another external client - this all works with the OpenSER and other clients behind their NAT, as Mediaproxy is installed What is actually not working is - another external client connetcing to an internal client (registered on OpenSER) because I have to adress the the To:URI as sip.mydomain.com:5062. - Whenever an external INVITE comed in, it ends up with "message too big" because OpenSER is cycling through its routing plan until it adds that many via headers that the message increases 2kb. - what I see is that openser does not try to find the client on sip.mydomain.com, instead I feel that it tries to find the client on sip.mydomain.com:5062 as uri. - as my operating system (debian etch) cannot resolve this domain, this runs through the loop My question: - is the a way to modify the destinaton header ($tu) that the port is cut off (name@sip.mydomain.com:5062 ==> name(a)sip.mydomain.com)? Best regards peter

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[Serusers] Call forward on no answer and busy failing
by Ranganath B 31 May '07

31 May '07

Hi, I could get the blind (unconditional) call forward option working with the configuration that is provided in the IPTEL website but using the same config file and the ideas provided I couldn't get the call forward on busy and no answer working. I get the following error in the /var/log/messages: May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: mk_proxy: could not resolve hostname: "ser.iptel.org" May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: uri2proxy: bad host name in URI <sip:1234@ser.iptel.org> May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: t_forward_nonack: failure to add branches May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: w_t_relay (failure mode): forwarding failed May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy May 16 15:49:52 localhost /usr/local/sbin/ser[17563]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) It would be of great help if anybody provide me input on how to resolve this issue or a configuration file using which I can get the call forward on busy and no response working. BTW, I am using the ser verison: 0.9.6. I have populated an entry for call forward no answer/busy in the mysql database for the table "usr_preferences". Thanks in advance. Regards, Ranganath B

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[Users] what algorithm does radiusclient-ng+openser use?(md5 or)
by Jason Ma 30 May '07

30 May '07

Hi All, I'm configuring radius support on openser using radiusclient-ng-0.5.5.1.tar.gz,and the radius server is Lucent AAA server,everytime openser sent the request to AAA server,the AAA server could not parse the request,the logs on AAA server is as below.I noticed that the the server said it was "Unsupported algorithm",and the algorithm was null,I'm wandering what kind of algorithm does radiusclient-ng use?MD5 or HA1......... Please help ! Thanks in advance. ~~~~~~~~~~~~~~~~~~~~~~~ 2007/05/29 16:51:45.685 <engine.item.setup> Initializing Radius Item: 192.168.1.42:33345->0.0.0.0:1812(77) 0 <engine.item.setup> Using dictionary: draft-sterman-aaa-sip-01 0 <engine.item.setup> Request decode: User-Name = "21230001(a)192.168.1.42" Digest-Attributes = User-Name = "21230001" Digest-Attributes = Realm = "192.168.1.42" Digest-Attributes = Nonce = "465c49b766fa11f4a9db29977bf16857d3372780" Digest-Attributes = URI = "sip:192.168.1.42" Digest-Attributes = Method = "REGISTER" Digest-Response = "689a0f89cd73751f61b12d04f585a224" Service-Type = IAPP-Register Anonymous = v0-a208-3231323330303031 NAS-Port = 5060 NAS-IP-Address = 192.168.1.42 0 <engine.item.setup> User-Name parsed: Base-User-Name = "21230001", User-Realm = "192.168.1.42" 0 <engine.item.setup> Item setup complete 0 <engine.worker.1> <setup> ==> ReadUserFile.auth:readUserFile 0 <plugin.ReadUserFile.auth:readUserFile> searchValue = ' 21230001(a)192.168.1.42'. 0 <plugin.ReadUserFile.auth:readUserFile> Found entry: 21230001(a)192.168.1.42 0 <plugin.ReadUserFile.auth:readUserFile> Check items for found entry: User-Password = <hidden> Auth-Type = Local 0 <plugin.ReadUserFile.auth:readUserFile> Reply items for found entry: 0 <plugin.ReadUserFile.auth:readUserFile> SUCCESS -- Read User 21230001(a)192.168.1.42 0 <engine.worker.1> ReadUserFile.auth:readUserFile ==> AuthHttpDigest.auth:checkDigest by SUCCESS -- Read User 21230001(a)192.168.1.42 0 <plugin.AuthHttpDigest.auth:checkDigest> response = 689a0f89cd73751f61b12d04f585a224 0 <plugin.AuthHttpDigest.auth:checkDigest> realm = 192.168.1.42 0 <plugin.AuthHttpDigest.auth:checkDigest> nonce = 465c49b766fa11f4a9db29977bf16857d3372780 0 <plugin.AuthHttpDigest.auth:checkDigest> method = REGISTER 0 <plugin.AuthHttpDigest.auth:checkDigest> uri = sip:192.168.1.42 0 <plugin.AuthHttpDigest.auth:checkDigest> qop = 0 <plugin.AuthHttpDigest.auth:checkDigest> algorithm = 0 <plugin.AuthHttpDigest.auth:checkDigest> entityBodyHash = 0 <plugin.AuthHttpDigest.auth:checkDigest> cNonce = 0 <plugin.AuthHttpDigest.auth:checkDigest> nonceCount = 0 <plugin.AuthHttpDigest.auth:checkDigest> username = 21230001 0 <plugin.AuthHttpDigest.auth:checkDigest> ERROR -- Error generating HTTP digest: java.io.IOException: Unsupported algorithm: : java.io.IOException: Unsupported algorithm: 1 <engine.worker.1> AuthHttpDigest.auth:checkDigest ==> End-Of-Methods by ERROR -- Error generating HTTP digest: java.io.IOException: Unsupported algorithm: 1 <engine.worker.1> 21230001(a)192.168.1.42 login discarded due to Error generating HTTP digest: java.io.IOException: Unsupported algorithm: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~

2 3

[Users] branch parameter in ACK set to zero?
by Tim Madorma 30 May '07

30 May '07

Hi, In testing with OpenSER 1.2 I noticed that the branch parameter of the Via in the ACK to 200 OK has a value of zero. This does not seem right. From 16.6 of RFC-3261: The proxy MUST insert a Via header field value into the copy before the existing Via header field values. The construction of this value follows the same guidelines of Section 8.1.1.7. This implies that the proxy will compute its own branch parameter, which will be globally unique for that branch, and contain the requisite magic cookie. Please advise. thanks, Tim

3 4

[Serusers] SER 2.0.0 RC 1 won't start on Fedora 6
by Umer Waqar 30 May '07

30 May '07

Hi, I downloaded the source and compiled SER 2.0.0 RC 1 as per instructions on http://www.iptel.org/ser_2_0_0_release_candidate_1 I get an error when try to check it with 'sercmd' and also can't see the ser process running Rgrds, Umer [root@umer ~]# /usr/local/sbin/ser Listening on udp: 192.168.1.103 [192.168.1.103]:5060 tcp: 192.168.1.103 [192.168.1.103]:5060 Aliases: tcp: phone03.tb.jasomi.com:5060 udp: phone03.tb.jasomi.com:5060 *: umer-sip.jasomi.com:* [root@umer ~]# /usr/local/sbin/sercmd ERROR: connect_unix_sock: connect(/tmp/ser_ctl): No such file or directory [2] [root@umer ~]# ps -A | grep ser 135 ? 00:00:00 kseriod 4931 ? 00:00:00 gam_server

2 1

[Serusers] SER as a boundary proxy - how to force routing?
by tzieleniewski 30 May '07

30 May '07

Hi, I have the following situation. I have one SER instance with public IP and LAN working as a boundary proxy. (its only role is to filter the traffic and authorize users) on the LAN side I have another SER instance with works as registrar and local sip proxy. How can I force the second proxy this in the LAN network to route the messages to the boundary proxy(no matter what transport is used). The best solution would be to route them only when the calls are made to the users which are localized outside the local LAN on the public NET. Thank You In advanced Cheers Tomasz

1 0

RE: [Users] BUG: qm_free ?
by Papadopoulos Georgios 30 May '07

30 May '07

No, I don't use t_newtran at all. Now I am trying to get this to happen again but no luck... Anyway, is it something to worry about or will the leaked transactions be cleaned up? Regards, George ps. I cannot connect to the SVN, is it down? > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, May 30, 2007 4:33 PM > To: Papadopoulos Georgios > Cc: Klaus Darilion; users(a)openser.org > Subject: Re: [Users] BUG: qm_free ? > > Perfect - thanks for testing - I applied the fix also on 1.2. > > regarding the Warning message: there were in 1.2 some fixes > related to detection of the transactions leak due scripting > issues. Probably 1.1 was not able to detect and report all cases. > > Do you use in your script t_newtran() function? typically, > when you use this function without terminating the > transaction (relay or reply in a stateful way), you will get > the warning. can you check this on your script? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > Solved! Thank you! > > However, before applying the patch, I increased private > memory to 2MB > > and it did not crash. Then I set it back to 1MB, applied > the patch and > > it worked. > > > > Now I have a ton of these messages: > > WARNING: script writer didn't release transaction What does > it mean? > > This was not present in 1.1 and my config script has not changed > > (except for the necessary changes for the migration) > > > > Regards > > > > George > > > > > > > > > > > >> -----Original Message----- > >> From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >> Sent: Wednesday, May 30, 2007 2:18 PM > >> To: Papadopoulos Georgios > >> Cc: Klaus Darilion; users(a)openser.org > >> Subject: Re: [Users] BUG: qm_free ? > >> > >> Hi George, > >> > >> I managed to find the bug without the backtrace. I applied > a fix for > >> development version (SVN trunk) and find attached a patch > for 1.2 - > >> could you please give it a try and see if it solves the bug? > >> > >> thanks and regards, > >> bogdan > >> > >> Papadopoulos Georgios wrote: > >> > >>> I cannot find a core file anywhere. Where should it be? I can > >>> reproduce the problem very easily, so please give me more > >>> > >> instructions > >> > >>> on how to get the coredump+backtrace that you need. > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> > >>> > >>>> -----Original Message----- > >>>> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >>>> Sent: Tuesday, May 29, 2007 8:34 PM > >>>> To: Papadopoulos Georgios > >>>> Cc: users(a)openser.org > >>>> Subject: Re: [Users] BUG: qm_free ? > >>>> > >>>> Maybe a bug? > >>>> > >>>> Can you provide a coredump+backtrace? I'm sure Bogdan or > >>>> > >> Daniel will > >> > >>>> take a look at it. > >>>> > >>>> regards > >>>> klaus > >>>> > >>>> Papadopoulos Georgios wrote: > >>>> > >>>> > >>>>> Hi Klaus, > >>>>> > >>>>> It crashes immediately, I have no time to do anything. I > >>>>> > >>>>> > >>>> tried again > >>>> > >>>> > >>>>> and I increased the log level. So now it is process 692. > >>>>> > >>>>> > >>>> Here is part > >>>> > >>>> > >>>>> of the output. I can send you the whole output if that helps. > >>>>> > >>>>> thank you > >>>>> > >>>>> George > >>>>> > >>>>> > >>>>> 16(692) DEBUG: timer routine:5,tl=0xb61abdc4 next=0xb61adfd8, > >>>>> timeout=3000000 > >>>>> 16(692) DEBUG: retransmission_handler : request resending > >>>>> (t=0xb61abc78, REGISTER ... ) > >>>>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>>>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61abdc4 (6000000) > >>>>> 16(692) DEBUG: retransmission_handler : done > >>>>> 16(692) DEBUG: timer routine:5,tl=0xb61adfd8 next=(nil), > >>>>> timeout=3000000 > >>>>> 16(692) DEBUG: retransmission_handler : request resending > >>>>> (t=0xb61ade8c, REGISTER ... ) > >>>>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>>>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61adfd8 (6000000) > >>>>> 16(692) DEBUG: retransmission_handler : done > >>>>> 16(692) DEBUG: timer routine:0,tl=0xb6182810 next=0xb61845ac, > >>>>> timeout=3 > >>>>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>>>> (0xb61826a8) > >>>>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>>>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>>>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>>>> > >>>>> > >>>> test: branch=0, > >>>> > >>>> > >>>>> last_recv=408, flags=0 > >>>>> 16(692) DEBUG:tm:t_should_relay_response: trying > >>>>> > >> DNS-based failover > >> > >>>>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>>>> add_rule_to_list(377) > >>>>> 16(692) qm_malloc(0xb5f99000, 72) returns address > 0xb6278238 frag. > >>>>> 0xb6278220 (size=72) on 1 -th hit > >>>>> 16(692) DBG:add_list_to_head: adding to bl dns > >>>>> > >> 0xb6278238,0xb6278238 > >> > >>>>> 16(692) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > >>>>> 16(692) DEBUG:tm:local_reply: local transaction completed > >>>>> 16(692) DEBUG: add_to_tail_of_timer[2]: 0xb61826f0 (8) > >>>>> 16(692) DEBUG: final_response_handler : done > >>>>> 16(692) DEBUG: timer routine:0,tl=0xb61845ac next=0xb6186c6c, > >>>>> timeout=3 > >>>>> 16(692) DEBUG: timer routine:0,tl=0xb6186c6c next=0xb6188d98, > >>>>> timeout=3 > >>>>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>>>> (0xb6186b04) > >>>>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>>>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>>>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>>>> > >>>>> > >>>> test: branch=0, > >>>> > >>>> > >>>>> last_recv=408, flags=0 > >>>>> 16(692) DEBUG:tm:t_should_relay_response: trying > >>>>> > >> DNS-based failover > >> > >>>>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>>>> add_rule_to_list(377) > >>>>> 16(692) qm_malloc(0xb5f99000, 72) returns address > 0xb62782b0 frag. > >>>>> 0xb6278298 (size=72) on 1 -th hit > >>>>> 16(692) DBG:add_list_to_head: adding to bl dns > >>>>> > >> 0xb62782b0,0xb62782b0 > >> > >>>>> 16(692) qm_free(0x8151c40, 0xb62782b0), called from > blacklists.c: > >>>>> rm_dups(440) > >>>>> 16(692) BUG: qm_free: bad pointer 0xb62782b0 (out of memory > >>>>> > >>>>> > >>>> block!) - > >>>> > >>>> > >>>>> aborting > >>>>> > >>>>> > >>>>> > >>>>>> -----Original Message----- > >>>>>> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >>>>>> Sent: Tuesday, May 29, 2007 5:48 PM > >>>>>> To: Papadopoulos Georgios > >>>>>> Cc: users(a)openser.org > >>>>>> Subject: Re: [Users] BUG: qm_free ? > >>>>>> > >>>>>> The bug was in process 32133 > >>>>>> > >>>>>> There are no log statements from this process. What what this > >>>>>> process? > >>>>>> (use "openserctl ps" to see the PIDs of the processes) > >>>>>> > >>>>>> regards > >>>>>> klaus > >>>>>> > >>>>>> Papadopoulos Georgios wrote: > >>>>>> > >>>>>> > >>>>>>> Hello all, > >>>>>>> > >>>>>>> I am having some trouble geting Openser 1.2 to work. It > >>>>>>> > >>>>>>> > >>>>>> crashes pretty > >>>>>> > >>>>>> > >>>>>>> much immediately. I have tried it successfully on a test > >>>>>>> > >>>>>>> > >>>>>> system but as > >>>>>> > >>>>>> > >>>>>>> soon as I start it in the production environment, it > >>>>>>> > >>>>>>> > >>>>>> crashes. Any ideas? > >>>>>> > >>>>>> > >>>>>>> > >>>>>>> Thank you for any help. > >>>>>>> > >>>>>>> George > >>>>>>> > >>>>>>> > >>>>>>> 0(0) INFO: statistics manager successfully initialized > >>>>>>> 0(0) StateLess module - initializing > >>>>>>> 0(0) TM - initializing... > >>>>>>> 0(0) Maxfwd module- initializing > >>>>>>> 0(0) INFO:ul_init_locks: locks array size 512 > >>>>>>> 0(0) TextOPS - initializing > >>>>>>> 0(0) AVPops - initializing > >>>>>>> 0(0) permissions - initializing > >>>>>>> 0(0) WARNING: File not found: > >>>>>>> > >>>>>>> > >>>>>> /root/openser/config/permissions.allow > >>>>>> > >>>>>> > >>>>>>> 0(0) Default allow file > >>>>>>> > >>>>>>> > >>>>>> (/root/openser/config/permissions.allow) not > >>>>>> > >>>>>> > >>>>>>> found => empty rule set > >>>>>>> 0(0) WARNING: File not found: > >>>>>>> > >>>>>>> > >>>> /root/openser/config/permissions.deny > >>>> > >>>> > >>>>>>> 0(0) Default deny file > >>>>>>> > >>>>>>> > >>>> (/root/openser/config/permissions.deny) not > >>>> > >>>> > >>>>>>> found => empty rule set > >>>>>>> 0(0) AUTH module - initializing > >>>>>>> 0(0) AUTH_DB module - initializing > >>>>>>> 0(0) ACC - initializing > >>>>>>> 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> > >>>>>>> 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> > >>>>>>> 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> > >>>>>>> 0(0) INFO:textops:hname_fixup: using hdr type (31) > instead of > >>>>>>> <Remote-Party-ID> > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>>>> 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo > >>>>>>> > >>>>>>> > >>>>>> listener processes > >>>>>> > >>>>>> > >>>>>>> created > >>>>>>> 2(32093) pre_auth(): Credentials received are not > >>>>>>> > >> filled properly > >> > >>>>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>>>> > >>>>>>> > >>>>>> header in > >>>>>> > >>>>>> > >>>>>>> state 22 > >>>>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>>>> 1(32087) ERROR: bad header field > >>>>>>> 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing > >>>>>>> > >>>>>>> > >>>>>> MAX_FORWARD header > >>>>>> > >>>>>> > >>>>>>> failed! > >>>>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>>>> > >>>>>>> > >>>>>> header in > >>>>>> > >>>>>> > >>>>>>> state 22 > >>>>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>>>> 1(32087) ERROR: bad header field > >>>>>>> 1(32087) find_first_route: Error while parsing headers > >>>>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>>>> > >>>>>>> > >>>>>> header in > >>>>>> > >>>>>> > >>>>>>> state 22 > >>>>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>>>> 1(32087) ERROR: bad header field > >>>>>>> 1(32087) find_credentials(): Error while parsing headers > >>>>>>> 1(32087) pre_auth(): Error while looking for credentials > >>>>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>>>> > >>>>>>> > >>>>>> header in > >>>>>> > >>>>>> > >>>>>>> state 22 > >>>>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>>>> 1(32087) ERROR: bad header field > >>>>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>>>> > >>>>>>> > >>>>>> header in > >>>>>> > >>>>>> > >>>>>>> state 22 > >>>>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>>>> 1(32087) ERROR: bad header field > >>>>>>> 1(32087) ERROR: build_res_buf_from_sip_req: alas, > >>>>>>> > >> parse_headers > >> > >>>>>>> failed > >>>>>>> 1(32087) pre_auth(): Error while sending 400 reply > >>>>>>> INFO: started (pid: 32073) > >>>>>>> [root@ser1 sbin]# 1(32087) pre_auth(): Credentials > >>>>>>> > >>>>>>> > >>>>>> received are not > >>>>>> > >>>>>> > >>>>>>> filled properly > >>>>>>> 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of > >>>>>>> > >>>>>>> > >>>>>> memory block!) > >>>>>> > >>>>>> > >>>>>>> - aborting > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) ERROR: parse_via: invalid char <=> in state 201 > >>>>>>> 3(32096) error: parse_via_param > >>>>>>> 3(32096) ERROR: parse_via on: <SIP/2.0/UDP > >>>>>>> 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>>>> To: <sip:gdt@sip.i-call.gr> > >>>>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>>>> <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> > >>>>>>> CSeq: 4471 REGISTER > >>>>>>> User-Agent: Asterisk PBX > >>>>>>> Max-Forwards: 70 > >>>>>>> Expires: 120 > >>>>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>>>> Event: registration > >>>>>>> Content-Length: 0 > >>>>>>> > >>>>>>> 3(32096) ERROR: parse_via parse error, parsed so > >>>>>>> > >>>>>>> > >>>> far:<SIP/2.0/UDP > >>>> > >>>> > >>>>>>> 87.202.97.60:59945;branc=> > >>>>>>> 3(32096) ERROR: get_hdr_field: bad via > >>>>>>> 3(32096) ERROR: bad header field > >>>>>>> 3(32096) ERROR: parse_msg: message=<REGISTER > sip:sip.i-call.gr > >>>>>>> SIP/2.0 > >>>>>>> Via: SIP/2.0/UDP > 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>>>> To: <sip:gdt@sip.i-call.gr> > >>>>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>>>> CSeq: 4471 REGISTER > >>>>>>> User-Agent: Asterisk PBX > >>>>>>> Max-Forwards: 70 > >>>>>>> Expires: 120 > >>>>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>>>> Event: registration > >>>>>>> Content-Length: 0 > >>>>>>> > >>>>>>> 3(32096) ERROR: receive_msg: parse_msg failed > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>>>> 0(32073) child process 32133 exited by a signal 6 > >>>>>>> 0(32073) core was generated > >>>>>>> 0(32073) INFO: terminating due to SIGCHLD > >>>>>>> 1(32087) INFO: signal 15 received > >>>>>>> 2(32093) INFO: signal 15 received > >>>>>>> 3(32096) INFO: signal 15 received > >>>>>>> 5(32102) INFO: signal 15 received > >>>>>>> 6(32103) INFO: signal 15 received > >>>>>>> 13(32123) INFO: signal 15 received > >>>>>>> 14(32127) INFO: signal 15 received > >>>>>>> 15(32130) INFO: signal 15 received > >>>>>>> 1(32087) 4(32099) 7(32104) INFO: signal 15 received > >>>>>>> 8(32109) INFO: signal 15 received > >>>>>>> 9(32112) INFO: signal 15 received > >>>>>>> 10(32115) INFO: signal 15 received > >>>>>>> 11(32118) INFO: signal 15 received > >>>>>>> 12(32121) INFO: signal 15 received > >>>>>>> INFO: signal 15 received > >>>>>>> INFO: signal 15 received > >>>>>>> 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child > >>>>>>> > >>>>>>> > >>>> is already > >>>> > >>>> > >>>>>>> dead! > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> Disclaimer > >>>>>>> The information in this e-mail and any attachments is > >>>>>>> > >>>>>>> > >>>>>> confidential. It is intended solely for the attention and > >>>>>> > >>>>>> > >>>> use of the > >>>> > >>>> > >>>>>> named addressee(s). If you are not the intended recipient, > >>>>>> > >>>>>> > >>>> or person > >>>> > >>>> > >>>>>> responsible for delivering this information to the intended > >>>>>> recipient, please notify the sender immediately. Unless > >>>>>> > >>>>>> > >>>> you are the > >>>> > >>>> > >>>>>> intended recipient or his/her representative you are not > >>>>>> > >>>>>> > >>>> authorized > >>>> > >>>> > >>>>>> to, and must not, read, copy, distribute, use or retain > >>>>>> > >>>>>> > >>>> this message > >>>> > >>>> > >>>>>> or any part of it. E-mail transmission cannot be > >>>>>> > >> guaranteed to be > >> > >>>>>> secure or error-free as information could be intercepted, > >>>>>> > >>>>>> > >>>> corrupted, > >>>> > >>>> > >>>>>> lost, destroyed, arrive late or incomplete, or contain viruses. > >>>>>> > >>>>>> > >>>>>>> > >>>>>>> > >> > --------------------------------------------------------------------- > >> > >>>> > >>>> > >>>>>> - > >>>>>> > >>>>>> > >>>>>>> -- > >>>>>>> > >>>>>>> _______________________________________________ > >>>>>>> Users mailing list > >>>>>>> Users(a)openser.org > >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>>>> > >>>>>>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users > >>> > >>> > >>> > >> > > > > > >

2 1

RE: [Users] BUG: qm_free ?
by Papadopoulos Georgios 30 May '07

30 May '07

Solved! Thank you! However, before applying the patch, I increased private memory to 2MB and it did not crash. Then I set it back to 1MB, applied the patch and it worked. Now I have a ton of these messages: WARNING: script writer didn't release transaction What does it mean? This was not present in 1.1 and my config script has not changed (except for the necessary changes for the migration) Regards George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, May 30, 2007 2:18 PM > To: Papadopoulos Georgios > Cc: Klaus Darilion; users(a)openser.org > Subject: Re: [Users] BUG: qm_free ? > > Hi George, > > I managed to find the bug without the backtrace. I applied a > fix for development version (SVN trunk) and find attached a > patch for 1.2 - could you please give it a try and see if it > solves the bug? > > thanks and regards, > bogdan > > Papadopoulos Georgios wrote: > > I cannot find a core file anywhere. Where should it be? I can > > reproduce the problem very easily, so please give me more > instructions > > on how to get the coredump+backtrace that you need. > > > > thank you > > > > George > > > > > > > >> -----Original Message----- > >> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >> Sent: Tuesday, May 29, 2007 8:34 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] BUG: qm_free ? > >> > >> Maybe a bug? > >> > >> Can you provide a coredump+backtrace? I'm sure Bogdan or > Daniel will > >> take a look at it. > >> > >> regards > >> klaus > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hi Klaus, > >>> > >>> It crashes immediately, I have no time to do anything. I > >>> > >> tried again > >> > >>> and I increased the log level. So now it is process 692. > >>> > >> Here is part > >> > >>> of the output. I can send you the whole output if that helps. > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> 16(692) DEBUG: timer routine:5,tl=0xb61abdc4 next=0xb61adfd8, > >>> timeout=3000000 > >>> 16(692) DEBUG: retransmission_handler : request resending > >>> (t=0xb61abc78, REGISTER ... ) > >>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61abdc4 (6000000) > >>> 16(692) DEBUG: retransmission_handler : done > >>> 16(692) DEBUG: timer routine:5,tl=0xb61adfd8 next=(nil), > >>> timeout=3000000 > >>> 16(692) DEBUG: retransmission_handler : request resending > >>> (t=0xb61ade8c, REGISTER ... ) > >>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61adfd8 (6000000) > >>> 16(692) DEBUG: retransmission_handler : done > >>> 16(692) DEBUG: timer routine:0,tl=0xb6182810 next=0xb61845ac, > >>> timeout=3 > >>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>> (0xb61826a8) > >>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>> > >> test: branch=0, > >> > >>> last_recv=408, flags=0 > >>> 16(692) DEBUG:tm:t_should_relay_response: trying > DNS-based failover > >>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>> add_rule_to_list(377) > >>> 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb6278238 frag. > >>> 0xb6278220 (size=72) on 1 -th hit > >>> 16(692) DBG:add_list_to_head: adding to bl dns > 0xb6278238,0xb6278238 > >>> 16(692) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > >>> 16(692) DEBUG:tm:local_reply: local transaction completed > >>> 16(692) DEBUG: add_to_tail_of_timer[2]: 0xb61826f0 (8) > >>> 16(692) DEBUG: final_response_handler : done > >>> 16(692) DEBUG: timer routine:0,tl=0xb61845ac next=0xb6186c6c, > >>> timeout=3 > >>> 16(692) DEBUG: timer routine:0,tl=0xb6186c6c next=0xb6188d98, > >>> timeout=3 > >>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>> (0xb6186b04) > >>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>> > >> test: branch=0, > >> > >>> last_recv=408, flags=0 > >>> 16(692) DEBUG:tm:t_should_relay_response: trying > DNS-based failover > >>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>> add_rule_to_list(377) > >>> 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb62782b0 frag. > >>> 0xb6278298 (size=72) on 1 -th hit > >>> 16(692) DBG:add_list_to_head: adding to bl dns > 0xb62782b0,0xb62782b0 > >>> 16(692) qm_free(0x8151c40, 0xb62782b0), called from blacklists.c: > >>> rm_dups(440) > >>> 16(692) BUG: qm_free: bad pointer 0xb62782b0 (out of memory > >>> > >> block!) - > >> > >>> aborting > >>> > >>> > >>>> -----Original Message----- > >>>> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >>>> Sent: Tuesday, May 29, 2007 5:48 PM > >>>> To: Papadopoulos Georgios > >>>> Cc: users(a)openser.org > >>>> Subject: Re: [Users] BUG: qm_free ? > >>>> > >>>> The bug was in process 32133 > >>>> > >>>> There are no log statements from this process. What what this > >>>> process? > >>>> (use "openserctl ps" to see the PIDs of the processes) > >>>> > >>>> regards > >>>> klaus > >>>> > >>>> Papadopoulos Georgios wrote: > >>>> > >>>>> Hello all, > >>>>> > >>>>> I am having some trouble geting Openser 1.2 to work. It > >>>>> > >>>> crashes pretty > >>>> > >>>>> much immediately. I have tried it successfully on a test > >>>>> > >>>> system but as > >>>> > >>>>> soon as I start it in the production environment, it > >>>>> > >>>> crashes. Any ideas? > >>>> > >>>>> > >>>>> Thank you for any help. > >>>>> > >>>>> George > >>>>> > >>>>> > >>>>> 0(0) INFO: statistics manager successfully initialized > >>>>> 0(0) StateLess module - initializing > >>>>> 0(0) TM - initializing... > >>>>> 0(0) Maxfwd module- initializing > >>>>> 0(0) INFO:ul_init_locks: locks array size 512 > >>>>> 0(0) TextOPS - initializing > >>>>> 0(0) AVPops - initializing > >>>>> 0(0) permissions - initializing > >>>>> 0(0) WARNING: File not found: > >>>>> > >>>> /root/openser/config/permissions.allow > >>>> > >>>>> 0(0) Default allow file > >>>>> > >>>> (/root/openser/config/permissions.allow) not > >>>> > >>>>> found => empty rule set > >>>>> 0(0) WARNING: File not found: > >>>>> > >> /root/openser/config/permissions.deny > >> > >>>>> 0(0) Default deny file > >>>>> > >> (/root/openser/config/permissions.deny) not > >> > >>>>> found => empty rule set > >>>>> 0(0) AUTH module - initializing > >>>>> 0(0) AUTH_DB module - initializing > >>>>> 0(0) ACC - initializing > >>>>> 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> > >>>>> 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> > >>>>> 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> > >>>>> 0(0) INFO:textops:hname_fixup: using hdr type (31) instead of > >>>>> <Remote-Party-ID> > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo > >>>>> > >>>> listener processes > >>>> > >>>>> created > >>>>> 2(32093) pre_auth(): Credentials received are not > filled properly > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing > >>>>> > >>>> MAX_FORWARD header > >>>> > >>>>> failed! > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) find_first_route: Error while parsing headers > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) find_credentials(): Error while parsing headers > >>>>> 1(32087) pre_auth(): Error while looking for credentials > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR: build_res_buf_from_sip_req: alas, > parse_headers > >>>>> failed > >>>>> 1(32087) pre_auth(): Error while sending 400 reply > >>>>> INFO: started (pid: 32073) > >>>>> [root@ser1 sbin]# 1(32087) pre_auth(): Credentials > >>>>> > >>>> received are not > >>>> > >>>>> filled properly > >>>>> 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of > >>>>> > >>>> memory block!) > >>>> > >>>>> - aborting > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) ERROR: parse_via: invalid char <=> in state 201 > >>>>> 3(32096) error: parse_via_param > >>>>> 3(32096) ERROR: parse_via on: <SIP/2.0/UDP > >>>>> 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>> To: <sip:gdt@sip.i-call.gr> > >>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>> <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> > >>>>> CSeq: 4471 REGISTER > >>>>> User-Agent: Asterisk PBX > >>>>> Max-Forwards: 70 > >>>>> Expires: 120 > >>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>> Event: registration > >>>>> Content-Length: 0 > >>>>> > >>>>> 3(32096) ERROR: parse_via parse error, parsed so > >>>>> > >> far:<SIP/2.0/UDP > >> > >>>>> 87.202.97.60:59945;branc=> > >>>>> 3(32096) ERROR: get_hdr_field: bad via > >>>>> 3(32096) ERROR: bad header field > >>>>> 3(32096) ERROR: parse_msg: message=<REGISTER sip:sip.i-call.gr > >>>>> SIP/2.0 > >>>>> Via: SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>> To: <sip:gdt@sip.i-call.gr> > >>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>> CSeq: 4471 REGISTER > >>>>> User-Agent: Asterisk PBX > >>>>> Max-Forwards: 70 > >>>>> Expires: 120 > >>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>> Event: registration > >>>>> Content-Length: 0 > >>>>> > >>>>> 3(32096) ERROR: receive_msg: parse_msg failed > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 0(32073) child process 32133 exited by a signal 6 > >>>>> 0(32073) core was generated > >>>>> 0(32073) INFO: terminating due to SIGCHLD > >>>>> 1(32087) INFO: signal 15 received > >>>>> 2(32093) INFO: signal 15 received > >>>>> 3(32096) INFO: signal 15 received > >>>>> 5(32102) INFO: signal 15 received > >>>>> 6(32103) INFO: signal 15 received > >>>>> 13(32123) INFO: signal 15 received > >>>>> 14(32127) INFO: signal 15 received > >>>>> 15(32130) INFO: signal 15 received > >>>>> 1(32087) 4(32099) 7(32104) INFO: signal 15 received > >>>>> 8(32109) INFO: signal 15 received > >>>>> 9(32112) INFO: signal 15 received > >>>>> 10(32115) INFO: signal 15 received > >>>>> 11(32118) INFO: signal 15 received > >>>>> 12(32121) INFO: signal 15 received > >>>>> INFO: signal 15 received > >>>>> INFO: signal 15 received > >>>>> 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child > >>>>> > >> is already > >> > >>>>> dead! > >>>>> > >>>>> > >>>>> > >>>>> Disclaimer > >>>>> The information in this e-mail and any attachments is > >>>>> > >>>> confidential. It is intended solely for the attention and > >>>> > >> use of the > >> > >>>> named addressee(s). If you are not the intended recipient, > >>>> > >> or person > >> > >>>> responsible for delivering this information to the intended > >>>> recipient, please notify the sender immediately. Unless > >>>> > >> you are the > >> > >>>> intended recipient or his/her representative you are not > >>>> > >> authorized > >> > >>>> to, and must not, read, copy, distribute, use or retain > >>>> > >> this message > >> > >>>> or any part of it. E-mail transmission cannot be > guaranteed to be > >>>> secure or error-free as information could be intercepted, > >>>> > >> corrupted, > >> > >>>> lost, destroyed, arrive late or incomplete, or contain viruses. > >>>> > >>>>> > >>>>> > >>>>> > >> > --------------------------------------------------------------------- > >> > >>>> - > >>>> > >>>>> -- > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)openser.org > >>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>> > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > >

3 2

[Users] nat problem
by Kro Kamil 30 May '07

30 May '07

Hi, I have a registered Linksys, behind a USR 9108 modem. I can detect its nat, call Linksys and speak normally, but when I try to make a call from Linksys, call breaks down right after call setup. Linksys sends the via and contact ip's as 0.0.0.0 in invite messages and fix_nated_contact does not work. This is possibly because of the modem but, there are no extra configurations that can cause this in the modem. I will be glad if you can direct me with your experiences. thank you... _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanýzý bir araya getirin; haberler, spor, hava durumu ve çok daha fazlasý. http://www.live.com/getstarted

6 8

[Users] AVP.pm
by Fero Vlkolinsky 30 May '07

30 May '07

Hi all, I just installed 1.2.1 on my test server and wanted to try out some of the new features. The perl module is exactly what I need. So I wrote a pl script to test avp manipulation. Now the problem is that there is no AVP.pm anywhere in the tarball. Can I get one somewhere? Thanks, Fero --- reklama ----------------------------------------------------- Si muž? A chceš porozumieť ženám? http://www.lesk.sk

3 2

[Serusers] SER rel 2_0_0 - Errors in msilo module
by tzieleniewski 30 May '07

30 May '07

Hi, When I try to store message in the database I get the following errors in the syslog (There are no messages stored in the database): May 14 15:26:40 sen ser[5853]: ERROR: parse_uri: uri too short: <> (0) May 14 15:26:40 sen ser[5853]: ERROR: uri2dst: bad_uri: May 14 15:26:40 sen ser[5853]: t_uac: no socket found And here is ser.cfg: xlog("L_INFO", " route[INBOUND]: (m_store (0, )) \n"); if (m_store ("0", "")) { xlog("L_INFO", " route[INBOUND]: offline message stored \n"); xlog("L_INFO", " route[INBOUND]: (!t_reply(202,Accepted) \n"); if (!t_reply("202", "Accepted")) { xlog("L_INFO", " route[INBOUND]: sl_reply_error() \n"); sl_reply_error(); } } the error appears in the following place accordin to my logging: May 14 15:26:40 sen ser[5853]: route[INBOUND]: (m_store (0, )) May 14 15:26:40 sen ser[5853]: ERROR: parse_uri: uri too short: <> (0) May 14 15:26:40 sen ser[5853]: ERROR: uri2dst: bad_uri: May 14 15:26:40 sen ser[5853]: t_uac: no socket found May 14 15:26:40 sen ser[5853]: route[INBOUND]: offline message stored May 14 15:26:40 sen ser[5853]: route[INBOUND]: (!t_reply(202,Accepted) May 14 15:26:40 sen ser[5853]: route[INBOUND]: drop <MESSAGE> <sip:hulk@voip.rd.touk.pl> Below is the Message transaction sent through SER: U 2007/05/14 15:26:40.746835 192.168.0.165:7089 -> 192.168.0.165:5060 MESSAGE sip:hulk@voip.rd.touk.pl SIP/2.0. CSeq: 64 MESSAGE. Via: SIP/2.0/UDP 192.168.0.165:7089;branch=z9hG4bK24d4276b-8c00-dc11-982b-00148539376e;rport. From: "Sen" <sip:sen@voip.rd.touk.pl>;tag=a0d0276b-8c00-dc11-982b-00148539376e. Call-ID: 32c6276b-8c00-dc11-982b-00148539376e@sen. To: "hulk@" <sip:hulk@voip.rd.touk.pl>. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE. Content-Type: text/plain;charset=UTF-8. Content-Length: 12. Max-Forwards: 70. . test message # U 2007/05/14 15:26:40.790793 192.168.0.165:5060 -> 192.168.0.165:7089 SIP/2.0 202 Accepted. CSeq: 64 MESSAGE. Via: SIP/2.0/UDP 192.168.0.165:7089;branch=z9hG4bK24d4276b-8c00-dc11-982b-00148539376e;rport=7089. From: "Sen" <sip:sen@voip.rd.touk.pl>;tag=a0d0276b-8c00-dc11-982b-00148539376e. Call-ID: 32c6276b-8c00-dc11-982b-00148539376e@sen. To: "hulk@" <sip:hulk@voip.rd.touk.pl>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-fd05. Server: Sip EXpress router (2.0.0-rc1 (x86_64/linux)). Content-Length: 0. Warning: 392 192.168.0.165:5060 "Noisy feedback tells: pid=5853 req_src_ip=192.168.0.165 req_src_port=7089 in_uri=sip:hulk@voip.rd.touk.pl out_uri=sip:hulk@voip.rd.touk.pl via_cnt==1". . Thanks for any feedback! Cheers Tomasz

3 13

RE: [Users] BUG: qm_free ?
by Papadopoulos Georgios 30 May '07

30 May '07

I cannot find a core file anywhere. Where should it be? I can reproduce the problem very easily, so please give me more instructions on how to get the coredump+backtrace that you need. thank you George > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Tuesday, May 29, 2007 8:34 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] BUG: qm_free ? > > Maybe a bug? > > Can you provide a coredump+backtrace? I'm sure Bogdan or > Daniel will take a look at it. > > regards > klaus > > Papadopoulos Georgios wrote: > > Hi Klaus, > > > > It crashes immediately, I have no time to do anything. I > tried again > > and I increased the log level. So now it is process 692. > Here is part > > of the output. I can send you the whole output if that helps. > > > > thank you > > > > George > > > > > > 16(692) DEBUG: timer routine:5,tl=0xb61abdc4 next=0xb61adfd8, > > timeout=3000000 > > 16(692) DEBUG: retransmission_handler : request resending > > (t=0xb61abc78, REGISTER ... ) > > 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > > 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61abdc4 (6000000) > > 16(692) DEBUG: retransmission_handler : done > > 16(692) DEBUG: timer routine:5,tl=0xb61adfd8 next=(nil), > > timeout=3000000 > > 16(692) DEBUG: retransmission_handler : request resending > > (t=0xb61ade8c, REGISTER ... ) > > 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > > 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61adfd8 (6000000) > > 16(692) DEBUG: retransmission_handler : done > > 16(692) DEBUG: timer routine:0,tl=0xb6182810 next=0xb61845ac, > > timeout=3 > > 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > > (0xb61826a8) > > 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > > 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > > 16(692) DEBUG:tm:t_should_relay_response: dns-failover > test: branch=0, > > last_recv=408, flags=0 > > 16(692) DEBUG:tm:t_should_relay_response: trying DNS-based failover > > 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > > add_rule_to_list(377) > > 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb6278238 frag. > > 0xb6278220 (size=72) on 1 -th hit > > 16(692) DBG:add_list_to_head: adding to bl dns 0xb6278238,0xb6278238 > > 16(692) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > > 16(692) DEBUG:tm:local_reply: local transaction completed > > 16(692) DEBUG: add_to_tail_of_timer[2]: 0xb61826f0 (8) > > 16(692) DEBUG: final_response_handler : done > > 16(692) DEBUG: timer routine:0,tl=0xb61845ac next=0xb6186c6c, > > timeout=3 > > 16(692) DEBUG: timer routine:0,tl=0xb6186c6c next=0xb6188d98, > > timeout=3 > > 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > > (0xb6186b04) > > 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > > 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > > 16(692) DEBUG:tm:t_should_relay_response: dns-failover > test: branch=0, > > last_recv=408, flags=0 > > 16(692) DEBUG:tm:t_should_relay_response: trying DNS-based failover > > 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > > add_rule_to_list(377) > > 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb62782b0 frag. > > 0xb6278298 (size=72) on 1 -th hit > > 16(692) DBG:add_list_to_head: adding to bl dns 0xb62782b0,0xb62782b0 > > 16(692) qm_free(0x8151c40, 0xb62782b0), called from blacklists.c: > > rm_dups(440) > > 16(692) BUG: qm_free: bad pointer 0xb62782b0 (out of memory > block!) - > > aborting > > > >> -----Original Message----- > >> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >> Sent: Tuesday, May 29, 2007 5:48 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] BUG: qm_free ? > >> > >> The bug was in process 32133 > >> > >> There are no log statements from this process. What what this > >> process? > >> (use "openserctl ps" to see the PIDs of the processes) > >> > >> regards > >> klaus > >> > >> Papadopoulos Georgios wrote: > >>> Hello all, > >>> > >>> I am having some trouble geting Openser 1.2 to work. It > >> crashes pretty > >>> much immediately. I have tried it successfully on a test > >> system but as > >>> soon as I start it in the production environment, it > >> crashes. Any ideas? > >>> > >>> Thank you for any help. > >>> > >>> George > >>> > >>> > >>> 0(0) INFO: statistics manager successfully initialized > >>> 0(0) StateLess module - initializing > >>> 0(0) TM - initializing... > >>> 0(0) Maxfwd module- initializing > >>> 0(0) INFO:ul_init_locks: locks array size 512 > >>> 0(0) TextOPS - initializing > >>> 0(0) AVPops - initializing > >>> 0(0) permissions - initializing > >>> 0(0) WARNING: File not found: > >> /root/openser/config/permissions.allow > >>> 0(0) Default allow file > >> (/root/openser/config/permissions.allow) not > >>> found => empty rule set > >>> 0(0) WARNING: File not found: > /root/openser/config/permissions.deny > >>> 0(0) Default deny file > (/root/openser/config/permissions.deny) not > >>> found => empty rule set > >>> 0(0) AUTH module - initializing > >>> 0(0) AUTH_DB module - initializing > >>> 0(0) ACC - initializing > >>> 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> > >>> 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> > >>> 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> > >>> 0(0) INFO:textops:hname_fixup: using hdr type (31) instead of > >>> <Remote-Party-ID> > >>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>> 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo > >> listener processes > >>> created > >>> 2(32093) pre_auth(): Credentials received are not filled properly > >>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >> header in > >>> state 22 > >>> 1(32087) ERROR: get_hdr_field: bad to header > >>> 1(32087) ERROR: bad header field > >>> 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing > >> MAX_FORWARD header > >>> failed! > >>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >> header in > >>> state 22 > >>> 1(32087) ERROR: get_hdr_field: bad to header > >>> 1(32087) ERROR: bad header field > >>> 1(32087) find_first_route: Error while parsing headers > >>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >> header in > >>> state 22 > >>> 1(32087) ERROR: get_hdr_field: bad to header > >>> 1(32087) ERROR: bad header field > >>> 1(32087) find_credentials(): Error while parsing headers > >>> 1(32087) pre_auth(): Error while looking for credentials > >>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >> header in > >>> state 22 > >>> 1(32087) ERROR: get_hdr_field: bad to header > >>> 1(32087) ERROR: bad header field > >>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >> header in > >>> state 22 > >>> 1(32087) ERROR: get_hdr_field: bad to header > >>> 1(32087) ERROR: bad header field > >>> 1(32087) ERROR: build_res_buf_from_sip_req: alas, parse_headers > >>> failed > >>> 1(32087) pre_auth(): Error while sending 400 reply > >>> INFO: started (pid: 32073) > >>> [root@ser1 sbin]# 1(32087) pre_auth(): Credentials > >> received are not > >>> filled properly > >>> 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of > >> memory block!) > >>> - aborting > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 3(32096) ERROR: parse_via: invalid char <=> in state 201 > >>> 3(32096) error: parse_via_param > >>> 3(32096) ERROR: parse_via on: <SIP/2.0/UDP > >>> 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>> To: <sip:gdt@sip.i-call.gr> > >>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>> <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> > >>> CSeq: 4471 REGISTER > >>> User-Agent: Asterisk PBX > >>> Max-Forwards: 70 > >>> Expires: 120 > >>> Contact: <sip:s@87.202.97.60:59945> > >>> Event: registration > >>> Content-Length: 0 > >>> > >>> 3(32096) ERROR: parse_via parse error, parsed so > far:<SIP/2.0/UDP > >>> 87.202.97.60:59945;branc=> > >>> 3(32096) ERROR: get_hdr_field: bad via > >>> 3(32096) ERROR: bad header field > >>> 3(32096) ERROR: parse_msg: message=<REGISTER sip:sip.i-call.gr > >>> SIP/2.0 > >>> Via: SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>> To: <sip:gdt@sip.i-call.gr> > >>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>> CSeq: 4471 REGISTER > >>> User-Agent: Asterisk PBX > >>> Max-Forwards: 70 > >>> Expires: 120 > >>> Contact: <sip:s@87.202.97.60:59945> > >>> Event: registration > >>> Content-Length: 0 > >>> > >>> 3(32096) ERROR: receive_msg: parse_msg failed > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 3(32096) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 1(32087) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 2(32093) WARNING: script writer didn't release transaction > >>> 0(32073) child process 32133 exited by a signal 6 > >>> 0(32073) core was generated > >>> 0(32073) INFO: terminating due to SIGCHLD > >>> 1(32087) INFO: signal 15 received > >>> 2(32093) INFO: signal 15 received > >>> 3(32096) INFO: signal 15 received > >>> 5(32102) INFO: signal 15 received > >>> 6(32103) INFO: signal 15 received > >>> 13(32123) INFO: signal 15 received > >>> 14(32127) INFO: signal 15 received > >>> 15(32130) INFO: signal 15 received > >>> 1(32087) 4(32099) 7(32104) INFO: signal 15 received > >>> 8(32109) INFO: signal 15 received > >>> 9(32112) INFO: signal 15 received > >>> 10(32115) INFO: signal 15 received > >>> 11(32118) INFO: signal 15 received > >>> 12(32121) INFO: signal 15 received > >>> INFO: signal 15 received > >>> INFO: signal 15 received > >>> 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child > is already > >>> dead! > >>> > >>> > >>> > >>> Disclaimer > >>> The information in this e-mail and any attachments is > >> confidential. It is intended solely for the attention and > use of the > >> named addressee(s). If you are not the intended recipient, > or person > >> responsible for delivering this information to the intended > >> recipient, please notify the sender immediately. Unless > you are the > >> intended recipient or his/her representative you are not > authorized > >> to, and must not, read, copy, distribute, use or retain > this message > >> or any part of it. E-mail transmission cannot be guaranteed to be > >> secure or error-free as information could be intercepted, > corrupted, > >> lost, destroyed, arrive late or incomplete, or contain viruses. > >>> > >>> > >>> > >>> > >> > --------------------------------------------------------------------- > >> - > >>> -- > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users >

2 2

RE: [Users] BUG: qm_free ?
by Papadopoulos Georgios 30 May '07

30 May '07

Hi Bogdan, I found /core.692. Strangely "core was generated" is not in the output. Do you want me to send the core file? thank you, George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, May 30, 2007 2:03 PM > To: Papadopoulos Georgios > Cc: Klaus Darilion; users(a)openser.org > Subject: Re: [Users] BUG: qm_free ? > > HI George, > > first check if the you get the message "core was generated" - > if yes, you should find the core file in the working > directory of openser; if non provided from CLI, check in the > root (/) directory. > > regards, > bogdan > > Papadopoulos Georgios wrote: > > I cannot find a core file anywhere. Where should it be? I can > > reproduce the problem very easily, so please give me more > instructions > > on how to get the coredump+backtrace that you need. > > > > thank you > > > > George > > > > > > > >> -----Original Message----- > >> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >> Sent: Tuesday, May 29, 2007 8:34 PM > >> To: Papadopoulos Georgios > >> Cc: users(a)openser.org > >> Subject: Re: [Users] BUG: qm_free ? > >> > >> Maybe a bug? > >> > >> Can you provide a coredump+backtrace? I'm sure Bogdan or > Daniel will > >> take a look at it. > >> > >> regards > >> klaus > >> > >> Papadopoulos Georgios wrote: > >> > >>> Hi Klaus, > >>> > >>> It crashes immediately, I have no time to do anything. I > >>> > >> tried again > >> > >>> and I increased the log level. So now it is process 692. > >>> > >> Here is part > >> > >>> of the output. I can send you the whole output if that helps. > >>> > >>> thank you > >>> > >>> George > >>> > >>> > >>> 16(692) DEBUG: timer routine:5,tl=0xb61abdc4 next=0xb61adfd8, > >>> timeout=3000000 > >>> 16(692) DEBUG: retransmission_handler : request resending > >>> (t=0xb61abc78, REGISTER ... ) > >>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61abdc4 (6000000) > >>> 16(692) DEBUG: retransmission_handler : done > >>> 16(692) DEBUG: timer routine:5,tl=0xb61adfd8 next=(nil), > >>> timeout=3000000 > >>> 16(692) DEBUG: retransmission_handler : request resending > >>> (t=0xb61ade8c, REGISTER ... ) > >>> 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 > >>> 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61adfd8 (6000000) > >>> 16(692) DEBUG: retransmission_handler : done > >>> 16(692) DEBUG: timer routine:0,tl=0xb6182810 next=0xb61845ac, > >>> timeout=3 > >>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>> (0xb61826a8) > >>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>> > >> test: branch=0, > >> > >>> last_recv=408, flags=0 > >>> 16(692) DEBUG:tm:t_should_relay_response: trying > DNS-based failover > >>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>> add_rule_to_list(377) > >>> 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb6278238 frag. > >>> 0xb6278220 (size=72) on 1 -th hit > >>> 16(692) DBG:add_list_to_head: adding to bl dns > 0xb6278238,0xb6278238 > >>> 16(692) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > >>> 16(692) DEBUG:tm:local_reply: local transaction completed > >>> 16(692) DEBUG: add_to_tail_of_timer[2]: 0xb61826f0 (8) > >>> 16(692) DEBUG: final_response_handler : done > >>> 16(692) DEBUG: timer routine:0,tl=0xb61845ac next=0xb6186c6c, > >>> timeout=3 > >>> 16(692) DEBUG: timer routine:0,tl=0xb6186c6c next=0xb6188d98, > >>> timeout=3 > >>> 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL > >>> (0xb6186b04) > >>> 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > >>> 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > >>> 16(692) DEBUG:tm:t_should_relay_response: dns-failover > >>> > >> test: branch=0, > >> > >>> last_recv=408, flags=0 > >>> 16(692) DEBUG:tm:t_should_relay_response: trying > DNS-based failover > >>> 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: > >>> add_rule_to_list(377) > >>> 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb62782b0 frag. > >>> 0xb6278298 (size=72) on 1 -th hit > >>> 16(692) DBG:add_list_to_head: adding to bl dns > 0xb62782b0,0xb62782b0 > >>> 16(692) qm_free(0x8151c40, 0xb62782b0), called from blacklists.c: > >>> rm_dups(440) > >>> 16(692) BUG: qm_free: bad pointer 0xb62782b0 (out of memory > >>> > >> block!) - > >> > >>> aborting > >>> > >>> > >>>> -----Original Message----- > >>>> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > >>>> Sent: Tuesday, May 29, 2007 5:48 PM > >>>> To: Papadopoulos Georgios > >>>> Cc: users(a)openser.org > >>>> Subject: Re: [Users] BUG: qm_free ? > >>>> > >>>> The bug was in process 32133 > >>>> > >>>> There are no log statements from this process. What what this > >>>> process? > >>>> (use "openserctl ps" to see the PIDs of the processes) > >>>> > >>>> regards > >>>> klaus > >>>> > >>>> Papadopoulos Georgios wrote: > >>>> > >>>>> Hello all, > >>>>> > >>>>> I am having some trouble geting Openser 1.2 to work. It > >>>>> > >>>> crashes pretty > >>>> > >>>>> much immediately. I have tried it successfully on a test > >>>>> > >>>> system but as > >>>> > >>>>> soon as I start it in the production environment, it > >>>>> > >>>> crashes. Any ideas? > >>>> > >>>>> > >>>>> Thank you for any help. > >>>>> > >>>>> George > >>>>> > >>>>> > >>>>> 0(0) INFO: statistics manager successfully initialized > >>>>> 0(0) StateLess module - initializing > >>>>> 0(0) TM - initializing... > >>>>> 0(0) Maxfwd module- initializing > >>>>> 0(0) INFO:ul_init_locks: locks array size 512 > >>>>> 0(0) TextOPS - initializing > >>>>> 0(0) AVPops - initializing > >>>>> 0(0) permissions - initializing > >>>>> 0(0) WARNING: File not found: > >>>>> > >>>> /root/openser/config/permissions.allow > >>>> > >>>>> 0(0) Default allow file > >>>>> > >>>> (/root/openser/config/permissions.allow) not > >>>> > >>>>> found => empty rule set > >>>>> 0(0) WARNING: File not found: > >>>>> > >> /root/openser/config/permissions.deny > >> > >>>>> 0(0) Default deny file > >>>>> > >> (/root/openser/config/permissions.deny) not > >> > >>>>> found => empty rule set > >>>>> 0(0) AUTH module - initializing > >>>>> 0(0) AUTH_DB module - initializing > >>>>> 0(0) ACC - initializing > >>>>> 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> > >>>>> 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> > >>>>> 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> > >>>>> 0(0) INFO:textops:hname_fixup: using hdr type (31) instead of > >>>>> <Remote-Party-ID> > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > >>>>> 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > >>>>> 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo > >>>>> > >>>> listener processes > >>>> > >>>>> created > >>>>> 2(32093) pre_auth(): Credentials received are not > filled properly > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing > >>>>> > >>>> MAX_FORWARD header > >>>> > >>>>> failed! > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) find_first_route: Error while parsing headers > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) find_credentials(): Error while parsing headers > >>>>> 1(32087) pre_auth(): Error while looking for credentials > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > >>>>> 1(32087) ERROR: parse_to: invalid To - unexpected end of > >>>>> > >>>> header in > >>>> > >>>>> state 22 > >>>>> 1(32087) ERROR: get_hdr_field: bad to header > >>>>> 1(32087) ERROR: bad header field > >>>>> 1(32087) ERROR: build_res_buf_from_sip_req: alas, > parse_headers > >>>>> failed > >>>>> 1(32087) pre_auth(): Error while sending 400 reply > >>>>> INFO: started (pid: 32073) > >>>>> [root@ser1 sbin]# 1(32087) pre_auth(): Credentials > >>>>> > >>>> received are not > >>>> > >>>>> filled properly > >>>>> 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of > >>>>> > >>>> memory block!) > >>>> > >>>>> - aborting > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) ERROR: parse_via: invalid char <=> in state 201 > >>>>> 3(32096) error: parse_via_param > >>>>> 3(32096) ERROR: parse_via on: <SIP/2.0/UDP > >>>>> 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>> To: <sip:gdt@sip.i-call.gr> > >>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>> <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> > >>>>> CSeq: 4471 REGISTER > >>>>> User-Agent: Asterisk PBX > >>>>> Max-Forwards: 70 > >>>>> Expires: 120 > >>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>> Event: registration > >>>>> Content-Length: 0 > >>>>> > >>>>> 3(32096) ERROR: parse_via parse error, parsed so > >>>>> > >> far:<SIP/2.0/UDP > >> > >>>>> 87.202.97.60:59945;branc=> > >>>>> 3(32096) ERROR: get_hdr_field: bad via > >>>>> 3(32096) ERROR: bad header field > >>>>> 3(32096) ERROR: parse_msg: message=<REGISTER sip:sip.i-call.gr > >>>>> SIP/2.0 > >>>>> Via: SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > >>>>> From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > >>>>> To: <sip:gdt@sip.i-call.gr> > >>>>> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > >>>>> CSeq: 4471 REGISTER > >>>>> User-Agent: Asterisk PBX > >>>>> Max-Forwards: 70 > >>>>> Expires: 120 > >>>>> Contact: <sip:s@87.202.97.60:59945> > >>>>> Event: registration > >>>>> Content-Length: 0 > >>>>> > >>>>> 3(32096) ERROR: receive_msg: parse_msg failed > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 3(32096) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 1(32087) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 2(32093) WARNING: script writer didn't release transaction > >>>>> 0(32073) child process 32133 exited by a signal 6 > >>>>> 0(32073) core was generated > >>>>> 0(32073) INFO: terminating due to SIGCHLD > >>>>> 1(32087) INFO: signal 15 received > >>>>> 2(32093) INFO: signal 15 received > >>>>> 3(32096) INFO: signal 15 received > >>>>> 5(32102) INFO: signal 15 received > >>>>> 6(32103) INFO: signal 15 received > >>>>> 13(32123) INFO: signal 15 received > >>>>> 14(32127) INFO: signal 15 received > >>>>> 15(32130) INFO: signal 15 received > >>>>> 1(32087) 4(32099) 7(32104) INFO: signal 15 received > >>>>> 8(32109) INFO: signal 15 received > >>>>> 9(32112) INFO: signal 15 received > >>>>> 10(32115) INFO: signal 15 received > >>>>> 11(32118) INFO: signal 15 received > >>>>> 12(32121) INFO: signal 15 received > >>>>> INFO: signal 15 received > >>>>> INFO: signal 15 received > >>>>> 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child > >>>>> > >> is already > >> > >>>>> dead! > >>>>> > >>>>> > >>>>> > >>>>> Disclaimer > >>>>> The information in this e-mail and any attachments is > >>>>> > >>>> confidential. It is intended solely for the attention and > >>>> > >> use of the > >> > >>>> named addressee(s). If you are not the intended recipient, > >>>> > >> or person > >> > >>>> responsible for delivering this information to the intended > >>>> recipient, please notify the sender immediately. Unless > >>>> > >> you are the > >> > >>>> intended recipient or his/her representative you are not > >>>> > >> authorized > >> > >>>> to, and must not, read, copy, distribute, use or retain > >>>> > >> this message > >> > >>>> or any part of it. E-mail transmission cannot be > guaranteed to be > >>>> secure or error-free as information could be intercepted, > >>>> > >> corrupted, > >> > >>>> lost, destroyed, arrive late or incomplete, or contain viruses. > >>>> > >>>>> > >>>>> > >>>>> > >> > --------------------------------------------------------------------- > >> > >>>> - > >>>> > >>>>> -- > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)openser.org > >>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>> > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > >

1 0

[Users] CPL module out of date?
by caco.vera＠unete.cl 30 May '07

30 May '07

Hi' All I'm have installed Openser 1.2.1 and CPL module. I'm having problems to set some parameters for CPL module: The following parameters can be loaded: loadmodule "cpl-c.so" modparam("cpl-c","db_url","mysql://openser:openserrw@localhost/openser") modparam("cpl-c","cpl_table","cpl") modparam("cpl-c","cpl_dtd_file","/usr/local/etc/openser/cpl-06.dtd") modparam("cpl-c","log_dir","/var/log/openser/cpl") modparam("cpl-c","proxy_recurse",2) modparam("cpl-c","proxy_route",0) modparam("cpl-c","lookup_domain","location") These parameters are showing an error "No module matching cpl_c found" #modparam("cpl_c","username_column","username") #modparam("cpl_c","domain_column","domain") #modparam("cpl_c","cpl_xml_column","cpl_xml") #modparam("cpl_c","cpl_bin_column","cpl_bin") #modparam("cpl_c","case_sensitive",0) #modparam("cpl_c","lookup_append_branches",1) #modparam("cpl_c","use_domain",0) LOG MESSAGES: 0(4660) loading module /usr/local/lib/openser/modules/cpl-c.so 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <db_url> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <cpl_table> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <cpl_dtd_file> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <log_dir> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <proxy_recurse> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <proxy_route> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: cpl-c matches module cpl-c 0(4660) set_mod_param_regex: found <lookup_domain> in module cpl-c [/usr/local/lib/openser/modules/] 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (105,18-19): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (106,18-19): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (107,18-19): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (108,18-19): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (109,17-18): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (110,17-18): Can't set module parameter 0(4660) set_mod_param_regex: No module matching cpl_c found | 0(4660) parse error (111,17-18): Can't set module parameter Is CPL module out of dated? somebody has an example of openser configuration file in order ti work with CPL Module? Thanks in advance for your help. Caco.

3 3

[Serusers] First steps understanding ser.cfg
by J. Alberto 30 May '07

30 May '07

Hello, I installed SER a few days ago and I am following the document "SER - Getting Started" to familiarize with SER and understand how it works. I've managed to run the first "hello world" application as well as including authentication through mysql. However, understanding ser.cfg has become more difficult than I thought. To help understand the main route script, I have inserted into the ser.cfg script several calls to xlog function to display information. Although the information displayed is very useful, I don't understand the result of the following sentence: if (uri!=myself) { ..... } I mean, uri is the request uri, for intance, sip:1001@192.168.1.10, and 'myself' is, according to the document, the sip proxy itself, so it can't be the same as uri. However, when the main route processes an INVITE message, the 'if' statement seems to return false because it doesn't execute the instrucion inside the 'if' block. And that would mean that uri is equal to 'myself'. I am sure that I don't undertand what the value of 'myself' is...... Thanks in advance

2 1

[Users] how to make IPv6-to-PSTN work?
by Lee Jacky 30 May '07

30 May '07

how to make IPv6-to-PSTN work? now I am using openser1.1 and rtpproxy. the IPV4 to IPV4/IPV6 works fine. But when I use IPV6 client call PSTN phone,I cann't hear from the PSTN phone. what is wrong? did any one ever meet this problem? thank you for any advice. ___________________________________________________________ 雅虎免费邮箱-3.5G容量，20M附件 http://cn.mail.yahoo.com/

1 0

RE: [Users] BUG: qm_free ?
by Papadopoulos Georgios 29 May '07

29 May '07

Hi Klaus, It crashes immediately, I have no time to do anything. I tried again and I increased the log level. So now it is process 692. Here is part of the output. I can send you the whole output if that helps. thank you George 16(692) DEBUG: timer routine:5,tl=0xb61abdc4 next=0xb61adfd8, timeout=3000000 16(692) DEBUG: retransmission_handler : request resending (t=0xb61abc78, REGISTER ... ) 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61abdc4 (6000000) 16(692) DEBUG: retransmission_handler : done 16(692) DEBUG: timer routine:5,tl=0xb61adfd8 next=(nil), timeout=3000000 16(692) DEBUG: retransmission_handler : request resending (t=0xb61ade8c, REGISTER ... ) 16(692) DEBUG:tm:set_timer: relative timeout is 3000000 16(692) DEBUG: add_to_tail_of_timer[6]: 0xb61adfd8 (6000000) 16(692) DEBUG: retransmission_handler : done 16(692) DEBUG: timer routine:0,tl=0xb6182810 next=0xb61845ac, timeout=3 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb61826a8) 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 16(692) DEBUG:tm:t_should_relay_response: dns-failover test: branch=0, last_recv=408, flags=0 16(692) DEBUG:tm:t_should_relay_response: trying DNS-based failover 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: add_rule_to_list(377) 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb6278238 frag. 0xb6278220 (size=72) on 1 -th hit 16(692) DBG:add_list_to_head: adding to bl dns 0xb6278238,0xb6278238 16(692) DEBUG:tm:local_reply: branch=0, save=0, winner=0 16(692) DEBUG:tm:local_reply: local transaction completed 16(692) DEBUG: add_to_tail_of_timer[2]: 0xb61826f0 (8) 16(692) DEBUG: final_response_handler : done 16(692) DEBUG: timer routine:0,tl=0xb61845ac next=0xb6186c6c, timeout=3 16(692) DEBUG: timer routine:0,tl=0xb6186c6c next=0xb6188d98, timeout=3 16(692) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb6186b04) 16(692) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 16(692) DEBUG:tm:t_pick_branch: picked branch 0, code 408 16(692) DEBUG:tm:t_should_relay_response: dns-failover test: branch=0, last_recv=408, flags=0 16(692) DEBUG:tm:t_should_relay_response: trying DNS-based failover 16(692) qm_malloc(0xb5f99000, 72) called from blacklists.c: add_rule_to_list(377) 16(692) qm_malloc(0xb5f99000, 72) returns address 0xb62782b0 frag. 0xb6278298 (size=72) on 1 -th hit 16(692) DBG:add_list_to_head: adding to bl dns 0xb62782b0,0xb62782b0 16(692) qm_free(0x8151c40, 0xb62782b0), called from blacklists.c: rm_dups(440) 16(692) BUG: qm_free: bad pointer 0xb62782b0 (out of memory block!) - aborting > -----Original Message----- > From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Sent: Tuesday, May 29, 2007 5:48 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] BUG: qm_free ? > > The bug was in process 32133 > > There are no log statements from this process. What what this > process? > (use "openserctl ps" to see the PIDs of the processes) > > regards > klaus > > Papadopoulos Georgios wrote: > > Hello all, > > > > I am having some trouble geting Openser 1.2 to work. It > crashes pretty > > much immediately. I have tried it successfully on a test > system but as > > soon as I start it in the production environment, it > crashes. Any ideas? > > > > Thank you for any help. > > > > George > > > > > > 0(0) INFO: statistics manager successfully initialized > > 0(0) StateLess module - initializing > > 0(0) TM - initializing... > > 0(0) Maxfwd module- initializing > > 0(0) INFO:ul_init_locks: locks array size 512 > > 0(0) TextOPS - initializing > > 0(0) AVPops - initializing > > 0(0) permissions - initializing > > 0(0) WARNING: File not found: > /root/openser/config/permissions.allow > > 0(0) Default allow file > (/root/openser/config/permissions.allow) not > > found => empty rule set > > 0(0) WARNING: File not found: /root/openser/config/permissions.deny > > 0(0) Default deny file (/root/openser/config/permissions.deny) not > > found => empty rule set > > 0(0) AUTH module - initializing > > 0(0) AUTH_DB module - initializing > > 0(0) ACC - initializing > > 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> > > 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> > > 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> > > 0(0) INFO:textops:hname_fixup: using hdr type (31) instead of > > <Remote-Party-ID> > > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > > 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > > 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > > 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > > 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > > 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 > > 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 > > 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo > listener processes > > created > > 2(32093) pre_auth(): Credentials received are not filled properly > > 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > > 1(32087) ERROR: parse_to: invalid To - unexpected end of > header in > > state 22 > > 1(32087) ERROR: get_hdr_field: bad to header > > 1(32087) ERROR: bad header field > > 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing > MAX_FORWARD header > > failed! > > 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > > 1(32087) ERROR: parse_to: invalid To - unexpected end of > header in > > state 22 > > 1(32087) ERROR: get_hdr_field: bad to header > > 1(32087) ERROR: bad header field > > 1(32087) find_first_route: Error while parsing headers > > 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > > 1(32087) ERROR: parse_to: invalid To - unexpected end of > header in > > state 22 > > 1(32087) ERROR: get_hdr_field: bad to header > > 1(32087) ERROR: bad header field > > 1(32087) find_credentials(): Error while parsing headers > > 1(32087) pre_auth(): Error while looking for credentials > > 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > > 1(32087) ERROR: parse_to: invalid To - unexpected end of > header in > > state 22 > > 1(32087) ERROR: get_hdr_field: bad to header > > 1(32087) ERROR: bad header field > > 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 > > 1(32087) ERROR: parse_to: invalid To - unexpected end of > header in > > state 22 > > 1(32087) ERROR: get_hdr_field: bad to header > > 1(32087) ERROR: bad header field > > 1(32087) ERROR: build_res_buf_from_sip_req: alas, parse_headers > > failed > > 1(32087) pre_auth(): Error while sending 400 reply > > INFO: started (pid: 32073) > > [root@ser1 sbin]# 1(32087) pre_auth(): Credentials > received are not > > filled properly > > 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of > memory block!) > > - aborting > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 3(32096) ERROR: parse_via: invalid char <=> in state 201 > > 3(32096) error: parse_via_param > > 3(32096) ERROR: parse_via on: <SIP/2.0/UDP > > 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > > From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > > To: <sip:gdt@sip.i-call.gr> > > Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > > <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> > > CSeq: 4471 REGISTER > > User-Agent: Asterisk PBX > > Max-Forwards: 70 > > Expires: 120 > > Contact: <sip:s@87.202.97.60:59945> > > Event: registration > > Content-Length: 0 > > > > 3(32096) ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP > > 87.202.97.60:59945;branc=> > > 3(32096) ERROR: get_hdr_field: bad via > > 3(32096) ERROR: bad header field > > 3(32096) ERROR: parse_msg: message=<REGISTER sip:sip.i-call.gr > > SIP/2.0 > > Via: SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport > > From: <sip:gdt@sip.i-call.gr>;tag=as52e610af > > To: <sip:gdt@sip.i-call.gr> > > Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx > > CSeq: 4471 REGISTER > > User-Agent: Asterisk PBX > > Max-Forwards: 70 > > Expires: 120 > > Contact: <sip:s@87.202.97.60:59945> > > Event: registration > > Content-Length: 0 > > > > 3(32096) ERROR: receive_msg: parse_msg failed > > 3(32096) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 3(32096) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 1(32087) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 2(32093) WARNING: script writer didn't release transaction > > 0(32073) child process 32133 exited by a signal 6 > > 0(32073) core was generated > > 0(32073) INFO: terminating due to SIGCHLD > > 1(32087) INFO: signal 15 received > > 2(32093) INFO: signal 15 received > > 3(32096) INFO: signal 15 received > > 5(32102) INFO: signal 15 received > > 6(32103) INFO: signal 15 received > > 13(32123) INFO: signal 15 received > > 14(32127) INFO: signal 15 received > > 15(32130) INFO: signal 15 received > > 1(32087) 4(32099) 7(32104) INFO: signal 15 received > > 8(32109) INFO: signal 15 received > > 9(32112) INFO: signal 15 received > > 10(32115) INFO: signal 15 received > > 11(32118) INFO: signal 15 received > > 12(32121) INFO: signal 15 received > > INFO: signal 15 received > > INFO: signal 15 received > > 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child is already > > dead! > > > > > > > > Disclaimer > > The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > > > > > ---------------------------------------------------------------------- > > -- > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users >

2 1

[Users] CDR capturing
by Tim 29 May '07

29 May '07

Hello, I was wondering if users out there would be willing to share what mechanism they have found success with in regards to capturing cdrs at a high rate. We recently ran into some problems writing CDRs to a postgres database. Openser threads seemed to block which led to incoming messages not being processed in a timely manner. Flatstore solved the problem but created others, primarily how to stream the data off box for real time processing. We are considering Mysql but first wanted to ask the community to weigh in on what is considered 'best practice' for cdr/transaction record capturing. Our goal is to handle 500 CPS without lost records or process blocking. Thanks. Tim

2 1

[Users] BUG: qm_free ?
by Papadopoulos Georgios 29 May '07

29 May '07

Hello all, I am having some trouble geting Openser 1.2 to work. It crashes pretty much immediately. I have tried it successfully on a test system but as soon as I start it in the production environment, it crashes. Any ideas? Thank you for any help. George 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing 0(0) AVPops - initializing 0(0) permissions - initializing 0(0) WARNING: File not found: /root/openser/config/permissions.allow 0(0) Default allow file (/root/openser/config/permissions.allow) not found => empty rule set 0(0) WARNING: File not found: /root/openser/config/permissions.deny 0(0) Default deny file (/root/openser/config/permissions.deny) not found => empty rule set 0(0) AUTH module - initializing 0(0) AUTH_DB module - initializing 0(0) ACC - initializing 0(0) INFO:xl_parse_name: using hdr type (4) instead of <From> 0(0) INFO:xl_parse_name: using hdr type (3) instead of <To> 0(0) INFO:textops:hname_fixup: using hdr type name <Nat-flag> 0(0) INFO:textops:hname_fixup: using hdr type (31) instead of <Remote-Party-ID> 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 262142 1(32087) INFO:mi_fifo:mi_child_init(1): extra fifo listener processes created 2(32093) pre_auth(): Credentials received are not filled properly 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 1(32087) ERROR: parse_to: invalid To - unexpected end of header in state 22 1(32087) ERROR: get_hdr_field: bad to header 1(32087) ERROR: bad header field 1(32087) ERROR:maxfwd:is_maxfwd_present : parsing MAX_FORWARD header failed! 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 1(32087) ERROR: parse_to: invalid To - unexpected end of header in state 22 1(32087) ERROR: get_hdr_field: bad to header 1(32087) ERROR: bad header field 1(32087) find_first_route: Error while parsing headers 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 1(32087) ERROR: parse_to: invalid To - unexpected end of header in state 22 1(32087) ERROR: get_hdr_field: bad to header 1(32087) ERROR: bad header field 1(32087) find_credentials(): Error while parsing headers 1(32087) pre_auth(): Error while looking for credentials 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 1(32087) ERROR: parse_to: invalid To - unexpected end of header in state 22 1(32087) ERROR: get_hdr_field: bad to header 1(32087) ERROR: bad header field 1(32087) ERROR: parse_to_param: spitting out [<] in status 22 1(32087) ERROR: parse_to: invalid To - unexpected end of header in state 22 1(32087) ERROR: get_hdr_field: bad to header 1(32087) ERROR: bad header field 1(32087) ERROR: build_res_buf_from_sip_req: alas, parse_headers failed 1(32087) pre_auth(): Error while sending 400 reply INFO: started (pid: 32073) [root@ser1 sbin]# 1(32087) pre_auth(): Credentials received are not filled properly 16(32133) BUG: qm_free: bad pointer 0xb648f38c (out of memory block!) - aborting 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 3(32096) ERROR: parse_via: invalid char <=> in state 201 3(32096) error: parse_via_param 3(32096) ERROR: parse_via on: <SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport From: <sip:gdt@sip.i-call.gr>;tag=as52e610af To: <sip:gdt@sip.i-call.gr> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx <mailto:1b7dabcf26487db47f520b5c7195958e@gdt.ath.cx> CSeq: 4471 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:s@87.202.97.60:59945> Event: registration Content-Length: 0 > 3(32096) ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 87.202.97.60:59945;branc=> 3(32096) ERROR: get_hdr_field: bad via 3(32096) ERROR: bad header field 3(32096) ERROR: parse_msg: message=<REGISTER sip:sip.i-call.gr SIP/2.0 Via: SIP/2.0/UDP 87.202.97.60:59945;branc==z9hG4bK2e302aa9;rport From: <sip:gdt@sip.i-call.gr>;tag=as52e610af To: <sip:gdt@sip.i-call.gr> Call-ID: 1b7dabcf26487db47f520b5c7195958e(a)gdt.ath.cx CSeq: 4471 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:s@87.202.97.60:59945> Event: registration Content-Length: 0 > 3(32096) ERROR: receive_msg: parse_msg failed 3(32096) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 3(32096) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 1(32087) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 2(32093) WARNING: script writer didn't release transaction 0(32073) child process 32133 exited by a signal 6 0(32073) core was generated 0(32073) INFO: terminating due to SIGCHLD 1(32087) INFO: signal 15 received 2(32093) INFO: signal 15 received 3(32096) INFO: signal 15 received 5(32102) INFO: signal 15 received 6(32103) INFO: signal 15 received 13(32123) INFO: signal 15 received 14(32127) INFO: signal 15 received 15(32130) INFO: signal 15 received 1(32087) 4(32099) 7(32104) INFO: signal 15 received 8(32109) INFO: signal 15 received 9(32112) INFO: signal 15 received 10(32115) INFO: signal 15 received 11(32118) INFO: signal 15 received 12(32121) INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received 0(32073) INFO:mi_fifo:mi_destroy: seems that fifo child is already dead! Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] Not able to add users
by kiran.bkumar＠wipro.com 29 May '07

29 May '07

Hi All, Even now I am still not able to add the users as I get this error when I am trying.. [root@mkd149633 sbin]# openserctl add wipro1 wipro1 wipro1(a)presence.mkd149633.wipro.com This command requires a database engine - none was loaded [root@mkd149633 sbin]# pwd /usr/local/sbin [root@mkd149633 sbin]# There seems to be some problem in opening the connection to the mysql database. I have done the changes in openserctlrc file which is under /usr/local/etc/openser directory. Please let me know what could be the problem. Is there any other workaround to add 1 0r 2 users for testing purpose. Thanks in advance. Regards, Kiran. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

4 7

[Serusers] ser/serweb acl
by flavio 29 May '07

29 May '07

Hi to all, I have successfully installed serweb and I can manage acl users grants as admin. Should I configure my ser.cfg anyway?how to? Any references will be well accepted. Thanks, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

2 7

[Serusers] Formal SIP message logging
by Sahria Hao 29 May '07

29 May '07

Hi All, I want to get ONLY formal SIP message (ex. INVITE, REGISTER and more) by SER, like Cisco AS5300 debug mode (ex. # debug ccsip message). How do I configure my ser.cfg? Now ser.cfg as follows: debug=3 fork=yes log_facility=LOG_LOCAL0 listen=MY.SER.IP.ADDRESS alias=MY.SER.FQDN.ADDRESS1 alias=MY.SER.FQDN.ADDRESS2 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" And /etc/syslog.conf as follows: # SIP Express Router log local0.* /var/log/ser.log Regards, Sahria

3 2

[Serusers] TECHNICAL BOARD: Closed for new nominations
by Greger V. Teigre 29 May '07

29 May '07

New nominations for the iptel.org technical board are no longer accepted. The nominees not having confirmed their candidacy yet have received the below email privately just a few minutes ago. We will shortly proceed to voting, which according to the rules will be open 7 days from its start. g-) ------------------------ Hi, You have been nominated as a candidate for the iptel.org technical board, but has not yet indicated your willingness (or not) to participate as a candidate in the election. Your nominations have all been posted quite some time ago, and you should have had time to consider it, so in order to proceed with voting, we need to quickly confirm the candidates. Please reply to this email within 24 hours to confirm your candidacy. If I do not hear anything, I assume that you are not willing to be a candidate. Cheers, Greger

1 0

[Serusers] Technical board: Secret vs. public voting, suggested amendment to the rules
by Greger V. Teigre 29 May '07

29 May '07

Several people have on the lists and privately indicated their concern about public voting and the risk that people may feel uncomfortable. I have had several discussions privately on the issue and suggest the following amendment to the rules for election of the technical board (http://www.iptel.org/technicalboard/rules): ------------------- - The parenthesis "(by posting to one of the lists)" is removed from item 6. New items: 9. Before nomination starts, a person and email address should be identified publicly as the coordinator of the election process. 10. Nominations are suggested either publicly on one of the lists identified in item 6 or privately to the coordinator. Each nominee must accept the nomination in order to be electable. 11. Voting is done according to item 8 either by posting who are voted for on one of the lists identified in item 6 or privately to the coordinator. 12. Nominations and votes shall be anonymously posted to an iptel.org webpage and kept updated regularly by the coordinator. The coordinator shall keep all posts and private emails with votes as an assurance that the result can be verified by a third party. 13. Voting shall be open for seven days. ------------------- I'm sorry this becomes elaborate, but I believe that by being very explicit and detailed on the technical board, we may avoid trouble later. Unless there are change suggestions, comments, or objections within the end of the nomination (end of day Monday), I will amend the technical board rules and proceed to the voting step. g-)

1 1

[Users] LinuxTag - Hall 12, Stand 11
by Daniel-Constantin Mierla 28 May '07

28 May '07

Hello, those visiting LinuxTag in Berlin, May 30 - June 2, can find OpenSER at Stand 11 in the Hall 12, it will be like the VoIP area: http://www.linuxtag.org/2007/en/visitors/ausstellung/ausstellerliste.html We hope to meet many of you, discuss about the project and go for some German beer. Cheers, Daniel

1 0

[Serusers] ser-to asterisk and asterisk to ser call
by Rjey Nomer 28 May '07

28 May '07

Hi, Good day to all, is anyone in the list successfully installed and configured on different machine an asterisk and ser server wherein ser sip user can call asterisk sip user and vice versa. Is it possible? Can you post configuration files for regarding this? Thank you in advanced! Regards, Rjey ____________________________________________________________________________________Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/

1 0

[Serusers] Accounting VIa IP Source IP addess
by Nhadie 28 May '07

28 May '07

Hi All, Can I account via IP address of the client? I have one client using a sip pbx appliance. they are using me as a gatewyay using one of my sip extension. I acocunt their calls based on the calls from that extension. But can i also do it such a way that he does need an extension from me, he will just send everything to me and i'll process it via his IP address. And then i will also do accounting via his IP address from the acc table. Is this possible? TIA Nhadie ________________________________________________ Message sent using UebiMiau 2.7

1 0

[Serusers] ser 2_0_0 - Can't import dlg (find_export_record: <bind_dlg_mod> not found)
by tzieleniewski 28 May '07

28 May '07

Hi, I am tring to set the presence server on SER according to the presence cookbook. when I start ser i encounter the following error with loading of the dlg library. What could be the reason for this? May 28 16:33:49 sen ser[19892]: ... common libraries May 28 16:33:49 sen ser[19892]: init the content May 28 16:33:49 sen ser[19892]: init the content May 28 16:33:49 sen ser[19892]: find_export_record: found <load_tm> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <register_tmcb> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_newtran> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_relay_to_tcp> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_relay_to_udp> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_relay> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_reply_with_body> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_is_local> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_get_trans_ident> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_lookup_ident> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_add_blind> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_forward_nonack> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_request_within> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_request_outside> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_request> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <new_dlg_uac> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <dlg_response_uac> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <new_dlg_uas> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <update_dlg_uas> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <dlg_request_uas> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <set_dlg_target> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <free_dlg> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <print_dlg> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_gett> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <calculate_hooks> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_uac> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_uac_with_ids> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_release> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <t_unref> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <run_failure_handlers> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: found <cancel_uacs> in module tm [/home/ser/ser/ser-2.0/modules/tm/tm.so] May 28 16:33:49 sen ser[19892]: find_export_record: <bind_dlg_mod> not found May 28 16:33:49 sen ser[19892]: Can't import dlg May 28 16:33:49 sen ser[19892]: init_mod(): Error while initializing module pa May 28 16:33:49 sen ser[19892]: PA module cleanup

2 1

[Serusers] Fw: Call Fwd Unconditional for PABX users
by Cyprian Clement 28 May '07

28 May '07

----- Forwarded Message ---- From: Cyprian Clement <cyprian_clement(a)yahoo.co.uk> To: serusers(a)lists.iptel.org Sent: Monday, 28 May, 2007 12:38:57 PM Subject: [Serusers] Call Fwd Unconditional for PABX users Hi, I can get an SER proxied CFU(serial forking) working ok for registered users with their corresponding data populated in the "usr_preferences" table. However, when use the same approach when an incoming INVITE(RURI) to the SER has a number that belongs to a PABX the new INVITE does not get generated by the SER. is there any specific area in any module that I need to look out for? Thanks in advance. Regards, Cyp. ------- Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

1 0

[Serusers] Fw: Call Fwd On No Answer
by Cyprian Clement 28 May '07

28 May '07

----- Forwarded Message ---- From: Cyprian Clement <cyprian_clement(a)yahoo.co.uk> To: serweb-users(a)lists.iptel.org; serusers(a)lists.iptel.org Sent: Monday, 28 May, 2007 12:30:08 PM Subject: [Serusers] Call Fwd On No Answer Hi, I have managed to get Call Fwd Unconditional working by making the SER re-route the call to a different from that of the original INVITE by simply using the lookup table "usr_preferences" of the Mysql server. However, am unable to get "Call Fwd On No Answer" work despite having the mentioned table populated with CFNA. The server sends CANCEL ok but does not generate a new INVITE after that. Any suggestion will be much appreciated. Thanks, Cyp. ------------ Inbox full of unwanted email? Get leading protection and 1GB storage with All New Yahoo! Mail. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

1 0

[Serusers] Configure Incoming in SER
by Arun Kumar 28 May '07

28 May '07

Hi I've configure two ser box in my local network: 1. one box with mysql auth (SER A) 2. without any auth (just make install) (SER B) in both boxes 2-2 users are registered and they are able to call each other. my SER A users are able to call SER B user (sip:2001@SER-B) but SER B user can't call SER A (sip:3001@SER-A) user. here is my config file for SER A box: ------------------ debug=5 fork=no log_stderror=yes listen=176.249.2.69 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost /ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden Cant cross 1st bridge"); break; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","1"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","1"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } ------------- please guide me how to do this. thanks arun

2 5

[Serusers] Call Fwd Unconditional for PABX users
by Cyprian Clement 28 May '07

28 May '07

Hi, I can get an SER proxied CFU(serial forking) working ok for registered users with their corresponding data populated in the "usr_preferences" table. However, when use the same approach when an incoming INVITE(RURI) to the SER has a number that belongs to a PABX the new INVITE does not get generated by the SER. is there any specific area in any module that I need to look out for? Thanks in advance. Regards, Cyp. ------- ___________________________________________________________ Inbox full of unwanted email? Get leading protection and 1GB storage with All New Yahoo! Mail. http://uk.docs.yahoo.com/nowyoucan.html

1 0

[Serusers] Call Fwd On No Answer
by Cyprian Clement 28 May '07

28 May '07

Hi, I have managed to get Call Fwd Unconditional working by making the SER re-route the call to a different from that of the original INVITE by simply using the lookup table "usr_preferences" of the Mysql server. However, am unable to get "Call Fwd On No Answer" work despite having the mentioned table populated with CFNA. The server sends CANCEL ok but does not generate a new INVITE after that. Any suggestion will be much appreciated. Thanks, Cyp. ------------ ___________________________________________________________ New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

1 0

[Users] OpenSER @ LinuxTAG @ Berlin
by Bogdan-Andrei Iancu 28 May '07

28 May '07

Hi, I have the great pleasure to inform that this year, OpenSER will be present at LinuxTAG expo in Berlin 30 May - 2 June 2007. LinuxTAG is one of the most important Open Source event in Europe, with the main purpose of promoting the Open Source projects to the public. About more details and history of this event, you can read at http://www.linuxtag.org/2007/ . OpenSER has a booth reserved - unfortunately the numbers are still subject of changes, but as soon as it is final, we will updated the news on the project's site. As the participation of the projects is for free (sponsored by LinuxTAG), each year, representative projects are selected to exhibit. OpenSER will be part of a larger "island" (group of booths) of "VoIP". The goal is to let more people know about OpenSER by presenting what is openser, what is good for, what makes it special, etc. We will bring publicity materials, as well as Polo shirts and hats with OpenSER ;). As we want to have at the booth a large representation for the project, once the project was approved, I contacted several people (non Voice System) to join me and Daniel at the booth. We are still working on the final team (it will be listed as soon as we get confirmations), but we welcome other OpenSER addicted to join to the booth. I will keep you posted (email and site) about the details of the event. Regards, Bogdan

2 3

[Serusers] SER and other SIP Provider
by Arun Kumar 27 May '07

27 May '07

Hi, I've done the config of my ser and is allowing me to call other user on my ser server. Using my ser server I can call ohter sip user on some other server but he is not able to call me. Please guide me how to do this ? what additional code do I've to write if someone has sample please share with me. thanks arun

1 1

[Users] How to setup the openser with presence function?
by Charles Wang 27 May '07

27 May '07

Hi all, I use two X-Lite SIP clients with Contact list supported. I want to use chat (text mode) with each sip clients(X-Lite) via Openser. I study the documents of Openser ( http://openser.org/docs/ ), I think the modules ( presence, pua, pua_usrloc, pua_mi ) should what I need. But I don't know how to set it up. I can't see any online or offline information for both Xlite1 and Xlite2. Of couse, I add both of them as friends in contact list on X-Lite software. Can anyone kind to give me some tips or any URL to reference please? Xlite1 ==chat==> Openser ==Chat==> Xlite2 # -- presence params -- loadmodule "presence.so" modparam("presence", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("presence", "max_expires", 3600) modparam("presence", "force_active", 1) modparam("presence", "server_address", "sip:127.0.0.1:5060" ) #### I don't know how to configure them in DB # loadmodule "pua.so" # modparam("pua", "db_url", "mysql://openser:openserrw@localhost/openser") # loadmodule "pua_mi.so" # loadmodule "pua_usrloc.so" route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if (is_method("NOTIFY")) { sl_send_reply("200", "OK"); exit; }; if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); if(!search("^Content-Length:[ ]*0")) { setbflag(6); } } if(is_method("BYE")) { setflag(24); # account failed transactions setflag(25); # account successful transactions } # mark as loose-routed for acc setflag(26); route(11); } if(is_method("REGISTER")) { route(10); } setflag(24); # account failed transactions setflag(25); # account successful transactions if(is_method("INVITE")) { route(12); } if(is_method("CANCEL") || is_method("ACK")) { route(16); } ## Added for Presence if(is_method("PUBLISH") || is_method("SUBSCRIBE") || is_method("MESSAGE")) { route(19); } route(17); } route[19] { xlog("L_INFO", "route[19] Presence Handling Router - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; } -- Best Regards Charles

1 0

[Users] FreeBSD ports tree updated to OpenSER 1.2.1
by Jesus Rodriguez 27 May '07

27 May '07

Hi, The update of OpenSER 1.2.1 to FreeBSD ports tree has been committed. The port will also appear as a package in the next official packages build. There are two ways to install a port in FreeBSD. Both methods require to update the ports tree (with cvsup or tarball). Once the port tree is updated: If portupgrade tool is installed just do: portupgrade -N openser If portupgrade is not used or you prefer the usual/old method: cd /usr/ports/net/openser make install clean The port allows to select some compile options like TLS support and database support for MySQL and/or Postgres. If you are upgrading the port, please run: cd /usr/ports/net/openser make config if you want to change some compile time option. When the package appears in the next official build, you will only need to do: pkg_add -r openser Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[Serusers] Simultaneous Connections
by Kapil Dhawan 27 May '07

27 May '07

Hi List When ser starts, how many incoming connections can it handle at one time (Queue).

1 0

[Serusers] Configure SER to allow external SIP Calling
by Arun Kumar 26 May '07

26 May '07

HI I want to configure my SER (public ip) to allow call from external SIP users that is other than my sip domain. what I've to write in my ser.cfg if some one has code sample please send me. thanks arun

1 0

[Users] XCAP Lite 1.0.2 released
by Adrian Georgescu 26 May '07

26 May '07

Change log xcap-lite (1.0.2) unstable; urgency=low * Added requireRefreshWatchers flag per XCAP application * Added Persistent publication of data (RFC 4827) * Updated documentation The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.2.tar.gz Regards, Adrian

1 0

[Users] How to change To and From URI in SIP message ?
by mika.saari＠wipsl.com 26 May '07

26 May '07

Hi, I have to try changing the To and From URI like below: At the moment: To: sip:test1@test.com;tag=..... To be like: To: "test1"<sip:test1@test.com>;tag=... What would be the easiest way for doing this for presence NOTIFY SIP message ? Or if this can not be done with configuration system, then from where in the code the SIP message header is generated ? Thanks a lot, -Mika

3 2

[Users] presence module and retransmissions
by Klaus Darilion 25 May '07

25 May '07

Hi! Does presence module handles PUBLISH/SUBSCRIBE stateful or stateless? Thus, how should retransmission be handled. Will they be detected by presence module or is it possible to wrap the handle_publish inside t_newtran()? regards klaus

6 35

[Users] Presence module, NOTIFY, Max-Forward
by mika.saari＠wipsl.com 25 May '07

25 May '07

Hi, I noticed that this was already asked in mailing-list, but couldn't find any response, so thought to ask what is the opinion about this. Seems that all SIP stacks can not handle the NOTIFY message without Max-Forward. What kind of by pass options could be done for this, and is this problem something which will be checked in the future ? Thank you for any information, -Mika

2 1

[Users] Handling refer method with a device which does not support it
by Sergio Gutierrez 25 May '07

25 May '07

Hi everybody. We are trying to integrate a telephony platform based on Openser 1.2.0, with a Class-4 third party softswitch which does not support REFER method. Provider asks to us to change this method within Openser to INVITE method, so that the softswitch does not have to handle REFER but INVITE messages- Does somebody have any idea how to do that? We still think that solution is weird and difficult (and tricky), but, might be we are wrong. Thanks in advance for your help and attention. Sergio Gutiérrez EPM Telecomunicaciones S.A. E.S.P. Medellín Colombia-

3 4

[Serusers] SER-2.0.0, how to get list of registered UAs?
by Maxim Sobolev 25 May '07

25 May '07

Hi, In 0.9.x we have had the command "serctl ul show". The serctl tool seems to be non-functional in 2.0.0, could somehow explain how one is supposed to get list of in-memory contacts from the usrloc module in 2.0.0? In general it would be nice to have some "migration guide" started in WiKi so that people could share information like this. Personally I've spend quite lot of time migrating my 0.9.6 config to 2.0.0 and have quite few tips to share. Thanks for any hints! -Maxim

4 5

[Serusers] TECHNICAL BOARD - Nomination and voting is open!
by Greger V. Teigre 25 May '07

25 May '07

Everybody should by now have had plenty of time to provide feedback, input, and objections to the proposed rules for an iptel.org technical board. I have noted one correction to the original summary on April 28: adding semsdev(a)iptel.org as a nomination and voting list. I have taken this correction into the rules and established a permanent page describing the rules for a technical board. I conclude that we have reached a consensus as described here: http://www.iptel.org/technicalboard/rules According to the rules, we shall nominate one candidate from SEMS, two candidates with SER CVS write access, and two candidates from the wider SER community. Feel free to nominate and vote on up to five candidates in the same post. Raphael Coeffic has been nominated as SEMS representative and already started to receive votes. g-)

7 14

[Serusers] Call Forward Redirection By server(SER)
by Cyprian Clement 25 May '07

25 May '07

Hello Community, I have managed to setup my SER to do serial forking(CFU) by simply intercepting the incoming INVITE and generating a new INVITE to a user that is different from the one contained in the original invite. This means the user ID of the Request URI generated by the SER is different from that of the original/received RURI. I am currently working on the Call Forward (CFU, CFB, CFNA) scenarios. I would like to configure the SER in such a way that it generates the 302 message back to the caller and this 302 contains the contact of the third party that the caller needs to get in touch with. Basically, the caller should generate a new INVITE to this third party since the intended target has had his phone activated with the CF feature. I currently managed to get the SER do the serial forking(i.e. without sending back the 302) using the AVPops madule as instructed in the User Guide. Does the AVPops do the redirection too? if yes, then I will really appreciate to hear your helping info on how to achieve this. Many thanks in advance. Cyp. --------PS: if anything is unclear I can clarify. ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

1 0

Re: [Users] How to log the username of caller and callee to acc table?
by Bogdan-Andrei Iancu 25 May '07

25 May '07

Charles, glad to help - all the changes between the versions are documented on the web site. See: http://www.openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x regards, bogdan Charles Wang wrote: > Dear Bogdan, > > Thank you for your important information. > It helps me to solve my problems. > I add four fields to DB and use db_extra to insert them. > > modparam("acc", "db_extra", "from_uri=$fu ; sip_from=$hdr(From) ; > to_uri=$tu ; sip_to=$hdr(To)") > > > > Best regards, > Charles > > 2007/5/18, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Charles, > > in openser 1.2 only a few mandatory info about the SIP transaction are > accounted. If you want something more, use the the extra accounting. > For, example, for db, to get the from uri, so: > > modparam("acc","db_extra","from_uri=$fu ; sip_from=$hdr(From)") > > Note, you need to create the new columns in the acc table. See > http://www.openser.org/docs/modules/1.2.x/acc.html#ACC-EXTRA-ID > > regards, > bogdan > > Charles Wang wrote: > > Hi all, > > > > I ever used SER and I know it will save the From information such as > > 101(a)192.168.11.200 <mailto:101@192.168.11.200> > <mailto:101@192.168.11.200 <mailto:101@192.168.11.200>> to > sip_from or from_uri > > in its acc table. > > But in my openser, I also setup the acc module. Where is my > caller info? > > > > I can find a lot of number such as "2806655523" in my from_tag. > > I wanna store the caller and callee info in my acc table when I > make > > an internal call or an external call. > > > > Can anybody kind to help me to solve it? or give me some tips > please. > > PS: I got this openser.cfg from http://www.sipwise.com/wizard > <http://www.sipwise.com/wizard>. > > > > Best regards, > > Charles > > > > > -- > > Best Regards > Charles

2 2

Re: [Users] Not able to add users
by Henning Westerholt 25 May '07

25 May '07

On Freitag, 25. Mai 2007, you wrote: > Hi Henning, > > I have added the mysql module into my configuration file. Can you please > tell me how to and where to look for log files, indicating any problems > with database connectivity > > Thanks, > Kiran. Hi, increase the debug level in your config, and look in your system logfile (/var/log/messages on debian, could differ on your system) for messages. Henning P.S: please CC to the list

3 2

Re: [Users] failure_route problem
by Klaus Darilion 25 May '07

25 May '07

openser uses syslog logging. The destination of the logs is distribution dependent. E.g. debian based -->/var/log/syslog, redhat based --> /var/log/messages. Read http://openser.org/dokuwiki/doku.php/tutorials:debug-syslog-messages for details about to configure logging. regards klaus ps: please cc the list Yago Bargalló wrote: > Sorry, but I've tried to use xlog but I don't know where logs are printed. In which file does logs saved?? > > Thanks...!! > > > -----Mensaje original----- > De: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] > Enviado el: jueves, 24 de mayo de 2007 14:42 > Para: Yago Bargalló > CC: users(a)openser.org > Asunto: Re: [Users] failure_route problem > > Add xlog statements to your config and watch syslog. This is how you can > trace if the failure route is executed. Further you can set debug=4 to > see debug messages of openser. > > Further watch the SIP traffic using ngrep and dump it here > > regards > klaus > > > > > Yago Bargalló wrote: >> Hello, >> >> First of all thaks to Kraus for his quick reply on my last question. However I can't do openser work fine. >> >> I'm trying to use failure_route as follows, can anybody tell me which could be the problem?? >> >> >> >> (...) >> >> modparam("tm", "fr_inv_timer" , 10) >> >> >> >> # timer for response to a request >> >> modparam("tm","fr_timer", 4) >> >> >> >> >> >> # ------------------------- request routing logic ------------------- >> >> >> >> # main routing logic >> >> >> >> route{ >> >> (...) >> >> if (uri==myself) { >> >> (...) >> >> # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 >> >> >> >> if(uri=~"sip:2[0-9]+@" && method=="INVITE") { >> >> t_on_failure("1"); >> >> } >> >> (...) >> >> route(1); >> >> } >> >> >> >> >> >> route[1] { >> >> # send it out now; use stateful forwarding as it works reliably >> >> # even for UDP2TCP >> >> if (!t_relay()) { >> >> sl_reply_error(); >> >> }; >> >> exit; >> >> } >> >> >> >> failure_route[1]{ >> >> rewritehostport("172.16.51.160"); #asterisk >> >> rewriteuser("2002"); >> >> append_branch(); >> >> t_relay(); >> >> exit; >> >> } >> >> >> >> >> >> With this openser.cfg the call is cancelled after fr_inv_timer and failure_route[1] doesn't work but, however, if I do this: >> >> >> >> route{ >> >> (...) >> >> if (uri==myself) { >> >> (...) >> >> # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 >> >> >> >> if(uri=~"sip:2[0-9]+@" && method=="INVITE") { >> >> à rewritehostport("172.16.51.160"); #asterisk >> >> à rewriteuser("2002"); >> >> t_relay(); >> >> exit; >> >> } >> >> (...) >> >> route(1); >> >> } >> >> >> >> If I use this the call is redirected to my asterisk but using the failure_route as my first openser.cfg it does't work. >> >> >> >> Which is my problem?? >> >> >> >> Thanks.... >> >> >> >> >> >> >> >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

RE: [Users] Interaction Presence Agent - Presence User Agent
by Schumann Sebastian 25 May '07

25 May '07

Perfect! Thanks for quick answer, Anca! Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Friday, May 25, 2007 11:14 AM To: Schumann Sebastian Cc: users(a)openser.org Subject: Re: [Users] Interaction Presence Agent - Presence User Agent Hello, There is already a means to do that but in the trunk version. You should just give the command with expires -1 and the pua module will refresh Publish until you give a command with expires 0. regards, Anca Schumann Sebastian wrote: > Dear all, > > > > I have a question for understanding: I have implemented one OpenSER as > Registrar and one OpenSER as Presence Agent plus XCAP Server 1.0.0. > This is all working so far with newest SVN Release (Big Thanks to > Ovidiu and Anca for the hint) of OpenSER. > > > > Now I want to deploy a Web-Front-End for user interaction with the > Presence Core, because not all clients used are SIMPLE capable. User > of non-capable devices should be able to just login and change their > presence state. They do not necessarily have to be registered at the > registrar, why PUA+ pua_usrloc on registrar side is no choice. > > > > I was thinking about another OpenSER deployed as PUA plus the pua_mi > module for a solution. It gets commands from Web-Frontend via PHP, > forwarded to the FIFO with pua_mi module. The problem I have now is > that the messages are sent just once. Means a user changes his > presence state, PUA sends a PUBLISH to PA but after the expiry the > user appears to be offline. > > > > Is there any possiblitiy to let the PUA send messages constantly, when > the user selects online - means all the time until offline is > selected? I was thinking about many possiblities (cronjob or > shellscript running all the time during online time and sending data > to FIFO) but they are more or less just work-arounds. > > > > Best regards > > Sebastian > > > > **Sebastian Schumann** > > Diploma Student Architecture and Design Department > > > > Slovak Telekom, a. s. > > T-Com, Operations Unit > > Production and Service Division > > Innovations and Enterprise Solution Subunit > > > > Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia > > Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia > > > > +421 910 643010 (mobile) > > > > sebastian.schumann(a)t-com.sk <mailto:sebastian.schumann@t-com.sk> > > http://www.t-com.sk > >----------------------------------------------------------------------- - > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

1 0

[Serusers] INFO: Overview page of nominations for technical board
by Greger V. Teigre 25 May '07

25 May '07

As requested, here is a page for tracking nominations and status for the technical board: http://www.iptel.org/nominations_and_voting_technical_board_2007

1 1

[Serusers] ANNOUNCEMENT: New serdoc mailing list for documentation work
by Greger V. Teigre 25 May '07

25 May '07

Dear all, We have just created a new mailing list called serdoc(a)iptel.org. The list will be used for coordination, questions, doc patches, and discussions/input on any iptel.org documentation. The main ongoing efforts right now are: - Migration guide to 2.0 - Reference guide - SER - Getting Started and the configuration files - Improve module documentation Other documentation work also for other projects like sems, serweb, rtpproxy etc will be covered on this list. Developer or user, please consider subscribing to the list and contribute to an improved iptel.org documentation! http://lists.iptel.org/mailman/listinfo/serdoc Thanks, g-)

1 0

Re: [Serusers] behind 2 routers
by karl lu 24 May '07

24 May '07

Hi Ram, Thank you for your reply. Do you mean read the "NAT Traversal with Media proxy" section in get started document or search for "NAT Traversal with Media proxy" in our emial archive? Thanks, Karl ram <talk2ram(a)gmail.com> wrote: Hi look NAT Traversal with Media proxy ram On 5/22/07, root <taishan02(a)yahoo.com> wrote: Hi, I have a DSL router, which has only one port. To connect multiple computers I have another 4 ports router behind this DSL router. So SER is behind two routers. How can I configure these two routers so that I can use SER in this situation? Thanks, _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Samuel, That was very helpful. The last line of the output made me wonder what was happening with the "ESTABLISHED" link. It was CounterPath Eyebeam UA that needed to be restarted. Somehow it was hung and was the source of the problem, not SER. Thanks for the help, M [root@vx1 ser]# lsof -i:5060 COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME . . ser 4739 root 10u IPv4 35606 TCP localhost.localdomain:sip (LISTEN) ser 4739 root 11u IPv4 35607 TCP vx1.v11.com:sip (LISTEN) . . ser 4741 root 10u IPv4 35606 TCP localhost.localdomain:sip (LISTEN) ser 4741 root 11u IPv4 35607 TCP vx1.v11.com:sip (LISTEN) ser 4741 root 45u IPv4 35925 TCP vx1.v11.com:sip->10.10.30.121:ispipes (ESTABLISHED) [root@vx1 ser]# ----- Original Message ---- From: samuel <samu60(a)gmail.com> To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> Cc: Andrey Kuprianov <andrey.kouprianov(a)gmail.com>; "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Thursday, May 24, 2007 9:27:16 AM Subject: Re: [Serusers] Frustrating error - please help Maybe you've got another software bind to the address SER is trying to bind to(softphone,asterisk,...). Can you check whether the port 5060 is in use? what's the output of the next comand? #lsof -i:5060 samuel. 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: Also a good idea. Still no effect. This installation was working great for a while. Is it possible that some other software I've installed is causing the problem? This is 2.0.0 rc2 from CVS. M ----- Original Message ---- From: Andrey Kuprianov <andrey.kouprianov(a)gmail.com> To: "serusers(a)iptel.org" < serusers(a)iptel.org> Sent: Thursday, May 24, 2007 8:23:14 AM Subject: Re: [Serusers] Frustrating error - please help Maybe just try setting fork=yes ?... On 5/24/07, Mahatma Sarasvati < gnoahhb(a)yahoo.com> wrote: > > A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only > added my 10.10.10.41 ip address. The "fork=no" command is commented out. > > Is there something else that would cause ser to start in non-forking mode? > > Thanks, > M > > > ----- Original Message ---- > From: samuel < samu60(a)gmail.com> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Thursday, May 24, 2007 12:40:41 AM > Subject: Re: [Serusers] Frustrating error - please help > > Are you starting SER in non-forking mode? It looks like SER is not binding > to the 10.10.10.41 IP, only to the local IP ( 127.0.0.1) and that's why you > see in the logs that it's not able to send to a "standard" IP address: > WARNING: xl_mod_init: more IP 10.10.10.41 not used > CRITICAL: invalid sendtoparameters one possible reason is the server is > bound to localhost and attempts to send to the net > > Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't > recall the exact syntax right now..). > > Hope it helps, > Samuel. > > 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: > > Hi All, > > > > This ser installation was working fine and I can't figure out why its not > working now. > > I'm getting an error (see last log line) every time I attempt a call to > local users. Calling out to PSTN works great. ngrep reveals that SER is not > sending any messages to the callee location. > > > > There are 2 registered local users, one registered twice. > > > > I have no idea what is going on. I'm using an unmodified version 1.4 > ser.cfg file on CentOS 5. > > > > I suspect some server configuration error? ? ? ? > > > > > > Thanks, > > M > > > > > > > > Here are the logs. > > > > > > May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method > (auto detected) > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more > IP 10.10.10.41 not used > > May 23 16:03:38 vx1 last message repeated 8 times > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY > is /var/run/rtpproxy.sock > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > . > > . > > . > > . > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy > <unix:/var/run/rtpproxy.sock> found, support for it re-enabled > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: > sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) > > > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid > sendtoparameters one possible reason is the server is bound to localhost > > and attempts to send to the net > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send > failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: > sending request on branch 0 failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: > Unfortunately error on sending to next hop occurred (477/SL) > > > > > > > > When I start SER I get this response: > > > > [root@vx1 ~]# service ser restart > > Stopping ser: > [ OK ] > > Starting ser: Listening on > > udp: 127.0.0.1 [127.0.0.1]:5060 > > udp: 10.10.10.41 [ 10.10.10.41]:5060 > > tcp: 127.0.0.1 [127.0.0.1]:5060 > > tcp: 10.10.10.41 [ 10.10.10.41]:5060 > > Aliases: > > tcp: vx1:5060 > > tcp: vx1.velocity11.com:5060 > > tcp: localhost:5060 > > tcp: localhost.localdomain:5060 > > udp: vx1:5060 > > udp: vx1.velocity11.com:5060 > > udp: localhost:5060 > > udp: localhost.localdomain:5060 > > > > > [ OK ] > > [root@vx1 ~]# > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Ready > for the edge of your seat? > > Check out tonight's top picks on Yahoo! TV. > > http://tv.yahoo.com/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________ > Sucker-punch spam with award-winning protection. > Try the free Yahoo! Mail Beta. > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222

1 0

[Users] question about the openserctl ping command
by Papadopoulos Georgios 24 May '07

24 May '07

Hello all, I am using the openserct ping command to periodically check whether openser is working properly (If there is a better way to do this, please let me know). My script looks like this: SIP_URI="sip:213.5.xxx.xxx:5060" OPENSER_DIR=/root/openser TMP_FILE=/tmp/openser_ping.result $OPENSER_DIR/bin/openserctl ping $SIP_URI > $TMP_FILE 2>&1 Sometimes in TMP_FILE I see the following type of message: mkfifo: cannot create fifo `/tmp/openser_receiver_26907': File exists anaconda-ks.cfg install.log install.log.syslog mbox openser ERROR: error opening read fifo /tmp/openser_receiver_26907 At the same time openser seems to be working fine, however because the script gets this error, it restarts the server (the idea is that if ping fails then restart). Any idea why this is happening? What is the best way to make sure openser is working properly? I am assuming that a succesful ping only proves that one child is working properly. Thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

3 2

[Users] One way communication on calls from PSTN, two way on outbound
by Aaron Seelye 24 May '07

24 May '07

Hello all, I've based my config on the one found at http://siprouter.onsip.org/doc/gettingstarted/ch10.html, coming up with the one found here http://209.213.8.4/openser.cfg (all the xlog's are so I can follow what's going on when I run openser in a non-forking mode). I'm trying to get a simple system working where clients behind nat'ing firewalls can both send and receive calls. Currently outbound calls work fine, with two way communication, but inbound calls only have one way communication, from the nat'd client to the pstn. If anyone could give me some pointers as to what's going wrong, I'd appreciate it. TIA, Aaron Seelye

1 0

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Also a good idea. Still no effect. This installation was working great for a while. Is it possible that some other software I've installed is causing the problem? This is 2.0.0 rc2 from CVS. M ----- Original Message ---- From: Andrey Kuprianov <andrey.kouprianov(a)gmail.com> To: "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Thursday, May 24, 2007 8:23:14 AM Subject: Re: [Serusers] Frustrating error - please help Maybe just try setting fork=yes ?... On 5/24/07, Mahatma Sarasvati <gnoahhb(a)yahoo.com> wrote: > > A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only > added my 10.10.10.41 ip address. The "fork=no" command is commented out. > > Is there something else that would cause ser to start in non-forking mode? > > Thanks, > M > > > ----- Original Message ---- > From: samuel <samu60(a)gmail.com> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Thursday, May 24, 2007 12:40:41 AM > Subject: Re: [Serusers] Frustrating error - please help > > Are you starting SER in non-forking mode? It looks like SER is not binding > to the 10.10.10.41 IP, only to the local IP (127.0.0.1) and that's why you > see in the logs that it's not able to send to a "standard" IP address: > WARNING: xl_mod_init: more IP 10.10.10.41 not used > CRITICAL: invalid sendtoparameters one possible reason is the server is > bound to localhost and attempts to send to the net > > Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't > recall the exact syntax right now..). > > Hope it helps, > Samuel. > > 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: > > Hi All, > > > > This ser installation was working fine and I can't figure out why its not > working now. > > I'm getting an error (see last log line) every time I attempt a call to > local users. Calling out to PSTN works great. ngrep reveals that SER is not > sending any messages to the callee location. > > > > There are 2 registered local users, one registered twice. > > > > I have no idea what is going on. I'm using an unmodified version 1.4 > ser.cfg file on CentOS 5. > > > > I suspect some server configuration error? ? ? ? > > > > > > Thanks, > > M > > > > > > > > Here are the logs. > > > > > > May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method > (auto detected) > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more > IP 10.10.10.41 not used > > May 23 16:03:38 vx1 last message repeated 8 times > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY > is /var/run/rtpproxy.sock > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is initially 109568 > > May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF > is finally 262142 > > . > > . > > . > > . > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy > <unix:/var/run/rtpproxy.sock> found, support for it re-enabled > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: > sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) > > > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid > sendtoparameters one possible reason is the server is bound to localhost > > and attempts to send to the net > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send > failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: > sending request on branch 0 failed > > May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: > Unfortunately error on sending to next hop occurred (477/SL) > > > > > > > > When I start SER I get this response: > > > > [root@vx1 ~]# service ser restart > > Stopping ser: > [ OK ] > > Starting ser: Listening on > > udp: 127.0.0.1 [127.0.0.1]:5060 > > udp: 10.10.10.41 [10.10.10.41]:5060 > > tcp: 127.0.0.1 [127.0.0.1]:5060 > > tcp: 10.10.10.41 [10.10.10.41]:5060 > > Aliases: > > tcp: vx1:5060 > > tcp: vx1.velocity11.com:5060 > > tcp: localhost:5060 > > tcp: localhost.localdomain:5060 > > udp: vx1:5060 > > udp: vx1.velocity11.com:5060 > > udp: localhost:5060 > > udp: localhost.localdomain:5060 > > > > > [ OK ] > > [root@vx1 ~]# > > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Ready > for the edge of your seat? > > Check out tonight's top picks on Yahoo! TV. > > http://tv.yahoo.com/ > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > ________________________________ > Sucker-punch spam with award-winning protection. > Try the free Yahoo! Mail Beta. > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________Get the free Yahoo! toolbar and rest assured with the added security of spyware protection. http://new.toolbar.yahoo.com/toolbar/features/norton/index.php

2 1

Re: [Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

A good hint, but no. I'm using the standard version 1.4 ser.cfg. I've only added my 10.10.10.41 ip address. The "fork=no" command is commented out. Is there something else that would cause ser to start in non-forking mode? Thanks, M ----- Original Message ---- From: samuel <samu60(a)gmail.com> To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> Cc: serusers(a)iptel.org Sent: Thursday, May 24, 2007 12:40:41 AM Subject: Re: [Serusers] Frustrating error - please help Are you starting SER in non-forking mode? It looks like SER is not binding to the 10.10.10.41 IP, only to the local IP (127.0.0.1) and that's why you see in the logs that it's not able to send to a "standard" IP address: WARNING: xl_mod_init: more IP 10.10.10.41 not used CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net Case you need to use 2 interfaces, I would recommend using mhomed=yes (don't recall the exact syntax right now..). Hope it helps, Samuel. 2007/5/24, Mahatma Sarasvati <gnoahhb(a)yahoo.com>: Hi All, This ser installation was working fine and I can't figure out why its not working now. I'm getting an error (see last log line) every time I attempt a call to local users. Calling out to PSTN works great. ngrep reveals that SER is not sending any messages to the callee location. There are 2 registered local users, one registered twice. I have no idea what is going on. I'm using an unmodified version 1.4 ser.cfg file on CentOS 5. I suspect some server configuration error? ? ? ? Thanks, M Here are the logs. May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method (auto detected) May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more IP 10.10.10.41 not used May 23 16:03:38 vx1 last message repeated 8 times May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY is /var/run/rtpproxy.sock May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 . . . . May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it re-enabled May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: sendto(sock,0xb62048f4,1773,0, 10.10.100.103:5060,16): Invalid argument(22) May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: sending request on branch 0 failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) When I start SER I get this response: [root@vx1 ~]# service ser restart Stopping ser: [ OK ] Starting ser: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.41 [10.10.10.41]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.41 [10.10.10.41]:5060 Aliases: tcp: vx1:5060 tcp: vx1.velocity11.com:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: vx1:5060 udp: vx1.velocity11.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 [ OK ] [root@vx1 ~]# ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

2 1

[Users] Sometime t_really() func seem to be endless
by Bodin Bruno 24 May '07

24 May '07

Hi, I don't now why, but when call is cancel or unavaible t_relay is endless and never return anything. anyone know why ? Thank

4 15

[Serusers] SER User Group Meeting In Oslo
by Martin Hoffmann 24 May '07

24 May '07

Heisann, we are currently in the planning stage of the Second SER User Group Meeting. We agreed back in Prague to try to have it in Oslo in summer (no point in coming here in the dark season). The preliminary date for the meeting is Friday, July 13, 2007. However, since this is in the middle of the holiday seasons, I'd rather ask if this suits you all or whether we should move it to September. A more formal invitation will follow later. Best regards, Martin

6 7

[Users] failure_route problem
by Yago Bargalló 24 May '07

24 May '07

Hello, First of all thaks to Kraus for his quick reply on my last question. However I can't do openser work fine. I'm trying to use failure_route as follows, can anybody tell me which could be the problem?? (...) modparam("tm", "fr_inv_timer" , 10) # timer for response to a request modparam("tm","fr_timer", 4) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { t_on_failure("1"); } (...) route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1]{ rewritehostport("172.16.51.160"); #asterisk rewriteuser("2002"); append_branch(); t_relay(); exit; } With this openser.cfg the call is cancelled after fr_inv_timer and failure_route[1] doesn't work but, however, if I do this: route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { à rewritehostport("172.16.51.160"); #asterisk à rewriteuser("2002"); t_relay(); exit; } (...) route(1); } If I use this the call is redirected to my asterisk but using the failure_route as my first openser.cfg it does't work. Which is my problem?? Thanks....

3 2

[Serusers] SIP Gateway
by David Borgesen 24 May '07

24 May '07

Hello, We are trying to find a SIP to SIP gateway and we fell across this project. Can someone tell me if we can accomplish the following with this software. 1) Our DID/SIP telephone provider does not support the Refer method, and as a result our SipX IP PBX can not transfer calls from extension to extension. 2) As a work around we are thinking that we could install a gateway between the DID provider and SipX where the gateway would support the Refer method and handle the transfers. This method is not ideal because all phone traffic would need to go through the gateway as opposed to a direct connection between the DID/SIP provider and the end user extension, but we can't see any other solution. So, can Ser act as a gateway as needed, or is there another solution that anyone of you can think of. Thanks DB

1 0

[Serusers] t_relay() TCP errors
by tzieleniewski 24 May '07

24 May '07

Hi, When I try to t_relay() a message through SER i get the following errors: May 22 10:31:21 rd ser[22137]: ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s May 22 10:31:21 rd ser[22137]: ERROR: tcpconn_connect: tcp_blocking_connect failed May 22 10:31:21 rd ser[22137]: ERROR: tcp_send: connect failed May 22 10:31:21 rd ser[22137]: msg_send: ERROR: tcp_send failed Please point me what could be the reason for this? Maybe such situation is possible in case of non tcp server socket listening?? Cheers Tomasz

2 1

[Serusers] mysql dies -- accounting stopped -- call continues
by Nhadie 24 May '07

24 May '07

Hi All How can i make sure that when mysql dies, ser will not allow calls anymore? Because i had a problem, when mysql died, ser still continued to allow calls and i was not able to account it. How can i resolve this. Thank You in Advance. Regards Ronald

5 11

[Users] Loop construct in openser config file
by Steve Blair 24 May '07

24 May '07

Hello: Is it possible to implement a loop in the openser config file? Something simple that can iterate through a list of R-URI of AVP values. Thanks,Steve

3 2

[Users] (no subject)
by Yago Bargalló 24 May '07

24 May '07

Hello, First of all thaks to Kraus for his quick reply on my last question. However I can't do openser work fine. I'm trying to use failure_route as follows, can anybody tell me which could be the problem?? (...) modparam("tm", "fr_inv_timer" , 10) # timer for response to a request modparam("tm","fr_timer", 4) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { t_on_failure("1"); } (...) route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1]{ rewritehostport("172.16.51.160"); #asterisk rewriteuser("2002"); append_branch(); t_relay(); exit; } With this openser.cfg the call is cancelled after fr_inv_timer and failure_route[1] doesn't work but, however, if I do this: route{ (...) if (uri==myself) { (...) # Redirecciono cap a l'asterisk si marquen una extensió que comença per 3 if(uri=~"sip:2[0-9]+@" && method=="INVITE") { à rewritehostport("172.16.51.160"); #asterisk à rewriteuser("2002"); t_relay(); exit; } (...) route(1); } If I use this the call is redirected to my asterisk but using the failure_route as my first openser.cfg it does't work. Which is my problem?? Thanks....

1 0

Re: [Users] Errors with openserctl
by Henning Westerholt 24 May '07

24 May '07

On Donnerstag, 24. Mai 2007, you wrote: > Hi Henning, > > Thanks for youe response, but can you please tell me how to load the mysql > database engine ? Hi, add ' loadmodule "mysql.so" ' to your config file. Then you must also add a db_url to the module that should use the database, look for example to the usrloc module documentation. A sample setup is described at in the installation notes at: http://www.openser.org/mos/view/-OpenSER-Installation-Notes/ Henning

1 0

Re: [Users] active_watchers table
by Jeremy George 24 May '07

24 May '07

Hi Henning, Perfect. Thanks again. - Jeremy >-----Original Message----- >From: Henning Westerholt [mailto:henning.westerholt@1und1.de] >Sent: Thursday, May 24, 2007 04:39 AM >To: users(a)openser.org, 'Jeremy George' >Subject: Re: [Users] active_watchers table > >On Mittwoch, 23. Mai 2007, Jeremy George wrote: >> Hello All, >> >> I'm getting this error message: >> >> 0(0) PRESENCE:mod_init: Wrong version v1 for table <active_watchers>, need >> v4 >> 0(0) init_mod(): Error while initializing module presence >> ERROR: error while initializing modules >> >> I downloaded and compiled the current svn head (2276), and created the >> mysql active_watchers table defined in the README (see below). >> [..] >> Can anyone tell me where to find the right table? Thanks much. > >Hello Jeremy, > >the table that is defined by the README is to old. You can extract at the >moment all tables from the mysqldb.sh script. > >In the next weeks the db stuff will be updated, then all db schemas would be >up to date in a central place. > >I've attached the current table too. > >################# >INSERT INTO version (table_name, table_version) values >('active_watchers','4'); >CREATE TABLE active_watchers ( > id INT(10) UNSIGNED AUTO_INCREMENT NOT NULL, > pres_user VARCHAR(64) NOT NULL, > pres_domain VARCHAR(128) NOT NULL, > to_user VARCHAR(64) NOT NULL, > to_domain VARCHAR(128) NOT NULL, > from_user VARCHAR(64) NOT NULL, > from_domain VARCHAR(128) NOT NULL, > event VARCHAR(64) NOT NULL DEFAULT 'presence', > event_id VARCHAR(64), > to_tag VARCHAR(128) NOT NULL, > from_tag VARCHAR(128) NOT NULL, > callid VARCHAR(128) NOT NULL, > local_cseq INT(11) NOT NULL, > remove_cseq INT(11) NOT NULL, > contact VARCHAR(128) NOT NULL, > record_route TEXT, > expires INT(11) NOT NULL, > status VARCHAR(32) NOT NULL DEFAULT 'pending', > version INT(11) NOT NULL DEFAULT '0', > socket_info VARCHAR(128) NOT NULL, > local_contact VARCHAR(255) NOT NULL, > UNIQUE KEY tt_watchers (to_tag), > KEY due_activewatchers (to_domain, to_user, event), > PRIMARY KEY (id) >) Type=MyISAM; >######## > > >Best regards, > >Henning >

1 0

[Users] General for openser-2 things
by Yago Bargalló 24 May '07

24 May '07

Hello, I'm trying to redirect a call from openser to an Asterisk extension when a call is no answer. I think that this wouldn't be so difficult but I can't do it. I know that I have to use the failure_route(); with a fr_inv_timer parameter and then do a rewritehostport and a rewriteuser. My problem is that when the fr_inv_timer pass the call is cancelled and nothing else happens. In addition, I don't know where I can find the logs that I put in my openser.cfg, I have read the xlog module but I can't understand how to do it. On other hand, I have seen that there are a lot of functions that are often used but I don't know what they do internally, for example t_relay(), t_onreply(), revert_uri(),t_was_cancelled() etc etc etc and I don't know how to use it. ¿Is there any document that explain all it WELL?? Also, I've checked that if I do a call and I put the lines t_relay() and exit that call works fine but if I erase one of the two lines it don't work. Why??? Please help, I want to do a lot of things with openser but there is nothing that explain the basic functionality to do interessant things... Thanks in advance, "It will arrive the day that we will control telephony"

2 1

Re: [Serusers] Ser + Asterisk on the same box??
by John Breen 24 May '07

24 May '07

SIP wrote: > Your biggest issue, bar none, is using the loopback address for > anything externally-facing. 127.0.0.1 is called a loopback address for > a reason -- it never leaves the host itself. Just how do you expect > the SIP phones to be able to contact a 127.0.0.1 address when that > address is, in essence, themselves? > > There's something inherently flawed in your design. > > Let's back up and explain, in detail, what it is exactly you're trying > to do, and why you feel the 127.0.0.1 address should be part of this > equation? Perhaps we can suggest something if we know what you're > trying to do. > There's a very simple reason. Asterisk and SER are on the same machine. So the traffic from SER to asterisk doesn't need to leave the box. This is why I'm using the loopback interface - there's no need to use the real world one. -- Regards, John Breen HGA IT&T Solutions Tel: 08 63113000 Fax: 08 63113144 http://www.hgait.net.au Level 9, 256 Adelaide Terrace, Perth, WA 6000 Postal PO Box 3133 Perth, Adelaide Tce WA 6832 The information in this E-mail is confidential. If you are not the named & intended recipient, you must not read, use, copy, disclose or disseminate this E-mail, any attachments or the information contained in this E-mail and must immediately notify the sender & delete this E-mail immediately. Although HGA IT&T Solutions believes that this E-mail and any attachments are free of any virus or other defect which may affect a computer, it is the responsibility of the recipient to ensure that it is virus free. HGA IT&T Solutions does not accept any responsibility for any loss or damage arising in any way from the use of this E-mail or any attachments. IMPORTANT INFORMATION This is not a SPAM. The SPAM Act 2003 commenced on the 10th April 2004 and requires consent to the sending of most commercial electronic messages. If you wish to continue receiving electronic communications from HGA IT&T Solutions, you do not need to take any action. If you DO NOT wish to continue receiving electronic communications from HGA IT&T Solutions please advise us at admin(a)hgait.net.au and we will update our records accordingly.

4 3

[Users] Errors with openserctl
by kiran.bkumar＠wipro.com 24 May '07

24 May '07

Hi all, I am new user to openser, and I am trying to install openser to work as a presence server with the openims. Firstly is it possible to use the openser as a presence server. Secondly, I am getting the below error when I try to run the openserctl. [root@mkd149633 scripts]# openserctl moni ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have line 'fifo=/tmp/openser_fifo' in your config Also I am not able to add users , [root@mkd149633 scripts]# openserctl add wipro1 wipro1 wipro1(a)presence.mkd149633.wipro.com This command requires a database engine - none was loaded [root@mkd149633 scripts]# Looking forward to your support. Thanks and Regards, Kiran. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

2 1

[Users] active_watchers table
by Jeremy George 24 May '07

24 May '07

Hello All, I'm getting this error message: 0(0) PRESENCE:mod_init: Wrong version v1 for table <active_watchers>, need v4 0(0) init_mod(): Error while initializing module presence ERROR: error while initializing modules I downloaded and compiled the current svn head (2276), and created the mysql active_watchers table defined in the README (see below). I'm attempting to run it against the sample bla config found at: http://www.openser.org/dokuwiki/doku.php/presence:pua_bla (with some changes for appropriate environment). Can anyone tell me where to find the right table? Thanks much. - Jeremy CREATE TABLE `active_watchers` ( `id` int(10) NOT NULL auto_increment, `pres_user` varchar(64) NOT NULL, `pres_domain` varchar(128) NOT NULL, `to_user` varchar(64) NOT NULL, `to_domain` varchar(128) NOT NULL, `from_user` varchar(64) NOT NULL, `from_domain` varchar(128) NOT NULL, `event` varchar(64) NOT NULL default 'presence', `event_id` varchar(64), `to_tag` varchar(128) NOT NULL, `from_tag` varchar(128) NOT NULL, `callid` varchar(128) NOT NULL, `local_cseq` int(11) NOT NULL, `remote_cseq` int(11) NOT NULL, `contact` varchar(128) NOT NULL, `record_route` text, `expires` int(11) NOT NULL, `status` varchar(32) NOT NULL default 'pending', `version` int(11) default '0', `socket_info` varchar(128) NOT NULL, `local_contact` varchar(255) NOT NULL, PRIMARY KEY (`id`), UNIQUE KEY `tt_watchers` (`to_tag`), KEY `due_activewatchers` (`to_domain`,`to_user`,`event`) ) ENGINE=MyISAM DEFAULT CHARSET=latin1 AUTO_INCREMENT=1 ;

2 1

[Serusers] Frustrating error - please help
by Mahatma Sarasvati 24 May '07

24 May '07

Hi All, This ser installation was working fine and I can't figure out why its not working now. I'm getting an error (see last log line) every time I attempt a call to local users. Calling out to PSTN works great. ngrep reveals that SER is not sending any messages to the callee location. There are 2 registered local users, one registered twice. I have no idea what is going on. I'm using an unmodified version 1.4 ser.cfg file on CentOS 5. I suspect some server configuration error? ? ? ? Thanks, M Here are the logs. May 23 16:03:38 vx1 ser: init_tcp: using epoll_lt as the io watch method (auto detected) May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: Maxfwd module- initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: WARNING: xl_mod_init: more IP 10.10.10.41 not used May 23 16:03:38 vx1 last message repeated 8 times May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: AVPops - initializing May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: DEB: IP address of RTPPROXY is /var/run/rtpproxy.sock May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: xlog null is "<null>" May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is initially 109568 May 23 16:03:38 vx1 /usr/local/sbin/ser[1692]: INFO: udp_init: SO_RCVBUF is finally 262142 . . . . May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it re-enabled May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: udp_send: sendto(sock,0xb62048f4,1773,0,10.10.100.103:5060,16): Invalid argument(22) May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: msg_send: ERROR: udp_send failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: t_send_branch: sending request on branch 0 failed May 23 16:09:15 vx1 /usr/local/sbin/ser[1781]: ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) When I start SER I get this response: [root@vx1 ~]# service ser restart Stopping ser: [ OK ] Starting ser: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.10.10.41 [10.10.10.41]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.10.10.41 [10.10.10.41]:5060 Aliases: tcp: vx1:5060 tcp: vx1.velocity11.com:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 udp: vx1:5060 udp: vx1.velocity11.com:5060 udp: localhost:5060 udp: localhost.localdomain:5060 [ OK ] [root@vx1 ~]# ____________________________________________________________________________________Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/

2 1

[Serusers] Ser + Asterisk on the same box??
by John Breen 24 May '07

24 May '07

Ok, let me see if I've got this right. I'm running SER and Asterisk on the same box. SER listens on the external IP of the box on port 5060, Asterisk on localhost (127.0.0.1) on port 5061. My starting point to get calls from SER able to use the real world is the SER PSTN gateway in the Getting Started document, with route[5] changed to: route[5] { rewritehostport("127.0.0.1:5061"); avp_write("i:45", "inv_timeout"); route(1); } I start SER and add accounts 306, 307 and 499. So far so good, and when I get it all up and running, SIP phones can call other SIP phones via SER. Now, if I read right, I then need to tell SER that 127.0.0.1 is a trusted account by doing INSERT INTO trusted VALUES('127.0.0.1', 'any', '^sip:.*$'); in mysql, and restarting SER. I can still dial between internal phones and that is all OK, but I cannot dial via asterisk. I don't even see calls getting there. If I then add into the trusted table the IP addresses of SIP clients, I can dial from them via asterisk. But I have to do this for every client. This is fine if there are only a limited number of IP addresses that can use the asterisk gateway, but I have users with laptops in the real world whose IP addresses change so it's obviously not an ideal solution. What am I missing? Is the Call Forwarding setup the answer? I need to use SER as a front end to asterisk because of NAT issues, but I don't want to have to add EVERY user to the trusted table individually. Regards John Breen -- Regards, John Breen HGA IT&T Solutions Tel: 08 63113000 Fax: 08 63113144 http://www.hgait.net.au Level 9, 256 Adelaide Terrace, Perth, WA 6000 Postal PO Box 3133 Perth, Adelaide Tce WA 6832 The information in this E-mail is confidential. If you are not the named & intended recipient, you must not read, use, copy, disclose or disseminate this E-mail, any attachments or the information contained in this E-mail and must immediately notify the sender & delete this E-mail immediately. Although HGA IT&T Solutions believes that this E-mail and any attachments are free of any virus or other defect which may affect a computer, it is the responsibility of the recipient to ensure that it is virus free. HGA IT&T Solutions does not accept any responsibility for any loss or damage arising in any way from the use of this E-mail or any attachments. IMPORTANT INFORMATION This is not a SPAM. The SPAM Act 2003 commenced on the 10th April 2004 and requires consent to the sending of most commercial electronic messages. If you wish to continue receiving electronic communications from HGA IT&T Solutions, you do not need to take any action. If you DO NOT wish to continue receiving electronic communications from HGA IT&T Solutions please advise us at admin(a)hgait.net.au and we will update our records accordingly.

2 1

[Users] CPU load, ERROR: append_branch: max nr of branches exceeded, etc..
by Dan Mechanic 24 May '07

24 May '07

Forgive me if this question is a little open-ended... I am new to openser and sip, and trying to learn all I can... I am looking for information regarding: ERROR: append_branch: max nr of branches exceeded I understand that this error can be avoided by increasing the value of MAX_BRANCHES and recompiling openser, but, I am curious as to what happens when this error has been reached. Here's our situation... We use openser, and have a separate openser presence server. We had a situation recently where it seemed our presence server was bombarding our proxy to the point where the load on the proxy jumped and phones became unusable. Taking down the presence server seemed to alleviate the troubles. Also, we have an intermittent issue with phone reboots and lost calls, particularly on extensions that are on more than 12 phones (MAX_BRANCHES is currently set to 12). So, my question is, is this failure (max nr of branches exceeded) an innocuous one? or could it lead to futher failures (and maybe sip message recents... thus the bombardment)... Are there known side affects of this failure? Any info would be appreciated. Thanks Much! Best, Dan Mechanic

2 1

[Users] Require Core Dump Analysis
by Martin Burns 23 May '07

23 May '07

Hi, After many many months of faultless operation, today we experienced a SEG FAULT on one of our proxies running OpenSER 1.1.0-notls: Below is a back trace (with some values masked with xxx's), however our initial analysis can not make any sense of what the problem is. Any ideas would be greatly appreciated. res.c:194 is =====> r = (db_res_t*)pkg_malloc(sizeof(db_res_t)); Here is the back trace: (gdb) bt #0 0x00000000 in ?? () #1 0x4016ad2c in new_result () at res.c:194 #2 0x4016a495 in store_result (_h=0x81301b8, _r=0xbffff198) at dbase.c:322 #3 0x404bfc0a in db_load_avp (uuid=0x81301b8, username=0x81301b8, domain=0x81301b8, attr=0x81301b8 "@X\022\bø\034\023\b", table=0x0, scheme=0x404ce9e0) at avpops_db.c:270 #4 0x404c05d6 in ops_dbload_avps (msg=0x8134cd8, sp=0x8128210, dbp=0x8128b18, use_domain=0) at avpops_impl.c:393 #5 0x404bef47 in w_dbload_avps (msg=0x81301b8, source=0x81301b8 "@X\022\bø\034\023\b", param=0x81301b8 "@X\022\bø\034\023\b") at avpops.c:1183 #6 0x0805071b in do_action (a=0x8128de8, msg=0x8134cd8) at action.c:701 #7 0x08050631 in do_action (a=0x8129478, msg=0x8134cd8) at action.c:89 #8 0x08050631 in do_action (a=0x81294a8, msg=0x8134cd8) at action.c:89 #9 0x080522c2 in run_actions (a=0x8126548, msg=0x8134cd8) at action.c:89 #10 0x08051f2e in run_top_route (a=0x81301b8, msg=0x81301b8) at action.c:151 #11 0x08071ec1 in receive_msg ( buf=0x80ea340 "REGISTER sip:xx.xxxxxxx.xxx SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK-f1cef63e\r\nFrom: xxxxx < sip:xxxxxx@xx.xxxxxxx.xxx>;tag=fe10f354c23eb963o0\r\nTo: xxxxx < sip:xxxxxx@xx.xxxxxxx.xxx>\r\nC"..., len=475, rcv_info=0xbffff9a0) at receive.c:155 #12 0x08089654 in udp_rcv_loop () at udp_server.c:465 #13 0x08063b64 in main_loop () at main.c:925 #14 0x08064fae in main (argc=9, argv=0xbffffb24) at main.c:1477 (gdb) quit Thanks in advance Martin

1 0

[Users] OpenSER v1.2.1 released
by Bogdan-Andrei Iancu 23 May '07

23 May '07

Hello, Version 1.2.1 of OpenSER has been released. It is an update of version 1.2.0 and includes the fixes for the issues discovered in version 1.2.0, therefore all people using 1.2.0 should upgrade to this new version. The compatibility of the configuration file and database has been preserved (you do not need to do database reinstallation/migration). Detailed changelog: http://www.openser.org/pub/openser/1.2.1/ChangeLog For people using SVN, a new TAG was created ( 1.2.1 ). The development of the 1.2 branch will continue on the SVN 1.2 branch. So, if you want to synchronize with the latest changes/release in 1.2, keep using the 1.2 branch. If you want to take the 1.2.1 snapshot, use the 1.2.1 SVN tag. Source tarball can be downloaded from: http://www.openser.org/pub/openser/latest/src/ Or SourceForge: http://sourceforge.net/project/showfiles.php?group_id=139143&package_id=224… If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to team(a)openser.org. Regards, Bogdan

1 0

[Serusers] acl
by flavio 23 May '07

23 May '07

---------- Forwarded message ---------- From: flavio <flavio.patria(a)gmail.com> Date: 23-mag-2007 17.35 Subject: acl To: serusers(a)list.iptel.org Cc: serweb-users(a)lists.iptel.org Hi to all, I have successfully installed serweb and I can manage acl users grants as admin. Should I configure my ser.cfg anyway?how to? Any references will be well accepted. Thanks, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

1 0

[Users] Content-Lenght problem with NOTIFYs
by Tobias Lindgren 23 May '07

23 May '07

Hi all, a customer of ours are having some trouble with the Content-Lenght header when a NOTIFY-message goes through our OpenSER 1.2.0. Here is what seem to happend. SIP NOTIFY from our PSTN-gateway to our OpenSER: --- CUT --- Content-Length: 21. . SIP/2.0 100 Trying. . --- CUT --- SIP NOTIFY from our OpenSER to Customer: --- CUT --- Content-Length: 21. . SIP/2.0 100 Trying. --- CUT --- Now, the customer has to manually do this in his OpenSER to make the UA on his side happy: if (method=="NOTIFY") { if (search_body("SIP/2\.0 100 Trying")) { xlog("L_WARN", "R: Setting Content-Length to 20\n"); remove_hf("Content-Length"); append_hf("Content-Length: 20\r\n"); } } Please advice on what we are doing wrong here? Br, /Tobias

5 10

[Users] How to use pdt module correctly?
by Charles Wang 23 May '07

23 May '07

Hi all, I try to rewrite 444111(a)aaa.com to 111(a)bbb.com with pdt module. ############# pdt table in DB ################## mysql> select * from pdt; +----+----------------------+----------+--------------+ | id | sdomain | prefix | domain | +----+----------------------+----------+--------------+ | 1 | bbb.com | 444 | bbb.com | +----+----------------------+----------+--------------+ ############## parts of my openser.cfg ##################################3 loadmodule "pdt.so" modparam("pdt", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("pdt", "db_table", "pdt") modparam("pdt", "domain_column", "sdomain") modparam("pdt", "prefix_column", "prefix") modparam("pdt", "prefix", "") modparam("pdt", "hsize_2pow", 4) modparam("pdt", "sync_time", 300) modparam("pdt", "clean_time", 600) route[0] { ........................(skip) if (is_method("INVITE")) { xlog("L_INFO", "Prefix To Domain I - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); prefix2domain("0","0"); xlog("L_INFO", "Prefix To Domain II - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); }; ........................(skip) } #################### /var/log/message ####################################### May 23 15:55:12 localhost openser[22172]: New request - M=INVITE RURI= sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP=60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 May 23 15:55:12 localhost openser[22172]: Prefix To Domain I - M=INVITE RURI=sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP= 60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 May 23 15:55:12 localhost openser[22172]: PDT:prefix2domain: no prefix found in [444111] May 23 15:55:12 localhost openser[22172]: Prefix To Domain II - M=INVITE RURI=sip:444111@aaa.com F=sip:222@aaa.com T=sip:444111@aaa.com IP= 60.244.xxx.xxx ID=88BEB1DB-B447-4DAF-A3D6-212DBA3EC628(a)10.6.1.185 -- Best Regards Charles

2 4

[Users] OpenSER Acc, Direction Detecting?
by Brandon Armstead 23 May '07

23 May '07

Hello everyone, I'm currently running into the following problem, I currently run openser SVN 1.3. With the following modparam("acc", "detect_direction", 1) ON INCOMING CALL FROM PSTN AND HANGUP ON UAC from/to username is flipped. (ERROR) ON INCOMING CALL FROM PSTN AND HANGUP ON PSTN from/to username remains 'detected' ON OUTGOING CALL FROM UAC AND HANGUP ON PSTN. from/to username is flipped. (ERROR) ON OUTGOING CALL FROM UAC AND HANGUP ON UAC from/tousername remains 'detected' Has anyone ran into this issue, or know what I can do to remain direction state? Thanks, all help is appreciated. -- Brandon Armstead

2 3

[Serusers] test
by Jan Janak 23 May '07

23 May '07

please ignore

1 0

[Users] Need starter information (newbie)
by Rizwan Hisham 23 May '07

23 May '07

Hi all, this is my first email to this list. hoping to mingle with you all from now on. I have been working on asterisk for about a year. Now my company is planning to use openser with asterisk having asterisk at backend for features and voicemail. So is there any book for starters for openser as there is for asterisk (Asterisk the future of telephony). i need a book or any helpfull link which should reveil the basic concepts of openser, how to install it, how to program it, and how to manage it. Thanx in advance -- Rizwan Hisham Software Engineer AXVOICE Inc.

2 1

[Users] dispatcher question
by Klaus Darilion 23 May '07

23 May '07

Hi! Let's assume I have a load balancer which distributes (using hash over R-URI) calls to 4 proxies. Further, ds_mark_dst() is used to marked broken proxys. Marking a destination as inactive - does this change the routing completely (as if there were only 3 proxies in the config file) or will it still hash over the 4 proxies, but choose automatically the second entry in the resulting list if the result is the inactive proxy? regards klaus

2 3

[Users] XCAP Lite 1.0.1 released
by Adrian Georgescu 23 May '07

23 May '07

Change log xcap-lite (1.0.1) unstable; urgency=low * Added support for multiple applications and detect them based on xcap-root * Added xcap-root setting in config.php * Added pidf-manipulation application (to allow persistent publication of data using xcap server) * Detect auth_domain from XCAP URI if SIP user is formatted as user@domain * Added DELETE method The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.1.tar.gz Regards, Adrian

1 0

[Users] WeSIP: RingTone
by Helmut Kuper 23 May '07

23 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I play around with WeSIP's ringtone app. I try to get ringtone running in this constellation: caller -> openser sip-proxy -> openser+wesip -> -> target sip-proxy -> callee openser+wesip is loaded with both c2c and ringtone app. When openser sip-proxy relay callers INVITE to wesip, wesip response with "SIP/2.0 481 Call Leg/Transaction Does Not Exist". I dunno wheter this is an answere from ringtone app or from wesip itself. I also dunno, whether I have to config ringtone with environment specific things, like sip port, ip address or something like that or whether my constallation from above is correct. Maybe someone can help me out here :) Here is WeSIP's log, after receiving caller's INVITE: Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip$ HttpConnector [main] - The HttpConnector [all:8080] has been created ExpressConnector [main] - SEAS module successfully connected at 212.6.123.118:5081 ExpressConnector [main] - trying to receive 2 socket descriptions from SEAS ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipConnector [main] - The SipConnector [212.6.123.118:5080] has been created SipProcessor [main] - Starting background thread ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[0] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[1] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[2] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[3] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[4] ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 StatisticsServlet [main] - Wrapper setted in StatisticsServlet!!! ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/home/hk/wesip/wesip/wesipapps/click2call-08_11_06 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.i18n,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Created org.apache.axis.i18n.resource, linked to parent null ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.utils,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.utils.resource, locale en_US ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachEnabled) JavaUtils [main] - Attachment support is enabled? false ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachDisabled) JavaUtils [main] - Unable to find required classes (javax.activation.DataHandler and javax.mail.internet.MimeMultipart). Attachment support is disabled. ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(oddDigits00) ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(badChars01) AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() AxisServlet [main] - In servlet init AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() Registrar [main] - Registering org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 as axis:type=server Registry [main] - Managed= axis:type=server Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.ServiceAdministrator MbeansDescriptorsIntrospectionSource [main] - Introspected attribute version public java.lang.String org.apache.axis.management.jmx.ServiceAdministrator.getVersion() null MbeansDescriptorsIntrospectionSource [main] - Setting name: ServiceAdministrator BaseModelMBean [main] - default constructor Registry [main] - Creating MBeanServer9 BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 axis:type=server Registrar$ModelerBinding [main] - Registered axis:type=server in ServiceAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 as axis:type=deploy Registry [main] - Managed= axis:type=deploy Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentAdministrator MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentAdministrator BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 axis:type=deploy Registrar$ModelerBinding [main] - Registered axis:type=deploy in DeploymentAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentQuery@edf730 as axis:type=query Registry [main] - Managed= axis:type=query Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentQuery MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentQuery BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentQuery@edf730 axis:type=query Registrar$ModelerBinding [main] - Registered axis:type=query in DeploymentQuery AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() ConvergedContext [main] - ConvergedContext [Click2Call SipServlet] has been started ConvergedHostDeployer [main] - Installing web application at context path /ringtone-22_02_07 from URL file:/home/hk/wesip/wesip/wesipapps/ringtone-22_02_07 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed ConvergedContext [main] - ConvergedContext [Ringtone] has been started ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 Contact: <sip:v101@212.6.95.198:25952> To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/sdp User-Agent: X-Lite release 1003l stamp 30942 P-Announcement-Code: 322 P-Hint: ringtone Content-Length: 300 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:46338 label:965718283 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK205b.8c060ce2.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 212.6.123.118 mapped to: ConvergedEngine[Standalone].StandardHttpHost[212.6.123.118] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:212.6.123.118 ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bK205b.8c060ce2.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 503 bytes into Action Socket SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[1]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[1]] - About to process ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[1]] - Done processing ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGU/w94tZeNddg3dwRAhs+AKCpq2ZrAwCatPUY3BCUJR/ZJYwuzQCfUEOd MuNZ3C48K5Ypzun/27JG6TI= =5nOc -----END PGP SIGNATURE-----

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[Users] WeSIP: RingTone
by Helmut Kuper 23 May '07

23 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I play around with WeSIP's ringtone app. I try to get ringtone running in this constellation: caller -> openser sip-proxy -> openser+wesip -> -> target sip-proxy -> callee openser+wesip is loaded with both c2c and ringtone app. When openser sip-proxy relay callers INVITE to wesip, wesip response with "SIP/2.0 481 Call Leg/Transaction Does Not Exist". I dunno wheter this is an answere from ringtone app or from wesip itself. I also dunno, whether I have to config ringtone with environment specific things, like sip port, ip address or something like that or whether my constallation from above is correct. Maybe someone can help me out here :) Here is WeSIP's log, after receiving caller's INVITE: Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip$ HttpConnector [main] - The HttpConnector [all:8080] has been created ExpressConnector [main] - SEAS module successfully connected at 212.6.123.118:5081 ExpressConnector [main] - trying to receive 2 socket descriptions from SEAS ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/udp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] ExpressConnector [main] - SER socket:[212.6.123.118:5080/tcp] SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasRouter [SeasRouter] - SeasDispatcher Thread starts running SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 SeasHashTable [main] - Creating Seas Hash Table, with tablePower=17 and total entries:131072 ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipConnector [main] - The SipConnector [212.6.123.118:5080] has been created SipProcessor [main] - Starting background thread ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false ExpressMessageChannel [ExpressMessageProcessor] - setting spiral=false SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[0] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[1] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[2] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[3] SipProcessor [main] - Starting background thread SipProcessor [main] - Background thread has been started SipConnector [main] - recycle: Recycling processor SipProcessor[4] ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 ConvergedEngine [main] - 212.6.123.118_5080 StatisticsServlet [main] - Wrapper setted in StatisticsServlet!!! ConvergedHostDeployer [main] - Installing web application at context path /click2call-08_11_06 from URL file:/home/hk/wesip/wesip/wesipapps/click2call-08_11_06 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.i18n,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Created org.apache.axis.i18n.resource, linked to parent null ProjectResourceBundle [main] - getBundle(org.apache.axis,org.apache.axis.utils,resource,null,...) ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.utils.resource, locale en_US ProjectResourceBundle$Context [main] - loadBundle: Ignoring MissingResourceException: Can't find bundle for base name org.apache.axis.resource, locale en_US ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - Root package not found, cross link to org.apache.axis.i18n.resource ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachEnabled) JavaUtils [main] - Attachment support is enabled? false ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(attachDisabled) JavaUtils [main] - Unable to find required classes (javax.activation.DataHandler and javax.mail.internet.MimeMultipart). Attachment support is disabled. ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(oddDigits00) ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(badChars01) AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() AxisServlet [main] - In servlet init AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() AxisServletBase [main] - Enter: getEngine() AxisServletBase [main] - Exit: getEngine() Registrar [main] - Registering org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 as axis:type=server Registry [main] - Managed= axis:type=server Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.ServiceAdministrator MbeansDescriptorsIntrospectionSource [main] - Introspected attribute version public java.lang.String org.apache.axis.management.jmx.ServiceAdministrator.getVersion() null MbeansDescriptorsIntrospectionSource [main] - Setting name: ServiceAdministrator BaseModelMBean [main] - default constructor Registry [main] - Creating MBeanServer9 BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.ServiceAdministrator@10ffb38 axis:type=server Registrar$ModelerBinding [main] - Registered axis:type=server in ServiceAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 as axis:type=deploy Registry [main] - Managed= axis:type=deploy Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentAdministrator MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentAdministrator BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentAdministrator@b40ec4 axis:type=deploy Registrar$ModelerBinding [main] - Registered axis:type=deploy in DeploymentAdministrator Registrar [main] - Registering org.apache.axis.management.jmx.DeploymentQuery@edf730 as axis:type=query Registry [main] - Managed= axis:type=query Registry [main] - Looking for descriptor Registry [main] - Introspecting Registry [main] - load class org.apache.axis.management.jmx.DeploymentQuery MbeansDescriptorsIntrospectionSource [main] - Setting name: DeploymentQuery BaseModelMBean [main] - default constructor BaseModelMBean [main] - preRegister org.apache.axis.management.jmx.DeploymentQuery@edf730 axis:type=query Registrar$ModelerBinding [main] - Registered axis:type=query in DeploymentQuery AxisServletBase [main] - In AxisServletBase init AxisServletBase [main] - Enter: getEngine() ProjectResourceBundle [main] - org.apache.axis.i18n.resource::handleGetObject(engineFactory) EngineConfigurationFactoryFinder$1 [main] - Got EngineFactory: org.apache.axis.configuration.EngineConfigurationFactoryServlet DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getServer DefaultAxisServerFactory [main] - Enter: DefaultAxisServerFactory::getEngineConfiguration DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getEngineConfiguration AxisEngine [main] - Enter: AxisEngine::init AxisEngine [main] - Exit: AxisEngine::init DefaultAxisServerFactory [main] - Exit: DefaultAxisServerFactory::getServer AxisServletBase [main] - Exit: getEngine() ConvergedContext [main] - ConvergedContext [Click2Call SipServlet] has been started ConvergedHostDeployer [main] - Installing web application at context path /ringtone-22_02_07 from URL file:/home/hk/wesip/wesip/wesipapps/ringtone-22_02_07 StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed ConvergedContext [main] - ConvergedContext [Ringtone] has been started ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT StandardSipManager [main] - Starting StandardSipManager [main] - Force random number initialization starting StandardSipManager [main] - Force random number initialization completed StandardSipManager [main] - Start: Loading persisted sessions StandardSipManager [main] - Loading persisted sessions from SIPSESSIONS.ser StandardSipManager [main] - No persisted data file found StandardAppSessionManager [main] - Starting StandardAppSessionManager [main] - Force random number initialization starting StandardAppSessionManager [main] - Force random number initialization completed processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[0]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[0]] - About to process INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 SipConnector [ExpressMChannel[0]] - createProcessor: Reusing existing processor SipProcessor [ExpressMChannel[0]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 Contact: <sip:v101@212.6.95.198:25952> To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,MESSAGE,SUBSCRIBE,NOTIFY,REFER Content-Type: application/sdp User-Agent: X-Lite release 1003l stamp 30942 P-Announcement-Code: 322 P-Hint: ringtone Content-Length: 300 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:46338 label:965718283 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK205b.8c060ce2.0] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 212.6.123.118 mapped to: ConvergedEngine[Standalone].StandardHttpHost[212.6.123.118] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:212.6.123.118 ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHostValve [SipProcessor[4]] - No servlet-mapping matched this Request. SipResponse [SipProcessor[4]] - >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 481 Call Leg/Transaction Does Not Exist Record-Route: <sip:212.6.123.118;lr=on;ftag=450c2a3b> Via: SIP/2.0/UDP 212.6.123.118;branch=z9hG4bK205b.8c060ce2.0,SIP/2.0/UDP 212.6.95.198:25952;branch=z9hG4bK-d87543-ed108070b87aca0e-1--d87543-;rport=25952 Max-Forwards: 69 To: "Bruening" <sip:v111@sip0.en.ewetel.de> From: "v101" <sip:v101@sip0.en.ewetel.de>;tag=450c2a3b Call-ID: 7315210f6c7d7741NTA0NmE0M2ZkM2Y4MTg0YmU4NjQzNTllM2Q1YzliZGE. CSeq: 1 INVITE Content-Length: 0 LingerList [SipProcessor[4]] - scheduled transaction z9hG4bK205b.8c060ce2.0 for deletion ExpressMessageChannel [SipProcessor[4]] - sending with spiral=false SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ActionRouter$ActionSender [ActionSender] - Writing 503 bytes into Action Socket SeasMessageEvent [SeasRouter] - New message received, from 212.6.123.118:5060 to 212.6.123.118:5080 over UDP SeasTransactionEvent [SeasRouter] - hash_index = 46338,label=965718283 flags=0 ExpressMessageChannel [ExpressMChannel[1]] - peerAddress = 212.6.123.118/5060 ExpressMessageChannel [ExpressMChannel[1]] - About to process ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[1]] - Done processing ACK sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 ExpressMessageChannel [ExpressMChannel[0]] - Done processing INVITE sip:v111@212.6.123.118:5080;rinstance=fe71c43d95b8211a SIP/2.0 regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGU/w94tZeNddg3dwRAhs+AKCpq2ZrAwCatPUY3BCUJR/ZJYwuzQCfUEOd MuNZ3C48K5Ypzun/27JG6TI= =5nOc -----END PGP SIGNATURE-----

1 0

[Users] With SER Subscription state:- pending and Publish data is not coming with Notify
by Ambed Kumar 23 May '07

23 May '07

Hi Friends, In SER I am subscribing and i am getting 202 Ok also notify but the publish method data is is not coming to me with notify although after sending publish i am getting 200 Ok. Can any one suggest me regarding this. Thanks and Regards, Ambed

2 1

Re: [Users] Newbie question about FIFO in Openser 1.2.x
by Andreas Granig 23 May '07

23 May '07

Forgot to CC to the list. > Hi, > >> I never worked with FIFO, and I need exec one command in fifo, the >> trusted_reload ! > > OSER_FIFO has to be "/tmp/openser_fifo" in your openserctlrc. Then just > try "openserctl fifo trusted_reload", and you should get a "200 OK" if > successful. > > Cheers, > Andreas >

1 0

[Users] Newbie question about FIFO in Openser 1.2.x
by Thiago Maluf 22 May '07

22 May '07

Hi list, I never worked with FIFO, and I need exec one command in fifo, the trusted_reload ! ********** openser.cfg ********************** alias="insta.voip.nce.ufrj.br" mpath="/lib/openser/modules" children=8 tcp_children=4 debug=3 fork=yes log_facility=LOG_DAEMON log_stderror=yes group="openser" user="openser" sock_group="openser" sock_user="openser" disable_tcp=no mhomed=no server_signature=yes sock_mode=0600 unix_sock="/tmp/openser.sock" unix_sock_children=1 reply_to_via=no sip_warning=no check_via=no dns=no rev_dns=no disable_core_dump=no dns_try_ipv6=yes dns_use_search_list=yes ....... loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) modparam("mi_fifo", "fifo_group", "openser") modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") ...... **************** openserctl ***************** # $Id: openserctlrc 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # openser control tool resource file # # here you can set variables used in the openserctl ## your SIP domain SIP_DOMAIN=insta.voip.nce.ufrj.br ## database type: MYSQL or PGSQL, by defaulte none is loaded DBENGINE=PGSQL ## database host DBHOST=146.164.247.209 ## database name DBNAME=openser ## database read/write user DBRWUSER=openser ## database read only user # DBROUSER=openserro ## password for database read only user DBROPW=openserro ## database super user DBROOTUSER="postgres" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE="DB" ## control engine: FIFO or UNIXSOCK ## - default FIFO CTLENGINE="UNIXSOCK" ## path to FIFO file OSER_FIFO="/tmp/" ## check ACL names; default on (1); off (0) VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## presence of serweb tables - default "no" # HAS_SERWEB="yes" ## verbose - debug purposes - default '0' VERBOSE=1 ## do (1) or don't (0) store plaintext passwords ## in the subscriber table - default '1' # STORE_PLAINTEXT_PW=0 *************************** My doubt is what my conf is correct and what is the command that I use in the Konsole!? Thanks in advanced for your dedication to someone that read to here. Thiago Maluf. ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

1 0

[Users] BYE relay issue
by kaiser 22 May '07

22 May '07

Dear sir, We install 3 openser 1.1.0 in public internet, and we find a BYE relay issue to caller SIP UA. UA(a)-------Proxy1 --------Proxy2 ----Proxy 3 ---------UA(b) We make call from UA(a) to UA (b), if UA(b) send Bye first, UA(a) can not get Bye. The packet is from Proxy2 to Proxy1 as following BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 67 Content-Length: 0 Then bye is looping in proxy1. BYE sip:1.2.3.4:5060;nat=yes;ftag=ooqjvbi7ql;lr=on SIP/2.0 Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Record-Route: <sip:5.6.7.8;lr=on;ftag=as1b41fbeb> Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bKc74c.1714a4e7.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.bd89ce13.0 Via: SIP/2.0/UDP 5.6.7.8;branch=z9hG4bKc74c.ad89ce13.0 Via: SIP/2.0/UDP 100.200.100.200;branch=z9hG4bKc74c.321ef5e3.0 Via: SIP/2.0/UDP 100.200.100.199:5060;branch=z9hG4bK75e8d6f3;rport=5060 From: <sip:0908900000@sip.edu.tw;user=phone>;tag=as1b41fbeb To: "0909802020 kl" <sip:0909802020@sip.edu.tw>;tag=ooqjvbi7ql Call-ID: 3c275da915f9-iylvmp2wjuh0@snom360-00041323058B CSeq: 102 BYE User-Agent: Gentrice_IPPBX Max-Forwards: 66 Content-Length: 0 We used to use default script for proxy1, but the same... And we do record_route, loose_route relay as well. Anyone know what happen? best regards Thrli

3 7

[Users] WeSIP: Click2Call Linefeed
by Helmut Kuper 22 May '07

22 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello WeSIPer, I found a little but annoying Error in WeSIP's C2C application. I send back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's sl-send_reply(). I expected that C2C terminates the call function, cause C2C has an error handling method for 4xx replies, but nothing happens. openser as wesip's frontend received definately the reply. By chance I found, that the reason for this was my reply message. This doesn't work with WeSIP's C2C application: sl_send_reply("483", "My message\n"); But this works: sl_send_reply("483", "My message"); It seems, that C2C's or WeSIPs reply parser has some problems with linefeeds at the end of reply's message text. I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06 regards Helmut PS: Is there already a fix for WeSIP to reconnect to openser's seas socket, when openser restarts? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUV2k4tZeNddg3dwRAsPcAKCNqGE+XTK7BvW69x4YT498Bg0dkgCbB3eL 6Me/ZKYaqxfo/qDGB8M3lg4= =jTel -----END PGP SIGNATURE-----

2 1

[Users] loose_route problem with Fritz
by Edoardo Serra 22 May '07

22 May '07

Hi guys, I have OpensER 1.1.0 in fornt of 2 Asterisks servers. My OpenSER act as load balancer for outbound calls and for registrar server for inbound calls coming from Asterisks PSTN GW to Users Some users, using AVM Fritz Fon Boxes cannot register on the proxy, they get a "483 Too Many Hops". They say it happens only if they configure the Fritx using FQDN of our proxy, if they put its IP addess in it they are having no problems... (it sounds very strange to me but I captured some trafic and had confirmation of what they're saying) Here is a SIP dialog with the error message Request-Line: REGISTER sip:sip.advancemedia.it SIP/2.0 Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120 Route: <sip:sip.advancemedia.it;lr> From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928 To: <sip:cirifax@sip.advancemedia.it> Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: AVM FRITZ!Box Fon ata 1020 11.04.01 (Jan 25 2006) Supported: 100rel, replaces Allow-Events: telephone-event, refer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE Accept: application/sdp, multipart/mixed Accept-Encoding: identity Content-Length: 0 Status-Line: SIP/2.0 483 Too Many Hops Message Header Via: SIP/2.0/UDP 192.168.1.121:5060;received=80.67.115.96;branch=z9hG4bKC0F2D683E6BE8AC5F1A4A260BE120 From: <sip:cirifax@sip.advancemedia.it>;tag=1898319928 To: <sip:cirifax@sip.advancemedia.it>;tag=b36cb132c8240044969e91db40b97d00.d5d1 Call-ID: 9ADCCD22187BDEC93B5953DD93E0F(a)192.168.1.121 CSeq: 1 REGISTER Server: OpenSer (1.1.0-notls (i386/linux)) Content-Length: 0 Warning: 392 213.92.79.131:5060 "Noisy feedback tells: pid=13326 req_src_ip=213.92.79.131 req_src_port=5060 in_uri=sip:sip.advancemedia.it out_uri=sip:sip.advancemedia.it via_cnt==71" It doesn't happen if the user put the ip of the proxy server in the configuration of the Fritz I guess it's a problem with loose routing but I cannot understand if it's a buggy implementation of SIP in the fritz or my misconfiguration Here is my openser.cfg, tnx in advance for help. Edoardo # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # ----------- global configuration parameters ------------------------ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/openser/modules/sl.so" [...] modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("usrloc", "timer_interval", 120) modparam("auth_db", "calculate_ha1", 0) modparam("auth_db", "db_url", "mysql://user:pass@192.168.252.5/db") modparam("uri_db", "db_url", "mysql://user:pass@192.168.252.5/openser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("registrar", "max_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) modparam("nathelper", "natping_interval", 20) # Ping interval 20 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("dispatcher", "force_dst", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); exit; }; if ( (method=="OPTIONS") || (method=="SUBSCRIBE") || (method=="NOTIFY") ) { sl_send_reply("405", "Method Not Allowed"); exit; } # loose-route processing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); t_relay(); exit; }; if (!method=="REGISTER") { record_route(); }; if ((src_ip==ip.of.asterisk.1) || (src_ip==ip.of.asterisk.1)) { if (!lookup("location")) { sl_send_reply("486", "Busy here"); exit; }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; exit; }; if (nat_uac_test("3")) { if ((method=="REGISTER") || (method=="INVITE") || (method=="OPTIONS")) { fix_nated_contact(); force_rport(); setflag(6); # Mark as NATed } } if (method=="REGISTER") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_to()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } save("location"); exit; }; if (method=="INVITE") { if (!proxy_authorize("exorsa", "openser_view")) { proxy_challenge("exorsa", "0"); exit; } if (!check_from()) { sl_send_reply("403", "Digest username and URI username do NOT match! Stay away!"); exit; } } if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); exit; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); exit; }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); if ((src_ip!=ip.of.asterisk.1) && (src_ip!=ip.of.asterisk.2)) { ds_select_dst("1", "0"); } if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

2 2

[Users] b2bua
by Kro Kamil 22 May '07

22 May '07

Hi, I need a b2bua to use with openser. I also don't want to use radius. what options do I have? thanks in advance. _________________________________________________________________ Live.com'u deneyin: çevrimiçi dünyanýzý bir araya getirin; haberler, spor, hava durumu ve çok daha fazlasý. http://www.live.com/getstarted

4 3

[Users] Presence, Presence_xml & NOTIFY
by mika.saari＠wipsl.com 22 May '07

22 May '07

Hi, At the moment I am using SVN version of OpenSER (2:2253M) also I have been using the 1.2 release. The problem has been the NOTIFY message which has been dropping time to time, or like in current svn release, I do not get the NOTIFY message at all anymore. I think my openser.cfg is having something wrong, but haven't been able to figure out what is the reason. I do use XCAP_Lite 1.0.0 with presence & presence_xml modules. The force_active is set to "0", because I want to have reactive buddy authorization. The XCAP side works nicely, also watchers and active_watchers are populated by OpenSER correctly. But once the SUBSCRIBE is received from client, no NOTIFY is sent to any clients which are active in watchers table. After reading the log file I found this message from it: 0(4415) PRESENCE: publ_notify: Could not get subs_dialog from database Thank you for tips, -Mika Saari

2 3

[Users] Distribution Calls with the same destination number to different servers
by Andreas Brodmann 22 May '07

22 May '07

Hi all, I am working on the following scenario: Our carrier sends all calls (SIP protocol) to my openser, which forwards the calls depending on their destination number to four asterisk servers (for load sharing reason - we're speaking of a roof of 300 concurrent calls). I have one number which leads to a conferencence (one speaker, multiple listeners), whoose number of callers will exceed the number of calls a single asterisk will hold. Therefore I want the openser to distribute the calls to this number to all four asterisk servers (to share the load). My question is: how do I configure this on openser, as I have to send all requests from the same SIP session to the same asterisk? Can anyone give me a hint on this? Thanks in advance. Andreas

1 0

[Serusers] Unable to download SER RC1
by Andrey Kuprianov 22 May '07

22 May '07

Hi, Is something wrong with siprouter.onsip.org FTP site? I've tried Firefox, IE7 and even wget, but still cant download ftp://siprouter.onsip.org/pub/ser/ser-2.0.0-rc1_src.tar.gz. What's wrong? Andrey.

2 3

[Serusers] ser.cfg buildsystem
by KUMAR 22 May '07

22 May '07

Hi all, I am having some problem with the configuration file buildsystem. Whatever changes I make, it always creates only the basic hello world configuration file only. I am using the latest package. It also gives the following message **** WARNING! Only helloworld is implemented!!!! How do I include other feature packages like NAT and authentication. Here's what i've done This configure script will generate a new local configuration to ./local/default/config.m4. ====================== Press ENTER to accept default values. SER Server IP address: IP address to listen on (202.52.247.221): 192.168.2.2 Port to listen on (5060): Public IP address of server (ENTER if same as listen IP): Domain name for this SER installation (i.e. mydomain.com, def. mytestser.org) : MYDOMAIN SER installation path (/usr/local, i.e. modules in /usr/local/lib/ser/modules): Usage: dbtype://readwriteuser:password@host/databasename Database URL (mysql://ser:heslo@localhost/ser): You may choose to turn on logging as found in the SER - Getting Started guide. This logging can be turned on and off in a live environment, and we recommend everybody to include the logging except in high-load environments. (You can also later disable it from your config.m4) Turn on logging (yes): How much log possibilities to include (low/medium/high, default high): You can base your configuration file on one of the feature packages available. Currently, the supported packages are (gettingstarted sems) gettingstarted : A set of basic configurations found in most setups sems : A configuration file that enables SER as a front-end for SEMS You can also choose to include specific components within a feature package (components). And finally, you can choose to include no prepared features, just use your own code in ./local. Your choice of feature, choose only one, default is gettingstarted (none, components or gettingstarted sems) ? You have chosen the SER - Getting Started feature package. (See SER - Getting Started document) Which feature set do you want to base your configuration on: helloworld : basic configuration auth : authentication using mysql database nat : NAT traversal handling pstn : Functionality for PSTN integration callfwd : basic call features like call forwarding **** WARNING! Only helloworld is implemented!!!! Your choice (default helloworld, auth, nat, pstn, callfwd): nat You have chosen NAT traversal, do you want to use rtpproxy (iptel.org) or mediaproxy (ag-projects)? Your choice (default rtpproxy, mediaproxy): rtpproxy Copying site templates to ./local/default... The resulting ser.config doesn't include anything except the helloworld. Thank you in advance. Kumar

2 1

Re: RES: [Users] Presence Problem - Publish Offline
by Anca Vamanu 22 May '07

22 May '07

Hello, The proper way to send Publish messages is described in RFC 3903 (see section 4.1 for event refreshing event state publication). If you don't include a Sip-If-Match header it is not considered a refresh of the previous state and thus it does not rewrite the open state previously published. regards, Anca Hildebrando Trannin wrote: >Hi Anca, > >Thanks for your answer. When I say that OFFLINE doesn't works is that I send >a PUBLISH with the basic status CLOSED (after already sent others PUBLISH >with status OPENED) and OpenSER doesn't send NOTIFY message to inform >subscribers that that contact is now OFFLINE. According to your answer it >happens because I'm not sending PUBLISH's with Sip-If-Match. Is it right? If >no, how is the correct way to send PUBLISH? Just to complement, PUBLISH with >expires = 0 that do UNPUBLISH is working fine. > >Regards, > Hildebrando Trannin > >-----Mensagem original----- >De: Anca-Maria Vamanu [mailto:anca@voice-system.ro] >Enviada em: segunda-feira, 21 de maio de 2007 06:12 >Para: Hildebrando Trannin >Cc: users(a)openser.org >Assunto: Re: [Users] Presence Problem - Publish Offline > >Hello, > >What happens here is that the presence module receives a first Publish >from your aplication (that is a Publish with no Sip-If-Match header) >having the basic status closed. In this situation, it does not send >Notify to its watches to prevent descovering that the user is in fact >online by tracking packets. > >Hildebrando Trannin wrote: > > > >>Hi all, >> >>I was testing OpenSER presence functionality using eyebeam and a >>softphone developed by my team. The OFFLINE state with eyebeam worked >>perfectly, but with my softphone it didn't work. I set force_active = >>1. XML sent by PUBLISH and OpenSER log are below. Can anyone help me? >> >> >> >I don't understand what you mean with the Offline state not working. >Could you please give more details? > > > >>Thanks, Hildebrando >> >>*PIDF XML:* >> >>* * >> >><?xml xmlns="urn:ietf:params:xml:ns:pidf" >> >>entity="sip:hildebrando@voip.maritaca"> >> >><presence> >> >><tuple id=""> >> >><status> >> >><basic>closed</basic> >> >></status> >> >><note>offline</note> >> >></tuple> >> >></presence> >> >>The basic difference between my application and eyebeam is that >>eyebeam uses *RPID XML *and I use *PIDF XML*. >> >>*OPENSER LOG:* >> >>* * >> >>3(26039) PRESENCE: update_presentity: The presentity status is >>offline; do not send notify >> >>------------------------------------------------------------------------ >> >> >> >> >> >regards, > >Anca Vamanu > > > >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > >

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Re: [Serusers] Does SER's MYSQL work in SUSE Linux?
by Cyprian Clement 22 May '07

22 May '07

Hi Ram, Thanks for the note. I thought there may be some issues with SUSE regarding mysql since all the talk is about other Linux flavours. I will give it a try and see. Thanks, Cyp. ==== ----- Original Message ---- From: ram <talk2ram(a)gmail.com> To: Cyprian Clement <cyprian_clement(a)yahoo.co.uk> Cc: "serusers(a)iptel.org" <serusers(a)iptel.org> Sent: Sunday, 20 May, 2007 5:06:55 PM Subject: Re: [Serusers] Does SER's MYSQL work in SUSE Linux? On 5/18/07, Cyprian Clement <cyprian_clement(a)yahoo.co.uk> wrote: Hello Community, Does anyone know whether it's possible to install the current recommended MYSQL on SUSE Linux. I have a Machine with a SUSE installed on it. I have successfully installed SER 0.9.6 but now I would like to have MYSQL for authentication support. Does the SER's recommended MYSQL work with SUSE?. I only see the talk about Redhat and Debian so unsure if it works with SUSE. Many thanks for in advance. Hi any linux with mysql and ser should support try using mysql authentication any problems post here ram Cyp. ---------- Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ___________________________________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

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Re: [Serusers] Problem running presence modules - SER won't start
by Vaclav Kubart 22 May '07

22 May '07

There was problem that sudo didn't propagate LD_LIBRARY_PATH settings or it did, but linker didn't care or something like that. Does your ser binary has setuid/setgid bit set? In this case linker may ignore the LD_LIBRARY_PATH settings too. You can try to add the path /usr/local/lib/ser to your /etc/ld.so.conf and run ldconfig. This might help... Vaclav On Mon, May 21, 2007 at 12:32:15PM -0700, Mahatma Sarasvati wrote: > No, I'm not using sudo. Does it matter? > > M > > ----- Original Message ---- > From: Vaclav Kubart <vaclav.kubart(a)iptel.org> > To: Mahatma Sarasvati <gnoahhb(a)yahoo.com> > Cc: serusers(a)iptel.org > Sent: Wednesday, May 16, 2007 9:47:40 PM > Subject: Re: [Serusers] Problem running presence modules - SER won't start > > Hello, > how did you start ser? Aren't you using sudo? > Vaclav > > On Wed, May 16, 2007 at 11:24:07AM -0700, Mahatma Sarasvati wrote: > > Hello All, > > > > Running SER 2.0.0 rc1 from CVS (updated yesterday) on CentOS 5. > > > > I'm trying to test/use the presence functionality in SER. > > I've compiled and installed with make group_include="standard standard-dep" all > > > > I've set the environment variable: export LD_LIBRARY_PATH=/usr/local/lib/ser > > [root@vx1 modules]# echo $LD_LIBRARY_PATH > > /usr/local/lib/ser > > > > My supporting modules are here: > > > > [root@vx1 modules]# ls -l /usr/local/lib/ser/ > > total 828 > > -rwxr-xr-x 1 root root 40768 May 14 15:19 libbinrpc.so > > -rwxr-xr-x 1 root root 179044 May 14 15:19 lib_ser_cds.so > > -rwxr-xr-x 1 root root 213028 May 14 15:19 lib_ser_presence.so > > -rwxr-xr-x 1 root root 371056 May 14 15:19 lib_ser_xcap.so > > drwxr-xr-x 2 root root 4096 May 14 15:19 modules > > [root@vx1 modules]# > > > > Load module commands are in ser.cfg: > > loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" > > loadmodule "/usr/local/lib/ser/modules/xcap.so" > > loadmodule "/usr/local/lib/ser/modules/pa.so" > > loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" > > loadmodule "/usr/local/lib/ser/modules/rls.so" > > loadmodule "/usr/local/lib/ser/modules/dialog.so" > > > > libXML2 and libXML2-devel are installed. > > > > Yet, when I try to start SER I get this: > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (81,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (82,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory > > May 15 13:09:24 vx1 ser: parse error (83,13-14): failed to load module > > May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared object file: No such file or directory > > > > Can anyone help? I think I've done everything right. Why won't SER start? > > > > Thanks, > > M > > > > Here's my modules directory: > > [root@vx1 modules]# ls -l > > total 7264 > > -rwxr-xr-x 1 root root 67000 May 14 15:19 acc_db.so > > -rwxr-xr-x 1 root root 73248 May 14 15:19 acc_syslog.so > > -rwxr-xr-x 1 root root 69946 May 14 15:19 auth_db.so > > -rwxr-xr-x 1 root root 58071 May 14 15:19 auth.so > > -rwxr-xr-x 1 root root 37345 May 14 15:19 avp_db.so > > -rwxr-xr-x 1 root root 168768 May 14 15:19 avpops.so > > -rwxr-xr-x 1 root root 79110 May 14 15:19 avp.so > > -rwxr-xr-x 1 root root 279385 May 14 15:19 ctl.so > > -rwxr-xr-x 1 root root 72896 May 14 15:19 db_ops.so > > -rwxr-xr-x 1 root root 218639 May 14 15:18 dbtext.so > > -rwxr-xr-x 1 root root 97287 May 14 15:19 dialog.so > > -rwxr-xr-x 1 root root 63003 May 14 15:19 dispatcher.so > > -rwxr-xr-x 1 root root 21604 May 14 15:19 diversion.so > > -rwxr-xr-x 1 root root 114536 May 14 15:19 domain.so > > -rwxr-xr-x 1 root root 45704 May 14 15:19 enum.so > > -rwxr-xr-x 1 root root 109624 May 14 15:19 eval.so > > -rwxr-xr-x 1 root root 84357 May 14 15:19 exec.so > > -rwxr-xr-x 1 root root 81398 May 14 15:19 fifo.so > > -rwxr-xr-x 1 root root 53442 May 14 15:19 flatstore.so > > -rwxr-xr-x 1 root root 43414 May 14 15:19 gflags.so > > -rwxr-xr-x 1 root root 87574 May 14 15:19 lcr.so > > -rwxr-xr-x 1 root root 35883 May 14 15:19 maxfwd.so > > -rwxr-xr-x 1 root root 112635 May 14 15:19 mediaproxy.so > > -rwxr-xr-x 1 root root 87485 May 14 15:19 msilo.so > > -rwxr-xr-x 1 root root 106696 May 14 15:19 mysql.so > > -rwxr-xr-x 1 root root 187822 May 14 15:19 nathelper.so > > -rwxr-xr-x 1 root root 21203 May 14 15:19 options.so > > -rwxr-xr-x 1 root root 741640 May 14 15:19 pa.so > > -rwxr-xr-x 1 root root 131893 May 14 15:19 pdt.so > > -rwxr-xr-x 1 root root 272113 May 14 15:19 permissions.so > > -rwxr-xr-x 1 root root 122158 May 14 15:19 pike.so > > -rwxr-xr-x 1 root root 232133 May 14 15:19 presence_b2b.so > > -rwxr-xr-x 1 root root 26172 May 14 15:19 print.so > > -rwxr-xr-x 1 root root 47597 May 14 15:19 ratelimit.so > > -rwxr-xr-x 1 root root 152843 May 14 15:19 registrar.so > > -rwxr-xr-x 1 root root 402652 May 14 15:19 rls.so > > -rwxr-xr-x 1 root root 114896 May 14 15:19 rr.so > > -rwxr-xr-x 1 root root 63276 May 14 15:19 sanity.so > > -rwxr-xr-x 1 root root 66942 May 14 15:19 sl.so > > -rwxr-xr-x 1 root root 222033 May 14 15:18 sms.so > > -rwxr-xr-x 1 root root 34415 May 14 15:19 speeddial.so > > -rwxr-xr-x 1 root root 120807 May 14 15:19 textops.so > > -rwxr-xr-x 1 root root 43276 May 14 15:19 timer.so > > -rwxr-xr-x 1 root root 848962 May 14 15:19 tm.so > > -rwxr-xr-x 1 root root 117593 May 14 15:19 uac.so > > -rwxr-xr-x 1 root root 38793 May 14 15:19 unixsock.so > > -rwxr-xr-x 1 root root 38979 May 14 15:19 uri_db.so > > -rwxr-xr-x 1 root root 38603 May 14 15:19 uri.so > > -rwxr-xr-x 1 root root 351468 May 14 15:19 usrloc.so > > -rwxr-xr-x 1 root root 41867 May 14 15:19 xcap.so > > -rwxr-xr-x 1 root root 100354 May 14 15:19 xlog.so > > -rwxr-xr-x 1 root root 98456 May 14 15:19 xmlrpc.so > > [root@vx1 modules]# > > > > > > > > > > > > > > > > > > ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. > > http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > ____________________________________________________________________________________ > Don't pick lemons. > See all the new 2007 cars at Yahoo! Autos. > http://autos.yahoo.com/new_cars.html

1 0

[Users] Customize messages [max_contacts]
by jprevedello 22 May '07

22 May '07

Hi gyus, I configure my openser to restrict the number of register with the same user using the function [max_contacts], and functioned perfectly. # Allow no more than 1 contacts per AOR modparam("registrar", "max_contacts", 1) Whenever that a user tries to register using the user that already register, the system return the following message: "Registration error:503 - Service Unavailable" I need to customize the message informing the user the reason of error, something as "user already registered" (softphone). I effected some tests, but I did not have success. Something as: if( t_check_status("503")) { t_reply("503","User already registered!!!"); exit; } Any sugestion ? Regards

2 1

[Serusers] VON Europe
by Greger V. Teigre 22 May '07

22 May '07

Hi guys! Anyone going to VON Europe in Stockholm?! I will be there, so if there it's interest, we can arrange an informal users' meeting (or maybe a SER pub meeting :-) Jan has signaled his interest of participating remotely if possible. Regardless of users' meeting or not, I'm open to meeting anyone interested, so ping me :-) I'm staying at the Rica hotel next to the fairground and can thus easily meet. g-) PS! For those of you who find it strange that SER is not present as an exhibitor: That costs money and resources (in terms of volunteer presence and preparation). As iptel.org only relies on sponsors and part-time volunteers, and nobody has raised the topic on the mailing lists, we won't be exhibiting :-) I don't know if any of the sponsors of SER are present as exhibitors, they may be.

1 0

Re: [Serusers] SER/PSTN Gateway conf
by Greger Viken Teigre 22 May '07

22 May '07

Copy the list! I don't understand your problem. The BYE will be caught by the loose route section of the config and relayed. The standard getting started should work fine. g-) ------- Original message ------- From: flavio <flavio.patria(a)gmail.com> Sent: 12.5.'07, 15:09 > 2007/5/12, Greger V. Teigre <greger(a)teigre.com>: > > Flavio, > > SER doesn't reply 200 OK to BYE message, the SIP UA does that. So, you > > are fine, the BYE reaches the UA. > > g-) > > > > That's right. > > If my gateway send INVITE message to SER, it does proxy it to UAC reached! > So UAC (SIP Phone) and Gateway (UAC in SIP scenario) are involved in a > media session. > If my gateway send BYE message to SER, it does not proxy it to UAC. > So I could send reply (200 OK) to Gateway (It works fine, but is not > solution!), but I cannot proxy BYE to UAC. > > How can proxy this? > > Thanks, > > Flavio

4 7

[Serusers] Registrar / Expires & reject frequent refreshes
by Kostas Marneris 22 May '07

22 May '07

Hello all, I have an issue with some SIP clients (like Linksys SPA 1001) that have not comply with the "expires" parameter of the Contact header of a "200 OK" Registrar Response. My SER conf : modparam("registrar", "default_expires", 1800) modparam("registrar", "min_expires", 300) modparam("registrar", "max_expires", 1800) Although the (200 OK) response contains the 'expires' parameter : Contact: <sip:userb2@192.168.2.10:5061>;expires=300;received="sip:1.2.3.4:5061" the SIP client ignores it and keeps sending REGISTERs every 5 sec (if configured so). Is there a way for SER to reject these registrations with (423 Interval Too Brief), according to RFC3261 - 21.4.17 (423 Interval Too Brief) & 10.3 (Processing REGISTER Requests). The 'retry_after()' registrar modparam is not my case. thank you, Kostas

2 1

[Serusers] Call pickup
by Stefano Blasco 22 May '07

22 May '07

Dear all, i'm trying to have call pickup with SER (i use ser 0.9.6), i mean i'd like to have the capability to pick up a call that is directed to another user while the other user telephone is ringing. Is there someone that suceeded in this features? If it is not possible to get this feature by configuriation of ser (i mean using already present ser functions), is there someone that can suggest how to implemente such features? (for istance with external scripts,or with avpops, or using fifo, or writing a new ser module). Maybe the use of fifo could help, in particular i saw the ctd.sh example that is able to iniziate a call for two users using t_uac_dlg fifo function. I tryed to use t_uac_dlg to send a CANCEL to a transaction in progress (in the 180 ringing status) (taking by hand parameters like Call-ID, tag, branch etc.), but it seems that t_uac_dlg is not able to interact with the transaction in progress because it creates one different transaction. Is it real, or am i wrong? This is some log for this attempt: ################################################################ 1(17187) INFO: fifo_server: command empty 1(17187) fifo_get_method: method: 'CANCEL' 1(17187) DEBUG: fifo_get_ruri: 'sip:801@x.x.x' 1(17187) DEBUG: fifo_get_nexthop: next hop empty 1(17187) fifo_get_headers: headers: To: <sip:801@x.x.x> From: Stefano<sip:803@x.x.x>;tag=237abc61 Call-ID: 262a660822777d26 CSeq: 2 CANCEL 1(17187) parse_headers: flags=-1 1(17187) end of header reached, state=9 1(17187) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 1(17187) DEBUG: to body [<sip:801@x.x.x> ] 1(17187) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 1(17187) DEBUG: fifo_uac: parse_headers succeeded 1(17187) fifo_get_body: body: 1(17187) DEBUG: add_param: tag=237abc61 1(17187) end of header reached, state=29 1(17187) DEBUG: fifo_uac: EoL -- proceeding to transaction creation 1(17187) DEBUG:tm:t_uac: next_hop=<sip:801@x.x.x> 1(17187) DEBUG: mk_proxy: doing DNS lookup... 1(17187) get_record: lookup(_sip._udp.x.x.x, 33) failed 1(17187) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 1(17187) DEBUG:destroy_avp_list: destroying list (nil) 1(17187) DEBUG: dlg2hash: 55633 1(17187) DEBUG: add_to_tail_of_timer[4]: 0xb5ba0ac0 1(17187) DEBUG: add_to_tail_of_timer[0]: 0xb5ba0ad0 1(17187) **** done consume 2(17188) SIP Request: 2(17188) method: <CANCEL> 2(17188) uri: <sip:801@x.x.x> 2(17188) version: <SIP/2.0> 2(17188) parse_headers: flags=1 2(17188) Found param type 232, <branch> = <z9hG4bK159d.9169633.0>; state=16 2(17188) end of header reached, state=5 2(17188) parse_headers: Via found, flags=1 2(17188) parse_headers: this is the first via 2(17188) After parse_msg... 2(17188) preparing to run routing scripts... 2(17188) parse_headers: flags=128 2(17188) end of header reached, state=9 2(17188) DEBUG: get_hdr_field: <To> [28]; uri=[sip:801@x.x.x] 2(17188) DEBUG: to body [<sip:801@x.x.x> ] 2(17188) get_hdr_field: cseq <CSeq>: <2> <CANCEL> 2(17188) DEBUG: get_hdr_body : content_length=0 2(17188) found end of header 2(17188) DEBUG: is_maxfwd_present: max_forwards header not found! 2(17188) DEBUG: add_param: tag=237abc61 2(17188) end of header reached, state=29 2(17188) parse_headers: flags=256 2(17188) find_first_route: No Route headers found 2(17188) loose_route: There is no Route HF 2(17188) is_local(): Realm 'x.x.x' is local 2(17188) DEBUG: t_newtran: msg id=9 , global msg id=8 , T on entrance=0xffffffff 2(17188) parse_headers: flags=-1 2(17188) parse_headers: flags=60 2(17188) t_lookup_request: start searching: hash=55633, isACK=0 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookup_request: no transaction found 2(17188) DBG: trans=0xb5ba1488, callback type 1, id 0 entered 2(17188) parse_headers: flags=44 2(17188) DEBUG: t_lookupOriginalT: searching on hash entry 55633 2(17188) DEBUG: RFC3261 transaction matching failed 2(17188) DEBUG: t_lookupOriginalT: no CANCEL matching found! 2(17188) DEBUG: t_lookupOriginalT completed 2(17188) DEBUG: mk_proxy: doing DNS lookup... 2(17188) get_record: lookup(_sip._udp.x.x.x, 33) failed 2(17188) sip_resolvehost: no SRV record found for x.x.x, trying 'normal' lookup... 2(17188) check_via_address(x.x.x.x, x.x.x.x, 0) ################################################################ Another thing, i saw that t_uac_cancel, instead, is able to interact with an existing transaction, by sending CANCEL to called and 487 to caller (using t_lookup_callid function to get transaction from callid and cseq). Does someone think that it is possible to modify t_uac_cancel to handle basic pickup functions? (that should be CANCEL for the called side and redirect for caller towards pickup user) This is some log for this case: ################################################################## 1(16115) INFO: fifo_server: command empty 1(16115) DEBUG: fifo_uac_cancel: ############### begin ############## 1(16115) DEBUG: fifo_uac_cancel: callid="0b29c5266716aa67" 1(16115) DEBUG: fifo_uac_cancel: cseq="2" 1(16115) created comparable call_id header field: >Call-ID: 0b29c5266716aa67 < 1(16115) created comparable cseq header field: >CSeq: 2 INVITE< 1(16115) just locked hash index 37328, looking for transactions there: 1(16115) we have a match: callid=>>Call-ID: 0b29c5266716aa67 << cseq=>>CSeq: 2<< 1(16115) DEBUG: t_lookup_callid: transaction found. 1(16115) DEBUG: fifo_uac_cancel: now calling cancel_uacs 1(16115) DEBUG: cancel_branch: sending cancel... 1(16115) DEBUG: add_to_tail_of_timer[4]: 0xb5b763f0 1(16115) DEBUG: add_to_tail_of_timer[0]: 0xb5b76400 1(16115) DEBUG: fifo_uac_cancel: ################ end ############## ################################################################## Thank you very much for your help Stefano Blasco

2 1

[Serusers] behind 2 routers
by root 22 May '07

22 May '07

Hi, I have a DSL router, which has only one port. To connect multiple computers I have another 4 ports router behind this DSL router. So SER is behind two routers. How can I configure these two routers so that I can use SER in this situation? Thanks,

1 0

[Serusers] SIP Security and Monitoring tool running on iptel.org
by Tudor Golubenco 21 May '07

21 May '07

Hi SER users and devs, During the past few months we, at Iptego, have been working on a security and monitoring application for SIP sites, one of the recurring subjects of the SER user meeting in Prague ;-). To be short, VSP is a network intrusion detection system, which can sniff the VoIP traffic, parse SIP messages and create alerts when something seems wrong. As the project evolved, we have presented our work to the iptel.org staff and, together with them, decided to make the monitoring extension of VSP available to all public. So, if you're interested in real time statistics of the iptel.org SIP site, or in the repartition of User-Agents, you can create your own graphs here: http://fly.iptel.org/vsp/ We have already received great feedback from the iptel.org guys and I hope to get even more from the rest of community. Disclaimer: The project is BETA for now and the graphs haven't been verified yet. I wouldn't use them for anything important. Cheers, Tudor Golubenco -- Dipl. Ing. Tudor Golubenco Software Engineer iptego GmbH Am Borsigturm 40, 13507 Berlin, Germany http://www.iptego.de

5 8

[Users] x-lite and nat_uac_test(16)
by Alexander Bergolth 21 May '07

21 May '07

Hi! I've noticed that (at least on my boxes) x-lite uses a different source-port for the sip-connection than the one that is announced in the Via-header. (See the example below.) This behavior results in nat_uac_test(16) always reporting that the client is behind a NAT. Can anyone confirm this behavior? Is it a bug in X-Lite or is there any good reason for it? Cheers, --leo -------------------- 8< -------------------- No. Time Source Destination Protocol Info 4 0.000808 137.208.90.164 137.208.89.45 SIP Request: REGISTER sip:wu-wien.ac.at Frame 4 (640 bytes on wire, 640 bytes captured) Ethernet II, Src: DellEsgP_85:83:ea (00:0b:db:85:83:ea), Dst: Cisco_f3:2e:80 (00:0d:29:f3:2e:80) Internet Protocol, Src: 137.208.90.164 (137.208.90.164), Dst: 137.208.89.45 (137.208.89.45) Transmission Control Protocol, Src Port: 1831 (1831), Dst Port: 5060 (5060), Seq: 1, Ack: 1, Len: 586 Source port: 1831 (1831) Destination port: 5060 (5060) Sequence number: 1 (relative sequence number) [Next sequence number: 587 (relative sequence number)] Acknowledgement number: 1 (relative ack number) Header length: 20 bytes Flags: 0x18 (PSH, ACK) Window size: 1028096 (scaled) Checksum: 0x491e [correct] Session Initiation Protocol Request-Line: REGISTER sip:wu-wien.ac.at SIP/2.0 Message Header Via: SIP/2.0/TCP 137.208.90.164:1676;branch=z9hG4bK-d87543-bd24cf102007467f-1--d87543-;rport Max-Forwards: 70 Contact: <sip:30001@137.208.90.164:1676;rinstance=7ce462f932219953;transport=TCP> To: "Alexander Bergolth"<sip:30001@wu-wien.ac.at> From: "Alexander Bergolth"<sip:30001@wu-wien.ac.at>;tag=b904b134 Call-ID: N2VjNGQxYzZlOWNiYWU5OTMwNzQ4OTZkMWU3NmRkYTE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: X-Lite release 1006e stamp 34025 Content-Length: 0 -------------------- 8< -------------------- -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

5 10

[Users] WeSIP: Click2Call Linefeed
by Helmut Kuper 21 May '07

21 May '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello WeSIPer, I found a little but annoying Error in WeSIP's C2C application. I send back a 483 reply on a C2C's first caller-INVITE to WeSIP via openser's sl-send_reply(). I expected that C2C terminates the call function, cause C2C has an error handling method for 4xx replies, but nothing happens. openser as wesip's frontend received definately the reply. By chance I found, that the reason for this was my reply message. This doesn't work with WeSIP's C2C application: sl_send_reply("483", "My message\n"); But this works: sl_send_reply("483", "My message"); It seems, that C2C's or WeSIPs reply parser has some problems with linefeeds at the end of reply's message text. I use openser 1.2.0, WeSIP 0.1.4 and click2call-08_11_06 regards Helmut PS: Is there already a fix for WeSIP to reconnect to openser's seas socket, when openser restarts? -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGUa8R4tZeNddg3dwRAscvAKDC49yJRkA4ALOCoC5naJzr9gavMwCff7vX oEc4vGa8tEP5r0IVMhvC+jA= =dMZ6 -----END PGP SIGNATURE-----

1 0

[Users] Openser and Mediaproxy behind NAT
by Maciek Kowalski 21 May '07

21 May '07

Hi All, I am new to Openser and propably answer to my question is very simple.. I,ve been trying to run Openser with Mediaroxy behind NAT (they both work on a same machine with a private IP - 192.168.1.x). I am connected to internet through ADSL router (dynamic IP). I ve tried few configurations and I cant get it to work - I have "no audio problem" with nated users who register from outside of my LAN. I ve read that mediaproxy must listen on a public IP address, but first I thought that this is just a matter of port forwarding. And here is my question: Is it possible for this to work with described configuration? (Openser with mediaproxy both behind NAT) I ve used similar solution with Asterisk and Xtunnels and it worked fine. Thanks in advance and sorry for my English:). Best regards, Maciek Kowalski

2 1

[Serusers] sip uri transform
by Ҷ�� 21 May '07

21 May '07

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Hi. I have a question. I use x_lite do client,and ser server do route. if user1 is 1001 and user2 is 1002, assumes i used 1001 to dial 999991002, I thinking in the ser.conf, first to remove the prefix(99999),then route to 1002. if used asterisk the idea is ok but I don't know ser whether to achieve? [...] Content analysis details: (9.8 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_HELO_PASS SPF: HELO matches SPF record 4.1 FROM_ILLEGAL_CHARS From: has too many raw illegal characters -0.0 SPF_PASS SPF: sender matches SPF record 0.5 HTML_40_50 BODY: Message is 40% to 50% HTML 0.0 HTML_MESSAGE BODY: HTML included in message 0.0 BAYES_50 BODY: Bayesian spam probability is 40 to 60% [score: 0.5035] 0.0 MIME_HTML_ONLY BODY: Message only has text/html MIME parts 0.2 DNS_FROM_RFC_ABUSE RBL: Envelope sender in abuse.rfc-ignorant.org 2.2 RCVD_IN_WHOIS_INVALID RBL: CompleteWhois: sender on invalid IP block [202.105.45.47 listed in combined-HIB.dnsiplists.completewhois.com] 1.7 DNS_FROM_RFC_POST RBL: Envelope sender in postmaster.rfc-ignorant.org 1.1 HTML_MIME_NO_HTML_TAG HTML-only message, but there is no HTML tag The original message was not completely plain text, and may be unsafe to open with some email clients; in particular, it may contain a virus, or confirm that your address can receive spam. If you wish to view it, it may be safer to save it to a file and open it with an editor.

2 1

[Users] Presence Problem - Publish Offline
by Hildebrando Trannin 21 May '07

21 May '07

Hi all, I was testing OpenSER presence functionality using eyebeam and a softphone developed by my team. The OFFLINE state with eyebeam worked perfectly, but with my softphone it didn't work. I set force_active = 1. XML sent by PUBLISH and OpenSER log are below. Can anyone help me? Thanks, Hildebrando PIDF XML: <?xml xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:hildebrando@voip.maritaca"> <presence> <tuple id=""> <status> <basic>closed</basic> </status> <note>offline</note> </tuple> </presence> The basic difference between my application and eyebeam is that eyebeam uses RPID XML and I use PIDF XML. OPENSER LOG: 3(26039) PRESENCE: update_presentity: The presentity status is offline; do not send notify

2 1

[Serusers] Does SER's MYSQL work in SUSE Linux?
by Cyprian Clement 20 May '07

20 May '07

Hello Community, Does anyone know whether it's possible to install the current recommended MYSQL on SUSE Linux. I have a Machine with a SUSE installed on it. I have successfully installed SER 0.9.6 but now I would like to have MYSQL for authentication support. Does the SER's recommended MYSQL work with SUSE?. I only see the talk about Redhat and Debian so unsure if it works with SUSE. Many thanks for in advance. Cyprian. ---------- ___________________________________________________________ What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

2 1

[Users] Dropped calls with OpenSER 1.1.x and Asterisk >= 1.2.14
by Edoardo Serra 19 May '07

19 May '07

Hi guys, I'm having a problem about asterisk dropping calls received from OpenSER The problems was spotted at Asterisk users mailing lists http://lists.digium.com/pipermail/asterisk-users/2007-April/184875.html Asterisk complains about an ACK not received and drops the calls. It expects OpenSER to send an ACK back to its '200 OK' when the call is set up Is that a problem OpenSER should solve ??? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

2 3

[Users] Getting start
by Mhmd 19 May '07

19 May '07

HI , Kindly provide me with the link , where can i get the OPENSER , thank you Mohammad

2 1

[Users] openser as b2bua
by A.M. 19 May '07

19 May '07

The openser website clearly states that OpenSER is not a B2BUA. Is there a particular architectural limitation that would prevent a module from implementing a B2BUA within OpenSER? Cheers, M

3 2

[Users] openser and postgresql
by zhaojunjie01＠st.lzu.edu.cn 19 May '07

19 May '07

Hello: I use postgresql to authorize the user.But it cann't work.Who can give me some suggestion? Thank you very much! My error log and openser.cfg fileas follows: #Error log May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: INFO: statistics manager successfully initialized May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: PostgreSQL - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: StateLess module - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: TM - initializing... May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: Maxfwd module- initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: INFO:ul_init_locks: locks array size 512 May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: TextOPS - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: AUTH module - initializing May 18 12:24:43 th016184 /usr/local/sbin/openser[4358]: AUTH_DB module - initializing May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: register_udomain(): Cannot open database connection file! May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: domain_fixup(): Error while registering domain May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: ERROR: fix_actions: fixing failed (code=-1) at cfg line 145 May 18 12:24:44 th016184 /usr/local/sbin/openser[4358]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill #openser.cfg loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri=~"localhost") { # mark routing logic in request # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"localhost") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("localhost", "subscriber")) { www_challenge("localhost", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

1 0

[Users] pseudovariables in rewriteuser() - OpenSER 1.1.x
by Edoardo Serra 18 May '07

18 May '07

Hi all, I have the need to call rewriteuser() with a pseudovariable. My need is to rewrite the user with the content of an extra header I put in the SIP packet coming from a PSTN gateway I tried with rewriteuser("$hdr(X-DNID)"); but it doesn't replace the string with the ocntent of the pseudovariable. Any idea ? Tnx in advance Edoardo Serra WeBRainstorm S.r.l.

3 5

[Users] "Register problem"
by Danielo Garcia 18 May '07

18 May '07

Hello all. I have a problem with register I dont know why, when I try to register a user behind NAT listening in port 5060 the evalution of the sentence "if (uri==myself){" is true and when change the port for another for example 5070 the evaluation of the same sentence is false could somebody explain me? or how can I do the sentence "if (uri==myself){" using patern matching My configuration is listen = IP-MYSERVER port = 5060 Thanks..

2 1

[Users] using of myself Core Values
by Victor Gamov 18 May '07

18 May '07

hi is it possible to do following in config if ( $avp(s:var_name) != myself ) { # do something } var_name is AVP looking like "192.168.0.1" If not -- how I can check that $avp(s:var_name) is not one of my IP-address (without using DOMAIN or other module but by core only) ? Thanks! -- CU, Victor Gamov

2 2

Re: [Users] x-lite and nat_uac_test(16)
by Alexander 'Leo' Bergolth 18 May '07

18 May '07

Hi! On 05/18/2007 05:12 PM, Andreas Granig wrote: > Alexander Bergolth wrote: >> I've noticed that (at least on my boxes) x-lite uses a different >> source-port for the sip-connection than the one that is announced in the >> Via-header. (See the example below.) > > Are you sure there isn't any NAT or ALG in between? I've done the packet dump on both ends, on a box running x-lite (with the windows-firewall disabled) and on the proxy-host. Both show the same source/destination ip-adresses and ports, so I assume there cannot be any kind of ALG. However, there is some packet filter (not doing NAT) between the box and the proxy, maybe x-lite does some firewall-detection on startup and uses the wrong port-number in the via header intentionally to enforce the use of an rtp-proxy? I'll do some tests on a box that isn't protected by a packet filter on Monday. > By default, x-lite binds to local port 5060, but you've some > non-standard ports in there. So my guess is either a non-standard > port setting in x-lite and NAT, or a faulty ALG on the NAT device. I don't have any special settings in X-Lite, ICE is disabled, "Use local IP address" is enabled. I've noticed that your version of x-lite uses UDP, mine seems to use TCP by default. Cheers, --leo > Here's a trace using x-lite 2.0 r1105d (Linux): > > U 192.168.123.129:5060 -> <public IP>:5060 > REGISTER sip:<some domain> SIP/2.0. > Via: SIP/2.0/UDP 192.168.123.129:5060;rport;branch=z9hG4<snip> > > Cheers, > Andreas -- e-mail ::: Alexander.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

2 1

[Users] "Register problem"
by Danielo Garcia 18 May '07

18 May '07

Hello all. I have a problem with register I dont know why the condition if (uri==myself) { My configuration is listen = 148.226.26.245 port = 5060 Thanks

1 0

[Users] Mediaproxy running with Openser
by Lesley Liu 18 May '07

18 May '07

Hi Users, I'm trying to setup a mediaproxy 1.7.2 running with operser 1.2.0 on Fedora 6. I can see all service running up. But when I make a call, the SDP does not change the owner address in the INVITE to callee. Openser only report following error: May 18 09:25:51 localhost /sbin/openser[2808]: error: mediaproxy/sendMediaproxyCommand(): can't connect to MediaProxy Configure as attachment, can anyone help? Thanks in advance. Regards, Lesley

3 10

[Users] Question about forking
by Tim Madorma 18 May '07

18 May '07

Hi, I want to start openser with runsv which requires that the starting process run in the foreground. My problem is that I also want to listen on a couple of different ports. When I set forking = yes, it will listen on multiple ports, but runsv won't work. When I set forking=no, then openser will only listen on the first specified port. Is there any way around this? Can I have the starting process run in the foreground and fork other processes that listen to the ports in the background? Here is the error message: WARNING: no fork mode and more than one listen address found(will use only the the first one) Here are the associated configuration lines: fork=no children=32 # Local IP address/port pairs to listen to listen=65.185.233.55:5061 listen=65.185.233.55:5062 # Alias IP address/port pairs alias=65.185.233.104:5061 alias=65.185.233.104:5062 thanks, Tim

3 14

[Users] How to log the username of caller and callee to acc table?
by Charles Wang 18 May '07

18 May '07

Hi all, I ever used SER and I know it will save the From information such as 101(a)192.168.11.200 to sip_from or from_uri in its acc table. But in my openser, I also setup the acc module. Where is my caller info? I can find a lot of number such as "2806655523" in my from_tag. I wanna store the caller and callee info in my acc table when I make an internal call or an external call. Can anybody kind to help me to solve it? or give me some tips please. PS: I got this openser.cfg from http://www.sipwise.com/wizard. Best regards, Charles ######################################################################## # Parts of my openser.cfg Configuration ######################################################################## loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "failed_transaction_flag", 24) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "log_flag", 0) modparam("acc", "log_missed_flag", 0) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 25) modparam("acc", "db_missed_flag", 0) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_time_column", "time") modparam("acc", "acc_totag_column", "to_tag") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "detect_direction", 1) modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); if(!search("^Content-Length:[ ]*0")) { setbflag(6); } } if(is_method("BYE")) { setflag(24); # account failed transactions setflag(25); # account successful transactions } # mark as loose-routed for acc setflag(26); route(11); } if(is_method("REGISTER")) { route(10); } setflag(24); # account failed transactions setflag(25); # account successful transactions if(is_method("INVITE")) { route(12); } if(is_method("CANCEL") || is_method("ACK")) { route(16); } route(17); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:callee_fr_inv_timer)/g"); avp_delete("$avp(s:cfu)/g"); } ######################################################################## # Request route 'usr-preferences-caller' ############################################################ ############ route[3] { route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(13); exit; } if(is_avp_set("$avp(s:ringtimeout)/n")) { xlog("L_INFO", "Setting ring timeout to $avp(s:ringtimeout) secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_copy("$avp(s:ringtimeout)", "$avp(s:callee_fr_inv_timer)/d"); } } ######################################################################## # Request route 'acc-caller' ############################################################ ############ route[5] { xlog("L_INFO", "Setting acc source-leg for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$fU|$fd"); } ######################################################################## # Request route 'acc-callee' ######################################################################## route[6] { xlog("L_INFO", "Setting acc destination-leg for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:902)", "$avp(s:callee_uuid)|$rU|$rd"); } ######################################################################## # Request route 'acc-failure' ######################################################################## route[7] { xlog("L_INFO", "Accounting failed request for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); resetflag(24); acc_db_request("404", "acc"); } ######################################################################## # Request route 'clir' ######################################################################## route[8] { if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous"," sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[9] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[10] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[11] { if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); } } route(8); if(isbflagset(6) && !isflagset(22)) { setflag(22); fix_nated_contact(); use_media_proxy(); t_on_reply("2"); } else { t_on_reply("1"); } xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!isflagset(21)) { t_on_failure("2"); if(!isflagset(26)) { route(5); route(6); } } if(!t_relay()) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[12] { sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid) = "0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(allow_trusted()) { if(is_avp_set("$avp(s:peer_uuid)/s")) { # use tag-column from trusted-table as uuid for this caller avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d"); } else { # if no uuid is set, use "0" as default uuid $avp(s:caller_uuid) = "0"; } xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!is_domain_local("$rd")) { xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Relaying Denied"); exit; } setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } consume_credentials(); } } route(3); if(nat_uac_test("19")) { setbflag(6); } route(13); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[13] { if(lookup("aliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(15); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(14); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[14] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[15] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:0[0-9]+@") { # only route numeric users to PSTN xlog("L_INFO", "Loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(11); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[16] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[17] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } # Request route 'base-filter-failover' route[18] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); exit; } } # Reply route 'base-standard-reply' onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } # Reply route 'base-nat-reply' onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } -- Best Regards Charles

2 2

[Serusers] No Audio
by Arun Kumar 18 May '07

18 May '07

Hi I've configured my ser server with dedicated Media proxy server. When I call from IP to IP it works fine without any problem but when I try to dialout using ser there is no audio. But if I'll enable STUN in my phone I'm able to hear audio. my phones are behind Nat( Port rest con nat). Please guide me how to setup my ser or mediaproxy so I don't have to use STUN server. thanks arun

2 5

[Users] ACK loop after receiving 404 from remote GW
by Taylor Carpenter 18 May '07

18 May '07

I am getting a ACK loop after receiving a 404 from a remote GW. I am starting the call using sipp for testing. sipp -r 1 -l 1 -d 5000 -s 8989 -p 5061 -sn uac 10.14.18.22 The ngrep trace is showing Via and From set to 127.0.0.1 for the communication between OpenSER and the sipp system. The ngrep output and the routes from the openser.cfg are at http://pastebin.ca/493807 I assume the Via/From being set to 127.0.0.1 may be the issue, which I assume I could change from sipp, but I would also like OpenSER to catch the loop. Taylor

2 1

[Users] Openser 1.2 Presence Module
by Curt Moore 18 May '07

18 May '07

Hello all. I've been experimenting with the new presence module in openser-1.2 and have a few questions which hopefully someone can help answer. I'm using an SVN checkout of the 1.2 branch, r2237, and am testing with Polycom SoundPoint IP phones, firmware 2.1.1.0052. >From what I can tell, openser 1.2 does not fully support the xpidf content type, it only supoorts the winfo content type. I've looked through the code for the presence module extensively and it appears that whenever a SUBSCRIBE comes in where the Event header is simply "Event: presence" and the Accept header is "Accept: application/xpidf+xml", the message is acceped and a 202 is sent in reply but in any NOTIFY sent from openser as a result, there is no message body and the message does not contain the XML information for pidf. Just as a test, I changed the event field in active_watchers to "presence.winfo" and noticed that with this change, openser generates a NOTIFY with a message body which contains watcherinfo. This, of course, generates a 481 from the phone since it does not know about the "Event: presence.winfo" event type. Also as a test, I tried to use the pa module as I'd had success with this module in the past since it _does_ support xpidf. In 1.2 it appears that the database structure specified in the databse setup scripts does not agree with the table structure being used in the module. As a result, I could not get the pa module to work either. Am I overlooking some small detail? Please let me know if further information is needed in order to help diagnose and resolve this issue. Cheers, -Curt

2 3

[Users] Failure_route help
by Yago Bargalló 18 May '07

18 May '07

Hello, I'm new in Openser and now I'm trying to use a failure_route but it doesn't work and I don't know why. The version that I'm using is Openser 1.2. What I'm trying is to forward a call after fr_inv_timer to Asterisk but it doesn't work. This is my openser.cfg : (...) modparam("tm", "fr_inv_timer" , 8) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("172.16.51.150", "subscriber")) { www_challenge("172.16.51.150", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if(is_method("INVITE") && (uri=~"sip:2[0-9]+@")) { t_on_failure("1"); } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { #redirect to Asterisk rewritehostport("172.16.51.160:5060"); rewriteuser("2002"); }

3 2

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

Hi Users, I tried to compile openser-1.2.0-tls on Solaris 10. The gcc, bison, flex, make, libiconv packages are added. However, when command "make all", the message shows the following without further information at beginning. When command "make install", it only shows the first 4 lines as following. Please advise how to solve it. Thanks a lot. ; # pwd /opt/openser-1.2.0-tls # make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce Compiling action.c gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.2.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Compiling blacklists.c .... ; Regards, Lesley

2 1

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

Hi Users, I tried to compile openser-1.2.0-tls on Solaris 10. The gcc, bison, flex, make, libiconv packages are added. However, when command "make all", the message shows the following without further information at beginning. When command "make install", it only shows the first 4 lines as following. Please advise how to solve it. Thanks a lot. ; # pwd /opt/openser-1.2.0-tls # make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce Compiling action.c gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.2.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Compiling blacklists.c .... ; Regards, Lesley

1 0

[Users] Call forward when no answer
by Howard Tang 18 May '07

18 May '07

Hi All, I have followed a tutorial and set up Asterisk as a voice mail server. http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op… It works fine when the UA is offline. Now, I want a call forwarded to the Voice mail server when there is no answer from the UA after 60 seconds(UA is registered on the openser). What should I do? Below is my config (copy from the above link). # requests for Media server if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") { route(3); exit; } # mark transaction if user is in voicemail group if(is_method("INVITE") && !has_totag() && is_user_in("Request-URI","voicemail")) { xdbg("user [$ru] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri", "i:10"); setflag(2); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { # route to Asterisk Media Server prefix("1"); rewritehostport("10.10.10.11:5060"); route(1); } else { sl_send_reply("404", "Not Found"); exit; } }; # voicemail access # - *98 - listen caller's voice messages, being prompted for pin # - *981 - listen voice messages, being promted for mailbox and pin # - *98XXXX - leave voice message to XXXX # route[3] { # direct voicemail if (uri =~ "sip:\*98@" ) { rewriteuser("1"); xdbg("voicemail access\n"); } else if (uri =~ "sip:\*981@" ) { strip(4); rewriteuser("11"); } else if (uri =~ "sip:\*98.+@" ) { strip(3); prefix("1"); } else { xlog("unknown media extension $rU\n"); sl_send_reply("404", "Unknown media service"); exit; } # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); route(1); } failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); resetflag(2); route(1); }

3 6

RE: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Thank you both very much for this hints, I will change the versions I currently use immediately. Regards Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Thursday, May 17, 2007 5:55 PM To: Ovidiu Sas Cc: Schumann Sebastian; users(a)openser.org Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule I was just about to tell you this myself. Please take the svn version, either 1.2 or the trunk. Ovidiu Sas wrote: > Hi Sebastian, > > You should use the 1.2 svn branch (or maybe the trunk). There are > several fixes already present in the tree. There's no point testing > against the original 1.2 tarball. > > > Regards, > Ovidiu Sas

1 0

RE: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Alright, I just used the Debian 4.0 OpenSER distribution files. I will compile it by myself now and send you the Logs again. Thanks so far. Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Thursday, May 17, 2007 5:37 PM To: Schumann Sebastian Cc: users(a)openser.org Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule I have not found the source either. It would really help if you would compile openser with debug simbols activated (they are activated in the default Makefile) to see exactly which line the segm fault occures. Then, please post again the backtrace. Thanks, Anca Schumann Sebastian wrote: >Hello, > >here is the stderr console output (just the last lines which should be important as I saw there is a message size limit), and the gdb backtrace. >Do you see any hint for the crash? > >Best regards >Sebastian > >________________________________ > >Von: Anca-Maria Vamanu [mailto:anca@voice-system.ro] >Gesendet: Do 5/17/2007 16:57 >An: Schumann Sebastian >Cc: users(a)openser.org >Betreff: Re: [Users] OpenSER crashed with Segmentation Fault using PresenceModule > > > >Hello, > >Please post the log and the gdb backtrace. > >regards, > >Anca Vamanu > >Schumann Sebastian wrote: > > > >>Dear all >> >>I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. >> >>My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. >> >>I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. >> >>After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. >> >>Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. >> >>Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. >> >>Thanks for your help. >> >>Sebastian >> >> >> >> >> > > > > > >----------------------------------------------------------------------- - > > > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:bob@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bK0a73.6d721e06.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-666f5148e9613c52-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:bob@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [25]; uri=[sip:bob@10.96.115.125] > 0(4591) DEBUG: to body [<sip:bob@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=14240, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe:Missing or unsupported event header field value > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:handle_subscribe: ERROR occured > 0(4591) DEBUG:tm:UNREF_UNSAFE: after is 0 > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) receive_msg: cleaning up > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:bob@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bKc989.6f967b16.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-4e0baf255d4f5c21-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:bob@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [25]; uri=[sip:bob@10.96.115.125] > 0(4591) DEBUG: to body [<sip:bob@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=39068, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe:Missing or unsupported event header field value > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:handle_subscribe: ERROR occured > 0(4591) DEBUG:tm:UNREF_UNSAFE: after is 0 > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) receive_msg: cleaning up > 0(4591) SIP Request: > 0(4591) method: <SUBSCRIBE> > 0(4591) uri: <sip:sebastian@10.96.115.125> > 0(4591) version: <SIP/2.0> > 0(4591) parse_headers: flags=2 > 0(4591) Found param type 232, <branch> = <z9hG4bKe413.3434b775.0>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=2 > 0(4591) parse_headers: this is the first via > 0(4591) After parse_msg... > 0(4591) preparing to run routing scripts... > 0(4591) parse_headers: flags=100 > 0(4591) Found param type 232, <branch> = <z9hG4bK-d87543-2a56156bb30b720d-1--d87543->; state=6 > 0(4591) Found param type 235, <rport> = <32800>; state=16 > 0(4591) end of header reached, state=5 > 0(4591) parse_headers: Via found, flags=100 > 0(4591) parse_headers: this is the second via > 0(4591) DEBUG:maxfwd:is_maxfwd_present: value = 69 > 0(4591) DEBUG: t_newtran: T on entrance=0xffffffff > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) DEBUG:parse_to:end of header reached, state=10 > 0(4591) DBUG:parse_to: display={}, ruri={sip:sebastian@10.96.115.125} > 0(4591) DEBUG: get_hdr_field: <To> [31]; uri=[sip:sebastian@10.96.115.125] > 0(4591) DEBUG: to body [<sip:sebastian@10.96.115.125> >] > 0(4591) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> > 0(4591) DEBUG: get_hdr_body : content_length=0 > 0(4591) found end of header > 0(4591) parse_headers: flags=78 > 0(4591) t_lookup_request: start searching: hash=12622, isACK=0 > 0(4591) DEBUG: RFC3261 transaction matching failed > 0(4591) DEBUG: t_lookup_request: no transaction found > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) PRESENCE: handle_subscribe: 'expires' found > 0(4591) PRESENCE: handle_subscribe: lexpire= 3600 > 0(4591) PRESENCE: handle_subscribe: 'To' header ALREADY PARSED: <sip:sebastian@10.96.115.125> > 0(4591) PRESENCE:handle_subscribe: 'From' header not parsed > 0(4591) DEBUG: add_param: tag=e035dc2e > 0(4591) DEBUG:parse_to:end of header reached, state=29 > 0(4591) DBUG:parse_to: display={"Bob"}, ruri={sip:bob@10.96.115.125} > 0(4591) PRESENCE:handle_subscribe: generating to_tag > 0(4591) PRESENCE :proces_rr: out rr [<sip:10.96.115.125;lr=on;ftag=e035dc2e>] > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE: update_subscribtion ... > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) expires: 3600 > 0(4591) PRESENCE:update_subscribtion:Inserting into database: >n_query_cols:14 > 0(4591) [0] = to_user sebastian > 0(4591) [1] = to_domain 10.96.115.125 > 0(4591) [2] = from_user bob > 0(4591) [3] = from_domain 10.96.115.125 > 0(4591) [4] = event presence > 0(4591) [5] = event_id > 0(4591) [6] = callid YTUwN2U3MmViZDM3NDNmZDdkMTllNTQ4MDgyN2ExMDk. > 0(4591) [7] = to_tag 10.4591.1179421870.7 > 0(4591) [8] = from_tag e035dc2e > 0(4591) [9] = contact sip:bob@10.96.100.128:32800 > 0(4591) [10] = status active > 0(4591) [11] = cseq 1 > 0(4591) parse_headers: flags=ffffffffffffffff > 0(4591) check_via_address(10.96.115.125, 10.96.115.125, 0) > 0(4591) PRESENCE:get_subs_dialog:querying database table = active_watchers > 0(4591) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= sebastian,[domain]= 10.96.115.125 for [event]= presence.winfo returned no result > 0(4591) PRESENCE:query_db_notify: Could not get subs_dialog from database > 0(4591) PRESENCE:update_subscribtion:Could not send notify for presence.winfo > 0(4591) PRESENCE:notify:dialog informations: > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE:get_xcap_tree:The query in table xcap for [username]=sebastian , domain=10.96.115.125 returned result 0(4591) PRESENCE:get_xcap_tree: xcap body: ><?xml version="1.0" encoding="UTF-8"?> ><ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules" xmlns:cp="www.counterpath.com/privacy-lists"> ><rule id="whitelist"> ><conditions> ><identity> ><id entity="sip:bob@10.96.115.125"/> ></identity> ></conditions> ><actions> ><pr:sub-handling>allow</pr:sub-handling> ></actions> ><transformations> ><pr:provide-services> ><pr:all-services/> ></pr:provide-services> ><pr:provide-persons> ><pr:all-persons/> ></pr:provide-persons> ><pr:provide-devices> ><pr:all-devices/> ></pr:provide-devices> ><pr:provide-all-attributes/> ></transformations> ></rule> ><rule id="blacklist"> ><conditions> ></conditions> ><actions> ><pr:sub-handling>block</pr:sub-handling> ></actions> ><transformations/> ></rule> ><cp:privacy-lists> ><cp:list name="whitelist_uri_phone"> ></cp:list> ><cp:list name="blacklist_uri_phone"> ></cp:list> ><cp:list name="whitelist_domain_phone"> ></cp:list> ><cp:list name="blacklist_domain_phone"> ></cp:list> ><cp:list name="whitelist_uri_IM"> ></cp:list> ><cp:list name="blacklist_uri_IM"> ></cp:list> ><cp:list name="whitelist_domain_IM"> ></cp:list> ><cp:list name="blacklist_domain_IM"> ></cp:list> ></cp:privacy-lists> ></ruleset> 0(4591) presence:uandd_to_uri: uri=sip:bob@10.96.115.125 > 0(4591) PRESENCE:is_watcher_allowed:node1->name= rule > 0(4591) PRESENCE:is_watcher_allowed:cond_node->name= conditions > 0(4591) PRESENCE:is_watcher_allowed:actions_node->name= actions > 0(4591) PRESENCE:is_watcher_allowed:sub_handling_node->name= sub-handling > 0(4591) PRESENCE:is_watcher_allowed:sub_handling_node->content= allow > 0(4591) PRESENCE:notify: [status]=active > 0(4591) PRESENCE:get_p_notify_body: querying presentity > 0(4591) PRESENCE: get_p_notify_body: The query returned no result >There is no presence information recorded forthe user > 0(4591) PRESENCE:notify: Could not get the notify_body > 0(4591) presence:uandd_to_uri: uri=sip:sebastian@10.96.115.125 > 0(4591) PRESENCE: notify: build notify to user= bob domain= 10.96.115.125 for event= presence > 0(4591) > [p_user]= sebastian [p_domain]= 10.96.115.125 > [w_user]= bob [w_domain]= 10.96.115.125 > 0(4591) [event]= presence > [staus]= active > [expires]= 3600 > 0(4591) [to_tag]= 10.4591.1179421870.7 > [from_tag]= e035dc2e > 0(4591) PRESENCE:build_str_hdr: expires = 3600 > 0(4591) PRESENCE:build_str_hdr: subs_expires : 3600 > 0(4591) PRESENCE: build_str_hdr: headers: >Event: presence >Contact: <sip:10.96.115.128:5060> >Subscription-State: active;expires=3600 >Content-Type: application/pidf+xml > > 0(4591) PRESENCE:notify: headers:Event: presence >Contact: <sip:10.96.115.128:5060> >Subscription-State: active;expires=3600 >Content-Type: application/pidf+xml > > 0(4591) CONTACT = sip:bob@10.96.100.128:32800 > 0(4591) presence:uandd_to_uri: uri=sip:bob@10.96.115.125 > 0(4591) DEBUG:tm:t_uac: next_hop=<sip:10.96.115.125;lr=on;ftag=e035dc2e> > 0(4591) DEBUG: mk_proxy: doing DNS lookup... > 0(4591) DEBUG:destroy_avp_list: destroying list (nil) > 0(4591) DEBUG: dlg2hash: 12619 > 0(4591) print_request_uri: sip:bob@10.96.100.128:32800 > 0(4591) DEBUG:tm:set_timer: relative timeout is 500000 > 0(4591) DEBUG: add_to_tail_of_timer[4]: 0xb5aaefec (41700000) > 0(4591) DEBUG:tm:set_timer: relative timeout is 30 > 0(4591) DEBUG: add_to_tail_of_timer[0]: 0xb5aaf008 (71) > 1(4592) DEBUG: timer routine:4,tl=0xb5aa825c next=0xb5aaefec, timeout=41700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(4592) DEBUG: add_to_tail_of_timer[5]: 0xb5aa825c (42700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:4,tl=0xb5aaefec next=(nil), timeout=41700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 1000000 > 1(4592) DEBUG: add_to_tail_of_timer[5]: 0xb5aaefec (42700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:5,tl=0xb5aa825c next=0xb5aaefec, timeout=42700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(4592) DEBUG: add_to_tail_of_timer[6]: 0xb5aa825c (44700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:5,tl=0xb5aaefec next=(nil), timeout=42700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 2000000 > 1(4592) DEBUG: add_to_tail_of_timer[6]: 0xb5aaefec (44700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:6,tl=0xb5aa825c next=0xb5aaefec, timeout=44700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (48700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:6,tl=0xb5aaefec next=(nil), timeout=44700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (48700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:2,tl=0xb5aa9bc8 next=(nil), timeout=46 > 1(4592) DEBUG: wait_handler : removing 0xb5aa9b80 from table > 1(4592) DEBUG: delete transaction 0xb5aa9b80 > 1(4592) DEBUG: wait_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=48700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (52700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=48700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (52700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=52700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (56700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=52700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (56700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=56700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (60700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=56700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (60700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=60700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (64700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=60700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (64700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=64700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (68700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=64700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (68700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=68700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aa8110, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aa825c (72700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=68700000 > 1(4592) DEBUG: retransmission_handler : request resending (t=0xb5aaeea0, NOTIFY si ... ) > 1(4592) DEBUG:tm:set_timer: relative timeout is 4000000 > 1(4592) DEBUG: add_to_tail_of_timer[7]: 0xb5aaefec (72700000) > 1(4592) DEBUG: retransmission_handler : done > 1(4592) DEBUG: timer routine:0,tl=0xb5aa8278 next=0xb5aaf008, timeout=71 > 1(4592) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb5aa8110) > 1(4592) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > 1(4592) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > 1(4592) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > 1(4592) DEBUG:tm:local_reply: local transaction completed > 1(4592) DBG: trans=0xb5aa8110, callback type 256, id 0 entered > 1(4592) PRESENCE:p_tm_callback: completed with status 408 [watcher_id:0xb5aaa9cc/10.4591.1179421870.4] > 1(4592) DEBUG: add_to_tail_of_timer[2]: 0xb5aa8158 (76) > 1(4592) DEBUG: final_response_handler : done > 1(4592) DEBUG: timer routine:0,tl=0xb5aaf008 next=(nil), timeout=71 > 1(4592) DEBUG: final_response_handler:stop retr. and send CANCEL (0xb5aaeea0) > 1(4592) DEBUG:tm:t_should_relay_response: T_code=0, new_code=408 > 1(4592) DEBUG:tm:t_pick_branch: picked branch 0, code 408 > 1(4592) DEBUG:tm:local_reply: branch=0, save=0, winner=0 > 1(4592) DEBUG:tm:local_reply: local transaction completed > 1(4592) DBG: trans=0xb5aaeea0, callback type 256, id 0 entered > 1(4592) PRESENCE:p_tm_callback: completed with status 408 [watcher_id:0xb5aaacf4/10.4591.1179421870.7] > 1(4592) DEBUG: add_to_tail_of_timer[2]: 0xb5aaeee8 (76) > 1(4592) DEBUG: final_response_handler : done > 1(4592) DEBUG: timer routine:7,tl=0xb5aa825c next=0xb5aaefec, timeout=72700000 > 1(4592) DEBUG: timer routine:7,tl=0xb5aaefec next=(nil), timeout=72700000 > 1(4592) DEBUG: timer routine:2,tl=0xb5aa8158 next=0xb5aaeee8, timeout=76 > 1(4592) DEBUG: wait_handler : removing 0xb5aa8110 from table > 1(4592) DEBUG: delete transaction 0xb5aa8110 > 1(4592) DEBUG: wait_handler : done > 1(4592) DEBUG: timer routine:2,tl=0xb5aaeee8 next=(nil), timeout=76 > 1(4592) DEBUG: wait_handler : removing 0xb5aaeea0 from table > 1(4592) DEBUG: delete transaction 0xb5aaeea0 > 1(4592) DEBUG: wait_handler : done > 1(4592) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > 1(4592) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information > 1(4592) PRESENCE:msg_presentity_clean:cleaning expired presentity information > 1(4592) PRESENCE: msg_watchers_clean:cleaning pending subscriptions > 1(4592) PRESENCE: msg_active_watchers_clean:cleaning expired watcher information > 1(4592) PRESENCE:msg_presentity_clean:cleaning expired presentity information > > >----------------------------------------------------------------------- - > > >... > >(no debugging symbols found) >Core was generated by `openser -f openser.cfg -w .'. >Program terminated with signal 11, Segmentation fault. >#0 0x0809e332 in fm_free () >(gdb) bt >#0 0x0809e332 in fm_free () >#1 0xb7b7d67a in handle_subscribe () from /usr/lib/openser/modules/presence.so >#2 0x08051224 in do_action () >#3 0x08052f8c in run_action_list () >#4 0x080521af in do_action () >#5 0x08052f8c in run_action_list () >#6 0x080525a1 in do_action () >#7 0x08052f8c in run_action_list () >#8 0x08053247 in run_top_route () >#9 0x08074e04 in receive_msg () >#10 0x0809832d in udp_rcv_loop () >#11 0x080664f4 in main_loop () >#12 0x08067f25 in main () >(gdb) quit > >

3 2

AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Hello, here is the stderr console output (just the last lines which should be important as I saw there is a message size limit), and the gdb backtrace. Do you see any hint for the crash? Best regards Sebastian ________________________________ Von: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Gesendet: Do 5/17/2007 16:57 An: Schumann Sebastian Cc: users(a)openser.org Betreff: Re: [Users] OpenSER crashed with Segmentation Fault using PresenceModule Hello, Please post the log and the gdb backtrace. regards, Anca Vamanu Schumann Sebastian wrote: >Dear all > >I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. > >My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. > >I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. > >After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. > >Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. > >Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. > >Thanks for your help. > >Sebastian > > >

2 1

[Users] OpenSER crashed with Segmentation Fault using Presence Module
by Schumann Sebastian 17 May '07

17 May '07

Dear all I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. Thanks for your help. Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 910 643010 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk

2 1

[Users] failure_route
by Yago Bargalló 17 May '07

17 May '07

Hello all, I don't know why my openser.cfg doesn't work correctly, I want my openser to catch the failure_route after 10 seconds to reroute a call to a Asterisk when nobody responds the phone on an openser. Does anybody know what I'm doing wrong?? Thanks in advance!!!!!!!!! Openser.cfg: modparam("tm","fr_timer",4) modparam("tm", "fr_inv_timer" ,10) # ------------------------- request routing logic ------------------- # main routing logic (...) route{ (...) # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("172.16.51.150", "subscriber")) { www_challenge("172.16.51.150", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # if(uri=~"sip:2[0-9]+@") { t_on_failure("1"); # rewritehostport("172.16.51.160:5060"); # t_relay(); # rewriteuser("2002"); # exit; } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { #reroute to asterisk rewritehost("172.16.51.150"); rewriteuser("2002"); t_relay(); exit; }

2 1

[Serusers] 407 Proxy Authentication Required to a subscribed user
by flavio 17 May '07

17 May '07

Hi all, I'm testing Blind Trasfer with my IP Phones, SER and GW, in this scenario: AnalogPhone (GW) is the Transferee IP Phone1, registred to SER, is the Transferor IP Phone 2, registred to SER, is the Transfer Target When the Transferor send INVITE to put on hold the Trasferee, SER reply with 407 Proxy Authentication Requrire, as follow reported: 2007/05/17 12:45:49.370808 10.28.19.124:5060 -> 10.28.19.202:5060 INVITE sip:06XXXXXX15@10.28.19.230 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. Route: <sip:10.28.19.202;ftag=1c1272914830;lr=on>. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Contact: <sip:06XXXXXX14@10.28.19.124;user=phone>. Supported: replaces. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. User-Agent: Grandstream BT110 1.0.8.12. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 206. . v=0. o=0660522014 8000 8001 IN IP4 10.28.19.124. s=SIP Call. c=IN IP4 0.0.0.0. t=0 0. m=audio 5004 RTP/AVP 8 18 0. a=sendonly. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2007/05/17 12:45:49.371204 10.28.19.202:5060 -> 10.28.19.124:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. Proxy-Authenticate: Digest realm="10.28.19.202", nonce="464c4fa95f251088d01666190944e53313bab22e". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=15940 req_src_ip=10.28.19.124 req_src_port=5060 in_uri=sip:0660522015@10.28.19.230 out_uri=sip:0660522015@10.28.19.230 via_cnt==1". My question is: why SER reply with an authentication request to INVITE, if the Transferor party is a registred user? Have you any suggestion about? I verify user subscribtion in my ser.cfg file as follow: if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } Thanks for support, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

2 4

[Serusers] looped requests, syn_branch==1 => branch value is repeated for 2 Ts
by JF 17 May '07

17 May '07

Hi all, I have observed when looping a request through SER (using t_relay_to("127.0.0.1", 5060)) that sometimes the via branch parameter value generated when the request goes out is kept equal to the value in the previous via (when this request was looped), and two equal vias go out. This causes problems with on_reply routes. On_reply routes are then called only once, instead of two. Since these are 2 independent transactions, this should happen twice. Browsing through tm code, it seems that this is caused by the usage of syn_branch == 1 in main.c (the default), leading to the following function calls when an initial INVITE loops: tm/t_fwd.c: t_forward_nonack add_uac print_uac_request tm/t_msgbuilder.c: t_calc_branch msg_translator.c: branch_builder <-- here, when syn_branch==1, the transaction 'label' value is used for the branch value, but according to tm/h_table.h, this value is "sequence number within hash collision slot". Is this the reason why some times the branch value is the same in two transactions looped through SER? The label value can perfectly be the same across 2 transactions not in the same hash slot... right? Maybe this has nothing to do with the request being looped, or it just makes this situation more probable? Anyone else can replicate this behaviour? Thanks, João Filipe Plácido -- PT Inovação

1 0

[Users] No answer when my openser 1.2.0 make a call to asterisk
by Charles Wang 17 May '07

17 May '07

Hi all, I try to make a call from my Openser 1.2.0 (SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip debug in my asterisk CLI. In the message on my openser, I can see as below: openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 But the To URI is still 00862123456789(a)openser.domain.name not 02123456789(a)asterisk.ip.addr. Is it correct? Does anybody kind to help me to solve it or give me some tips please? Best regards, Charles ##################### message on my Openser ################################# openser[16802]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: Proxy authentication failed - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[3] Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[1] Clear Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear caller preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Load caller preferences for uuid '1' - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[13] Invite Find Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[2] Clear Usr Preferences Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear callee preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Callee is not local - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[15] Invite To External - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Loading PSTN gateways - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[8] Clir - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 localhost mediaproxy[1905]: command request 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 192.168.11.9:57538:audio 61.217.xxx.xxx openser.domain.name local asterisk.ip.addr remote X-PRO=20build=201082 info=from:101@openser.domain.name, to:00862123456789@openser.domain.name,fromtag:1111281943,totag: localhost mediaproxy[1905]: session 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9: started. listening on 60.250.132.204:35076 localhost mediaproxy[1905]: command execution time: 3.04 ms openser[16801]: Request leaving server - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 ################# my asterisk CLI ######################## [root@asterisk ~]# asterisk -rvvvvvvvvvvvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding iaxusers to mysql/asterisk/iaxfriends == Binding iaxpeers to mysql/asterisk/iaxfriends == Binding queues to mysql/asterisk/queue_table == Binding queue_members to mysql/asterisk/queue_member_table Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster(a)digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.16 currently running on asterisk (pid = 26311) Verbosity is at least 14 -- Remote UNIX connection asterisk*CLI> sip debug SIP Debugging re-enabled asterisk*CLI> ######### my command running on asterisk machine: "ngrep -t -W byline -d any port 5060" #################### interface: any filter: (ip) and ( port 5060 ) # U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536 ;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: < sip:0028863939749001@my.openser.domain.name>. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9 . CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9 . s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:37.325722 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:39.325425 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Best Regards Charles -- Best Regards Charles

2 2

[Serusers] Problem running presence modules - SER won't start
by Mahatma Sarasvati 17 May '07

17 May '07

Hello All, Running SER 2.0.0 rc1 from CVS (updated yesterday) on CentOS 5. I'm trying to test/use the presence functionality in SER. I've compiled and installed with make group_include="standard standard-dep" all I've set the environment variable: export LD_LIBRARY_PATH=/usr/local/lib/ser [root@vx1 modules]# echo $LD_LIBRARY_PATH /usr/local/lib/ser My supporting modules are here: [root@vx1 modules]# ls -l /usr/local/lib/ser/ total 828 -rwxr-xr-x 1 root root 40768 May 14 15:19 libbinrpc.so -rwxr-xr-x 1 root root 179044 May 14 15:19 lib_ser_cds.so -rwxr-xr-x 1 root root 213028 May 14 15:19 lib_ser_presence.so -rwxr-xr-x 1 root root 371056 May 14 15:19 lib_ser_xcap.so drwxr-xr-x 2 root root 4096 May 14 15:19 modules [root@vx1 modules]# Load module commands are in ser.cfg: loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" libXML2 and libXML2-devel are installed. Yet, when I try to start SER I get this: May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (81,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (82,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (83,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared object file: No such file or directory Can anyone help? I think I've done everything right. Why won't SER start? Thanks, M Here's my modules directory: [root@vx1 modules]# ls -l total 7264 -rwxr-xr-x 1 root root 67000 May 14 15:19 acc_db.so -rwxr-xr-x 1 root root 73248 May 14 15:19 acc_syslog.so -rwxr-xr-x 1 root root 69946 May 14 15:19 auth_db.so -rwxr-xr-x 1 root root 58071 May 14 15:19 auth.so -rwxr-xr-x 1 root root 37345 May 14 15:19 avp_db.so -rwxr-xr-x 1 root root 168768 May 14 15:19 avpops.so -rwxr-xr-x 1 root root 79110 May 14 15:19 avp.so -rwxr-xr-x 1 root root 279385 May 14 15:19 ctl.so -rwxr-xr-x 1 root root 72896 May 14 15:19 db_ops.so -rwxr-xr-x 1 root root 218639 May 14 15:18 dbtext.so -rwxr-xr-x 1 root root 97287 May 14 15:19 dialog.so -rwxr-xr-x 1 root root 63003 May 14 15:19 dispatcher.so -rwxr-xr-x 1 root root 21604 May 14 15:19 diversion.so -rwxr-xr-x 1 root root 114536 May 14 15:19 domain.so -rwxr-xr-x 1 root root 45704 May 14 15:19 enum.so -rwxr-xr-x 1 root root 109624 May 14 15:19 eval.so -rwxr-xr-x 1 root root 84357 May 14 15:19 exec.so -rwxr-xr-x 1 root root 81398 May 14 15:19 fifo.so -rwxr-xr-x 1 root root 53442 May 14 15:19 flatstore.so -rwxr-xr-x 1 root root 43414 May 14 15:19 gflags.so -rwxr-xr-x 1 root root 87574 May 14 15:19 lcr.so -rwxr-xr-x 1 root root 35883 May 14 15:19 maxfwd.so -rwxr-xr-x 1 root root 112635 May 14 15:19 mediaproxy.so -rwxr-xr-x 1 root root 87485 May 14 15:19 msilo.so -rwxr-xr-x 1 root root 106696 May 14 15:19 mysql.so -rwxr-xr-x 1 root root 187822 May 14 15:19 nathelper.so -rwxr-xr-x 1 root root 21203 May 14 15:19 options.so -rwxr-xr-x 1 root root 741640 May 14 15:19 pa.so -rwxr-xr-x 1 root root 131893 May 14 15:19 pdt.so -rwxr-xr-x 1 root root 272113 May 14 15:19 permissions.so -rwxr-xr-x 1 root root 122158 May 14 15:19 pike.so -rwxr-xr-x 1 root root 232133 May 14 15:19 presence_b2b.so -rwxr-xr-x 1 root root 26172 May 14 15:19 print.so -rwxr-xr-x 1 root root 47597 May 14 15:19 ratelimit.so -rwxr-xr-x 1 root root 152843 May 14 15:19 registrar.so -rwxr-xr-x 1 root root 402652 May 14 15:19 rls.so -rwxr-xr-x 1 root root 114896 May 14 15:19 rr.so -rwxr-xr-x 1 root root 63276 May 14 15:19 sanity.so -rwxr-xr-x 1 root root 66942 May 14 15:19 sl.so -rwxr-xr-x 1 root root 222033 May 14 15:18 sms.so -rwxr-xr-x 1 root root 34415 May 14 15:19 speeddial.so -rwxr-xr-x 1 root root 120807 May 14 15:19 textops.so -rwxr-xr-x 1 root root 43276 May 14 15:19 timer.so -rwxr-xr-x 1 root root 848962 May 14 15:19 tm.so -rwxr-xr-x 1 root root 117593 May 14 15:19 uac.so -rwxr-xr-x 1 root root 38793 May 14 15:19 unixsock.so -rwxr-xr-x 1 root root 38979 May 14 15:19 uri_db.so -rwxr-xr-x 1 root root 38603 May 14 15:19 uri.so -rwxr-xr-x 1 root root 351468 May 14 15:19 usrloc.so -rwxr-xr-x 1 root root 41867 May 14 15:19 xcap.so -rwxr-xr-x 1 root root 100354 May 14 15:19 xlog.so -rwxr-xr-x 1 root root 98456 May 14 15:19 xmlrpc.so [root@vx1 modules]# ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow

3 2

[Users] Help (handle status)
by Jeferson Prevedello 17 May '07

17 May '07

Hi guys, I able the control of simultaneous users with the function [max_contacts], but whenever the user is blocked the softphone (X-lite) show the following message: [Registration error: 503 - Service Unavailable]. I tried to adapt the message conform code below, but not resolve: if (t_check_status("503")) { sl_send_reply("503","my new reply"); exit; } Any sugestion ? Regards Jeferson

2 1

[Users] changing log level of certain messages
by Dimo 17 May '07

17 May '07

Hi guys, I was wondering if it is possible to change the log level of some messages, namely: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed rc_send_server: no reply from RADIUS server xxxx.:1812 I have alot of clutter from this one and i would like to lower its severity so it does not get logged, but i still need other messages from radius to be logged. Apart from going into the code and changing the log level do you know of any other way to do it? Best. Dimo

2 1

[Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in you package , in you readme , you have explain to how I install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can I get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan

2 1

RE: [Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

hi can anyone advise . how can i configure the mediaproxy with sipxfoundry sipxces ... if possible. as long as some files needs openser or ser . thanks Mohammad ________________________________ From: Lesley Liu [mailto:lesley1025@gmail.com] Sent: Thu 5/17/2007 12:00 PM To: Mhmd; Users(a)openser.org Subject: Re: [Users] Mediaproxy Mohammad, mediaproxy.so is one module of SER or OpenSER, it can be find in /usr/local/lib/openser/modules/ or simular path. Regards, Lesley 2007/5/17, Mhmd <mohammad(a)guestware.net>: Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in you package , in you readme , you have explain to how I install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can I get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Users] Mediaproxy
by Mhmd 17 May '07

17 May '07

Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in your package , in your readme , you have explain how do i install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can i get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan

1 0

[Users] Openser trusted cache table
by Thiago Maluf 16 May '07

16 May '07

I would like to work with Module Parameter DB_MODE in Permissions Module set to 1 so I can use cache to optimize the request to function allow_trusted in trusted table . But this table may change once in a while. Taking a look in the source, I didn't find how the cache is updated without my interaction. My doubt is if Cache is updated automatically or if I have to use MI_FIFO or change the source to do this. Thanks in advanced. Thiago Maluf. -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

1 0

[Users] 1.2.0 LCR schema error or my method of load_gw?
by Charles Wang 16 May '07

16 May '07

Hi all, I got this openser.cfg from http://www.sipwise.com/wizard . I tried to make a offnet call prefix with 00. I try to load_gws and next_gw for lcr. But I can't find out such as "select xxx from lcr" or "select xxx from gw" during these steps in my mysql.log ( I opened the log file myself ). And also it will not redirect to the voice gateway. Is it the error of my schemas ? Any idea for it please. #################### /var/log/message #################### May 16 22:52:05 XXX /usr/local/sbin/openser[29112]: New request - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29112]: Proxy authentication failed - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXXT= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: New request - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Clear caller preferences - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Load caller preferences for uuid '1' - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Clear callee preferences - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Callee is not local - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to Invite To External - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to 00 - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to Load PSTN gateways - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: Ready to NextGW - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F=sip:101@XXX.XXX.XXX T= sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 May 16 22:52:05 XXX /usr/local/sbin/openser[29117]: No PSTN gateways available - M=INVITE RURI=sip:00862123456789@XXX.XXX.XXX F= sip:101@XXX.XXX.XXX T=sip:00862123456789@XXX.XXX.XXX IP=61.217.XXX.XXX ID=C8F50846-1452-4EDC-9726-5A3AFECDF533(a)192.168.11.9 #################### /tmp/mysql.log #################### 070516 22:52:05 755 Query select password,uuid from subscriber where username='101' AND domain='xxx.xxx.xxx' 755 Query select value,attribute,type from usr_preferences where uuid='1' 755 Query select uuid from subscriber where username = '00862123456789' ===> gw table | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | | 1 | my-test-gw | 1 | XXX.XXX.XXX.XXX | 5060 | 1 | 1 | 0 | NULL | ===> gw schema | Field | Type | Null | Key | Default | Extra | | id | int(10) | NO | PRI | NULL | auto_increment | | gw_name | varchar(128) | NO | UNI | NULL | | | grp_id | int(10) unsigned | NO | MUL | NULL | | | ip_addr | varchar(15) | NO | | NULL | | | port | smallint(5) unsigned | YES | | NULL | | | uri_scheme | tinyint(3) unsigned | YES | | NULL | | | transport | tinyint(3) unsigned | YES | | NULL | | | strip | tinyint(3) unsigned | YES | | NULL | | <=== it should be varchar ? | prefix | varchar(16) | YES | | NULL | | <=== default value should be Empty or NULL ? ===> gw_grp table | grp_id | grp_name | | 1 | my-test-gw | ===> gw_grp schema | Field | Type | Null | Key | Default | Extra | | grp_id | int(10) unsigned | NO | PRI | NULL | auto_increment | | grp_name | varchar(64) | NO | | NULL | | ===> lcr table | id | prefix | from_uri | grp_id | priority | | 1 | 00862123456789 | % | 1 | 10 | ===> lcr schema | Field | Type | Null | Key | Default | Extra | | id | int(10) | NO | PRI | NULL | auto_increment | | prefix | varchar(16) | NO | MUL | NULL | | | from_uri | varchar(128) | YES | MUL | NULL | | | grp_id | int(10) unsigned | NO | MUL | NULL | | | priority | tinyint(3) unsigned | NO | | NULL | | ################ parts of my openser.cfg####################################### loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") modparam("lcr", "contact_avp", "1401") modparam("lcr", "fr_inv_timer_avp", "s:fr_inv_timer_avp") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "s:rpid") ######################################################################## # Request route 'invite-to-external' ######################################################################## route[15] { xlog("L_INFO", "Ready to Invite To External - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded xlog("L_INFO", "Ready to 00 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(uri =~ "^sip:00[0-9]+@") { xlog("L_INFO", "Ready to Load PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # only route numeric users to PSTN if(!load_gws()) { xlog("L_INFO", "Ready to LoadGW - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_INFO", "Ready to NextGW - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(11); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } -- Best Regards Charles

3 7

Re: [Serusers] Configuration examples
by Mahatma Sarasvati 16 May '07

16 May '07

Hello, I prefer one big example configuration. I like the current default ser.cfg that has nothing but calls to other route blocks in the main route block. That way its easy to turn off route features that you don't want or need to use. I like the new build system. When its more complete it should be extremely useful. Thanks, M ----- Original Message ---- From: SIP <sip(a)arcdiv.com> To: Nils Ohlmeier <nils(a)iptel.org> Cc: serdev(a)iptel.org; serusers(a)iptel.org Sent: Monday, May 14, 2007 4:03:09 AM Subject: Re: [Serusers] Configuration examples Nils Ohlmeier wrote: > Hello everybody, > > as we want to provide more examples for the configuration the following the > question came up: > > Do you prefer > - one big example configuration > - probably harder to understand > - easier to maintain for us > - one example config file per "feature" > - probably easier to understand your desired "feature" > - you might have problems to integrate all your desired "features" into one > config without breaking anything > - harder to maintain for us, as we have to update several files :-) > > As the configuration examples are indented to be for the community, I would > like to know what you prefer or if you see other options. > > Thanks > Nils > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > Ideally, I'd like to see: -One big configuration, well documented (per line or per block at the VERY least). -Smaller configuration examples for options that might not fit into a larger overall config (block-level, which blocks/modparams/etc in the large config would be replaced with/added by the option blocks provided; detailed documentation on the blocks and modparams and what they do/mean (perhaps with links to the proper lines in the module documentation). That way you're not keeping separate configs for each option (unless it's something that completely replaces all the rest of the config), and as more tricks/suggestions come up, it's easier to add an option or two to the options sections since they're just small blocks of code. Just an idea. N. _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

1 0

[Serusers] Problem running SER behind firewall
by Wei Wang 16 May '07

16 May '07

I have a problem running SER behind firewall. Here is the network diagram: |UA1|--|FW1| +--|FW|--|SER| \ / +--+ / \ |UA2|--|FW2| +--|MediaProxy| Where, UA1 and UA2 are Xlite soft-phones behind their own firewalls. SER is listening on private IP address 192.168.4.217. FW has public IP address 66.134.1.34 and forwards port 5060 to SER. The ser.cfg file is pretty much copied from SER getting start guide. When UA1 calling UA2, the call established fine but UA1 will hang up by itself after ~30 seconds. The captured IP packages on SER revealed that the last ACK received from UA2 by SER was sent to FW IP address. Since port 5060 is forwarded to SER on the FW, it caused a looping situation. The ser.cfg is listed at the end. Thanks in advance. Wei Wang wwang(a)m1global.com ======== ser.cfg ============ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=3 #debug=9 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=192.168.4.217 #listen=66.134.1.36 port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:s3rv1c3@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:s3rv1c3@localhost/ser") modparam("auth_db", "calculate_ha1", 1) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("xlog", "buf_size", 8192) # ------------------------- request routing logic ------------------- # main routing logic route{ # xlog("L_INFO", "Main route [From]%fu,[To]%tu,[Req-Method]%rm,[Req-RURI]%ru[IP-src]%is ...\n"); if(method != "SUBSCRIBE") { xlog("L_INFO", "\r\n===========SIP MSG==================\r\n%mb\r\n_____END SIP MSG________________________\r\n"); }; # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 4086 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (method == "INVITE" && client_nat_test("3")) { # xlog("L_INFO", "method==INVITE and nated: calling record_route_preset\n"); # IP ADDRESS Here record_route_preset("66.134.1.34:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ------------------------- # Call Tear Down Section #------------------------- if(method=="BYE" || method=="CANCEL") { #xlog("L_INFO", "RECEIVED BYE or CANCEL..."); end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { xlog("L_INFO", "DEBUG: loose_route..."); if((method=="INVITE" || method == "REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if(method == "INVITE") { if(!allow_trusted()) { if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if(!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } if(client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; # mark routing logic in request #append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { route(4); # mark routing logic in request #append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if(method == "ACK") { route(1); break; } else if(method=="CANCEL") { route(1); break; } else if(method == "INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (!uri==myself) { route(4); #append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if(method=="INVITE" || method == "ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ############################ # REGISTER Message Handler ########################### sl_send_reply("100", "Trying"); if(!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if(!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if(!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if(!save("location")) { sl_reply_error(); }; } route[3] { ############################ # INVITE Message Handler ########################### if(client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(!allow_trusted()) { if(!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if(!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; }; consume_credentials(); lookup("aliases"); if(uri != myself) { route(4); route(1); break; }; if(!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { #---------------------------- # NAT Traversal Section #---------------------------- if(isflagset(6) || isflagset(7)) { if(!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if((isflagset(6) || isflagset(7)) && (status =~ "(180)|183)|2[0-9][0-9]")) { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if(client_nat_test("1")) { fix_nated_contact(); }; } =============== END ser.cfg ================

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Re: [Serusers] dns failover
by samuel 16 May '07

16 May '07

I'm sorry but I don't know enough about the internals . Maybe Andrei (DNS) and Dragos (openIMS) can help with it. Probably the most important lines of the log are: [00m 1(17870) [01;30mdns_hash_find(scscf.open-ims.test(19), 1), h=915 [00m 1(17870) [01;30mdns_a_resovle(scscf.open-ims.test, 0) returning 0 [00m 1(17870) [01;30mdns_srv_resolve_ip("_sip._udp.scscf.open-ims.test", 0, 0), ret=0, ip=127.0.0.1 [00m 1(17870) [01;30mdns_sip_resolve(scscf.open-ims.test, 0, 0), srv0, ret=0 It looks like there's only one returning record...while in the DNS there are two. Try to post last message to openims and serdev lists and maybe developers can help with it... Just for trying, can you add a second fake SRV entry differing not only in the port but also in the hostname? Luck, Samuel. 2007/5/16, Mário Ferreira <mariojvp(a)gmail.com>: > > Hi, > > I tried with SER version ser 2.1.0-dev1, with dns cache and failover > compile options, and with use_dns_cache and > use_dns_failover turned on in the config files ( of P and I-CSCF, the > previous hop of S-CSCF) , but it doesn't work. > I attached the log corresponding to the REGISTER reception by I-CSCF and > delivery to S-CSCF. > I have two S-CSCFs and when i kill one, the message never goes to the > other S-CSCF. > > Thanks, > > Mário Ferreira > > On 5/14/07, Mário Ferreira <mariojvp(a)gmail.com> wrote: > > > > Hi, > > > > The SER version was updated, but only in the trunk of openimscore, and i > > had a working copy of a branch (with the old SER version) . > > Now, i have checked out the trunk and i will test it soon. > > > > Thanks, > > Mário Ferreira > > > > On 5/14/07, samuel < samu60(a)gmail.com> wrote: > > > > > > It looks like SER is compiled with all compilation flags for DNS > > > failover... are these features set in the config file? > > > I guess you're using openIMS SER branch, isn't it? I don't know if it > > > has been sync with latest SER 2.0 and I can not assure whether ser > > > 0.10.99-dev53-tm_fixes has SRV failover support (which I think it does > > > althought with maybe some bugs). > > > > > > From the dig output I guess you're trying to load balance between the > > > two servers responding for scscf.open-ims.test, isn't it? > > > Can you please attach the log file section where SER is trying to send > > > to this destination? > > > > > > Thanks, > > > Samuel > > > > > > 2007/5/14, Mário Ferreira < mariojvp(a)gmail.com>: > > > > > > > > Hi, > > > > > > > > # ser -V > > > > version: ser 0.10.99-dev53-tm_fixes (i386/linux) > > > > flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, > > > > DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, > > > > FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_DST_BLACKLIST > > > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN > > > > 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > > > @(#) $Id: main.c 136 2007-02-13 17:58:42Z vingarzan $ > > > > main.c compiled on 22:49:24 Mar 26 2007 with gcc 3.4.6 > > > > > > > > # dig NAPTR open-ims.test > > > > > > > > ; <<>> DiG 9.2.4 <<>> NAPTR open-ims.test > > > > ;; global options: printcmd > > > > ;; Got answer: > > > > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 56440 > > > > ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, > > > > ADDITIONAL: 3 > > > > > > > > ;; QUESTION SECTION: > > > > ;open-ims.test. IN NAPTR > > > > > > > > ;; ANSWER SECTION: > > > > open-ims.test. 86400 IN NAPTR 10 50 "s" "SIP+D2U" > > > > "" _sip._udp.open- ims.test. > > > > open-ims.test. 86400 IN NAPTR 20 50 "s" "SIP+D2T" > > > > "" _sip._tcp.open-ims.test. > > > > > > > > ;; AUTHORITY SECTION: > > > > open-ims.test. 86400 IN NS > > > > 127.0.0.1.open-ims.test . > > > > > > > > ;; ADDITIONAL SECTION: > > > > icscf.open-ims.test. 86400 IN A 127.0.0.1 > > > > _sip._udp.open-ims.test . 86400 IN SRV 0 0 5060 > > > > icscf.open-ims.test. > > > > _sip._tcp.open- ims.test. 86400 IN SRV 0 0 5060 > > > > icscf.open-ims.test. > > > > > > > > ;; Query time: 0 msec > > > > ;; SERVER: 127.0.0.1#53(127.0.0.1 ) > > > > ;; WHEN: Mon May 14 18:24:18 2007 > > > > ;; MSG SIZE rcvd: 259 > > > > > > > > # dig _sip._udp.scscf.open-ims.test. srv > > > > > > > > ; <<>> DiG 9.2.4 <<>> _sip._udp.scscf.open-ims.test. srv > > > > ;; global options: printcmd > > > > ;; Got answer: > > > > ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 21445 > > > > ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 1, > > > > ADDITIONAL: 1 > > > > > > > > ;; QUESTION SECTION: > > > > ;_sip._udp.scscf.open-ims.test. IN SRV > > > > > > > > ;; ANSWER SECTION: > > > > _sip._udp.scscf.open-ims.test. 86400 IN SRV 0 1 6060 > > > > scscf.open-ims.test. > > > > _sip._udp.scscf.open-ims.test. 86400 IN SRV 0 1 6061 > > > > scscf.open-ims.test. > > > > > > > > ;; AUTHORITY SECTION: > > > > open-ims.test. 86400 IN NS > > > > 127.0.0.1.open-ims.test. > > > > > > > > ;; ADDITIONAL SECTION: > > > > scscf.open-ims.test. 86400 IN A 127.0.0.1 > > > > > > > > ;; Query time: 0 msec > > > > ;; SERVER: 127.0.0.1#53( 127.0.0.1) > > > > ;; WHEN: Mon May 14 18:26:33 2007 > > > > ;; MSG SIZE rcvd: 165 > > > > > > > > > > > > regards, > > > > > > > > Mário Ferreira > > > > > > > > On 5/13/07, samuel <samu60(a)gmail.com> wrote: > > > > > > > > > > I have tried DNS failover and can assert is working like charm for > > > > > SER version 2.0 and further. > > > > > Which SER version are you using? > > > > > > > > > > can you check wheter you have compiled SER with DNS failover? > > > > > which is the ouput of > > > > > #ser -V > > > > > what is the ouput of > > > > > #dig NAPTR yourdomain.name.com > > > > > (replace yourdomain.name.com for the appropriate valie) > > > > > > > > > > regards, > > > > > sam. > > > > > > > > > > 2007/5/10, Mário Ferreira <mariojvp(a)gmail.com >: > > > > > > > > > > > > Hi, > > > > > > > > > > > > I am interested in using SER DNS failover, but I am getting some > > > > > > problems. > > > > > > > > > > > > A failure can be a 503 error response, ICMP errors or a timeout > > > > > > due to a not received response. > > > > > > > > > > > > Does anyone know if SER is able to detect these failures and > > > > > > then select an alternative destination for the message? > > > > > > > > > > > > If the ICMP errors can't be detected, can this be solved with > > > > > > timers? > > > > > > I turned on use_dns_cache and use_dns_failover, but when the > > > > > > message fails to be delivered to the first destination, > > > > > > SER never tries the second destination, which was configured > > > > > > with DNS SRV. > > > > > > > > > > > > regards, > > > > > > > > > > > > Mário Ferreira > > > > > > > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > >

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[Serusers] ANNOUNCEMENT: SER 2.0 Release Candidate 1 is available
by Greger V. Teigre 16 May '07

16 May '07

If you have 30 minutes to spare, you can set up a working SER 2.0! If you haven't taken the time to test SER 2.0 yet, it has never been simpler. Find full instructions on: http://www.iptel.org/ser_2_0_0_release_candidate_1 Questions and feedback go to these lists! Have fun, Greger

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[Users] Openser - LCS 2005 incoming call
by Tsiferana 16 May '07

16 May '07

Hi, I've tried to test this scenario: windows messenger>>lcs2005>>openser>>x-lite and it works great but this scenario x-lite>>openser>>lcs2005>>windows messenger the call can't be etablished., in the network traffic is see that the lcs does not relay the invite request to windows messenger. what do I need for running this scenario? Best Regards Tsiferana --------------------------------- Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

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[Users] Max-Forward header in Notify messages
by Toni Barata 16 May '07

16 May '07

Hi Anka, In Notify messages generated by Presence module the Max-Forward Header is missing. Some sip stacks have a strange behaviour when receiving Notify messages without Max-Forward header. Best regards, Toni

1 0

[Users] purposely dropping reply message
by Tim 16 May '07

16 May '07

Hello, How/Where would be the best place to purposely drop a particular reply message ( > 400 ) I have armed both the reply route and failure route and attempted to drop the message by calling the drop() function but this doesn't seem to work. Is there a simple method for conditionally dropping a reply message? My goal is to force the UAC that created the original request to retransmit (UDP) the request in certain scenarios, such as when ACK to an initial invite has not been fully processed by the downstream UAS but a re-invite has been received by the UAS causing the UAS to return a 400 for the re-invite. I do not want the UAC to see the 400 reply, instead I simply want the UAC to resend the re-invite because no final response has been received, presumably by this time the ACK processing will be finished and the retransmitted re-invite would be handled smoothly. I realize replies are routed based on via headers. Is there some way to short circuit this routing other than drop()? Thanks Tim Tried this from failure route if(t_check_status("400")) { xlog( "L_ERR", "400 message in failure route 2 ********** \n" ); drop(); return; } Tried this from reply route if(status == "400") { xlog( "L_ERR", "400 in reply route 2 .... dropping\n" ); drop(); return; }

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[Users] trying to get 2-way audio in multi-homed openser configuration
by Taylor Carpenter 16 May '07

16 May '07

I have tried all sorts of configurations with both rtpproxy and media proxy, but I continue to end up with audio only working on one end of the call. The latest I tried to turn on rtpproxy for all calls. The call goes through but audio is only heard from the callee to the caller. Any help is greatly appreciated. Taylor

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[Users] XCAP Lite 1.0 released
by Adrian Georgescu 16 May '07

16 May '07

Change log xcap-lite (1.0.0) unstable; urgency=low * Added control interface for OpenSER to inform the Presence agent about changes in the presrules document. This allows the presence server to imediately send Notify to the updated buddy list after the end-user changes his presence policy document. Thanks to Anca Vamanu and Juha Heinanen for the support provided for adding this critical functionality * The configuration config.php has changed to allow for definition of the OpenSER management interface URL (refreshWatchersURL) * OpenSER must have the mi_xmlrpc module loaded and listening on the port configured in config.php * Fixed query in updateWatchers function * OpenSER trunk version 2230 or higher is required when refreshWatchersURL setting is enabled in config.php The software can be downloaded from: http://download.dns-hosting.info/XCAP/xcap-lite-1.0.0.tar.gz Regards, Adrian

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[Users] openser setup as a telephony gateway (to PSTN) for NAT'd clients
by Taylor Carpenter 16 May '07

16 May '07

I may be misunderstanding things (very probably so), but all the examples for both SER and OpenSER that I have seen either do not do NAT or NAT but are not the "end point" for a UAC to the PSTN. From what I can tell they are just sending on the SIP request to the next destination as is. The setup I am trying to accomplish is * SIP clients are all on a private network with connectivity to OpenSER directly on one interface (on the same private network) * OpenSER's other interface is on the external network (internet facing) * SIP clients are only sending telephone numbers (sip:telephone_number@*... do not know about PSTN providers) * OpenSER connects to the PSTN provider and send the number to dial that came from the SIP client * OpenSER "proxies" the entire call (with the help of rtpproxy or mediaproxy for RTP of course) * No incoming calls from the internet to OpenSER (no support for that is needed) * Registration not required for sip clients (they are all on same private network and authorized) I have found several posts, example configs, documents that have pieces of what I need (from what I can tell).. and I have tried to put it together, but it does not quite work... So is there some example that fits this type of usage? If not one then possibly several pieces from a few documents? I have been thinking that OpenSER setup as a outbound proxy configured for multihome, but everything I have seen on that just routes the calls through to where ever the SIP client was requesting as a final destination and I need to send to one destination (the PSTN provider). Any help is greatly appreciated. BTW, I was thinking the NAThelper or outbound proxy example from http://www.voip-info.org/wiki-SER+tips+and+tricks Looked close to what is needed, but have a bunch of stuff (seemingly) unneeded for my scenario and have nothing about PSTN connectivity. Thanks for taking the time to read this long post (if you made it this far). Taylor

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[Serusers] Race Condition on CANCEL?
by Gustavo Passos Tourinho 16 May '07

16 May '07

Hello, Im having some problems with cancelled calls. This is the scenario: U1 Proxy U2 INVITE -->>> <<--- 100 Trying INVITE -->>> <<--- 100 Trying <<--- 100 Trying CANCEL ->>> <<-- 200 Cancelling CANCEL ->> <<-- 180 Ringing <<-- 487 Cancelled <<-- 180 Ringing <<-- 200 OK (Wrong??) <<-- 200 OK My problem is that after some time waiting for "ringing", the user cancel the call. Even that proxy responses "487" it still forward the late 200 OK. Should it forward? I guess not because the transaction was destroyed, right? Can it be a configuration problem on my ser.cfg ou it can be in t_relay implementation? Thanks in advanced. Regards, Gustavo

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[Serusers] ser_fifo gives timeout
by Andrey Kuprianov 16 May '07

16 May '07

Hi guys, I dont know what the problem is.. Really.. Im out of ideas with permissions and reinstallations for serweb's CTD and IM. During my 1st installation with serweb they both worked fine. Then I had to reinstall my OS (Ubuntu 7.x btw) and now they give both timeouts.. I have even copied send_im() and write2fifo() functions separately into a PHP file and ran it from command line. Same old "408 Request Timeout".. Please, suggest something. Im basically using nat-mediaproxy.cfg for config file and I have tried setting various ownerships and permissions to serweb dir and for /tmp/serweb and /tmp/ser_fifo.. Nothing helps really. All packages are installed for php4 including php4-cli, pear, pear-log, pear-DB, XML_RPC and even classkit (which is useless here anyway).

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[Serusers] Configuration examples
by Nils Ohlmeier 15 May '07

15 May '07

Hello everybody, as we want to provide more examples for the configuration the following the question came up: Do you prefer - one big example configuration - probably harder to understand - easier to maintain for us - one example config file per "feature" - probably easier to understand your desired "feature" - you might have problems to integrate all your desired "features" into one config without breaking anything - harder to maintain for us, as we have to update several files :-) As the configuration examples are indented to be for the community, I would like to know what you prefer or if you see other options. Thanks Nils

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[Users] SIP SIMPLE / XCAP test bed - feedback request
by Adrian Georgescu 15 May '07

15 May '07

Hello, We setup a SIP SIMPLE testbed using the latest OpenSER / XCAP-lite. To make use of this testbed you need a recent Eyebeam commercial version for Apple or Windows. Other SIP/SIMPLE/XCAP client implementations are of course more then welcome and desired. To use this test bed register a SIP account at: http://sip2sip.info Instructions about the SIP/Presence/XCAP settings are sent by email to the registrant. We appreciate any feedback regarding the software behavior so that we can fix any bugs in the used implementations and update the wiki pages with proper configuration samples. Thank you for your support, Adrian

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Re: [Serusers] SER/PSTN Gateway conf
by flavio 15 May '07

15 May '07

My Record_Route and Loose section are the follow: # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (method=="INVITE") { if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; }; route(1); break; }; As follow reported, SER Record Route INVITE original INVITE message, that lack of 'Route:' header, and forward it to IP Phone. Why not for BYE? =_=" Thanks for support and big patience. ^_^ U 2007/05/15 16:14:20.789400 10.28.52.105:5060 -> 10.28.19.202:5060 INVITE sip:06605XXXXX@10.28.19.202 SIP/2.0. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 70. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 2007/05/15 16:14:20.790166 10.28.19.202:5060 -> 10.28.52.105:5060 SIP/2.0 100 trying -- your call is important to us. To: <sip:06605XXXXX@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Call-ID: 238757681179426(a)10.28.52.105. CSeq: 23392 INVITE. Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=3803 req_src_ip=10.28.52.105 req_src_port=5060 in_uri=sip:0660522014@10.28.19.202 out_uri=sip:06605XXXXX@10.28.19.124;user=phone via_cnt==1". . # U 2007/05/15 16:14:20.790257 10.28.19.202:5060 -> 10.28.19.124:5060 INVITE sip:06605XXXXX@10.28.19.124;user=phone SIP/2.0. Record-Route: <sip:10.28.19.202;ftag=DXsf22387576811Yaz079427;lr=on>. To: <sip:0660522014@10.28.19.202>. From: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;tag=DXsf22387576811Yaz079427. Via: SIP/2.0/UDP 10.28.19.202;branch=z9hG4bK4639.ea946db1.0. Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2387576811429488787179425. Remote-Party-ID: "06720XXXXX" <sip:06720XXXXX@10.28.52.105>;party=calling;screen=yes;privacy=off. Contact: <sip:10.28.52.105:5060>. Call-ID: 238757681179426(a)10.28.52.105. Max-Forwards: 16. User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, REGISTER, NOTIFY. CSeq: 23392 INVITE. Content-Length: 274. Content-Type: application/sdp. . v=0. o=NetsyntSIP-GW-UserAgent 34373 1 IN IP4 10.28.52.105. s=SIP Call. c=IN IP4 10.28.52.105. t=0 0. m=audio 16572 RTP/AVP 18 8 0. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 PCMA/8000. a=silenceSupp:off - - - -. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. 2007/5/15, Kostas Marneris <K.Marneris(a)otenet.gr>: - Nascondi testo tra virgolette - > As far as I know and understand it seems that the > R-URI of the BYE mesg (1st line) which GW sends to SER is wrong. > > I expected to see : > BYE sip:user@SIP_Phone_IP_Address SIP/2.0 > > or the problem is that the 'Route:' header is missing from the BYE mesg > so it's not loose_routed. > > Do you Record_Route the original INVITE ? > > Check also : rfc3261 / 16.12.1 > > > Kostas > >

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sr-users May 2007