- sr-users - kamailio.org

[Serusers] How to investigate/filter the sdp contents if Content type is text/plain
by Tomasz Zieleniewski 07 Feb '08

07 Feb '08

Hi, is it possible to check message contents if they contain some sting in case of content type text/plain?? will it work if I use search('some string'); from textops module? Regards Tomasz

3 2

Re: [OpenSER-Users] newbie questions
by Doug McLetchie 07 Feb '08

07 Feb '08

Thanks for the reply Alex! I really appreciate it. I've been reading manuals & sip specs all day, and I think that I found my answer, and it really is an easy one. here's my own answer to my question, can you let me know if this sounds right? When a message is send from a user agent (on the public internet) to my gatweway & then to the proxy (on a private network), the gateway will add a via on to the message. that's how open SER (or any other proxy) can send a response message back to user agent on the public internet via the gateway. It's really the same thing as setting up a registrar, or any other kind of proxy/user agent that's going to communicate via a SBC of any kind. So, it that right? is that how the SIP via header is used? I understand the part about registrations. We're not currently doing that in our network, we're using all static IP addressing. I plan on using registrations in the future. the gateway that I'm using can act as a session border controller, but it can also act as a class 4/5 switch (TDM or IP). so it's some serious overkill for what I want to do here. :o) Thanks again for the reply, --Doug ----- Original Message ---- From: Alex Balashov <abalashov(a)evaristesys.com> To: Doug McLetchie <dougmcletchieatwork(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Wednesday, February 6, 2008 1:23:51 AM Subject: Re: [OpenSER-Users] newbie questions Doug, Doug McLetchie wrote: > For inbound calls (calls coming from another carrier to my big expensive > gateway & destined for a specific PBX), I'd like my gateway to send the > call to the proxy, who will determine which PBX to send the call to, and > then send the call to to correct PBX via the Gateway. I don't > understand how to set up OpenSER to send the call via the Gateway. If I > provision the static ip address of the PBX in the proxy, won't it try to > send directly to that IP address instead of sending it to the Gateway? > I think that the feature that I'm looking for is something like an > outbound proxy, for the proxy. (does that make sense?) OpenSER can certainly do what you are trying to accomplish. SIP routing is done by URI, which consists of a "user" part and a "domain" part. The "user" part is the number (or alphanumeric identification, in the case of pure-VoIP peering) and the "domain" part is the IP "place" at which the "user" part is reachable. When you route a call to some URI, what you are really saying is, "Here, domain, you must know what to do with this 'user' part - i.e. have reachability information for it (a SIP contact bound from a registration, for example)." A proxy by itself isn't enough. If you need to reach these PBXs, you clearly need to know how to reach them. This requires a SIP registrar somewhere, so that the PBXs can register against it and say, "Here, you can reach me at such and such IP and port." Or, I suppose, you can define these contacts statically with a database interface from the proxy, in which case you don't need to use a registrar. I don't know what this Big Expensive Gateway is, but if it's something like a Session Border Controller, it should be able to forward SIP REGISTER requests to your proxy/registrar. Or do they register against the gateway? Either way, you can perform this resolution with OpenSER. The problem you *might* run into is sending the same logical call leg back to the Gateway. Depending on what it is, it may not like that and perceive a call routing loop, because the call that went through it to the proxy is the "same" call (in terms of SIP Call-ID, and other things that make up a logical call "leg") that is now being sent back around to it. This problem is often solved with the introduction of a back-to-back user agent which can re-originate a different call leg. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

3 2

[OpenSER-Users] Accounting through Mediaproxy
by Abdul 07 Feb '08

07 Feb '08

Good Day All, I am trying to get accrue AcctSessionTime of calls. I am using Openser+Asterisk but some time asterisk CDR are not accoutered, record missing + if some one call 15 min it will come 2 hrs in cdr. Is Mediaproxy will be best choice to get trust able AcctSessionTime? If Mediaproxy AcctSessionTime is trustable, i have to use ACC module of Openser or only dbaccounting configuration will be enough to store records direct from Mediaproxy into MySQL backend? Thank You Abdul Latef --------------------------------- Never miss a thing. Make Yahoo your homepage.

2 1

[OpenSER-Users] How to handle messages received in WeSIP?
by Rajat Dudeja 06 Feb '08

06 Feb '08

Hi, I'm using SEAS module to interact with WeSIP Application Server. I'm using the version wesip-02-04-07-beta-0.1.4.tar.gz of WeSIP. I've managed to configure it and I'm also seeing that the messages are received by WeSIP. I'ld like to know: 1. Are there any special files to which WeSIP writes the received messages? 2. In case, some one wants to modify the received message then is there a possibilty to send the modified message back to OpenSER? 3. I have used the above tar file, to me it appears that this file only contains the binary form of WeSIP. So where can I get the complete source code of WeSIP? Many thanks, Rajat =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[OpenSER-Users] renaming of database modules
by Henning Westerholt 06 Feb '08

06 Feb '08

Hi all, the database modules and the supporting infrastrucure in the OpenSER core are heavily reworked in the trunk version at the moment. Common functionality that is present in all database modules were integrated in the core, new functionality have been added. Its planned to unify the overall behaviour and structure of all modules that are interfacing with databases, to make them more maintainable, more safe and also more bug-free. Important foundations for this changes have already been created. As part of this process the database modules names were changed. All these module names are now prefixed with 'db_'. The new names of the modules are now db_mysql, db_postgres, db_unixodbc, db_berkeley, db_text, db_flatstore. If you use one database module with the trunk version, you must change some parts of the configuration. Old module configuration: loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) New module configuration: loadmodule "db_mysql.so" modparam("db_mysql", "ping_interval", 300) modparam("db_mysql", "auto_reconnect", 1) The database URLs for modules that connect to a database, e.g. usrloc don't need to be changed, if you use mysql, postgres, unixodbc or flatstore DBs. For the dbtext and db_berkeley modules the new database URLs are: modparam("usrloc", "db_url", "text://my-database-path")" modparam("usrloc", "db_url", "berkeley://my-database-path")" (you need to adapt the path to your setup, of course). If you encounter any bugs that were introduced during this change, please open a bug report on the tracker. Best regards, Henning

1 0

[Serusers] (SER + OpenXCAP) Presence Authorization problem
by Guhan P 06 Feb '08

06 Feb '08

Hi All, I am using SER 2.0 and OpenXCAP 0.9.7 and facing problem in implementing Presence Authorization using XCAP in my project. I have done the following:- User A (User_A.pcap(attachment)): 1.Registered with SER. 2.Published A's info into SER. 3.Subscribed for it's 'presence.watcher info' 4.Transferred presence_rules.xml(which contains rules to allow B to see his presence status) to OpenXcap. Following debugging messages were printed (while transferring presence_rules file ) in OpenXcap: ************************************** Feb 6 12:15:40 sips openxcap[2557]: [-] Starting factory <twisted.web.xmlrpc._QueryFactory instance at 0xb3cde68c> Feb 6 12:15:40 sips openxcap[2557]: [Uninitialized] error: Error while notifying OpenSER management interface for 'user' a@: Connection was refused by other side: 111: Connection refused. Feb 6 12:15:40 sips openxcap[2557]: [Uninitialized] a@ from 172.16.7.187 "PUT http://172.16.7.204/xcap-root/pres-rules/users/a/presence-rules.xml" 201 0 - unknown Feb 6 12:15:40 sips openxcap[2557]: [Uninitialized] Stopping factory <twisted.web.xmlrpc._QueryFactory instance at 0xb3cde68c> ************************************* User B (User_B.pcap(attachment)): 5.Registered with SER. 6.Published B's info into SER. 7.Sent Subscribe request for A's presence status to SER. 8.Received 200 ok for subscribe request. 9.Received Notify request with status as pending 10.SER intimated User A about User B's subscription. Following debugging messages were printed in SER: *************************** 5(5114) find_export_record: found <xcap_query> in module xcap [/home/ser/SER2.0/lib/ser/modules/xcap.so] 5(5114) curl error: 22 5(5114) XCAP problems for uri 'http://172.16.7.204:8000/xcap-root/pres-rules/users/a/presence-rules.xml' *************************** SER did not send User A's presence status to User B(Notify status was pending). It seems to be SER did not interact with OpenXcap server. This is my Presence Authorization configuration settings in ser.cfg ************************************************************************************************** modparam("pa", "use_db", 1) modparam("pa", "offline_winfo_timer", 600) #modparam("pa", "use_offline_winfo", 1) modparam("pa", "offline_winfo_expiration", 600) modparam("pa", "db_url", "mysql://ser:heslo@127.0.0.1:3306/ser") # mode of PA authorization: none, implicit or xcap modparam("pa", "auth", "xcap") modparam("pa", "winfo_auth", "none") modparam("pa", "use_callbacks", 1) modparam("pa", "accept_internal_subscriptions", 0) modparam("pa", "max_subscription_expiration", 120) modparam("pa", "timer_interval", 1) modparam("pa", "watcherinfo_notify", 1) modparam("pa", "async_auth_queries", 1) modparam("pa", "pres_rules_file", "presence-rules.xml") modparam("xcap", "xcap_root", "http://172.16.7.204:8000/xcap-root") modparam("usrloc", "db_mode", 1) ************************************************************************************************** Please find the attached file NW_Captures.zip which contains two ethereal captures(User_A.pcap & User_B.pcap). Could you help me to resolve this problem? Regards, Guhan

1 0

[OpenSER-Users] Call Session Time
by VoIP Forums www.Go4Calls.com 06 Feb '08

06 Feb '08

Hi all, How i can store Call Session Time in acc table? I am using Mediaproxy so i hope the call duration will be correct event if openser does not receive the BYE. I tried to use ACC but inside the table all information is storing only i cannot see the filed name for session or call duration. Could you please help me how i can figure out to get call duration to charge our customers? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

Re: [OpenSER-Users] Mediaproxy listening but handling zero bytes of audio
by A.smith 06 Feb '08

06 Feb '08

Seems like this may be the same issue Im experiencing, anyone any experience of this type of issue?? Anything? :P http://www.voipuser.org/index.php?name=PNphpBB2&file=viewtopic&t=11052&high… ________________________________________________ Message sent using UK Grid Webmail 2.7.9

1 0

[OpenSER-Users] Sending Text Response from Openser to client
by Ali Jawad 06 Feb '08

06 Feb '08

Dear All I would like to change the messages that are sent back from the openser server to the client is this possible using sl_send_reply for example ? I have tried sl_send_reply in the route[1] but it pops up a dialog box and it seems to break the calls. Can anyone tell me where to get more info about such a scenario. Or suggest any solutions ? Thanks

4 7

[OpenSER-Users] Store Encypted Pwds in MySQL DB
by Ali Jawad 06 Feb '08

06 Feb '08

Hi All I am migrating a couple of thousand users from an old sip server with mysql support to openser with mysql support, the thing is that in the old server the passwords were encrypted using 3DES and stored encypted in the MySQL db ...the old SIP server received the passwords using plaintext converted them to 3DES using perl script and the compared the result to the password stored in the MySQL DB. Any ideas about how to achieve something like this in OpenSER "? I.e. hash the passwords ? Or any other solution ? I have seen some hashed entries in the MySQL db could they be used for such purposes ? Thanks

2 1

[OpenSER-Users] [OT] 1000 subscribers exceeded on the list
by Bogdan-Andrei Iancu 06 Feb '08

06 Feb '08

Hi, Just proud to announce that the users mailing list just exceeded 1000 subscribers (1001 right now). I will just take the opportunity to thanks again to all people answering on this list and providing valuable knowledge to all interested and working with openser. Regards, Bogdan

1 0

Re: [OpenSER-Users] MSILO back notify not working
by Iñaki Baz Castillo 06 Feb '08

06 Feb '08

On Monday 04 February 2008 19:02:30 Dan-Cristian Bogos wrote: > Iñaki, > > this code should do the magic: > > if (!t_reply("202", "Accepted")) > { > sl_reply_error(); > }; I use exactly that code: t_newtran(); if (m_store("$ru")) { if (!t_reply("202", "Accepted")) sl_reply_error(); } But there is not a created MESSAGE by OpenSer to the sender. I've set "debug=4" and no errors appear, the message storage is done correctly and OpenSer doesn't try to send back a new MESSAGE as notification. I'd like to insist a bit in the fact that what I'm trying to get is receiving a **new** MESSAGE generated by OpenSer as notification to the sender, not just the "202, Accept" reply (that is a reply, not a MESSAGE). Please Dan, can you confirm that MSILO module should generate and send a new MESSAGE as notification to the sender, and not just a "202 Accepted" reply? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

7 12

[OpenSER-Users] can't get openser to start
by Bill Brownrigg 06 Feb '08

06 Feb '08

My openser.cfg runs and the modules load but then I get this message: "ERROR: tcp_init: bind(7, 0x811d384, 16) on 127.0.0.1:5060 : Address already in use" Environment is Red Hat and OpenSER 1.1.0 We do not need TCP as we only use UDP. Is there a simple way to remove TCP from the OpenSER environment? If not, how do we figure out what is causing the conflict? The box used to run Asterisk but it was shut down. TIA, Bill

3 2

[OpenSER-Users] newbie questions
by Doug McLetchie 06 Feb '08

06 Feb '08

Hi all, I've been reading through the documentation online fore a few days, and I've read through a lot of the user & developer artchives. I'm trying to figure out if OperSER is the right proxy solution for the problem that I'm trying to solve. I was hoping that I could get a few pointes from some more experienced users. My first question is really simple... I must have the terminology wrong, otherwise I'm sure that I would have found the answer by now. I have a big expensive gateway, but it can't do a few things that I really want to do. I'm hoping to augment my big expensive gateway with OpenSER, so that I can do some specific routing. I want my network to look something like this: -------------- (Public IP network) (Private IP network) .----------. | carriers |-----. '----------' | .-----------. '----| Big | .-------. | Expensive |----| Proxy | .------. .----| Gateway | '-------' | PBXs |---------' '-----------' '------' ---------- my solution involves customer-premise based PBXs that my company makes in house. I want the gateway to act as a session border controller. It'll take care of forwarding me media, and it'll take care of protecting my proxy from the public internet. For inbound calls (calls coming from another carrier to my big expensive gateway & destined for a specific PBX), I'd like my gateway to send the call to the proxy, who will determine which PBX to send the call to, and then send the call to to correct PBX via the Gateway. I don't understand how to set up OpenSER to send the call via the Gateway. If I provision the static ip address of the PBX in the proxy, won't it try to send directly to that IP address instead of sending it to the Gateway? I think that the feature that I'm looking for is something like an outbound proxy, for the proxy. (does that make sense?) Sorry to write such a long e-mail to the mailing list! I wanted this to be descriptive enough that you all would know what I mean. THANKS! --Doug ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] openser+diameter communication problem-------->kindly pay attentionusers
by bhargav.b＠renovau.com 06 Feb '08

06 Feb '08

hellow everybody, i have to authorise clients connected to openser with the help of opendiameter. can u guide me to do the needed thing. actually i have installed opendiameter,openser,xlite[soft phone].right now xlite is communicating and authorised by openser.i have added most of all modules in openser configuration file provided by openser group.even openser is communicating with open diameter client.but the problem is the user name and password are not identified by opendiameter client. could u tell me how to add user information in opendiameter[username,passwd,etc]so that server can authorise him to do further task. in clients dump i can see messages like"code not found,avp not found "sort of things. i really need u peoples help in developing a working model. your's faithfully bhargav.b

1 0

[OpenSER-Users] Is it possible to add a header to a reply?
by Iñaki Baz Castillo 05 Feb '08

05 Feb '08

Hi, it could be nice the possibility of adding headers in OpenSer replies. I mean replies generated by OpenSer with "t_reply()" or "sl_reply()". For example it could be cool adding a header: "Success: User offline, message stored" when using MSILO before: if (m_store("$ru")) t_reply("202", "Accepted")) AFAIK this is not possible, is it? If not, maybe I'd report it as feature request, what's your opinion about it? Thanks. -- Iñaki Baz Castillo

2 2

Re: [OpenSER-Users] Remote Mediaproxy Server
by Abdul 05 Feb '08

05 Feb '08

Actually i have one extra server in our Rack so i wanted to make Mediaproxy in different server then we may get better and more peformance. Currently MediaProxy is installed in my same Openser Server and it is working well in production. The Server where we need to install Mediaproxy, it is on same Network of Openser and i can ping well the difference is only that it is in NAT mode it does not have any public IP we only can connect that server through Openser or using some VPN connection --------------------------------- Never miss a thing. Make Yahoo your homepage.

1 0

[OpenSER-Users] AVPOPS question
by Mouncif Benniane 05 Feb '08

05 Feb '08

How Do I re-write the IP, Port of Request URI using avp_pushto() ?

3 4

[OpenSER-Users] Mediaproxy listening but handling zero bytes of audio
by Andy Smith 05 Feb '08

05 Feb '08

Hi, Im trying to test mediaproxy for the first time, I am forcing all traffic through mediaproxy in the openser.cfg (there's nothing wrong with that in principal is there?). I have one SIP handset hanging directly off OpenSER and another connected to Asterisk which has a channel configured to OpenSER. I can get the call answered but there is audio only in one direction and even more strangely I see that mediaproxy has not handled any data at all in the audio stream. This is what I see from mediaproxy: Feb 5 21:26:05 server proxydispatcher[13968]: request 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 10.1.1.1:16042:audio 10.1.1.1 10.1.1.1 remote server.domain.net local Asterisk=20PBX info=from:0042*001@10.1.1.1,to:0044208888888@server.domain.net,fromtag:as1a72639d,totag: Feb 5 21:26:05 server mediaproxy[13965]: request 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 10.1.1.1:16042:audio 10.1.1.1 10.1.1.1 remote server.domain.net local Asterisk=20PBX info=totag:,to:0044208888888@server.domain.net,from:0042*001@10.1.1.1,fromtag:as1a72639d Feb 5 21:26:05 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: started. listening on 10.1.2.2:60006 Feb 5 21:26:05 server mediaproxy[13965]: execution time: 1.49 ms Feb 5 21:26:05 server proxydispatcher[13968]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '10.1.2.2 60006' Feb 5 21:26:05 server proxydispatcher[13968]: execution time: 5.03 ms Feb 5 21:26:53 server proxydispatcher[13968]: delete 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 info= Feb 5 21:26:53 server mediaproxy[13965]: delete 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1 info= Feb 5 21:26:53 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Feb 5 21:26:53 server mediaproxy[13965]: session 75c5e34d0e567d5f1fa22f583ecc7af0(a)10.1.1.1: ended. Feb 5 21:26:53 server mediaproxy[13965]: execution time: 0.60 ms Feb 5 21:26:53 server proxydispatcher[13968]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: '' Feb 5 21:26:53 server proxydispatcher[13968]: execution time: 1.51 ms This "0/0/0 packets, 0/0/0 bytes (caller/called/relayed)" being the part that I dont understand, well it proves its not handling the audio stream, but other than put a use_media_proxy command in the openser route there isnt anything else I can do to get the data stream to route via media proxy is there? My openser route is very simple: route{ if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="OPTIONS") { exit; }; if (method=="REGISTER") { record_route(); if (!save("location")) { sl_reply_error(); }; exit; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); end_media_session(); } route(1); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags record_route(); use_media_proxy(); setflag(1); # setflag(2); $avp(s:can_uri) = $ru; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (!lookup("location")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # SECTION 5 - SECONDARY ROUTE BLOCKS route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } Anyone any ideas what might be going on or how I can investigate the issue further? thanks for any help! Andy.

1 0

[OpenSER-Users] Remote Mediaproxy Server
by Abdul 05 Feb '08

05 Feb '08

Hi All, I read readme file of Mediaproxy that there is possibility to install mediaproxy on different server than Openser. I have doubt how i can connect to that server my Mediaproxy server is in NAT mode and the IP is : 192.168.1.77 and my openser server is NO-NAT Mode and IP is : 212.XXX.XXX.XXX Is there any possibility to communicate to Mediaserver? I tried the following in openser.cfg but i got error "CANNOT CONNECT TO MEDIAPROXY SERVER" I forced media proxy to use 30001 port to accept remote command. loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "192.168.1.77:30001") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/src/sip-server/modules/mediaproxy/config/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/src/sip-server/modules/mediaproxy/config/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) -------- Regard, --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo! Search.

2 1

Re: [OpenSER-Users] Remote Mediaproxy Server
by Andy Smith 05 Feb '08

05 Feb '08

Hi Abdul, looking at the documentation this isnt a valid option: modparam("mediaproxy", "mediaproxy_socket", "192.168.1.77:30001")Check http://openser.org/docs/modules/1.3.x/mediaproxy.htmlThe mediaproxy docs included with the mediaproxy software gives details of how to configure remote mediaproxy servers, from my understanding you need to have a local proxydispatcher.sock socket (defined in openser.cfg) then you need to follow the INSTALL and/or README files included with mediaproxy to correctly configure your remote mediaproxy servers (and how they are located by the proxydispatcher) cheers Andy.

1 0

[Serusers] commercial help requested
by Kenneth 05 Feb '08

05 Feb '08

We need someone who can quickly assist us with creating the P-asserted header on SER. RFC 3325. Configuration. If there is an expert out there, please contact us with your qualifications and rates. Kenneth

2 3

[Serusers] Multiple Subscription (PA)
by Carlos Angeles 05 Feb '08

05 Feb '08

Hi... I am using the SER Presence agent for polling information from the server by sending SUBSCRIBE with expiration = 0 (just fetching the information). However, the server only responds to SUBSCRIBES with a time interval of 5 secondes between them. Is there any timer or parameter I can modify in order to reduce this time. Thanks Carlos _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] ser rel_2_0_0_rc6 from cvs doesn't start and breaks down
by Tomasz Zieleniewski 05 Feb '08

05 Feb '08

Hi, I have just downloaded the latest cvs sources of rel_2_0_0. I compiled and installed it as usually: make proper make prefix=/homedir install after this each time I start ser it breaks down here my log from console: Starting SER SIP Proxy: ser $Listening on udp: 127.0.0.1 [127.0.0.1]:7060 udp: 192.168.129.38 [192.168.129.38]:7060 tcp: 127.0.0.1 [127.0.0.1]:7060 tcp: 192.168.129.38 [192.168.129.38]:7060 Aliases: tcp: venom:7060 tcp: venom.touk.pl:7060 tcp: localhost:7060 udp: venom:7060 udp: venom.touk.pl:7060 udp: localhost:7060 *: sip.rd.touk.pl:* WARNING: no fork mode and more than one listen address found (will use only the the first one) 0(11117) init_tcp: using epoll_lt as the io watch method (auto detected) 0(11117) Maxfwd module- initializing 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) WARNING: xl_mod_init: more IP 192.168.129.38 not used 0(11117) AVPops - initializing 0(11117) using 'none' subscription authorization for presence! 0(11117) TRACE: qsa_interface.c:55: PA: PA_CONTENT_TYPE: 0x2ba1844692e0 0(11117) INFO: xcap_mod.c:55: xcap module initialization 0(11117) TRACE: virtual_subscription.c:55: RLS: RLS_PRESENCE_INFO: 0x2ba1844692e0 0(11117) TRACE: virtual_subscription.c:63: RLS: RLS_RAW: 0x2ba184479598 0(11117) using 'none' rls-subscription authorization! 0(11117) TRACE: qsa_events.c:331: presence_b2b: b2b_CONTENT_TYPE: 0x2ba1844692e0 0(11117) WARNING: tls: mod_init: tls support is disabled (set enable_tls=1 in the config to enable it) exec - initializing ERROR: error -1 while trying to fix configuration Best regards tomasz

3 4

[OpenSER-Users] Modifing the SIP header fields in OpenSER
by Skender Benattia 05 Feb '08

05 Feb '08

Hello everyone; I would like to rewrite the "To" field of all the SIP messages OpenSER receives in order to make some tests. Such a function exists for the originating part using rewriteuri or rewritehost, but I could not find anything for the destination part. Is it possible ? If yes, does somebody have a clue ? It would be very helpful. Thank you very much in advances. Best Regards, Skander

3 3

Re: [OpenSER-Users] CDRTool rating on test system (without PSTN)
by Andy Smith 05 Feb '08

05 Feb '08

:P This was simple to fix, turns out my admin account in CDRTool didnt have full admin privileges as far as CDRTool was concerned. Not sure if Id accidentically messed it up or if the default is like that... Well anyway, in case anyone else is as dumb as me in future :D now the answer is on the list, cheers Andy. >However via the web interface in the CDR Section for the cost it is >displayed as "x.xxx" and no data is displayed in the detail section "Rating >Information" either... Any ideas what the issue is?

1 0

[OpenSER-Users] Accurate CDR with RTPProxy
by VoIP Forums www.Go4Calls.com 05 Feb '08

05 Feb '08

Hi all, Is it possible to get accurate CDR of openser using RTPProxy? If ATA did not send BYE message to openser can RTPProxy disconnect the call using some rtptimeout then ACCT module log the correct duration in MySQL database? Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 2

[Serusers] ser 2.1.0-dev16-tcp avpops module compilation error
by Tomasz Zieleniewski 05 Feb '08

05 Feb '08

Hi, I was trying to compile sources from trunk and I encoutered following error: make[1]: Entering directory `/home/sip/src/ser-trunk/modules/avpops' gcc -fPIC -DPIC -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev16-tcp"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.2.3"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/home/sip/ser/trunk/test/etc/ser/"' -DMODS_DIR='"/home/sip/ser/trunk/test/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c avpops.c -o avpops.o In file included from avpops_impl.h:41, from avpops_parse.h:38, from avpops.c:50: avpops_db.h:66: error: expected ')' before '*' token make[1]: *** [avpops.o] Error 1 make[1]: Leaving directory `/home/sip/src/ser-trunk/modules/avpops' make: *** [modules] Error 1 Best regards tomasz

1 0

[OpenSER-Users] Missing information in Migration Guide from 1.2.2 to 1.3.0.
by Sergio Gutierrez 05 Feb '08

05 Feb '08

Hello everybody. Trying to migrate from 1.2.2 version to 1.3.0 version, I discovered there is a lost item in migration guide; The structure of gw table changes, because in 1.3.0 version, it has an additional column, so this table must be rebuilt. Best regards. Sergio Gutiérrez

4 8

[OpenSER-Users] Viewing OpenSER debug output
by Brian Heath 04 Feb '08

04 Feb '08

Hi all, Is there a way that you can 're-connect' to an openser debug output process, so you don't have to restart openser every time you want to view the debug data? I've currently got it set accordingly: ################# # openser.cfg # ################ debug=9 fork=yes log_stderror=yes log_facility=LOG_LOCAL1 Alternately, if there's a way I can pipe it all into a text file, and then use tail -f that would be optimal. But I don't mind it as it all flies down my screen - it's at least all in my buffer. Thanks in advance for your help, Brian

2 3

Re: [OpenSER-Users] MSILO back notify not working
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

On Monday 04 February 2008 18:27:28 Dan-Cristian Bogos wrote: > Hi Iñaki, > > in order to send message confirmation, you need to instruct openser to do > so, it will not do automatically this for you. > > In the documentation you have some sample config how to do it ( > http://www.openser.org/docs/modules/1.3.x/msilo.html#AEN390): > > # MSILO - storing as offline message > if (m_store("$ru")) > { > log("MSILO: offline message stored\n"); > if (!t_reply("202", "Accepted")) > { > sl_reply_error(); > }; > }else{ > log("MSILO: offline message NOT stored\n"); > if (!t_reply("503", "Service Unavailable")) > { > sl_reply_error(); > }; > }; > break; > }; Hi Dan, I've read the fully MSILO module documentation, and my openser script implements the code above. But it does not send any notification MESSAGE to the sender. In fact I don't see in the code you paste where the notification MESSAGE is generated by OpenSer. As I said, message storage works ok for me, as well the message retrieving when destination user registers, but there is not notification MESSAGE (and I've set all the required parameters). Any suggestion? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] XMPP GW Error "invalid content-type 0x100ff"
by Ali Jawad 04 Feb '08

04 Feb '08

Hi All I have the following setup X-Lite Client -> Openser With XMPP GW Module <-> ejabberD on the same server <- exodus XMPP client I have enabled presence on the Openser server this allowed me to chat using the Openser server. The next step was to use the XMPP GW module to enable XMPP features using an ejabberd server. I changed the configuration as shown below and logged in using test2 on the exodus XMPP client. I added test2 to the x-lite client and chose send instant message, the x-lite client immediately returned "Error: Your contact is not online." The Openser logs return the following: Feb 4 17:05:52 [11929] DBG:xmpp:cmd_send_message: cmd_send_message Feb 4 17:05:52 [11929] DBG:core:parse_headers: flags=ffffffffffffffff Feb 4 17:05:52 [11929] ERROR:xmpp:cmd_send_message: invalid content-type 0x100ff My Config is below: http://pastebin.com/m7707753a

3 9

Re: [OpenSER-Users] Mediaproxy error no attribute 'autocommit' RESOLVED
by Andy Smith 04 Feb '08

04 Feb '08

Ok, simple fix... Updated to MySQL_python-1.2.2 and all ok now! :D ----- Original Message ----- From: Andy Smith To: users(a)lists.openser.org Sent: Monday, February 04, 2008 4:01 PM Subject: Mediaproxy error no attribute 'autocommit' Hi there, Im trying to run a Mediaproxy and Im recieving this error, Feb 4 14:53:14 server proxydispatcher[22800]: 'Connection' object has no attribute 'autocommit' I´ve had a google for this and its seems like it may be a bug python-sqlite or sqlite bug , but also I tried searching for it on the python issue traker and didnt find anything. Is anyone else aware of this issue and any solution? Im running: Python 2.3.4 MySQL-python.i386 1.0.0-1.RHEL4.1 python-sqlite.i386 1.1.7-1.2.1 sqlite.i386 3.3.6-2 mysql server 5.1.22-rc thanks for any help! Andy.

1 0

[OpenSER-Users] Problem with OpenSER when MySQL database fails
by Arturo Díaz Almagro 04 Feb '08

04 Feb '08

Hello all, Hope anybody could help me. We have a SIP telephony system based on OpenSER with a redundant MySQL database ring. When the "master" database dies, a monit process detect it and modify properly the database entry in the DNS zone file. Two tests can be perform to check the availability of the system when MySQL crashes: - kill mysql process in the database host machine. In this case OpenSER asks the DNS inmediately for the database IP address and the service keeps going as expected. No problem here. - shutdown the interface of the machine where mysql is hosted. In this case OpenSER does not ask the DNS for the new address. It keeps trying over and over the connection with the master and until the interface is not turn on the system keeps hung. So, if the mysql's host interface is shutdown the system hangs but if the mysql process dies the system works properly. In the first case both machines (OpenSER and MySQL) has IP visibility, not in the second, so the problem should be related to that. Has anybody experienced this behaviour? Does anybody know how to solve it? Thanks a lot. -- Arturo Díaz Contact me on FWD: 870436 Skype: arturo.diaz.almagro

1 0

[OpenSER-Users] MSILO back notify not working
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

Hi, I've configured MSILO module. It works well except the fact that I don't get any notification back to the sender about the storage of the message sent by him. This is: when user_A sends a MESSAGE to user_B and this is stored by MSILO module I don't see any MESSAGE from OpenSer to user_A notifing it. I have set: ------------------------------ # -- msilo -- modparam("msilo", "registrar", "sip:user-offline@mydomain.org") modparam("msilo", "reminder", "sip:reminder@mydomain.org") modparam("msilo", "expire_time", 86400) modparam("msilo", "check_time", 30) modparam("msilo", "send_time", 60) modparam("msilo", "clean_period", 5) modparam("msilo", "use_contact", 1) modparam("msilo", "add_date", 1) modparam("msilo", "max_messages", 20) ------------------------------ As I read in the doc: ------------------------------ 1.3.3. registrar (string) The SIP address used to inform users that destination of their message is not online and the message will be delivered next time when that user goes online. If the parameter is not set, the module will not send any notification. ------------------------------ But I've set that parameter, but when a message is stored I don't see any MESSAGE originated by OpenSer to notify the sender. Shoul I see a MESSAGE from OpenSer? I don't understant too what "reminder" parameter means: ------------------------------ 1.3.4. reminder (string) The SIP address used to send reminder messages. If this value is not set, the reminder feature is disabled. ------------------------------ Any explanation please? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] ERROR:mysql:db_mysql_convert_row: no private memory
by Iñaki Baz Castillo 04 Feb '08

04 Feb '08

Hi, why could this error occur? It has ocurred me now several times but never seen it before ¿? Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_row: no private memory left Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_rows: error while converting row #1 Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_convert_result: error while converting rows Feb 4 15:31:31 [27228] ERROR:mysql:db_mysql_store_result: error while converting result Feb 4 15:31:31 [27228] ERROR:core:db_do_raw_query: error while storing resultFeb 4 15:31:31 [27228] ERROR:usrloc:get_all_db_ucontacts: raw_query failed I will investigate it but any comment will be welcome. Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

4 9

[OpenSER-Users] Mediaproxy error no attribute 'autocommit'
by Andy Smith 04 Feb '08

04 Feb '08

Hi there, Im trying to run a Mediaproxy and Im recieving this error, Feb 4 14:53:14 server proxydispatcher[22800]: 'Connection' object has no attribute 'autocommit' I´ve had a google for this and its seems like it may be a bug python-sqlite or sqlite bug , but also I tried searching for it on the python issue traker and didnt find anything. Is anyone else aware of this issue and any solution? Im running: Python 2.3.4 MySQL-python.i386 1.0.0-1.RHEL4.1 python-sqlite.i386 1.1.7-1.2.1 sqlite.i386 3.3.6-2 mysql server 5.1.22-rc thanks for any help! Andy.

1 0

[OpenSER-Users] Call transfer
by Jan ONDREJ (SAL) 04 Feb '08

04 Feb '08

Hello, can somebody give me a working example of configuration file, supporting "Call transfer" or proper "REFER" passing? Call transfer does not work with default openser.cfg and also with my extended configuration or sipwise generated configuration. My phones supports call transfer and it works well with asterisk, but no success with openser. Can somebody give me an example config or hints, what to do to make a working call transfer? Thank you. SAL

4 9

Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by A.smith 04 Feb '08

04 Feb '08

Ok, Rating engine seems to be working ok when I have a canonicalURI starting "00" and in the database I see this for a rated call: Span: 1 Duration: 75 s StartTime: 2008-02-02 12:33:15 (UK) ProfileId: 442 for weekend RateId: 442 for 0-24h Rate: 0.0200 / 60 s Price: 0.0250 However via the web interface in the CDR Section for the cost it is displayed as "x.xxx" and no data is displayed in the detail section "Rating Information" either... Any ideas what the issue is? thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

1 0

Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by A.smith 04 Feb '08

04 Feb '08

Hey Dan! Yep ur right! I had to have a leading "00" on the canonicalURI for to rate the call! Thanks alot for the info! cheers Andy! :) --------- Original Message -------- From: Dan-Cristian Bogos <dan.bogos(a)gmail.com> To: Andy Smith <a.smith(a)ukgrid.net> Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem Date: 04/02/08 11:28 > Hi Andy, > > I suspect that your problem is related to > var $intAccessCode = '00'; inside cdrlib.phtml. > Based on that all the numbers which should not be altered and be > considered billable should contain "00" as international prefix. > > Have a look on that, perhaps it helps. > > Cheers, > DanB > > On Feb 3, 2008 7:17 PM, Andy Smith <a.smith(a)ukgrid.net> wrote: > > > > > > Hi all, > > > > im trying to "trick" CDRTool into rating calls when my OpenSER isnt actually > > routing anything to a PSTN, Im trying just putting the handset username as > > an internation number without the leading zera (ie 44208888888) but CDRTool > > is rating ever call as zero cost. > > > > Maybe its something to do with this, from the CDRTool rating document: > > > > "The rating logic checks if the TLD exists in the billing_enum_tlds table. > > If it does, the rating engine tries to match the regexp field against the > > username part of the destination from the ENUM response, which has been > > saved in the Canonical URI. The match must return a fully qualified E164 > > number otherwise the call is considered to have ended to a no E164 > > destination and the call will be free of charge." > > > > What constitutes a fully qualified E164 number? 44208888888(a)mydomain.com > > should be ok shouldnt it? I also tried > > +44208888888(a)mydomain.com also without any success... > > > > > > Its quite furstrating cos I cant see the logic as its applied by the rating > > system which would explain how it reaches the zero cost > > result. It is certainly looking at the calls as they are made as its logging > > info to the syslog saying so, eg: > > > > Feb 1 16:11:39 server CDRTool[22869]: ShowPrice Id=94 Timestamp=1201882245 > > Duration=6 inputTraffic=0 outputTraffic=0 From=003(a)domain.net > > To=44208888888(a)domain.net Gateway=x.x.x.x Application=audio ENUMtld= > > Feb 1 16:11:39 server CDRTool[10751]: Normalization done in 0 s, memory > > usage: 4.96 MB > > Feb 1 16:11:39 server CDRTool[10751]: Unlock ser_radius:radacct > > > > > > Any ideas people?? Mediaproxy isnt actually required for billing is it? I > > understood it is optional for the documentation...?? > > > > Thanks for any help! Andy. > > > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > ________________________________________________ Message sent using UK Grid Webmail 2.7.9

1 0

[OpenSER-Users] CDRTool rating on test system (without PSTN) zero cost problem
by Andy Smith 04 Feb '08

04 Feb '08

Hi all, im trying to "trick" CDRTool into rating calls when my OpenSER isnt actually routing anything to a PSTN, Im trying just putting the handset username as an internation number without the leading zera (ie 44208888888) but CDRTool is rating ever call as zero cost. Maybe its something to do with this, from the CDRTool rating document: "The rating logic checks if the TLD exists in the billing_enum_tlds table. If it does, the rating engine tries to match the regexp field against the username part of the destination from the ENUM response, which has been saved in the Canonical URI. The match must return a fully qualified E164 number otherwise the call is considered to have ended to a no E164 destination and the call will be free of charge." What constitutes a fully qualified E164 number? 44208888888(a)mydomain.com should be ok shouldnt it? I also tried +44208888888(a)mydomain.com also without any success... Its quite furstrating cos I cant see the logic as its applied by the rating system which would explain how it reaches the zero cost result. It is certainly looking at the calls as they are made as its logging info to the syslog saying so, eg: Feb 1 16:11:39 server CDRTool[22869]: ShowPrice Id=94 Timestamp=1201882245 Duration=6 inputTraffic=0 outputTraffic=0 From=003(a)domain.net To=44208888888(a)domain.net Gateway=x.x.x.x Application=audio ENUMtld= Feb 1 16:11:39 server CDRTool[10751]: Normalization done in 0 s, memory usage: 4.96 MB Feb 1 16:11:39 server CDRTool[10751]: Unlock ser_radius:radacct Any ideas people?? Mediaproxy isnt actually required for billing is it? I understood it is optional for the documentation...?? Thanks for any help! Andy.

2 1

[OpenSER-Users] Recommended distribution for OpenXCAP
by Sergio Gutierrez 04 Feb '08

04 Feb '08

Hello everybody. What is the linux distribution you recommend to run OpenXCAP Server? What distribution allows an easy installation, and offers better performance? Thanks in advance for your answer. Best regards. Sergio Gutiérrez

2 1

[OpenSER-Users] MSRP Relay release 0.9.1
by Adrian Georgescu 04 Feb '08

04 Feb '08

Hello, AG Projects launches MSRPRelay, an Open Source implementation of IETF standard RFC 4976. MSRP relay is an extension to the MSRP protocol (RFC 4975). Its main role is to help NAT traversal of Interactive Messaging and file transfer sessions for SIP/MSRP endpoints located behind NAT. Background ----------- A series of related instant messages between two or more parties can be viewed as part of a "message session", that is, a conversational exchange of messages with a definite beginning and end. This is in contrast to individual messages each sent independently. Messaging schemes that track only individual messages can be described as "page-mode" messaging, whereas messaging that is part of a "session" with a definite start and end is called "session-mode" messaging. Page-mode messaging is enabled in SIP via the SIP MESSAGE method, as defined in RFC 3428. Session-mode messaging has a number of benefits over page-mode messaging, however, such as explicit rendezvous, tighter integration with other media-types, direct client-to-client operation, and brokered privacy and security. Message Session Relay Protocol (MSRP) is a protocol for transmitting a series of related instant messages in the context of a session. Message sessions are treated like any other media stream when set up via a rendezvous or session creation protocol such as the Session Initiation Protocol (SIP). MSRP Sessions are defined in RFC 4975. Out of the box, the software is designed to work in combinations with the subscriber database used by OpenSER (see http://openser.org). For more information visit http://msrprelay.org Regards, Adrian

2 1

[OpenSER-Users] Openser problem in logs
by Ali Jawad 04 Feb '08

04 Feb '08

Hi All When I start openser using /usr/local/sbin/openser it loads then I login using one user and I keep gettingthe logs below all the time is this normal Feb 1 14:03:54 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:03:54 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:03:54 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:03:54 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:03:54 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:03:54 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:03:54 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:03:54 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=12700000 Feb 1 14:03:54 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:03:54 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:03:54 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (16700000) Feb 1 14:03:54 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:03:58 [29075] DBG:core:parse_msg: SIP Request: Feb 1 14:03:58 [29075] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:03:58 [29075] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:03:58 [29075] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=2 Feb 1 14:03:58 [29075] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:03:58 [29075] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:03:58 [29075] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:03:58 [29075] DBG:core:parse_headers: via found, flags=2 Feb 1 14:03:58 [29075] DBG:core:parse_headers: this is the first via Feb 1 14:03:58 [29075] DBG:core:receive_msg: After parse_msg... Feb 1 14:03:58 [29075] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=100 Feb 1 14:03:58 [29075] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=200 Feb 1 14:03:58 [29075] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:03:58 [29075] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: content_length=0 Feb 1 14:03:58 [29075] DBG:core:get_hdr_field: found end of header Feb 1 14:03:58 [29075] DBG:rr:find_first_route: No Route headers found Feb 1 14:03:58 [29075] DBG:rr:loose_route: There is no Route HF Feb 1 14:03:58 [29075] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:03:58 [29075] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:03:58 [29075] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:03:58 [29075] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:03:58 [29075] DBG:core:db_free_rows: row[0]=0x81841e0 Feb 1 14:03:58 [29075] DBG:core:db_free_rows: 0x81841e0=pkg_free() RES_ROWS Feb 1 14:03:58 [29075] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:03:58 [29075] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:03:58 [29075] DBG:core:parse_headers: flags=78 Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:03:58 [29075] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:03:58 [29075] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:03:58 [29075] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:03:58 [29075] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:03:58 [29075] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:03:58 [29075] DBG:core:receive_msg: cleaning up Feb 1 14:03:58 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=16700000 Feb 1 14:03:58 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:03:58 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:03:58 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (20700000) Feb 1 14:03:58 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:02 [29077] DBG:core:parse_msg: SIP Request: Feb 1 14:04:02 [29077] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:02 [29077] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:02 [29077] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=2 Feb 1 14:04:02 [29077] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:02 [29077] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:02 [29077] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:02 [29077] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:02 [29077] DBG:core:parse_headers: this is the first via Feb 1 14:04:02 [29077] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:02 [29077] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=100 Feb 1 14:04:02 [29077] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=200 Feb 1 14:04:02 [29077] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:02 [29077] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:02 [29077] DBG:core:get_hdr_field: found end of header Feb 1 14:04:02 [29077] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:02 [29077] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:02 [29077] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:02 [29077] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:02 [29077] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:02 [29077] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:02 [29077] DBG:core:db_free_rows: row[0]=0x817fca0 Feb 1 14:04:02 [29077] DBG:core:db_free_rows: 0x817fca0=pkg_free() RES_ROWS Feb 1 14:04:02 [29077] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:02 [29077] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:02 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:02 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:02 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:02 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:02 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:02 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:02 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:04:02 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=20700000 Feb 1 14:04:02 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:02 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:02 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (24700000) Feb 1 14:04:02 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:06 [29075] DBG:core:parse_msg: SIP Request: Feb 1 14:04:06 [29075] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:06 [29075] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:06 [29075] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=2 Feb 1 14:04:06 [29075] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:06 [29075] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:06 [29075] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:06 [29075] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:06 [29075] DBG:core:parse_headers: this is the first via Feb 1 14:04:06 [29075] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:06 [29075] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=100 Feb 1 14:04:06 [29075] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=200 Feb 1 14:04:06 [29075] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:06 [29075] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:06 [29075] DBG:core:get_hdr_field: found end of header Feb 1 14:04:06 [29075] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:06 [29075] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:06 [29075] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:06 [29075] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:06 [29075] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:06 [29075] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:06 [29075] DBG:core:db_free_rows: row[0]=0x817fcc8 Feb 1 14:04:06 [29075] DBG:core:db_free_rows: 0x817fcc8=pkg_free() RES_ROWS Feb 1 14:04:06 [29075] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:06 [29075] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:06 [29075] DBG:core:parse_headers: flags=78 Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:06 [29075] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:06 [29075] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:06 [29075] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:06 [29075] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:06 [29075] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:06 [29075] DBG:core:receive_msg: cleaning up Feb 1 14:04:06 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=24700000 Feb 1 14:04:06 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:06 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:06 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (28700000) Feb 1 14:04:06 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:10 [29077] DBG:core:parse_msg: SIP Request: Feb 1 14:04:10 [29077] DBG:core:parse_msg: method: <SUBSCRIBE> Feb 1 14:04:10 [29077] DBG:core:parse_msg: uri: <sip:ali@jabber.splendor.net> Feb 1 14:04:10 [29077] DBG:core:parse_msg: version: <SIP/2.0> Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=2 Feb 1 14:04:10 [29077] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-d87543-5c484340d1308708-1--d87543->; state=6 Feb 1 14:04:10 [29077] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Feb 1 14:04:10 [29077] DBG:core:parse_via: end of header reached, state=5 Feb 1 14:04:10 [29077] DBG:core:parse_headers: via found, flags=2 Feb 1 14:04:10 [29077] DBG:core:parse_headers: this is the first via Feb 1 14:04:10 [29077] DBG:core:receive_msg: After parse_msg... Feb 1 14:04:10 [29077] DBG:core:receive_msg: preparing to run routing scripts... Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=100 Feb 1 14:04:10 [29077] DBG:maxfwd:is_maxfwd_present: value = 70 Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=200 Feb 1 14:04:10 [29077] DBG:core:parse_to: end of header reached, state=10 Feb 1 14:04:10 [29077] DBG:core:parse_to: display={"ssafass"}, ruri={sip:ali@jabber.splendor.net} Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: <To> [40]; uri=[sip:ali@jabber.splendor.net] Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: to body ["ssafass"<sip:ali@jabber.splendor.net> ] Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: content_length=0 Feb 1 14:04:10 [29077] DBG:core:get_hdr_field: found end of header Feb 1 14:04:10 [29077] DBG:rr:find_first_route: No Route headers found Feb 1 14:04:10 [29077] DBG:rr:loose_route: There is no Route HF Feb 1 14:04:10 [29077] DBG:core:grep_sock_info: checking if host==us: 19==15 && [jabber.splendor.net] == [193.227.186.153] Feb 1 14:04:10 [29077] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Feb 1 14:04:10 [29077] DBG:uri_db:does_uri_exist: User in request uri does exist Feb 1 14:04:10 [29077] DBG:core:db_free_rows: freeing 1 rows Feb 1 14:04:10 [29077] DBG:core:db_free_rows: row[0]=0x8179d78 Feb 1 14:04:10 [29077] DBG:core:db_free_rows: 0x8179d78=pkg_free() RES_ROWS Feb 1 14:04:10 [29077] DBG:core:rewrite_uri: rewriting Request-URI with 'sip:ali@193.227.186.146:6242;rinstance=63a8660ec1b35169' Feb 1 14:04:10 [29077] DBG:tm:t_newtran: transaction on entrance=0xffffffff Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:10 [29077] DBG:core:parse_headers: flags=78 Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: start searching: hash=50458, isACK=0 Feb 1 14:04:10 [29077] DBG:tm:matching_3261: RFC3261 transaction matched, tid=-d87543-5c484340d1308708-1--d87543- Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: REF_UNSAFE: after is 1 Feb 1 14:04:10 [29077] DBG:tm:t_lookup_request: transaction found (T=0x4069bb80) Feb 1 14:04:10 [29077] DBG:tm:t_retransmit_reply: nothing to retransmit Feb 1 14:04:10 [29077] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Feb 1 14:04:10 [29077] DBG:core:destroy_avp_list: destroying list (nil) Feb 1 14:04:10 [29077] DBG:core:receive_msg: cleaning up Feb 1 14:04:10 [29078] DBG:tm:utimer_routine: timer routine:7,tl=0x4069bccc next=(nil), timeout=28700000 Feb 1 14:04:10 [29078] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x4069bb80, SUBSCRIBE ... ) Feb 1 14:04:10 [29078] DBG:tm:set_timer: relative timeout is 4000000 Feb 1 14:04:10 [29078] DBG:tm:insert_timer_unsafe: [7]: 0x4069bccc (32700000) Feb 1 14:04:10 [29078] DBG:tm:retransmission_handler: retransmission_handler : done Feb 1 14:04:11 [29075] DBG:core:udp_rcv_loop: probing packet received from 193.227.186.146 11518 Feb 1 14:04:12 [29078] DBG:tm:timer_routine: timer routine:0,tl=0x4069bce8 next=0x4069bdfc, timeout=31 Feb 1 14:04:12 [29078] DBG:tm:final_response_handler: stop retr. and send CANCEL (0x4069bb80) Feb 1 14:04:12 [29078] DBG:tm:t_should_relay_response: T_code=0, new_code=408 Feb 1 14:04:12 [29078] DBG:tm:t_pick_branch: picked branch 0, code 408 Feb 1 14:04:12 [29078] DBG:tm:is_3263_failure: dns-failover test: branch=0, last_recv=408, flags=0 Feb 1 14:04:12 [29078] DBG:tm:t_should_relay_response: trying DNS-based failover Feb 1 14:04:12 [29078] DBG:tm:relay_reply: branch=0, save=0, relay=0 Feb 1 14:04:12 [29078] DBG:core:parse_headers: flags=ffffffffffffffff Feb 1 14:04:12 [29078] DBG:core:check_via_address: params 193.227.186.146, 192.168.0.176, 0 Feb 1 14:04:12 [29078] DBG:core:_shm_resize: resize(0) called Feb 1 14:04:12 [29078] DBG:tm:insert_timer_unsafe: [2]: 0x4069bbc8 (36) Feb 1 14:04:12 [29078] DBG:tm:relay_reply: sent buf=0x8179e70: SIP/2.0 4..., shmem=0x40699c70: SIP/2.0 4 Feb 1 14:04:12 [29078] DBG:tm:final_response_handler: done Feb 1 14:04:12 [29078] DBG:tm:timer_routine: timer routine:0,tl=0x4069bdfc next=(nil), timeout=31

2 1

[OpenSER-Users] anonymous LDAP bind issue
by antalsia＠free.fr 04 Feb '08

04 Feb '08

Hi, I'm trying to implement LDAP authentication with anonymous LDAP bind. I set the ldap configuration file without ldap_bind_dn, ldap_bind_password attributes. This step works fine. Unfortunately, I can't figure out how to set the openser.cfg file. I need to pass the bind DN and the user password to the ldap_search function ; that's ok for the bind DN but I don't know how to procede for the password. Can someone post an example please ? Regards,

2 4

Re: [OpenSER-Users] Mediaproxy Vs RTP Proxy
by Andy Smith 03 Feb '08

03 Feb '08

Hi Ali, there is a good description on the mediaproxy module doc page, near the bottom (nathelper). Should clear things up for you! :P http://openser.org/docs/modules/1.3.x/mediaproxy.html cheers Andy.

1 0

[Serusers] Help needed with SER imprementations
by voip termination 02 Feb '08

02 Feb '08

Hello, I´m new to SER Express and I needed to use SER, on the following imprementations, but do not know which to use. I want to use this imprementations for free in network internet calling, calling card via asterisk (a2billing), and PSTN calls for all the network users. 1. I need to setup SER, to routes all my incoming and outgoing sip calls, (in network users to users) free sip calls 2. Routes all PSTN calls to Asterisk for billing via a2billing and then Asterisk will do the PSTN call terminations of the calls. 3. All the sip user to be authenticated by Asterisk / a2billing for the billing via their user name and password. 4. SER to use asterisk for the users voicemail and other pbx features. Can anyone please help me out? Looking to hearing from you. Regards, Henry -- ================================ "Never Let Go Of A Dream Untill You Can Wakeup And Make It Happens" ================================

1 0

[OpenSER-Users] match src_ip
by Andreas Heise 02 Feb '08

02 Feb '08

Hi all, I search for a simple way to match src_ip from 192.168.1.6 - 192.168.1.11. Any ideas, the following aren't working :-( ... if(src_ip==192.168.1.[6-11]) if(src_ip==192.168.1.[6|7|8|9|10|11]) thanks, Andreas

3 5

[OpenSER-Users] Openser, inbound SIP "trunks" and REFER
by Robby Dermody 01 Feb '08

01 Feb '08

All, I have a situation where I have a OpenSER server frontending a set of Asterisk boxes (call them box A1 and A2). Calls come in and are routed to one of the boxes (after doing some custom HTTP lookups using a custom perl module, rewriting the RURI, etc). This works fine, however at some point, the call will most likely need to be transferred from one internal Asterisk box to the other Asterisk box (e.g. A1 to A2). At this point, what we have set to happen is that A1 would send a REFER up through the proxy, the proxy caches an auth token record from info in the outgoing REFER and writes the token ID into the Refer-To header (along with its own address as the recipient) as the REFER goes out to the SIP trunk, and then when the subsequent invite is made from the sip trunk (to something like route_34983948-34343@<sipProxyIP> with that 34983948-34343 being the token ID) the SIP proxy will look up that auth token, rewrite the RURI with the cached info and direct the call to the appropriate internal asterisk box...A2 in this case. So that all is done and would normally work. However our SIP provider says that it only provides "inbound DID service" and thus do not have their dialplan set up to handle REFERs with custom URIs (even if it is back down to the same openser proxy, as it always is in our case). So looking at this, I have two questions: 1. Could resolving this problem just be as simple as finding a new SIP DID provider? I am a bit worried that providers may consider use of REFERs in any way as "SIP termination" (e.g. outgoing service) and either not office REFER support or charge accordingly for outbound access, even though this is clearly still inbound. 2. Those Asterisk boxes in our environment are behind a NAT, so our openser proxy does employ nathelper/rtpproxy. Given this, another alternative I see is for the openser proxy to intercept the REFER for the DID->A2 message, craft an INVITE for proxy->A2, negotiate the conversation, and reconstruct the internal RTP leg between it and A2 instead of A1. This way, the SIP trunk is none the wiser and does not even know this transfer has happened, it just gets some dead air for a second or two and everything is done behind the scenes. I know that a SIP proxy shouldn't inject new SIP messages into the stream, and the REFER way seems to be the most RFC-compliant and technically sound method. However, there is the real-world threat in my mind that it may not be very well supported among inbound DID providers, so I am interested in the community's opinion on the options. Thanks, Robby

1 0

[OpenSER-Users] Route calls to PSTN Gateway
by Ali Jawad 01 Feb '08

01 Feb '08

Hi All I have successfully setup OpenSer with a mysql backend. I am able to make calls using sip phones between registered users. What I need to do know is routing calls that start with 00 to my PSTN gateway. But first I need to setup OpenSer to route calls to my PSTN ..! I am using OpenSer1.3 and I have checked sipwise.com but honestly I prefer changing my own config file. But from sipwise.com I concluded that I need the config below But are the nathelper and mediaproxy compiled by default or do I have to recompile with special options ? Other than that where do I set the IP of my PSTN gateway ? Any suggestions or help would be welcome thanks. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket","/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics","/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics","/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so"

2 2

[OpenSER-Users] Wrong To: Field In SIP Packet when Sending to PSTN GW
by Ali Jawad 01 Feb '08

01 Feb '08

Hi All I am using OpenSER as a proxy to make outbound calls my config is very simple if the number dialled is not an openser account route it to the PSTN gw. if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } } else { # probably a call to pstn.... route(2); return; } and route[2] { # pstn handling, simply route out to pstn. sethostport("xx.xx.xx.xx:5060"); route(1); } The problem is that once the SIP packet arrives at the PSTN GW it does NOT have the correct TO: set. Therefore the call does not get routed . In the example below TO: is sip:calledNumber@myOpenserDomain instead of sip:calledNumber@PSTN.GATEWAY.IP.IP Caller: ali [!at] jabber.splendor.net (replace the [!at] with a @) Callee: 009613041708 OpenSerDomain: jabber.splendor.net U +0.289348 PSTN.GW.IP.IP:5060 -> 193.237.226.252:5060 SIP/2.0 404 Not Found . Via: SIP/2.0/UDP 193.237.226.252;rport;branch=z9hG4bK6828.10c0315.0. Via: SIP/2.0/UDP 192.168.0.176:65068;received=193.227.186.146;branch=z9hG4bK-d87543-be62c 55d821be10d-1--d87543-;rport=65068. Record-Route: <sip:193.237.226.252;lr=on>. From: "ssafass" <sip:ali@jabber.splendor.net>;tag=f36d6608. To: "009613041705" <sip:009613041705@jabber.splendor.net>;tag=GR52RWG346-34. Call-ID: 0942e159a72eab40ZmViZWY4YTVlOTRlOGJmZTM5ZDdkZGJiZjFmMTlmMjk.. CSeq: 1 INVITE. Contact: "0000" <sip:PSTN.GW.IP.IP:5060>. User-Agent: eyeBeam release 1003s stamp 31159. Content-Length: 0. I did a siptrace on the interface of the SIP proxy http://pastebin.com/d56426d63 <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/d56426d63> This is my config: http://pastebin.com/m128ca16e <http://www.voipuser.org/ship_to.php?url=http://pastebin.com/m128ca16e>

2 4

[OpenSER-Users] Not able to connect Application Server to the SEAS module of OpenSER.
by Rajat Dudeja 01 Feb '08

01 Feb '08

Hi Folks, My openser is running on SCTP. I'm using SEAS module. I am using a simple client socket (TCP Socket) program as Application Server(AS) to connect to SEAS module. As is written to connect to port 5080 of server. But it does not connect as is evident from the line (openser' console log): Feb 1 17:30:47 [12031475] INFO:seas:dispatcher_main_loop: polling [2 ServSock] [1 pipe] [0 App Servers] [0 Uncomplete AS] I'm sending a REGISTER message from a SCTP Client to OpenSER on port 5070. I'm expecting the same REGISTER to be passed to the AS. I'm using the following openser.cfg file : ####### Global Parameters ######### #debug=3 #log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=1 dns=no /* uncomment the following lines to enable debugging */ debug=9 #fork=no log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "//etc/openser/tls/user/user-cert.pem" #tls_private_key = "//etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "//etc/openser/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=SCTP:157.227.110.27:5070 ####### Modules Section ######## #set module path #mpath="//lib/openser/modules/" mpath="/home/ops/openser/lib/openser/modules/" 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 "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" #Rajat-Jiten added 01.02.08 loadmodule "seas.so" # ----------------- setting module-specific parameters --------------- #Rajat-Jiten added 01.02.08 # ----- seas params ----- modparam("seas", "listen_sockets","157.227.110.27:5080") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/abc/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- modparam("usrloc", "db_mode", 0) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") #modparam("auth_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ #modparam("domain", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## route{ if(!as_relay_t("app_server")){ t_reply("500","Application Server error"); } } =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

1 0

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION
by bhargav.b＠renovau.com 01 Feb '08

01 Feb '08

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser for clients authorization. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

4 4

[OpenSER-Users] openser+opendiameter integration problem----->KINDLY PAY ATTENTION
by bhargav.b＠renovau.com 01 Feb '08

01 Feb '08

hello everybody, i have some problem regarding the integratoion b/w open diameter and openser. i have recently installed openser and opendiameter as needed.both are working fine independently.but for the integration i need somebodies kind help.i have run all the sample examples of both. even my softphone[xlite]is doing fine with openser.can anybody please tell me how can i configure the openser.cfg to support diameter? please also sujjest me regarding avp's creation for the diameter+openser support. thank you very much bhargav.b bhargav.b(a)renovau.com

1 0

Re: [OpenSER-Users] Reg. Proxy domains
by Padmaja 01 Feb '08

01 Feb '08

Hi.. That is the whole issue,,, The proxy 1 and proxy2 do not know each other. How can I configure openser so that it recognizes proxy1 as another proxy and sends/receives invites? I do not want any of the proxies acting as out bound proxy. Both should be primary proxies for their respective domains and yet they should be able to recognize the other proxy and forward the requests to it. In other words, what I am asking how do the proxies talk to each other? Thanks, Padmaja ----- Original Message ----- From: raviprakash sunkara To: Padmaja Sent: Thursday, January 31, 2008 1:55 PM Subject: Re: [OpenSER-Users] Reg. Proxy domains Hello Padmaja Akka, IS urs Proxy1 sending the INVITE to Proxy2.. Can u trace by using ngrep or trace and Send to u .... On Jan 31, 2008 12:39 PM, Padmaja <padmaja.rv(a)vodcalabs.com> wrote: Hi all, Please see the situation below: UA1 ----------------------> Proxy 1(not openser)---------------------- Proxy 2(openser)<----------------------UA2 UA1 is registered to proxy 1 and UA 2 to openser. I wish to make a call from UA 1 to UA 2. Proxy 1 and proxy 2 are in the LAN but do not know of each other. Is there a way we can configure the openser to send/receive the invites to proxy 1? I think the calls will be handled on domain basis, so openser will treat all the UAs registered to it as local and allow them to call each other. But how to make it allow inter domain calls? Please let me know. Thanks, Padmaja _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

2 1

[OpenSER-Users] XMPP Gateway
by Ali Jawad 31 Jan '08

31 Jan '08

Dear All I am trying to achieve a setup where users can both chat using XMPP and do voice calls through VOIP. I have already setup eJabberD and I like it's features and I am almost done setting up OpenSer. I have read about the XMPP Gateway module ? If I do use this plugin will this synchronize the users created between the two systems if I do create them on the OpenSer server ? Has anyone used the module ? If I do use it will the contact lists for the users be stored on the SIP server or the XMPP server ? What about presence will the XMPP be responsible for the presence changes or the OpenSer server ? Thanks

1 0

[OpenSER-Users] Mediaproxy Vs RTP Proxy
by Ali Jawad 31 Jan '08

31 Jan '08

Dear All I know that there are two possible setups one usng mediaproxy and another one using rtpproxy. How do these two proxies differ from each other ? And if I chose NOT to use any of these proxies would I be able to make calls through a PSTN gateway located on the Internet ? Thanks.

1 0

[OpenSER-Users] Help Sending calls to PSTN
by Ali Jawad 31 Jan '08

31 Jan '08

Hi there my config file is http://pastebin.com/d33669ee8 I have set modparam("nathelper", "ping_nated_only", 1) modparam("registrar", "nat_flag", 6) This gives me the following error: Jan 31 16:47:26 [16380] ERROR:core:set_mod_param_regex: parameter <nat_flag> not found in module <registrar> Jan 31 16:47:26 [16380] CRITICAL:core:yyerror: parse error in config file, line 173, column 19-20: Can't set module parameter If I set modparam("nathelper", "ping_nated_only", 0) and ### modparam("registrar", "nat_flag", 6) This allows me to run openser but when I make a call to PSTN the following warning is sent out Jan 31 16:55:45 [16484] ERROR:nathelper:send_rtpp_command: can't connect to RTP proxy Jan 31 16:55:45 [16484] ERROR:nathelper:send_rtpp_command: proxy </var/run/rtpproxy.sock> does not respond, disable it Jan 31 16:55:45 [16484] WARNING:nathelper:rtpp_test: can't get version of the RTP proxy Jan 31 16:55:45 [16484] WARNING:nathelper:rtpp_test: support for RTP proxy </var/run/rtpproxy.sock> has been disabled temporarily Calls between softphones work but a call to the PSTN ends up with the following Jan 31 16:57:26 [16509] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 International PSTN Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=80 Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=ffffffffffffffff Jan 31 16:57:26 [16509] DBG:nathelper:check_content_type: type <application/sdp> found valid Jan 31 16:57:26 [16509] DBG:core:parse_headers: flags=40 ===>Jan 31 16:57:26 [16509] ERROR:nathelper:force_rtp_proxy2_f: no available proxies Routed to PSTN gateway Jan 31 16:57:26 [16509] DBG:tm:t_unref: UNREF_UNSAFE: after is 0 Jan 31 16:57:26 [16509] DBG:core:destroy_avp_list: destroying list (nil) Jan 31 16:57:26 [16509] DBG:core:receive_msg: cleaning up Jan 31 16:57:27 [16514] DBG:tm:utimer_routine: timer routine:4,tl=0x406ddf7c next=(nil), timeout=25100000 Jan 31 16:57:27 [16514] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x406dde30, INVITE si ... ) Jan 31 16:57:27 [16514] DBG:tm:set_timer: relative timeout is 1000000 Jan 31 16:57:27 [16514] DBG:tm:insert_timer_unsafe: [5]: 0x406ddf7c (26100000) Jan 31 16:57:27 [16514] DBG:tm:retransmission_handler: retransmission_handler : done Jan 31 16:57:28 [16514] DBG:tm:utimer_routine: timer routine:5,tl=0x406ddf7c next=(nil), timeout=26100000 Jan 31 16:57:28 [16514] DBG:tm:retransmission_handler: retransmission_handler : request resending (t=0x406dde30, INVITE si ... ) If I use tcpdump I don't see any traffic leaving for my PSTN GW. With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 4

[OpenSER-Users] dp and dP pseudo-vars
by Victor Gamov 31 Jan '08

31 Jan '08

Hi gentlemen! I use openser-1.2.2 and I want to access to dp & dP variables but sometimes they are empty. I try to access it before and after t_relay() and result is the same -- variable is empty. When this variables are set? Is it possible that they are not set at all? Thanks! -- CU, Victor Gamov

3 4

[OpenSER-Users] Reg. Proxy domains
by Padmaja 31 Jan '08

31 Jan '08

Hi all, Please see the situation below: UA1 ----------------------> Proxy 1(not openser)---------------------- Proxy 2(openser)<----------------------UA2 UA1 is registered to proxy 1 and UA 2 to openser. I wish to make a call from UA 1 to UA 2. Proxy 1 and proxy 2 are in the LAN but do not know of each other. Is there a way we can configure the openser to send/receive the invites to proxy 1? I think the calls will be handled on domain basis, so openser will treat all the UAs registered to it as local and allow them to call each other. But how to make it allow inter domain calls? Please let me know. Thanks, Padmaja

1 0

[Serusers] SER/FIFO not accessible
by Steve B. 31 Jan '08

31 Jan '08

I am new to SER. I have installed SER 0.9.6-2 on a fresh Kubuntu server using apt. I have been able to get SER to run using the sample ser.cfg files (only changing module paths from /usr/local/lib... to /usr/lib/...). I am trying to setup the mysql support. In a nutshell, when I run serctl I get an error: # serctl add sbelt(a)ser.mytestsite.com sbelt sbelt(a)mytestsite.com MySql password: (here I enter 42serro11) error: SER/FIFO not accessible: 2 # Am I using the correct password? I am not sure which Mysql Password is being requested here. I have also tried the mysql root password with the same result. I have seen other posts similar that seem to indicate it is permissions on the /tmp directory. I tried adding to ser.cfg fifo_mode=0777. I have verified the current user can write to the /tmp directory. Frankly, I do not know if this /tmp/ser_fifo is all a red herring. Thanks for any help on how I can figure out what I am doing wrong. Steve B. Here is my ser.cfg # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 fork=no log_stderror=yes listen=192.168.32.103 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0777 fifo_db_url="mysql://serro:47serro11@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://serro:47serro11@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()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } else 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("","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 (!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(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }

1 0

[Serusers] SER and user=phone parameter
by simon 30 Jan '08

30 Jan '08

Hi all, i have some issue with my SER server and somes IP phones. I'm using a SER 2.0 Somes phones doesn't what login in the server because its send their username <sip:login@domain.tld:5060;user=phone> The SER server try to search the domain named : "domain.tld:5060;user=phone" and doesn't find it. How can I configure my SER server to make the difference between domain and parameter ? Thanks, Simon

3 2

[OpenSER-Users] Disconnect A SIP User, how ?
by Ali Jawad 30 Jan '08

30 Jan '08

Dear All I need a way to detect if a user is logged in and disconnect him if possible. A better solution would be to detect if he is in a call and disconnect him. My questions would be 1. How to detect if a user is connected ? 2. How to detect if a user is currently in a call ? 3. How to disconnect a user 4. How to stop the call for a user (If possible) 5. If dialog is needed is my install below sufficient ? I was told in the IRC channel with regard to Q 3 that I can use the dialog module to achieve that through sending bye using dlg_end_dlg mi command. I checked the documentation at http://www.openser.org/docs/modules/1.3.x/dialog.html#AEN578 but I do not have modparam("dialog" // and so on lines in my openser.cfg file I have installed 1.3 with MySQL, Jabber and Presence support using the following First I edited the MakeFile and deleted the modules I want from the exclude_modules entry, it showed up as follows exclude_modules?= cpl-c postgres osp unixodbc \ avp_radius auth_radius group_radius uri_radius xmpp \ pua pua_bla pua_mi \ pua_usrloc pua_xmpp rls mi_xmlrpc perl perlvdb \ ldap carrierroute h350 xcap_client db_berkeley seas NICER=1 make all make prefix=/usr/local install Does that mean that the dialog module is installed? I could not find a loadmodule "dialog" line!! With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

1 0

Re: [OpenSER-Users] CFG problem with 3rd party tool SOLVED(ish)
by Andy Smith 30 Jan '08

30 Jan '08

Hi, ok, I found and solved one issue. The acc radius_extra entries seem to have a maximum lenght that openser can parse so when I break the configuration into two "lines" like this: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat)") modparam("acc", "radius_extra", "User-Agent=$hdr(user-agent); \ Event=$hdr(event); \ Contact=$hdr(contact); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") #Billing-Party=$avp(billing_party); \ It all works, I hope OpenSER isnt ignoring the first entry in the cfg file!!!! :S. But its only working if I remove the entry "Billing-Party=$avp(billing_party)", if I change this to "Billing-Party=$avp(s:billing_party)", then all starts fine. Just hope thats how its meant to be! :P Andy.

2 2

Re: [OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 30 Jan '08

30 Jan '08

If I was guessing (without fully understanding the syntax!) Id say the problem was Billing-Party=$avp(billing_party); should be Billing-Party=$avp(s:billing_party); Although in theory both of the above config entries seem to be valid, though different, openser.cfg entries. According to this: http://www.openser.org/docs/modules/1.2.x/avpops.html :S Also, I tried with the second format (with s:) and that then gives me another error when I start openser: 0(0) INFO:xl_parse_name: using hdr type (27) instead of <user-agent> 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(s:billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 319 0(0) acc:parse_acc_extra: error 0(0) ERROR:acc:mod_init: failed to parse rad_extra param 0(0) init_mod(): Error while initializing module acc ERROR: error while initializing modules

2 1

Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 30 Jan '08

30 Jan '08

Hi Iñaki, ah yeah, I hadnt noticed this before, and diameter. However have you tested this to see if it writes data in exactly the same format to a DB as Freeradius would? For sure CDRTool has been written to work with freeradius and will therefore need the data in a freeradius format/schema... Andy. ----- Original Message ----- @Andy: > but Radius will do all other updates to DB accounting data. But the point here is that accounting is done by OpenSer ACC module, which can use radius or mysql to fill "radacct" table.

2 1

[Serusers] Question on Accounting
by Nhadie 30 Jan '08

30 Jan '08

Hi, I'm recently added voip on our services using ser. i bill my users using the time of INVITE and BYE. but got a complain from one of my users, why even if the call was not answered they still get billed, what can i use on ser to check whether a call has been answered or not, then i will bill only base on the answered call. Thanks in advanced. Regards, Ron

2 2

[OpenSER-Users] Does CDRTool require Radius server?
by Iñaki Baz Castillo 30 Jan '08

30 Jan '08

Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;) -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[Serusers] ser.cfg (forward to ser-sems)
by dim 30 Jan '08

30 Jan '08

hi.. I have attached my "running" ser.cfg...in which i want to include something like : if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } ... in order to uris start from 1 to be forwarded to my other ser (stack for sems).. Where exactly in my ser.cfg, should i put this ??? thank you very much... dimitris

2 1

[Serusers] Check on-connected status on SER
by roayu 30 Jan '08

30 Jan '08

Hi, I would like to know whether SER can know the following status? For instance, (1) UA1(1000) is on conversation with UA2(1005) ? (2) UA1(1000) is on conversation with UA2(1005), then UA1 transfer to UA(1006)? Is there any logs that we can keep track? Thanks Roayu

2 1

[Serusers] SER Config
by Rami Abualhasan 30 Jan '08

30 Jan '08

Hi, I'm trying to configure ser to be able to act as a sip registerer and proxy sip simple message between the sip UAC and UAS. I'd like also to enable mediaproxy and natping. It will be really appreciated if some one can give the correct version of ser or openser that can fulfil my requirement with a config file which will load the modules I need. Regards

2 1

[OpenSER-Users] CFG problem with 3rd party tool
by Andy Smith 29 Jan '08

29 Jan '08

Hi list, can someone more familiar with openser CFG syntax tell me why follong the instructions for CDRTool, here: http://download.dns-hosting.info/CDRTool/doc/INSTALL.txt , breaks OpenSER?? With the following acc module settings OpenSER won't start up: modparam("acc", "radius_extra", "User-Name=$Au; \ Calling-Station-Id=$from; \ Called-Station-Id=$to; \ Sip-Translated-Request-URI=$ruri; \ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp(s:can_uri); \ Billing-Party=$avp(billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$hdr(X-RTP-Stat); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event); \ SIP-Proxy-IP=$avp(s:sip_proxy_ip); \ ENUM-TLD=$avp(s:enum_tld)") I get this when starting OpenSER: 0(0) xl_parse_vname: error - bad avp name [billing_party] 0(0) ERROR:acc:parse_acc_extra: parse failed in <User-Name=$Au; Calling-Station-Id=$from; Called-Station-Id=$to; Sip-Translated-Request-URI=$ruri; Sip-RPid=$avp(s:rpid); Source-IP=$si; Source-Port=$sp; Canonical-URI=$avp(s:can_uri); Billing-Party=$avp(billing_party); Divert-Reason=$avp(s:divert_reason); X-RTP-Stat=$hdr(X-RTP-Stat); User-Agent=$hdr(user-agent); \Contact=$hdr(contact); Event=$hdr(event); SIP-Proxy-IP=$avp(s:sip_proxy_ip); ENUM-TLD=$avp(s:enum_tld)> around position 198 0(0) acc:parse_acc_extra: error Thanks for any help! Andy.

2 1

Re: [OpenSER-Users] Does CDRTool require Radius server?
by Andy Smith 29 Jan '08

29 Jan '08

Hi Iñaki, my understanding of the documentation is that mediaproxy can update the (Radius) DB in theevent of a missing SIP BYE, but Radius will do all other updates to DB accounting data. Doesthat sound right to you? Do you have any documentation to the contrary etc?cheers Andy.Hi, I'm starting reading about CDRTool and as AFAIK: - MediaProxy can update call media info directly into database (without using radius). - OpenSer can log accounting in database (with "exrta-accounting"). So my question is: does CDRTool require a radius server? why exactly? Thanks for any tip ;)

1 0

[Serusers] SER/OpenSER Consultant
by Brian Peters 29 Jan '08

29 Jan '08

Hi, We are looking for someone for offer consultancy services for a SER/OpenSER deployment. Please advise if you offer those services or know of anyone that does. Thank you, Brian Peters mailto: brian(a)proximosystems.com

1 0

[OpenSER-Users] ABout Diameter
by dilip.m＠renovau.com 29 Jan '08

29 Jan '08

hello everyone, Anyone has used opendiameter with openser. i can run diameter client and diameter server but about the users i dont know.. How can we add users in the diameter? SO please anyone have any idea then tell me. Regards, Dilip

2 2

[OpenSER-Users] OpenSER XMPP Gateway Feature
by Ali Jawad 29 Jan '08

29 Jan '08

Dear OpenSER Users I would like some info about the XMPP gateway feature, I do have a running XMPP server ejabbard, and I would like to use a client that uses both XMPP and SIP. So I am setting up the OpenSER server and I noticed the XMPP gateway feature. What does it do ? I what scenarios can it be applied ? Thanks With Regards Ali Jawad

1 1

[OpenSER-Users] how to implement handoff?
by Aaash shah 29 Jan '08

29 Jan '08

hi i want to implement soft handoff using SIP, so please give me some information regarding it. And also i dont know how to use openSER and how to register it! i am trying to do register by kphone but registration failed. so please give me some information regarding this problems. thanks shital -- "Never Ignore Anyone Who Cares 4 u, Becoz one day u will Realize that u have lots a diamond while u were buzy in collecting stones...."

2 1

[OpenSER-Users] Strip Digits
by Jonathan Creasy 29 Jan '08

29 Jan '08

I am sending an INVITE to Box A for "F+18007622265@boxa" Part of GW Table: gw_name grp_id ip_addr port uri_scheme transport strip prefix toll-free failover 5 XXXXXXXX /NULL/ /NULL/ /NULL/ 1 Part of LCR Table: id prefix from_uri grp_id priority 170 F+1800 5 1 This sends the call to box B XXXXXXXX per the GW table: Jan 29 04:03:16 boxb-sip /usr/local/sbin/openser[28118]: Req 41070 INVITE From <sip:asterisk@XXX.XXX.XXX.XXX> To <sip:F+18007622265@boxa> RURI <sip:+18007622265@boxb> It strips the F off in the RURI field but not in the TO field so boxb then fails the call as it doesn't know how to route a call to F+1800NXXXXXX. What can I do to make it re-format the to: field? The original problem I am trying to solve is that toll-free calls are routed to a system. If that system cannot handle the call I want that system to then place the call back to the OpenSER box (boxa) but instead of using the grp_id of 2 that +1800 calls would go to (and route to the system in question), I want to indicate that boxa should use group 5 and send the calls to boxb. There is probably a very easy way of doing this which doesn't involve a mangled dial string but I'm missing it so if anyone would like to assist me with this that would be great.

1 0

[OpenSER-Users] [OT] Any UAC sending INVITE with empty SDP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

Hi, in order to try managing NAT when the SDP negoziation is done via "200 OK" - "ACK" (instead of "INVITE - "200 OK") I need a UAC that can send INVITE with empty SDP in order to force the called to generate the SDP offer in the reply. Is there any UAC allowing it? Thanks a lot. -- Iñaki Baz Castillo

4 6

[OpenSER-Users] Increased heap size.
by Sergio Gutierrez 28 Jan '08

28 Jan '08

Hello everybody. Currently, I am watching a non normal behaviour in OpenSER 1.2.1, no tls. My operating System is Solaris 10, running on Sparc The process which appears as timer, is consuming almost the 50% of the whole physical memory of the machine. After a deep inspection of memory distribution, appears that the most of memory consumption is at process heap, as presented with pmap(1M) command: 1978: /toip/sbin/openser -n 9 -m 256 -f /toip/etc/openser/prueba1.cfg Address Kbytes RSS Anon Locked Mode Mapped File 00010000 760 752 - - r-x-- openser 000DC000 24 24 24 - rwx-- openser 000E2000 56 32 24 - rwx-- [ anon ] 000F0000 192 128 - - rwx-- openser 00120000 3906432 3906432 3627200 - rwx-- [ heap ] EE800000 262144 3624 - - rw-s- [ anon ] ... Thanks in advance for any hint about it. Best regards. Sergio Gutiérrez. EPM Telecomunicaciones S.A. E.S.P. Medellin, Colombia.

2 5

[OpenSER-Users] dual nic at single machine openser
by flatworm 28 Jan '08

28 Jan '08

I got some routing problem configuring two NICs at debian. Openser is listening all possible network interface. I've configured one for static global IP address and one for internal private IP address. The reason why I've configured like this is for maintenance and development purpose. The static global IP line is for the service and the private IP line is for monitoring and administrations. But when I try to connect to openser of static global IP address, it was failed to register. 1) My machine was using private IP address of same subnet of openser connected to our private network. 2) My machine is trying to connect openser of static global IP address. In case of configuring dual NICs on Openser machine, is there any routing problem ? So I restricted the openser only listening static global IP address then it allows connection and succeeds in registration.

2 1

[OpenSER-Users] Implementing the Ringing Groups
by raviprakash sunkara 28 Jan '08

28 Jan '08

Hello Users Like Asterisk PBX I implemented the Ringing Group like Ring all, Round Robin and Memory Hunt , RingAll and Round Robin Ringing is working fine with the following UserAgent LinkSys 941 and X-lite, GrandStream, And SJphones, with these is working, When comes to Memory Hunt Ring feature only with Sjphone are working fine, but with User Agent not working the problem is Previous User-Agent number is Not ringing. Can any one tell me solution Why these user-Agents are not working ........ Here is my Memory Hunt logic First A rings Second time A and B rings ( in Failure route Ist time ) Third Time A, B and C ( in Failure route 2 nd time ) Fourth time A, B,C and D rings ( in Failure route 3 time ) -- Thanks &Regards Ravi Prakash Sunkara VoIP Development Tech Lead

3 2

[OpenSER-Users] Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

2 1

Re: [OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 28 Jan '08

28 Jan '08

El Sábado, 26 de Enero de 2008, escribió: > Hello, > > Why not just make allowed_from_numbers a regex, and feed it directly? > I.e. > > $avp(allowed_from_numbers) = "(100|200|300|400)"; > > if($avp(allowed_from_numbers) =~ $fU){ > # allowed > } else { > # not allowed > } > > > Hope this helps! Yes, sure, it's a good idea ;) Thanks. -- Iñaki Baz Castillo

2 1

[OpenSER-Users] Fw: Are there are any special settings to run openser over SCTP?
by rajat.dudeja＠tcs.com 28 Jan '08

28 Jan '08

1 0

[OpenSER-Users] rtpproxy seems to be idle
by Kovács Zoltán 26 Jan '08

26 Jan '08

Hi, my plan is to give the possibility for multiple VoIP users within the same LAN to call each other using OpenSER as the SIP server. I found that if there is a NATed LAN, two different VoIP users inside the NATed LAN cannot talk to each other properly, because IP parts of the RTP packets are of the form of the local network and it is not acceptable by the VoIP software which is prepared for the IP of the router instead. (I hope I understand clearly what is going on here.) As a workaround, I found a suggestion in the internet: using rtpproxy with OpenSER. As I can understand, all RTP packets are sent to the rtpproxy process via OpenSER (nathelper module). rtpproxy will behave as a bridge between the two NATed VoIP clients lying in the same LAN. Both clients will initiate their RTP/UDP traffic on their own side and answers will come from the rtpproxy process just for the same UDP pipe. Using this method the clients do not need to allow incoming UDP traffic, only outgoing traffic should be enabled (which is usually allowed by default in most LANs). Here comes my question. I successfully set up OpenSER (version 1.1.x, but I will migrate to 1.2.x soon). I think I also successfully set up rtpproxy, but it seems that there is no real traffic between OpenSER and rtpproxy. Here I attach my command line for running rtpproxy ( 200.200.200.200 is my public IP of the OpenSER server host; I run rtpproxy on the same server). -------------------------8X snip here------------------------------- # rtpproxy -f -l 200.200.200.200 -s unix:/var/run/rtpproxy.sock -r /home/rtpproxy -S /home/rtpproxy.work rtpproxy started, pid 24568 received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "V" sending reply "20040107 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "VF 20050322" sending reply "1 " received command "U 3331228473621-29341876517920(a)160.114.144.65 160.114.144.65 10004 196215176;1" new session 3331228473621-29341876517920(a)160.114.144.65 , tag 196215176;1 requested, type strong new session on a port 35000 created, tag 196215176;1 pre-filling caller's address with 160.114.144.65:10004 sending reply "35000 212.92.3.218 " session timeout RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped session on ports 35000/0 is cleaned up -------------------------8X snip here------------------------------- My debug=7 output for OpenSER, filtered for the string "rtp" comes here (I had to chown openser.openser /var/run/rtpproxy* for OpenSER to make it possible to communicate with rtpproxy at all): -------------------------8X snip here------------------------------- # tail -f daemon.log | grep rtp Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_sock> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_disable_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_tout> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: set_mod_param_regex: found <rtpproxy_retr> in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: find_cmd_export_t: found <force_rtp_proxy>(0) in module nathelper [/usr/lib/openser/modules/] Jan 15 00:27:39 firewall openser: f<force_rtp_proxy> Jan 15 00:27:39 firewall /usr/sbin/openser[25093]: fixing force_rtp_proxy, line 443 Jan 15 00:27:39 firewall /usr/sbin/openser[25095]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25096]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25098]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25100]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25106]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25105]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25117]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25103]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25108]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25104]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25112]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25094]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25113]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25093]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25111]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25122]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25118]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25123]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25097]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25107]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25125]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25120]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25115]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25110]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25119]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25116]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25124]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25114]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25121]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25099]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25102]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25109]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:27:40 firewall /usr/sbin/openser[25101]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Jan 15 00:28:21 firewall /usr/sbin/openser[25112]: force_rtp_proxy2: proxy reply: 35000 212.92.3.218 Jan 15 00:28:27 firewall /usr/sbin/openser[25114]: build_res_from_sip_res: copied size: orig:77, new: 16, rest: 780 msg= SIP/2.0 200 Ok^M Via: SIP/2.0/UDP 160.114.144.65:1024;branch=z9hG4bK27994379851422672^M From: 06212527755 <sip:06212527755@sip.ephone.hu:5060>;tag=196215176^M To: 06212527722 < sip:06212527722@sip.ephone.hu:5060>;tag=7816428^M Contact: < sip:06212527722@160.114.144.65:5060>^M Record-Route: <sip:200.200.200.200;lr;ftag=196215176>^M Call-ID: 3331228473621-29341876517920(a)160.114.144.65^M CSeq: 1 INVITE^M Content-Type: application/sdp^M Server: X-Lite release 1105d^M Content-Length: 320^M ^M v=0^M o=06212527722 2057424946 2057430520 IN IP4 160.114.144.65^M s=X-Lite^M c=IN IP4 160.114.144.65^M t=0 0^M m=audio 8000 RTP/AVP 0 8 3 98 97 101^M a=rtpmap:0 pcmu/8000^M a=rtpmap:8 pcma/8000^M a=rtpmap:3 gsm/8000^M a=rtpmap:98 iLBC/8000^M a=rtpmap:97 speex/8000^M a=rtpmap:101 telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv^M -------------------------8X snip here------------------------------- My nathelper related part of openser.cfg comes here: -------------------------8X snip here------------------------------- loadmodule " nathelper.so" modparam("nathelper", "natping_interval", 60) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "received_avp", 42) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") -------------------------8X snip here------------------------------- Of course, directories /home/rtpproxy* are empty. Could somebody please help me why there is no media stream between OpenSER and rtpproxy? Your help is greatly appreciated in advance. I am using Debian Etch. Best regards, Zoltan

2 4

[OpenSER-Users] Is possible to store various values in an AVP?
by Iñaki Baz Castillo 25 Jan '08

25 Jan '08

Hi, I'd like to store in an AVP various posible values and load it with "load_credentials" parameter of "auth" module. The DB column would be: "allowed_from_numbers" AndI'd like to store there the valid values a subscribe can use as From username. For example: allowed_from_numbers = "200;201;200" So in OpenSer I do: if ($avp(allowed_from_numbers) == $fU) route(FROM_VALID); But unfortunatelly I think this is not possible, is it? which would be the required syntax to store various values? Thanks. -- Iñaki Baz Castillo

1 0

[OpenSER-Users] Revealing Number Dialled
by Jon Farmer 25 Jan '08

25 Jan '08

Hi I have some users successfully using openser and they are able to make and receive calls with any problems. I now want to provide inbound service from other telephone numbers to these registed users and want to be able to put the number dialled into the SIP header. However I am unsure how to represent this in the SIP message. Can anyone help me? TIA Jon __________________________________________________________ Sent from Yahoo! Mail - a smarter inbox http://uk.mail.yahoo.com

2 1

Re: [OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Thanks Christian! I will use this module then in addition, didn't actually realize it is available for OpenSER :-( So far, I just checked the LDAP module, and on http://www.openser.org/dokuwiki/doku.php/tutorials:openser-auth-ldap was no hint on the H.350.4 module when showing the sample LDAP tree. I will try to implement the H.350.4 into my LDAP server and authenticate against it with OpenSER and LDAP+H.350 module. Thanks so far! Best regards Sebastian -----Original Message----- From: Christian Schlatter [mailto:cs@unc.edu] Sent: Friday, 25. January 2008 4:57 PM To: Schumann Sebastian Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] Schema for OpenSER LDAP module Sebastian, There is an ITU-T Recommendation called H.350 that standardizes LDAP schemas for storing SIP (and other session signaling protocols) account data. The documentation of the OpenSER 1.3 H350 module (http://www.openser.org/docs/modules/1.3.x/h350.html) further describes this standard. The "H.350 Cookbook" includes LDAP schema files ready for use with different LDAP server implementations: http://metric.it.uab.edu/vnet/cookbook/v2.1/node143.html /Christian Schumann Sebastian wrote: > Dear all > > I am wondering whether there is a schema file for OpenSER LDAP module, > to use at the LDAP server for data storage? > > On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and > SIPPassword are used, but no standard schema actually defines them acc. > my research. > > Any help with this is as always appreciated! > > Thanks and a nice weekend to all of you! > > Regards > Sebastian > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users

1 0

[OpenSER-Users] Schema for OpenSER LDAP module
by Schumann Sebastian 25 Jan '08

25 Jan '08

Dear all I am wondering whether there is a schema file for OpenSER LDAP module, to use at the LDAP server for data storage? On the tutorial from OpenSER Wiki, e.g. the attributes SIPUserName and SIPPassword are used, but no standard schema actually defines them acc. my research. Any help with this is as always appreciated! Thanks and a nice weekend to all of you! Regards Sebastian

2 1

[OpenSER-Users] OpenSUSE packages required
by Dowling, Dan (Com US) 25 Jan '08

25 Jan '08

Hi, Can anyone tell me how to best determine which OpenSUSE packages are required for openser when used with asterisk? Best regards, Dan

1 0

[OpenSER-Users] presence: difference between watchers and active_watchers
by Klaus Darilion 25 Jan '08

25 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

2 1

[OpenSER-Users] Intial INVITES without SDP
by Patrick Miccio 25 Jan '08

25 Jan '08

Hi @ all, I am fighting a pretty annoying problem at the moment. As described in rfc3261 / 13.2.1 Creating the Initial INVITE o The initial offer MUST be in either an INVITE or, if not there, in the first reliable non-failure message from the UAS back to the UAC. In this specification, that is the final 2xx response. My problem is the following: I receive a initial INVITE without SDP, forward it to a PSTN Gateway. >From the Gateway I receive the 200 OK with SDP offer. In the onreply_route I make a call to the mediaproxy use_media_proxy(); Unfortunately use_media_proxy() seems to make a lookup command when used in a reply route. I would need to make a request command which would create the media session and then do a lookup with the ACK from the UAC. Is there a way around this? cheers, Patrick. This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

3 6

[OpenSER-Users] test
by NP南頻電信黃明杰網路電話部 jeff 25 Jan '08

25 Jan '08

南頻電信股份有限公司 NP Telecom Co., Ltd. 網路電話部 VOIP Dept 黃明杰 Jeff Huang E-Mail：jeff(a)np.net.tw 公司電話(Tel)：+886-2-8226-5008 Ext：867 公司網路代表號：070-288-1122 公司傳真(FAX)：+886-2-8226-5558 公司地址：23562台北縣中和市建八路十六號七樓之一 7F-1, NO. 16, Jian-ba Rd., Chung-ho City 23562, Taipei County, Taiwan, R.O.C. 公司網址：www.np.net.tw 統一編號：89459253 匯款銀行：彰化銀行雙和分行匯款帳號：5678-01-00999-1-00

1 0

[OpenSER-Users] WeSIP: Not SER connector available - Check the Configuration, new user issue help!
by rahul kadian 25 Jan '08

25 Jan '08

Hi, I am trying to configure WESIP and trying to run demo application "Click2dial" Everything seems to ve connected properly according to logs But when I call the manager Web-GUI I get as a status of OpenSER: "Not SER connector available - Check the Configuration" What might be the reason for this. Also while trying to run Click2dial , i get the error code "Call failed with code 2" or sometimes with code 3. Have checked the confi and server.xml according to the docs and seem to be running fine. If someone can let me know any solution to this , it would be of great help. -- Regards, Rahul.

1 0

[OpenSER-Users] SVN checkout of 1.2.3?
by Tobias Lindgren 24 Jan '08

24 Jan '08

Hi, I must be getting really tired, but checking out 1.2 from svn only brings me up to 1.2.2, not 1.2.3? Same with latest snapshot from 1.2. # $Id: ChangeLog 2609 2007-08-16 14:02:25Z miconda $ Br, /Tobias

2 2

Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 24 Jan '08

24 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Thanks Henning.... > Please see my answers below : > > [..] > According to this,in the "failure route" is checked if receives from the > gateway the messages "408" or "5XX", but since the gateway is offline there > is no way to receive any response from it. So, what is the behaviour of > the module here?. It keeps sending the calls to the de-2.carrier1 and > de-3.carrier1 gateways? or keeps sending the calls between the 3 gateways? Hi, this has nothing to do with the behaviour of carrierroute.. OpenSER will try to resend the invite, and then generate a local 408 after the timer of the tm module run out. And then you can enter a failure route. > > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > > probably have a reply from the gateway (maybe a "480" message"), so the > > carrierroute module now tries with a failover route (defa.carrier1?) or > > tries with any of the other two gateways still not full? > > I suppose you mean that 'de-1' is overloaded. No, the next domain is not > automatically entered. This deficiency is known, and will be addressed in > the future. But this code is not ready yet for a release. > > Yes sorry.. i meant "de-1" is overloaded, so in this case, i can receive a > "408" Message from the gateway indicating maybe "overload". So i handle > the answer with the failure route[2], so all the calls answered with the > "408" are re-routed to the "domain 1" route isn't?. Wouldn't be more > accurate to keep sending calls to the other gateways (de-2 and de-3)? > insted of make a failover to a default route? since the other two gateways > are not overloaded...... Well, 408 is not the right error for overloading, more appropriate would be 503. Carrierroute has no knowledge of the meaning and your policy regarding a 408/ 503, let alone that the two other gws are not overloaded.. A hardcoded logic in this module would not fit to everybody. At the moment you must go with the failure_routes, or use the lcr module instead. In the future there will be probably some better solution, as i wrote in my first mail. Cheers, Henning

1 0

[OpenSER-Users] Openser 1.3 How to rewrite transport protocol from udp to tcp after redirect
by Peter P GMX 24 Jan '08

24 Jan '08

Hello, I have the following scenario: SIP UA <=UDP=> OpenSER <=TCP> Microsoft Unified Messaging 2007 When I invite an account on the UM 2007 server the following happens 1 SIP UA ==Invite UDP=> OpenSER 2 SIP UA <== 407 Proxy Authentication Required UDP= OpenSER 3 SIP UA ==ACK UDP=> OpenSER 4 SIP UA ==Invite UDP=> OpenSER 5 OpenSER ==Invite TCP=> UM2007 6 OpenSER <=Trying TCP== UM2007 7 OpenSER <= 302 Moved Temporarily TCP== UM2007 8 OpenSER == ACK TCP=> UM2007 9 OpenSER == INVITE UDP=> UM2007 .....no answer from UM2007 (I left out some steps between Openser and UA) In Step 5 OpenSER invites "sip:11111@10.10.2.13;transport=UDP>" with TCP protocol (checked with ngrep) and UM2007 answers. In Step 7 UM2007 answers with CONTACT: <sip:11111@10.10.2.13:5065;transport=UDP> but I need to answer on TCP to UM2007. (I can change the message with add_uri_param("transport=tcp") in step 5 and I get a <sip:11111@10.10.2.13:5065;transport=TCP> back from UA2007, but in fact the message is corrupted and OpenSER replies to the UA with Error 500 then, see below) The point is: - I can invite UM2007 with TCP (although I was not able to produce correct headers as there is still "transport=udp;>" mentioned in the INVITE part) - but in the failure route when I force OpenSER to reinvite with get_redirects("*") and t_relay() I am not able to swith to TCP I tried several ways as described in discussions which I found in http://osdir.com/ml/voip.openser.user/2005-12/msg00256.html - add_uri_param("transport=tcp"); ==> added transport=tcp additionally to transport=udp ==> transport=udp;transport=tcp ==> 500 Internal Server error - $avp(s:foo)="sip:10.10.2.13;transport=tcp"; avp_pushto("$du","$avp(s:foo)"); ==>had actually no effect - replace_all("transport=udp", "transport=tcp"); ==> produced invalid headers as it simply replaced the headers wrongly as described in http://openser.org/pipermail/devel/2007-May/007702.html Questions: 1. How can I change the initial INVITe message (Step5) in order to have the right headers. (transport=TCP instead of transport=UDP)? 2. How can I answer to a redirect and change to a different transport protocol (Step 9)? Your help is very much appreciated. Best regards Peter

4 10

[OpenSER-Users] Lookinfg for an SCTP compliant SIP phone
by Rajat Dudeja 24 Jan '08

24 Jan '08

Hi folks, I'm new to openser. I'm using the sctp feature of it. I'm looking to test the openser's sctp functionality. For this I have used the sctp compliant kphone. However, it does not work. The phone does not even register. Please suggest if there is any other tool to it's functionality. Thanks and regards, Rajat =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] presence: difference between watchers and active_watchers
by Klaus Darilion 24 Jan '08

24 Jan '08

Hi! Can someone explain me please the difference between between watchers and active_watchers? If I understand correctly, active_watchers are the SIP clients which have SUBSCRIBEd and thus will receive NOTIFYs. But what are "watchers"? regards klaus

3 3

[OpenSER-Users] User Agent (Client)
by J Bhanot 24 Jan '08

24 Jan '08

Hi, I am running openser Server... I want to know that do we have any inbuit user agent (client) in openser which we can use to send messages to this running server.. Thanks, jb ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[Serusers] ser , rtpproxy and pstn gateway
by mashudi 24 Jan '08

24 Jan '08

I have two separate network, for public and vpn , i have install ser and rtpproxy on different machine, all user from public network will register to proxy server on ip 10.44.9.18 and the rtp will be handle by another machine using ip 10.44.9.17 , all call comming from client on public ip will route to pstn gateway using cantata media gateway on ip 10.177.107.90 (vpn ip). When client call to pstn gateway, the signalling reach the callee, but after callee pickup the phone, the client using x-lite still in calling status, and rtpproxy not receive rtp from pstn gateway. the rtpproxy have two nic using two ip for public and vpn, and ser as proxy using two nic public ip and vpn ip. any one can help? here is the log for rtpproxy and ser. the rtpproxy log : Starting rtpproxy: rtpproxy started, pid 25745 received command "2573_0 V" sending reply "2573_0 20040107 " received command "2574_0 V" sending reply "2574_0 20040107 " received command "2572_0 V" sending reply "2572_0 20040107 " received command "2572_6 U NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI. 10.11.11.152 30126 9d08dc55" new session NjQzNmQzNzFjZTBmMzM5MDUxYjQwODFlOTFmMTdhMzI., tag 9d08dc55 requested, type strong new session on a port 35010 created, tag 9d08dc55 pre-filling caller's address with 10.11.11.152:30126 sending reply "2572_6 35010 10.44.9.17 " the ser log: 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=461 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=55 , global msg id=54 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24011, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bKbcd5.79134365.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=56 global id=55 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-9533d56650536c16-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <13> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24011 label 1446261143 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=56 global id=56 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-9533d56650536c16-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 13 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <1> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=16384 0(2572) pre_auth(): Credentials with given realm not found 0(2572) build_auth_hf(): 'Proxy-Authenticate: Digest realm="10.44.9.18", nonce="4798817c850ce0e3c233a35804d9be1afe7fe2a3" ' 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-2623ee4d6908a453-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=4 0(2572) DEBUG: add_param: tag=a3f1a545ddce5205bb83afef24613e34.78aa 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 0(2572) SIP Request: 0(2572) method: <INVITE> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) end of header reached, state=9 0(2572) DEBUG: get_hdr_field: <To> [43]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18> ] 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <2> <INVITE> 0(2572) DEBUG: get_hdr_body : content_length=420 0(2572) found end of header 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) check_nonce(): comparing [4798817c850ce0e3c233a35804d9be1afe7fe2a3] and [4798817c850ce0e3c233a35804d9be1afe7fe2a3] 0(2572) HA1 string calculated: 0f7e2f302e87dfe46c5ca6566ad32b47 0(2572) check_response(): Our result = '82c47bef0adf71299b8fb157a15e0131' 0(2572) check_response(): Authorization is OK 0(2572) save_rpid(): rpid value is '' 0(2572) check_username(): Digest username and URI username match 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) parse_headers: flags=64 0(2572) lookup(): '02170890010' Not found in usrloc 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=-1 0(2572) DEBUG:check_content_type: type <application/sdp> found valid 0(2572) parse_headers: flags=32 0(2572) ERROR: extract_mediaip: no `c=' in SDP 0(2572) DEBUG: t_newtran: msg id=59 , global msg id=56 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) SER: new INVITE 0(2572) parse_headers: flags=-1 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: reply sent out. buf=0x6c9930: SIP/2.0 1..., shmem=0x2ab57bba3ad8: SIP/2.0 1 0(2572) DEBUG: _reply_light: finished 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6bebb8 0(2572) WARNING: get_send_socket: not listening on the requested socket, no fork mode? 0(2572) parse_headers: flags=2048 0(2572) parse_headers: flags=-1 0(2572) clen_builder: content-length: 436 (436) 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:5,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[6]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:6,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) udp_rcv_loop: probing packet received from 10.11.11.152 19710 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, INVITE si ... ) 1(2573) DEBUG: add_to_tail_of_timer[7]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 1(2573) DEBUG: timer routine:0,tl=0x2ab57bba4228 next=(nil) 1(2573) DEBUG: final_response_handler:stop retr. and send CANCEL (0x2ab57bba4030) 1(2573) ->>>>>>>>> T_code=100, new_code=408 1(2573) DEBUG: relay_reply: branch=0, save=0, relay=0 1(2573) parse_headers: flags=-1 1(2573) check_via_address(10.11.11.152, 10.11.11.152, 0) 1(2573) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4150 1(2573) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4170 1(2573) DEBUG: reply relayed. buf=0x6c84d0: SIP/2.0 4..., shmem=0x2ab57bba3ad8: SIP/2.0 4 1(2573) DEBUG: final_response_handler : done 1(2573) DEBUG: timer routine:7,tl=0x2ab57bba4208 next=(nil) 0(2572) SIP Request: 0(2572) method: <ACK> 0(2572) uri: <sip:02170890010@10.44.9.18> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-363a1c17b37ac509-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(2572) parse_headers: flags=128 0(2572) DEBUG: add_param: tag=d548cf07be509c4dc64b42f06e6fb880-a911 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [85]; uri=[sip:02170890010@10.44.9.18] 0(2572) DEBUG: to body ["02170890010"<sip:02170890010@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <2> <ACK> 0(2572) DEBUG: get_hdr_body : content_length=0 0(2572) found end of header 0(2572) DEBUG: is_maxfwd_present: max_forwards header not found! 0(2572) DEBUG: add_param: tag=9d08dc55 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) find_first_route: No Route headers found 0(2572) loose_route: There is no Route HF 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) DEBUG: t_newtran: msg id=60 , global msg id=59 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=48512, isACK=1 0(2572) DEBUG: RFC3261 transaction matched, tid=-d87543-363a1c17b37ac509-1--d87543- 0(2572) DEBUG: t_lookup_request: transaction found (T=0x2ab57bba4030) 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4150 next=(nil) 1(2573) DEBUG: timer routine:2,tl=0x2ab57bba40b0 next=(nil) 1(2573) DEBUG: wait_handler : removing 0x2ab57bba4030 from table 1(2573) DEBUG: delete transaction 0x2ab57bba4030 1(2573) DEBUG: wait_handler : done 0(2572) SIP Request: 0(2572) method: <NOTIFY> 0(2572) uri: <sip:02170743627@10.11.11.152:65100> 0(2572) version: <SIP/2.0> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <n/a>; state=17 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) preparing to run routing scripts... 0(2572) parse_headers: flags=128 0(2572) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2572) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2572) parse_headers: flags=8 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) DEBUG: add_param: tag=59420d70 0(2572) end of header reached, state=29 0(2572) parse_headers: flags=256 0(2572) is_preloaded: No 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==10 && [10.11.11.152] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) grep_sock_info - checking if host==us: 12==14 && [10.11.11.152] == [10.177.107.124] 0(2572) grep_sock_info - checking if port 5060 matches port 65100 0(2572) check_self: host != me 0(2572) grep_sock_info - checking if host==us: 10==10 && [10.44.9.18] == [10.44.9.18] 0(2572) grep_sock_info - checking if port 5060 matches port 5060 0(2572) after_loose: Topmost route URI: 'sip:10.44.9.18;lr;ftag=9e3f287c' is me 0(2572) parse_headers: flags=256 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: get_hdr_body : content_length=425 0(2572) found end of header 0(2572) find_next_route: No next Route HF found 0(2572) after_loose: No next URI found 0(2572) DEBUG: t_newtran: msg id=61 , global msg id=60 , T on entrance=0xffffffffffffffff 0(2572) parse_headers: flags=-1 0(2572) parse_headers: flags=60 0(2572) t_lookup_request: start searching: hash=24006, isACK=0 0(2572) DEBUG: RFC3261 transaction matching failed 0(2572) DEBUG: t_lookup_request: no transaction found 0(2572) DEBUG: mk_proxy: doing DNS lookup... 0(2572) DEBUG: get_out_socket: socket determined: 0x6be9c8 0(2572) check_via_address(10.11.11.152, 10.11.11.152, 0) 0(2572) DEBUG: add_to_tail_of_timer[4]: 0x2ab57bba4208 0(2572) DEBUG: add_to_tail_of_timer[0]: 0x2ab57bba4228 0(2572) SER: new transaction fwd'ed 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up 1(2573) DEBUG: timer routine:4,tl=0x2ab57bba4208 next=(nil) 1(2573) DEBUG: retransmission_handler : request resending (t=0x2ab57bba4030, NOTIFY si ... ) 1(2573) DEBUG: add_to_tail_of_timer[5]: 0x2ab57bba4208 1(2573) DEBUG: retransmission_handler : done 0(2572) SIP Reply (status): 0(2572) version: <SIP/2.0> 0(2572) status: <200> 0(2572) reason: <OK> 0(2572) parse_headers: flags=1 0(2572) Found param type 232, <branch> = <z9hG4bK6cd5.c487f5f5.0>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=1 0(2572) parse_headers: this is the first via 0(2572) After parse_msg... 0(2572) forward_reply: found module tm, passing reply to it 0(2572) DEBUG: t_check: msg id=62 global id=61 T start=0xffffffffffffffff 0(2572) parse_headers: flags=17 0(2572) Found param type 232, <branch> = <z9hG4bK-d87543-3217dc739e312401-1--d87543->; state=6 0(2572) Found param type 235, <rport> = <65100>; state=16 0(2572) end of header reached, state=5 0(2572) parse_headers: Via found, flags=17 0(2572) parse_headers: this is the second via 0(2572) DEBUG: add_param: tag=9e3f287c 0(2572) end of header reached, state=29 0(2572) DEBUG: get_hdr_field: <To> [56]; uri=[sip:02170743627@10.44.9.18] 0(2572) DEBUG: to body ["02170743627"<sip:02170743627@10.44.9.18>] 0(2572) get_hdr_field: cseq <CSeq>: <14> <NOTIFY> 0(2572) DEBUG: t_reply_matching: hash 24006 label 1600092236 branch 0 0(2572) DEBUG: t_reply_matching: reply matched (T=0x2ab57bba4030)! 0(2572) DEBUG: t_check: msg id=62 global id=62 T end=0x2ab57bba4030 0(2572) DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0) 0(2572) parse_headers: flags=64 0(2572) parse_headers: flags=64 0(2572) ->>>>>>>>> T_code=0, new_code=200 0(2572) DEBUG: relay_reply: branch=0, save=0, relay=0 0(2572) old size: 531, new size: 472 0(2572) build_res_from_sip_res: copied size: orig:267, new: 208, rest: 264 msg= SIP/2.0 200 OK Via: SIP/2.0/UDP 10.11.11.152:65100;branch=z9hG4bK-d87543-3217dc739e312401-1--d87543-;rport=65100 Record-Route: <sip:10.44.9.18;lr;ftag=59420d70> Contact: <sip:02170743627@10.11.11.152:65100> To: "02170743627"<sip:02170743627@10.44.9.18>;tag=9e3f287c From: "02170743627"<sip:02170743627@10.44.9.18>;tag=59420d70 Call-ID: MWNiZWE5ODMyN2U0MTJmMWMyNjQzMTNiNGJjZmRjYjA. CSeq: 14 NOTIFY User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 0(2572) WARNING:vqm_resize: resize(0) called 0(2572) DEBUG: add_to_tail_of_timer[2]: 0x2ab57bba40b0 0(2572) DEBUG: reply relayed. buf=0x6c9930: SIP/2.0 2..., shmem=0x2ab57bba6b10: SIP/2.0 2 0(2572) DEBUG:destroy_avp_list: destroying list (nil) 0(2572) receive_msg: cleaning up # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=9 fork=no log_stderror=yes mhomed=yes listen=10.44.9.18 # INSERT YOUR IP ADDRESS HERE listen=10.177.107.124 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/uri_db.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/avpops.so" loadmodule "/usr/lib/ser/modules/permissions.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("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "udp:10.44.9.17:12345") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") 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(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; 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); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); 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 (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (uri=~"^sip:0217[0-9]*@") { route(4); route(5); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { rewritehost("10.177.107.90"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; }

1 0

[Serusers] SIP Express Bundle - let's get started
by Mike Trest - Personal 23 Jan '08

23 Jan '08

Hi, The holidays and the new year work load seems to have taken all my time. Anyway, next week, I should be clear of regulatory filings and paperwork. That should give me time to resume building stand alone SER to test prototype of the SER Express Bundle. Any first-draft materials from anyone yet? ..mike..

4 5

[OpenSER-Users] UAC module (openser 1.2.1)
by Antonio Reale 23 Jan '08

23 Jan '08

Hi all, I'm trying to use UAC module and I'd like to know if it is possible to configure openser to register on another proxy on start up and wait for incoming calls, instead of use UAC module to terminate calls on an authenticated trunk. Thanks. Regards, Antonio

4 4

[Serusers] forking error - branch response sent to uac
by Tomasz Zieleniewski 23 Jan '08

23 Jan '08

hi, I encountered following error. I have two user agents register under the same uid. When there is an invite request both ua's ring. When one of them UA1 answers the call, by sending 200 response some strange error occurs. UA1's 200 response is immediately sent to UAC who initiated a call. after this ser sends cancel message to other UA2. The other UA2 responses with 200 to the cancel message and sends response to the initial invite: 487 Request Terminated. After this ser should response with ACK to the second UA2 but it isn't instead it forwards 487 response to UAC who sent the initial invite. Please tell why ser behaves this way? What could be the reason for such behavior? Regards Tomasz

1 0

[OpenSER-Users] Performance of Openser as Presence Server
by Noble Antony T 23 Jan '08

23 Jan '08

Hi, Do any one has an idea about the Performance benchmark of Openser as Presence Server..? Like how many Subscribers, how many subscribe/publish/notify messages, it can handle.. [ I want to know how much Subscribers/Subscriptions & how many Subscribe/Notify/Publish messages, Openser can handle. I am using Openser as only Presence Server ( not proxy/registrar )] If anybody have any figures about this performance, please let me know. Thanks -- Noble Antony Thattil.

1 0

[OpenSER-Users] Reg. Proxy authentication
by Padmaja 23 Jan '08

23 Jan '08

Hi all, Please see the call flow below: GW1 Proxy GW2 |-----invite----------->| | |<------407-----------| | |-----invite w/cred--->| | | |-----Invite------->| |<------------200 Ok---------------------| |----------------Ack--------------------->| | | | |<----ReInvite -----| | When the proxy receives the Reinvite below from GW 2, should it again challenge it for proxy authentication or simply forward the call to the GW1? The openser proxy in our lab simply forwards the Invite without asking for authentication this time. However, I need to test the situation, where the proxy asks for authentication for the Reinvite. Is this possible? Please let me know. Thanks, Padmaja

2 2

[OpenSER-Users] Forwarding to SRV records not working
by James Holden 23 Jan '08

23 Jan '08

Hi, I'm having problems with OpenSER forwarding calls to a domain where there are only SRV records. OpenSER seems not to perform any balancing or failover when a target server is unavailable or gives an error. I wish to forward calls to a number of onward gateways that at times may not respond at all, or may respond with something like "503 Service Unavailable". The actual destination address is derived from an enum lookup, but it may be for example 123456(a)example.com There are SRV records as follows: _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. The behaviour I'd expect/like would be that OpenSER will distribute the calls between the two targets specified in the SRV records, and will fail over to the other target if one fails to respond or gives a fatal error such as a 503. What actually happens, is that the calls all go to the first one regardless. Are my expectations reasonable? Does OpenSER's SRV record implementation work like this or am I misled? Kind regards, James

3 3

Re: [OpenSER-Users] Users Digest, Vol 32, Issue 70
by Padmaja 23 Jan '08

23 Jan '08

----- Original Message ----- From: <users-request(a)lists.openser.org> To: <users(a)lists.openser.org> Sent: Wednesday, January 23, 2008 3:28 AM Subject: Users Digest, Vol 32, Issue 70 > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Forwarding to SRV records not working (James Holden) > 2. Re: Serial fork terminated after 5 call (Bogdan-Andrei Iancu) > 3. How to handle INVITES with different To: header (Peter P GMX) > 4. Re: How to handle INVITES with different To: header > (Bogdan-Andrei Iancu) > 5. Re: Forwarding to SRV records not working (Bogdan-Andrei Iancu) > 6. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 7. OpenSER: no NOTIFY when updating PUBLISH if integrating with > OpenXCAP? (KevinKinnan) > 8. Re: Problems trying to build an Active/Active OSer 1.3 > cluster (Stefan Sayer) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 22 Jan 2008 16:41:42 +0000 > From: James Holden <james.holden(a)uk.aql.com> > Subject: [OpenSER-Users] Forwarding to SRV records not working > To: users(a)lists.openser.org > Message-ID: <20080122164142.GD4315@james-desktop> > Content-Type: text/plain; charset=us-ascii > > Hi, > > I'm having problems with OpenSER forwarding calls to a domain where > there are only SRV records. > > OpenSER seems not to perform any balancing or failover when a target > server is unavailable or gives an error. > > I wish to forward calls to a number of onward gateways that at times may > not respond at all, or may respond with something like "503 Service > Unavailable". > > The actual destination address is derived from an enum lookup, but it > may be for example 123456(a)example.com > > There are SRV records as follows: > > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. > _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. > > The behaviour I'd expect/like would be that OpenSER will distribute the > calls between the two targets specified in the SRV records, and will > fail over to the other target if one fails to respond or gives a fatal > error such as a 503. > > What actually happens, is that the calls all go to the first one > regardless. > > Are my expectations reasonable? > > Does OpenSER's SRV record implementation work like this or am I misled? > > Kind regards, > > James > > > > ------------------------------ > > Message: 2 > Date: Tue, 22 Jan 2008 18:49:21 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Serial fork terminated after 5 call > To: Sergey Babichev <zaikini(a)yandex.ru> > Cc: users(a)lists.openser.org > Message-ID: <47961E91.7080003(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Sergey, > > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > > Regards, > Bogdan > > Sergey Babichev wrote: >> Okey:)) And last question about documentation, I searched documentation >> about function nat_uac_test from nathelper module for version 1.2.x and >> I'm find it in documentation for version 0.9. Openser 1.2.x not support >> function nat_uac_test or documentation 1.2.x not full? >> >> >> >> >>> Hi Sergey, >>> >> >> >>> I'm afraid there is no docs about the errors - the idea is to have >>> >> >> >>> sef-explaining error messages and to avoid documenting them ;) >>> >> >> >>> Regards, >>> >> >> >>> bogdan >>> >> >> >>> Sergey Babichev wrote: >>> >> >> >>>> Thanks, where i can read about this error (discarding fwd for a >>>> cancelled transaction) and others? >>>> >> >> >> >> >> >> >> >> >>>> >>>> >> >> >>>>> Hi Sergey, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> the "discarding fwd for a cancelled transaction" error is generated >>>>> when >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> you try to send a request for a transaction that was already cancelled >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> from the client side. So probably the PSTN GW cancels the call while >>>>> you >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> still want to fork a new branch. >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> regards, >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Bogdan >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>> Sergey Babichev wrote: >>>>> >> >> >>>>> >>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Problem is appear when i calling from PSTN. It's work fine when i >>>>>> calling from sip account to huntgroup. I think this is problem whith >>>>>> timer in my PSTN gateway. >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Hi all! >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> I want enable hunting for my extensions. It's work fine, but if my >>>>>>> hunt group consists of more then 5 extensions, call terminated and I >>>>>>> see in syslog error: >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:t_forward_nonack: discarding fwd for a cancelled >>>>>>> transaction >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> ERROR:tm:w_t_relay: t_forward_nonack failed >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: >>>>>>> DEBUG:tm:relay_reply: branch=5, save=0, relay=5 >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> What is problem may be? >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> -- >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Best regards, >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Sergey Babichev >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> _______________________________________________ >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users mailing list >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> Users(a)lists.openser.org >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>>> >>>>>>> >> >> >>>>>>> >>>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> -- >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Best regards, >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Sergey Babichev >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> _______________________________________________ >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users mailing list >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> Users(a)lists.openser.org >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >>>> >>>> >> >> >> >> >>>> >>>> >> >> >>>>>> >>>>>> >> >> >>>>>> >>>>>> >> >> >> >> >> >> >> >> >>>> -- >>>> >> >> >> >> >>>> Best regards, >>>> >> >> >> >> >>>> Sergey Babichev >>>> >> >> >> >> >>>> >>>> >> >> >> >> -- >> >> Best regards, >> >> Sergey Babichev >> >> > > > > > ------------------------------ > > Message: 3 > Date: Tue, 22 Jan 2008 19:20:44 +0100 > From: Peter P GMX <Prometheus001(a)gmx.net> > Subject: [OpenSER-Users] How to handle INVITES with different To: > header > To: users(a)lists.openser.org > Message-ID: <479633FC.2090506(a)gmx.net> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > I have the following scenario: > > I receive a request with PSTN_number(a)my.ip.add.res from my VoIP > In alias_db I have defined a rule that points to yyy(a)myip.com which is > a registered account > > After alias_db_lookup OpenSER invites as follows: > Invite yyy(a)myip.com > To: xxxxxxx(a)my.ip.add.res > > The UA (actually a PBX with VoIP support) answers with 404 not found. > > When I invite as follows: > Invite yyy(a)myip.com > To: yyy(a)myip.com > > everything works fine. > > My question: How do you handle these cases? > Rewriting the To: part seems dificult and will mess up a lot - hein? > > Kind regards > Peter > > > > > ------------------------------ > > Message: 4 > Date: Tue, 22 Jan 2008 21:41:32 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] How to handle INVITES with different To: > header > To: Prometheus001(a)gmx.net > Cc: users(a)lists.openser.org > Message-ID: <479646EC.5090301(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi Peter, > > According to RFC3261, in SIP, routing is exclusively done based on RURI > (and eventually Route hdrs). TO header has 0 implication in routing. > So, your PBX is deprecated and out of standards and I suggest either > upgrade, either replacement. > > Regards, > Bogdan > > Peter P GMX wrote: >> I have the following scenario: >> >> I receive a request with PSTN_number(a)my.ip.add.res from my VoIP >> In alias_db I have defined a rule that points to yyy(a)myip.com which is >> a registered account >> >> After alias_db_lookup OpenSER invites as follows: >> Invite yyy(a)myip.com >> To: xxxxxxx(a)my.ip.add.res >> >> The UA (actually a PBX with VoIP support) answers with 404 not found. >> >> When I invite as follows: >> Invite yyy(a)myip.com >> To: yyy(a)myip.com >> >> everything works fine. >> >> My question: How do you handle these cases? >> Rewriting the To: part seems dificult and will mess up a lot - hein? >> >> Kind regards >> Peter >> > > > > ------------------------------ > > Message: 5 > Date: Tue, 22 Jan 2008 23:00:28 +0200 > From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> > Subject: Re: [OpenSER-Users] Forwarding to SRV records not working > To: James Holden <james.holden(a)uk.aql.com> > Cc: users(a)lists.openser.org > Message-ID: <4796596C.7000004(a)voice-system.ro> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi James, > > Openser does not do balancing over records with same priority - it will > just use the first (as returned by DNS server). > > But failover based on NAPTR and SRV records is in place. > > Regards, > Bogdan > > > James Holden wrote: >> Hi, >> >> I'm having problems with OpenSER forwarding calls to a domain where >> there are only SRV records. >> >> OpenSER seems not to perform any balancing or failover when a target >> server is unavailable or gives an error. >> >> I wish to forward calls to a number of onward gateways that at times may >> not respond at all, or may respond with something like "503 Service >> Unavailable". >> >> The actual destination address is derived from an enum lookup, but it >> may be for example 123456(a)example.com >> >> There are SRV records as follows: >> >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw1.example.com. >> _sip._udp.example.com. 600 IN SRV 10 50 5060 gw2.example.com. >> >> The behaviour I'd expect/like would be that OpenSER will distribute the >> calls between the two targets specified in the SRV records, and will >> fail over to the other target if one fails to respond or gives a fatal >> error such as a 503. >> >> What actually happens, is that the calls all go to the first one >> regardless. >> >> Are my expectations reasonable? >> >> Does OpenSER's SRV record implementation work like this or am I misled? >> >> Kind regards, >> >> James >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > > > ------------------------------ > > Message: 6 > Date: Wed, 23 Jan 2008 05:49:18 +0800 > From: KevinKinnan <kinnan2224(a)hotmail.com> > Subject: [OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if > integrating with OpenXCAP? > To: <users(a)lists.openser.org> > Message-ID: <BAY126-W641A1FEB850E880B28F56C43E0(a)phx.gbl> > Content-Type: text/plain; charset="gb2312" > > > Thank you so much, Franz! > > Indeed, the problem is coming from xml. It is well-formed but seems the > server can't understand it. > The whole xml document I used is the similar with yours except the root > element with namespace which > is showing below: > > <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" > xmlns:pr="urn:ietf:params:xml:ns:pres-rules" > xmlns:cr="urn:ietf:params:xml:ns:common-policy"> > > > I got no error sending the xml doc with this as beginning. Only no Notify > message after updating Publish, > and sometimes the first Subscribe can't get through that is no Notify > sending back from server. It's really > weired! Anyway, I finally fix it with your advise, Thanks Indeed. > > Kevin, > > _________________________________________________________________ > ????? MSN ?????????? > http://mobile.msn.com.cn/ >

1 0

Re: [OpenSER-Users] Serial fork terminated after 5 call SOLVED
by Sergey Babichev 23 Jan '08

23 Jan '08

Thanks ))) > Hi Sergey, > the most important thing when searching is to know where to search ;)... > See: > http://www.openser.org/docs/modules/1.2.x/nathelper.html#AEN370 > Regards, > Bogdan > Sergey Babichev wrote: > > Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> I'm afraid there is no docs about the errors - the idea is to have > >> > > > > > >> sef-explaining error messages and to avoid documenting them ;) > >> > > > > > >> Regards, > >> > > > > > >> bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi Sergey, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> the "discarding fwd for a cancelled transaction" error is generated when > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> you try to send a request for a transaction that was already cancelled > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> from the client side. So probably the PSTN GW cancels the call while you > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> still want to fork a new branch. > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Bogdan > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev wrote: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Hi all! > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> What is problem may be? > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> -- > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Best regards, > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Sergey Babichev > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> _______________________________________________ > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users mailing list > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> Users(a)lists.openser.org > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > >>>>>> > >>>>>> > > > > > >>>>>> > >>>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> -- > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Best regards, > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> Sergey Babichev > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> _______________________________________________ > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users mailing list > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> Users(a)lists.openser.org > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > >>>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > >>> > >>> > > > > > > > > > >>> > >>> > > > > > >>>>> > >>>>> > > > > > >>>>> > >>>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] Problems trying to build an Active/Active OSer 1.3 cluster
by Raúl Alexis Betancor Santana 22 Jan '08

22 Jan '08

Hi all, I'm trying to build an Active/Active OpenSer 1.3 cluster with 2 nodes. All it's working ok for the REGISTER messages throught the use of t_replicate and add_path, but I've problems with the INVITE messages. Both servers share the same location table on a PostgreSQL Multimaster cluster (but that's irrelevant, the PostgreSQL cluster I mean...), using db_mode = 2 An that's the logic I used: - On REGISTER ->> - Auth Request - NAT ? - add_path - t_replicate(oser2) - On INVITE ->> - Auth Request - lookup - Load prefs from DB - relay The problem I'm having is that the lookup function changes the RURI of the message on node1 BEFORE taking into account the "Path:" element present inside the AoR, so when the INVITE gets forwarded to node2 (where the NATed UAC it's registered), node2 deny the relaying of the message because $rD of the RURI it's not local (remember that the RURI whas changed on node1). Is there any way to know on node1 that the INVITE request MUST obey the "Path:" so I could to something like: $old_ru = $ru lookup("location") if(path_exists()) { $ru = $old_ru } ... t_relay() ?? Any other aproach ? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual S.L.

3 3

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Franz, Have you tested 'validity' rule and 'sphere' rule under Openser + Openxcap? Regards, Kevin _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 22 Jan '08

22 Jan '08

Thank you so much, Franz! Indeed, the problem is coming from xml. It is well-formed but seems the server can't understand it. The whole xml document I used is the similar with yours except the root element with namespace which is showing below: <cr: ruleset xmlns="urn:ietf:params:xml:ns:common-policy" xmlns:pr="urn:ietf:params:xml:ns:pres-rules" xmlns:cr="urn:ietf:params:xml:ns:common-policy"> I got no error sending the xml doc with this as beginning. Only no Notify message after updating Publish, and sometimes the first Subscribe can't get through that is no Notify sending back from server. It's really weired! Anyway, I finally fix it with your advise, Thanks Indeed. Kevin, _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] How to handle INVITES with different To: header
by Peter P GMX 22 Jan '08

22 Jan '08

I have the following scenario: I receive a request with PSTN_number(a)my.ip.add.res from my VoIP In alias_db I have defined a rule that points to yyy(a)myip.com which is a registered account After alias_db_lookup OpenSER invites as follows: Invite yyy(a)myip.com To: xxxxxxx(a)my.ip.add.res The UA (actually a PBX with VoIP support) answers with 404 not found. When I invite as follows: Invite yyy(a)myip.com To: yyy(a)myip.com everything works fine. My question: How do you handle these cases? Rewriting the To: part seems dificult and will mess up a lot - hein? Kind regards Peter

2 1

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Okey:)) And last question about documentation, I searched documentation about function nat_uac_test from nathelper module for version 1.2.x and I'm find it in documentation for version 0.9. Openser 1.2.x not support function nat_uac_test or documentation 1.2.x not full? > Hi Sergey, > I'm afraid there is no docs about the errors - the idea is to have > sef-explaining error messages and to avoid documenting them ;) > Regards, > bogdan > Sergey Babichev wrote: > > Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > > > > > > > > > >> Hi Sergey, > >> > > > > > >> the "discarding fwd for a cancelled transaction" error is generated when > >> > > > > > >> you try to send a request for a transaction that was already cancelled > >> > > > > > >> from the client side. So probably the PSTN GW cancels the call while you > >> > > > > > >> still want to fork a new branch. > >> > > > > > >> regards, > >> > > > > > >> Bogdan > >> > > > > > >> Sergey Babichev wrote: > >> > > > > > >>> Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > >>> > > > > > > > > > > > > > > > > > >>> > >>> > > > > > >>>> Hi all! > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> What is problem may be? > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> -- > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Best regards, > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Sergey Babichev > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> _______________________________________________ > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users mailing list > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> Users(a)lists.openser.org > >>>> > > > > > >>>> > >>>> > > > > > > > > > >>> > >>> > > > > > >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>>> > > > > > >>>> > >>>> > > > > > > > > > > > > > > > > > >>> -- > >>> > > > > > > > > > >>> Best regards, > >>> > > > > > > > > > >>> Sergey Babichev > >>> > > > > > > > > > >>> _______________________________________________ > >>> > > > > > >>> Users mailing list > >>> > > > > > >>> Users(a)lists.openser.org > >>> > > > > > >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >>> > > > > > > > > > >>> > >>> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] Strating up with openser
by J Bhanot 22 Jan '08

22 Jan '08

Hi, I am just starting up with this stack.. Could you kindly let me know what strategy i should follow so as to use this stack for building applications rapidly. Thanking in anticipation. Regards, Jagmohan Bhanot Tata Consultancy Services Mailto: j.bhanot(a)tcs.com Website: http://www.tcs.com ____________________________________________ Experience certainty. IT Services Business Solutions Outsourcing ____________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

[OpenSER-Users] cdr records in openser - how to get them
by tloginbr-openserbr＠yahoo.com.br 22 Jan '08

22 Jan '08

Hi all, I'm pretty new to OpenSER, and I'm trying to get it working here. I have one OpenSER machine and a few asterisk machines behind this one. All my call go the OpenSER that does load balancing and fail over between my asterisk servers. Up to now I was getting the CDR from all my asterisk, but that is not going very well since I have to search in all asterisk machines to get the cdr for one extension. So I want to get the cdr directly from the openser thats where all calls have to pass. I found some people talking about CDRTool, but I think it does a little too much for me. I'm just looking for some tool that saves the information I need inside mysql so I can build my cdr report with some simple sql queries. Does anyone in here get the cdr out of the openser or is that not even possible? All help is very welcome here. Thanks in advance, Thiago Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

4 6

Re: [OpenSER-Users] Carrierroute help..
by Henning Westerholt 22 Jan '08

22 Jan '08

On Tuesday 22 January 2008, Ricardo Martinez wrote: > Hello. > I would like to make a question about the carrierroute module. > How does it work when one of the "level 0" gw's are down?. For example , > this is my carrierroute table : > > +----+---------+-------------+--------+------+-------+---------------+ > | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | > +----+---------+-------------+--------+------+-------+---------------+ > | 1 | 1 | 49 | 0 | 0.3 | 0 | de-1.carrier1 | > | 2 | 1 | 49 | 0 | 0.3 | 0 | de-2.carrier1 | > | 3 | 1 | 49 | 0 | 0.3 | 0 | de-3.carrier1 | > | 4 | 1 | | 1 | 1 | 0 | defa.carrier1 | > +----+---------+-------------+--------+------+-------+---------------+ > > > > 1- Let's suppose "de-1.carrier1" gateway is down. So.. any INVITE to > that gateway will not have any answer (for the failure_route failover), how > the module handle this?, It tries with the others+.? Hi Ricardo, no, at the moment there is no automatically failover done from the carrierroute module. So if you want to route your calls to another gw, e.g. 'de-2', you must manual re-route it to another gw/ domain in the failure_route block. E.g. route to domain 0 in route[..], route to domain 1 in failure_route. > 2.- Now, suppose that "de-1.carrier1" gateway is only full, so i can > probably have a reply from the gateway (maybe a "480" message"), so the > carrierroute module now tries with a failover route (defa.carrier1?) or > tries with any of the other two gateways still not full? I suppose you mean that 'de-1' is overloaded. No, the next domain is not automatically entered. This deficiency is known, and will be addressed in the future. But this code is not ready yet for a release. You could use the lcr module, there exist a 'next_gw' function, if you don't want to manually specify the failure_routes. Cheers, Henning

1 0

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Thanks, where i can read about this error (discarding fwd for a cancelled transaction) and others? > Hi Sergey, > the "discarding fwd for a cancelled transaction" error is generated when > you try to send a request for a transaction that was already cancelled > from the client side. So probably the PSTN GW cancels the call while you > still want to fork a new branch. > regards, > Bogdan > Sergey Babichev wrote: > > Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > > > > > > > > > >> Hi all! > >> > > > > > >> I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > >> > > > > > >> Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > >> > > > > > >> What is problem may be? > >> > > > > > >> -- > >> > > > > > >> Best regards, > >> > > > > > >> Sergey Babichev > >> > > > > > >> _______________________________________________ > >> > > > > > >> Users mailing list > >> > > > > > >> Users(a)lists.openser.org > >> > > > > > >> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >> > > > > > > > > -- > > > > Best regards, > > > > Sergey Babichev > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] SCTP enabled stack
by J Bhanot 22 Jan '08

22 Jan '08

Hi, I am trying to build the stack with SCTP enabled... I am doing following SCTP=1 make all but I am getting error - In file included from forward.h:49, from action.c:49: sctp_server.h:32:26: error: netinet/sctp.h: No such file or directory make: *** [action.o] Error 1 I am using FC8 - have inserted the SCTP module using /sbin/modprobe sctp without any error.. To overcome this error.... I chaged the absolute path of sctp.h in sctp_server.h to #include "/usr/src/kernels/2.6.23.1-42.fc8-i686/include/net/sctp/sctp.h"*/ instead of #include <netinet/sctp.h> but even then i am getting many erros of redeclration of many variables and files not being found in sctp.h Kindly suggest what should be done so that server can work with SCTP. Thanks, jb __________________________________________ =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

4 7

[OpenSER-Users] register two AOR
by Victor Gamov 22 Jan '08

22 Jan '08

Hi gentlemens! Is it possible to set modparam("registrar", "aor_avp", "$avp(s:sip-aor)") and call save("location") to save AOR from $avp(s:sip-aor) If not, how I can register two AOR when REGISTER message processed? -- CU, Victor Gamov

2 2

[OpenSER-Users] Launch of Soap-Simple-Proxy version 1.0.2
by Adrian Georgescu 22 Jan '08

22 Jan '08

Hello, This software exposes a simple to use SOAP interface for the OpenXCAP server available at http://openxcap.org It is a transparent proxy for SOAP to XCAP requests relying on XCAP server for authenticating requests. Applications ------------ You may use the provided functions to: 1. Publish presence information (using pidf-manipulation application) 2. Set the presence policy for a given presentity (pres-rules application) 3. Retrieve the OpenSER watchers for a given presentity (watchers application) The software can be used as: - Stand-alone data publisher for persistent Presence documents. - Policy manager. You can build a web interface to manage the pres-rules documetn and use it in combination with Eyebeam-lite as a complete SIP SIMPLE client. - To retreive and display the active watchers from OpenSER in a web interface. Usage ----- To use the software you must use a SOAP client. First you must generate the client library from the supplied WSDL file using the SOAP tools available in your programming language. The SOAP functions usage and syntax are self explanatory. Available functions are described in the wsdl/ssp.wsdl file. An example of PHP client library is provided with CDRTool version 6 or higher, the soap client functions are available in the file: /var/www/CDRTool/provisioning/ngnpro_soap_library.phtml in the class WebService_SoapSIMPLEProxy_PresencePort Support ------- The support is provided by AG Projects on a best effort basis. Report problems via the ticketing system from openxcap.org or openser users mailing lists. License ------- Soap-Simple-Proxy is licensed under GNU PUBLIC LICENSE version 2.0. The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[OpenSER-Users] New OpenXCAP release 0.9.9
by Adrian Georgescu 22 Jan '08

22 Jan '08

Hello, There is a new OpenXCAP release available. Changes in version 0.9.9 ------------------------ * Fixed node selector decoding * Save presence rules documents under the same file name ("index.xml") * Moved trusted_peers to [Authentication] config section * Default xcap_table to "xcap" in config.ini.sample * Improved URI logging * DELETE of a full document doesn't produce an Etag in the 200 response * Added SIP Thor backend Changes in version 0.9.8 ------------------------ * Support subdirectories by using doc_uri column in xcap table * Updated MySQL xcap table structure The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[Serusers] SER service: registration problem (load balancing quirk?)
by Tom van der Geer 22 Jan '08

22 Jan '08

Dear SER users, I'm experiencing occasional problems with my SIP clients registration to the (free) iptel.org SER service. I'm not sure if this is the place to ask questions regarding this service. If not, please give me a pointer who/what I should contact. I've made 2 tcpdumps. 1 from a failing registration request and from a successful registration request. The setup for both situations is the same. These captures were made within 40 minutes from each other. In both cases the first registration attempt is replied with a 401 Unauthorized (obviously) which request for authentication. In both cases my SIP client (SJPhone) replies correctly (I assume) with a new registration and the requested account credentials. In the case of a successful registration the SER replies with a 200 OK. In the case of a failing registration my SIP client receives another 401 Unauthorized message. What struck me is that this second 401 is sent from another SER system. (The "Server:" header from the first registration attempt is different from the second registration attempt.) The first header has "Server: Sip EXpress router (2.1.0-dev16-tcp (i386/linux))" while the second 401 has "Server: Sip EXpress router (2.0.0-rc5 (i386/linux))". (note the difference in SER versions) I've verified that in the case of a successful registration the Server header in the first (401) and second (200) response are the same: "Server: Sip EXpress router (2.1.0-dev16-tcp (i386/linux))" Is this some kind of weird load balancing quirk? From a security point of view I don't want to send my TCP Dumps (with my authentication) to the mailing list, but I can send it to a personal email address when required. Thanks, Tom

2 1

Re: [OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Problem is appear when i calling from PSTN. It's work fine when i calling from sip account to huntgroup. I think this is problem whith timer in my PSTN gateway. > Hi all! > I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed > Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 > What is problem may be? > -- > Best regards, > Sergey Babichev > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users -- Best regards, Sergey Babichev

2 1

[OpenSER-Users] Serial fork terminated after 5 call
by Sergey Babichev 22 Jan '08

22 Jan '08

Hi all! I want enable hunting for my extensions. It's work fine, but if my hunt group consists of more then 5 extensions, call terminated and I see in syslog error: Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: ERROR:tm:w_t_relay: t_forward_nonack failed Jan 21 20:54:16 ser1-2 /usr/local/sbin/openser[23728]: DEBUG:tm:relay_reply: branch=5, save=0, relay=5 What is problem may be? -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] Why RPID related functions instead of PAI?
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, "Remote-Party-ID" is a draft ("draft-ietf-sip-privacy-04") [1] expired in 2002 while "P-Asserted-Identity" is a official RFC (3325) [2]. I'd like to know why OpenSer manages RPID in some functions ("auth" module) instead of PAI. I assume that some gateways just implement RPID and so, but can't understand why PAI is ignored (the use of PAI is extended in gateways). As a curiosity, both draft and RFC have really **common** parts, as chapter "Introduction" (4 in draft, 3 in RFC): "3/4 Introduction Various providers offering a telephony service over IP networks have selected SIP as a call establishment protocol. Their environments require a way for trusted network elements operated by the service providers (for example SIP proxy servers) to communicate the identity of the subscribers to such a service, yet also need to withhold this information from entities that are not trusted when necessary. Such networks typically assume some level of transitive trust amongst providers and the devices they operate. These networks need to support certain traditional telephony services and meet basic regulatory and public safety requirements. These include Calling Identity Delivery services, Calling Identity Delivery Blocking, and the ability to trace the originator of a call ... " So my question is again:why OpenSer supports RPID related functions and not PAI functions? Thanks for any explanation. [1] RPID: http://tools.ietf.org/html/draft-ietf-sip-privacy-04 [2] PAI: http://www.faqs.org/rfcs/rfc3325.html -- Iñaki Baz Castillo

3 3

[OpenSER-Users] Openser + Mediaproxy not working with 2 domains
by Andreti 21 Jan '08

21 Jan '08

Hi everybody, I'm working with Openser + Mediaproxy 1.9.0 and it seems that everything is working when the calls are establised between users attached to the same proxy server, even with different kind of NATs. However It doen't work in 2 different scenarios, and the result is exactly the same , the video and audio is only sent in one way. Scenario 1 ======== User A attached to the SIP proxy xxx.xxx.xxx.13 (Public IP) calls to a GW xxx.xxx.xxx.11 (Public IP) with several users internally associated. In this case the user A can see the video and audio sent by the GW, but the GW doesn't receive any RTSP stream. It seems that the mediaproxy doesn't do anything, why? maybe because the GW blongs to other domain (xxx.xxx.xxx.11) ? What can I do? If the GW calls to user A, it works fine (I can see the session in the mediaproxy with sessions.py) Scenario 2 ======== In this case, I have another GW with Public IP address xxx.xxx.xxx.14, but it doesn't include in the INVITE message the SDP body. The GW calls to the same user attached to the SIP proxy xxx.xxx.xxx.13 , and the behaviour is exactly the same as scenario 1, the calling site can sse the video and audio but the called can't. Unlike the previous scenario, the signalling is: INVITE without SDP --> 200 OK (SDP) -- > ACK (SDP) In theory, Mediaproxy 1.9.0 should support this procedure since it's a SIP standard mechanism, however the called party doesn't receive RTP stream. In my opinion, the problem could be related to scenario 1, I mean , the calling party is not attached to the SIP proxy (belongs to other domain) and when the 200 OK (SDP) message arrives to the SIP proxy, the mediaproxy doesn't do anything Sorry for the complex explanation. I've waste a lot of time trying to solve this solution and honestly I don't know what to do. Please, could somebody help?? I attach my openser.conf. I hope it helps. Andreti debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes listen=xxx.xxx.10.12 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- mpath = "/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "uri_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.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) modparam("usrloc", "timer_interval", 20) # -- 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", "password_column", "password") # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) #cambio 16_04_07 modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") #puesto para el CDRTool modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) #modparam("acc", "radius_config", "/etc/openser/radius/client.conf") modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp;\ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp($can_uri); \ Billing-Party=$avp($billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event)") # SIP-Proxy-IP=$avp(s:sip_proxy_ip)") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("avpops","avp_aliases","can_uri=i:34") modparam("avpops","avp_aliases","billing_party=i:1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- 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; }; # setflag(ACCOUNTING_FLAG); # avp_write("SER_IP","$avp(s:sip-proxy)"); # avp_write("$ru", "$avp(can_uri)"); # 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"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); setflag(1); } route(1); }; if (src_ip==193.36.177.227) { fix_nated_sdp("2"); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (method=="MESSAGE") { setflag(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); 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==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.com", "subscriber")) { # www_challenge("sip.com", "0"); # exit; # }; if (!radius_www_authorize("")) { www_challenge("","1"); exit; } if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; 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")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; }; if (method=="INVITE") { t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri=="sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; t_relay(); return; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri=="sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; 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] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; } -- View this message in context: http://www.nabble.com/Openser-%2B-Mediaproxy-not-working-with-2-domains-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[OpenSER-Users] Calls disconnect automatically
by VoIP Forums www.Go4Calls.com 21 Jan '08

21 Jan '08

Hi Friends, I start getting one problem, the calls disconnect automatically in 30 and 32 sec. I am using openser + rtpproxy before with the same openser.cfg it was running smoothly and once traffic increased this problem appeared. Could you please help me to solve this issue because i put openser in production and now no one can make long call. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

Re: [OpenSER-Users] xlite with openser / hello world posts SOLVED
by A.smith 21 Jan '08

21 Jan '08

Hi Moujane, In case you haven´t worked it out yet, I managed to get an openser.cfg to work for simple testing between XLite or other SIP client. Its a shame there isnt a working config easily findable on the openser site or other internet resource for those people new to SIP and OpenSER.... :S Try this, for me it works on OpenSER 1.2.2 # SECTION 1 - GLOBAL DEFINTIONS debug=3 fork=yes log_stderror=no #fork=no #log_stderror=yes listen=10.10.10.10 #put your server IP here port=5060 disable_dns_blacklist=yes dns=no rev_dns=no children=4 # SECTION 2 - MODULE DEFINTION mpath="/usr/local/lib/openser/modules" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "registrar.so" # SECTION 3 - MODULE CONFIGURATION modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) # main routing logic FOR RADIUS ACC ACCOUNTING route{ if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; if (method=="REGISTER") { record_route(); if (!save("location")) { sl_reply_error(); }; exit; }; if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } # SECTION 5 - SECONDARY ROUTE BLOCKS route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } ________________________________________________ Message sent using UK Grid Webmail 2.7.9

3 3

[OpenSER-Users] Adding "Reason: SIP ; cause=200" header when CANCEL in forking scenarios
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, according to RFC 3326 [1] (Reason Header): 3.1 Call Completed Elsewhere A proxy forks an INVITE request and one of the branches returns a 200 (OK). The forking proxy includes this status code in a Reason header field in the CANCEL request that it sends to the rest of the branches. Reason: SIP ;cause=200 ;text="Call completed elsewhere" With this behaviour, if an AoR is registered in two phones and one of them answers a call, that call will not appear as MISSED in the other phone (if the phone supports "Reason" headers). Is it possible OpenSer to add this header when cancelling the rest of ringing branches? If not, do you think it could be a feature request? [1] http://www.faqs.org/rfcs/rfc3326.html -- Iñaki Baz Castillo

3 2

[OpenSER-Users] Routing logic Check
by SP GLE 21 Jan '08

21 Jan '08

Hi, Is there any tool to validate an OpenSer Routing Configuration Logic ? We would like to validate a configuration to avoid regressions on routing behaviour. Regards. _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail http://mail.yahoo.fr

2 1

Re: [OpenSER-Users] Calls disconnect automatically
by VoIP Forums www.Go4Calls.com 21 Jan '08

21 Jan '08

Hi sorry, I forget to give my openser.cfg, there is one more point if i am using STUN server in our linksys device the call goes normal for long time till user finish the call. It seems something wrong in NAT configuration. # # sample config file to be used with nathelper/rtpproxy # # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=212.XXX.XXX.XXX:5064 #port=5064 children=4 disable_dns_blacklist = yes # --- module loading mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "nathelper.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("usrloc|auth_db","db_url","mysql://proxy:MoHaKa21@192.168.1.50/emafone") # -- usrloc params -- modparam("usrloc", "nat_bflag", 6) # -- registrar params -- modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # -- auth params -- # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- nathelper modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("nathelper", "rtpproxy_disable", 0) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") #modparam("nathelper", "rtpproxy_sock", "udp:212.100.235.229:22222") #modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") #modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "sipping_bflag", 7) #modparam("nathelper", "sipping_from", "sip:pinger@openser.org") # --- main routing logic route{ 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; }; # NAT detection route(2); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { #append_hf("P-hint: outbound\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("212.XXX.XXX.XXX", "subscriber")) { www_challenge("212.XXX.XXX.XXX", "0"); exit; }; if (isflagset(5)) { setbflag(6); # if you want OPTIONS natpings uncomment next # setbflag(7); }; save("location"); exit; }; #########PSTTN CALL ############################################# if (uri=~"sip:00[1-9][0-9]+@.*") { strip(2); rewritehostport("212.XXX.XXX.XXX:5060"); route(1); exit; }; ################################################################### if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (subst_uri('/(sip:.*);nat=yes/\1/')){ setbflag(6); }; if (isflagset(5)||isbflagset(6)) { route(3); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(6); }; } route[3] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); t_on_failure("1"); }; if (isflagset(5)) search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); t_on_reply("1"); } failure_route[1] { if (isbflagset(6) || isflagset(5)) { unforce_rtp_proxy(); } } onreply_route[1] { if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); if (isbflagset(6)) { fix_nated_contact(); } exit; } Regards, www.Go4Calls.Com VoIP Forums From: go4calls(a)hotmail.com To: users(a)lists.openser.org; users(a)openser.org Subject: Calls disconnect automatically Date: Sun, 20 Jan 2008 18:01:48 +0800 Hi Friends, I start getting one problem, the calls disconnect automatically in 30 and 32 sec. I am using openser + rtpproxy before with the same openser.cfg it was running smoothly and once traffic increased this problem appeared. Could you please help me to solve this issue because i put openser in production and now no one can make long call. Regards, www.Go4Calls.Com VoIP Forums Express yourself instantly with MSN Messenger! MSN Messenger _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

5 9

[OpenSER-Users] hor to register ?
by Aaash shah 21 Jan '08

21 Jan '08

hi i am making project on SIP handoffs i already install openSER on fedora i dont know how to register it so please give me reply thanks -- "Never Ignore Anyone Who Cares 4 u, Becoz one day u will Realize that u have lots a diamond while u were buzy in collecting stones...."

1 0

[Serusers] lookup_contacts for natted clients returns local ip
by Tomasz Zieleniewski 21 Jan '08

21 Jan '08

Hi, I have the followning natted contact in my location table: mysql> select * from location where uid ='tw(a)sip.touk.pl'; +--------------------+-------+-----------------------------------------------------------+ | contact | flags | received | +--------------------+-------+-----------------------------------------------------------+ | sip:tw@192.168.7.2 | 1 | sip:80.53.110.142:61066;dstip=192.168.129.74;dstport=5060 | +--------------------+-------+-----------------------------------------------------------+ when I invoke lookup_contacts the ruri has local address 192.168.7.2instead of public one. Is this correct behaviour? Please point me is this should work as I think. Thank You for any help Tomasz

2 3

[OpenSER-Users] Obtaining the domain of "Refer-to" URI
by Iñaki Baz Castillo 21 Jan '08

21 Jan '08

Hi, in order to match the domain of "Refer-to" header I must do a dirty string substitution: $avp(s:rt) = $rt; avp_subst("$avp(s:rt)", "/(.*(a)|[;?].*)//gi"); If for example: Refer-To: <sip:bob@biloxi.example.net?Accept-Contact=sip:bobsdesk. biloxi.example.net&Call-ID%3D55432%40alicepc.atlanta.example.com> Then $avp(s:rt) is "biloxi.example.net". Couldn't be possible to have generic string transformations to get part of a URI in any header/variable? Is there any other way I don't know to get it easily? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

6 14

[OpenSER-Users] [OT] Annoying empty lines when using "M4 Macro Processor"
by Iñaki Baz Castillo 20 Jan '08

20 Jan '08

Hi, as suggested here: http://www.openser.org/docs/scripting I'm using M4 to declare "define" in my OpenSer script. I've an annoying issue since M4 adds some empty lines at the top of the script. If my script has any error, the logs will show me "ERROR in line 235 in openser.cfg" but that line correspondes to line 187 in my openser.cfg.m4 file, so makes it very difficult to debug. I've read the entire doc and have no idea of how to fix it. For example, consider this simple case: ---- local.m4 ---- define(`IP',`127.0.0.1') define(`PORT' ,`5061') ----------------------- ---- file.txt.m4 ---- My ip is IP and my port is PORT. -------------------------- ~# m4 local.m4 file.txt.m4 > file.txt ---- file.txt ---- <-- Empty line <-- Empty line <-- Empty line <-- Empty line My ip is 127.0.0.1 and my port is 5061. --------------------- Does someone know how to solve it? Thanks a lot. -- Iñaki Baz Castillo

2 5

[OpenSER-Users] mysql error shuts down openser 1.3
by Christian Schlatter 20 Jan '08

20 Jan '08

Hi all, Is it normal that a simple mysql error causes the worker process to die which in turn shuts down all other openser processes? If I remember correctly, this never happened to me with the 1.2 version. The following log comes from an 1.3 installation: [30812]: ERROR:mysql:db_mysql_submit_query: driver error: Can't lock file (errno: 4009) [30812]: ERROR:mysql:db_mysql_raw_query: error while submitting query [30812]: ERROR:usrloc:get_all_db_ucontacts: raw_query failed [30792]: INFO:core:handle_sigs: child process 30812 exited by a signal 11 [30792]: INFO:core:handle_sigs: core was not generated [30792]: INFO:core:handle_sigs: terminating due to SIGCHLD [30797]: INFO:core:sig_usr: signal 15 received [30814]: INFO:core:sig_usr: signal 15 received [30798]: INFO:core:sig_usr: signal 15 received [30800]: INFO:core:sig_usr: signal 15 received [30802]: INFO:core:sig_usr: signal 15 received [30804]: INFO:core:sig_usr: signal 15 received [30806]: INFO:core:sig_usr: signal 15 received [30808]: INFO:core:sig_usr: signal 15 received Jan 14 12:54:45 proxy-2 ser[30810]: INFO:core:sig_usr: signal 15 received /Christian

3 5

[OpenSER-Users] [OT] Nortel gateway allows "tel:1234@host" but not "tel:1234" ¿?¿?¿
by Iñaki Baz Castillo 20 Jan '08

20 Jan '08

Hi, I'm trying "P-Asserted-Identity" header against a Nortel CS2000_NGSS/8.0. It works (the number in PAI header is shown as PSTN callerid number) in those cases: P-Asserted-Identity: <sip:+34666555444@host/IP> P-Asserted-Identity: <tel:+34666555444@host/IP> As you can see, the second case is wrong since a "tel" uri has no host part according to RFC 3966 [1]. And the worst thing is that Nortel disallows or doesn't understand a correctly formed "tel" uri in PAI header as this: P-Asserted-Identity: <tel:+34666555444> My experience with gateways is very limited and new, but I should assume that it is a sad bug in the Nortel SIP stack, is it? Thanks for any help. [1] RFC 3966: http://www.faqs.org/rfcs/rfc3966.html -- Iñaki Baz Castillo

1 1

[Serusers] opensource prepaid system
by Nhadie 19 Jan '08

19 Jan '08

Hi, Has anyone tested an opensource prepaid system to work on a ser/freeradius/asterisk-b2bua/asterisk setup? TIA Regards, Nhadie

1 0

[OpenSER-Users] unixodbc (ARRGH!)
by Douglas Garstang 19 Jan '08

19 Jan '08

Can someone tell me what the correct syntax is for the avp_url command? I've been at this for hours and it's driving me CRAZY! this: modparam("avpops","avp_url","unixodbc://user:password@127.0.0.1/database") gives: unixodbc:extract_error: unixodbc:SQLDriverConnect=IM002:1:0:[unixODBC][Driver Manager]Data source name not found, and no default driver specified this: modparam("avpops","avp_url","unixodbc://dsn" ) gives: core:new_db_id: error while parsing database URL: unixodbc://dsn and I've tried about a HUNDRED other combinations. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

[OpenSER-Users] Problems with NAPTR records and failover
by Robert McGilvray 19 Jan '08

19 Jan '08

Greetings list, I'm trying to setup uri rewriting and failover using NAPTR records. I've gone wrong somewhere since instead of trying the first uri and only failing over if there's a transactional and/or communications problem it's creating two branches and doing parallel forking on the records. Example: >From my test SIP phone I dial 3759. Openser receives this INVITE and does a private ENUM lookup on it which returns two NAPTR records: 9.5.7.3 NAPTR 10 10 "u" "E2U+sip" "!^.*$!sip:3759@mygateway1.whoever.com!" . NAPTR 20 10 "u" "E2U+sip" "!^.*$!sip:7100@myfailovergateway1.whoever.com!" . >From what I understand this should first try to contact my gateway (a cisco box) using a uri of sip:3759@mygateway1.whoever.com. If this fails it should then take the second higher order and rewrite the URI to sip:7100@myfailovergateway1.whoever.com. Instead what I end up with is two parallel calls to gateway and failover, the second NAPTR record is a conference bridge so the cisco immediately sends back a 200 OK and openser cancels the first branch. First gateway barely gets off a ring to my phone. I haven't adjusted any of the SIP timers so the defaults should allow plenty of time for call setup etc. Debug output from the call is in http://www.idled.net/~bobm/openser-debug.txt . Am I handling the INVITE incorrectly in my config? Any assistance would be appreciated. route[1] { if (!t_relay()) { xlog("L_NOTICE", "sipprx-wpf1.globeop.com: $mi route[$rm][0] $fu -> $ru -> Error with t_relay\n"); sl_reply_error(); } exit; } INVITE routing block route[6] { sl_send_reply("100", "Trying"); if (isflagset(1)) { xlog("L_NOTICE", "sipprx-wpf1.globeop.com: $mi route[$rm][0] $fu -> $ru Processing INVITE\n"); } if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); if ((uri =~ "sip:[0-9]{4}@.*") || (uri =~ "sip:[0-9]{10}@.*")) { prefix ("+"); if (enum_query("e164.globeop.com")) { xlog("L_INFO", "sipprx-wpf1.globeop.com: Number found in ENUM, new uri is $ru\n"); } else { xlog("L_INFO", "sipprx-wpf1.globeop.com: Number not found in ENUM\n"); revert_uri(); } } if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); route(1); } else if (lookup("location")) { append_hf("P-hint: usrloc applied\r\n"); route(1); } else { sl_send_reply("404","Not found"); exit; } exit; } -- Bob This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediatley delete this email in its entirety and any attachments thereto.

1 1

[OpenSER-Users] Sippy b2bua
by TCB 18 Jan '08

18 Jan '08

Anyone managed to setup sippy b2bua . Pointers most welcome. I have tried setting it up and this is as far as i have gone. I have done the necessary radius config in my /etc/radiusclient-ng/{ radiusclient.conf,servers}. When i send a call from my proxy i get 403 (Auth Failed) Find attached the log file . regards -- TC

1 0

[OpenSER-Users] Sippy b2bua
by TCB 18 Jan '08

18 Jan '08

Anyone suceeded in implementing sippy b2bua. Pointers most welcome. -- TC

1 0

[OpenSER-Users] ERROR: presence_xml:get_final_notify_body?
by KevinKinnan 18 Jan '08

18 Jan '08

Did I miss something on configuration? or it simply does not send NOTIFY message when changes occur in the published data after integrating OpenXCAP? Thx in advance! Kevin _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

1 0

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 18 Jan '08

18 Jan '08

Did I miss something on configuration? or it simply does not send NOTIFY message when changes occur in the published data after integrating OpenXCAP? Thx in advance! Kevin _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] OpenSER: Table "subscriber", only one new user can be added?
by KevinKinnan 18 Jan '08

18 Jan '08

Only one new entry can be added in the "subscriber" table. Or, "Duplicate entry '' for key 3" error responses. Is there any way to create more than one user in this table? Kevin, _________________________________________________________________ MSN圣诞礼物火热登场，免费发放中，快来领取吧！ http://im.live.cn/emoticons/?ID=18

6 6

[OpenSER-Users] OpenXCAP application support
by Richard Good 18 Jan '08

18 Jan '08

Hi, I am using the OpenXCAP implementation in co-ordination with OpenSER. Is there any way to support any (or at least a few more) applications than those currently supported by OpenXCAP? I previously used XCAP-Lite and stored various XML documents ranging from address books and TV guides to privacy and domain policies. I've found that with OpenXCAP if you use any application not supported ( anything besides resource-lists, pidf-manipulation, pres-rules, etc.) the Document Selector is not recognised and the documents can not be uploaded or downloaded. Is there a way to bypass this? Apologies if this is not the correct mailing list - the OpenXCAP page points to this list. Thanks and Regards, Richard

1 0

[OpenSER-Users] Help with this mediaproxy + Openser nat problem... thankssss
by Mario Bedialauneta 17 Jan '08

17 Jan '08

Hi everybody, I'm working with Openser + Mediaproxy 1.9.0 and it seems that everything is working when the calls are establised between users attached to the same proxy server, even with different kind of NATs. However It doen't work in 2 different scenarios, and the result is exactly the same , the video and audio is only sent in one way. Scenario 1 ======== User A attached to the SIP proxy xxx.xxx.xxx.13 (Public IP) calls to a GW xxx.xxx.xxx.11 (Public IP) with several users internally associated. In this case the user A can see the video and audio sent by the GW, but the GW doesn't receive any RTSP stream. It seems that the mediaproxy doesn't do anything, why? maybe because the GW blongs to other domain (xxx.xxx.xxx.11) ? What can I do? If the GW calls to user A, it works fine (I can see the session in the mediaproxy with sessions.py) Scenario 2 ======== In this case, I have another GW with Public IP address xxx.xxx.xxx.14, but it doesn't include in the INVITE message the SDP body. The GW calls to the same user attached to the SIP proxy xxx.xxx.xxx.13 , and the behaviour is exactly the same as scenario 1, the calling site can sse the video and audio but the called can't. Unlike the previous scenario, the signalling is: INVITE without SDP --> 200 OK (SDP) -- > ACK (SDP) In theory, Mediaproxy 1.9.0 should support this procedure since it's a SIP standard mechanism, however the called party doesn't receive RTP stream. In my opinion, the problem could be related to scenario 1, I mean , the calling party is not attached to the SIP proxy (belongs to other domain) and when the 200 OK (SDP) message arrives to the SIP proxy, the mediaproxy doesn't do anything Sorry for the complex explanation. I've waste a lot of time trying to solve this solution and honestly I don't know what to do. Please, could somebody help?? I attach my openser.conf. I hope it helps. Andreti debug=5 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL0 # Uncomment these lines to enter debugging mode #fork=no log_stderror=yes listen=xxx.xxx.10.12 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- mpath = "/usr/local/lib/openser/modules/" # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "domain.so" loadmodule "mediaproxy.so" loadmodule "uri_db.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "avpops.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "auth_radius.so" loadmodule "group_radius.so" loadmodule "avp_radius.so" loadmodule "nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.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) modparam("usrloc", "timer_interval", 20) # -- 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", "password_column", "password") # -- acc params -- modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 1) #cambio 16_04_07 modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 1) modparam("acc", "service_type", 15) modparam("acc|auth_radius|group_radius|avp_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") #modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp") #puesto para el CDRTool modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_cancels", 0) modparam("acc", "report_ack", 0) modparam("acc", "early_media", 0) modparam("acc", "log_level", 1) #modparam("acc", "radius_config", "/etc/openser/radius/client.conf") modparam("acc", "radius_extra", "Sip-Src-IP=$si;Sip-Src-Port=$sp;\ Sip-RPid=$avp(s:rpid); \ Source-IP=$si; \ Source-Port=$sp; \ Canonical-URI=$avp($can_uri); \ Billing-Party=$avp($billing_party); \ Divert-Reason=$avp(s:divert_reason); \ X-RTP-Stat=$avp(s:rtp_statistics); \ From-Header=$hdr(from); \ User-Agent=$hdr(user-agent); \ Contact=$hdr(contact); \ Event=$hdr(event)") # SIP-Proxy-IP=$avp(s:sip_proxy_ip)") # -- group_radius params -- modparam("group_radius", "use_domain", 1) # -- avpops params -- #modparam("avpops", "avp_aliases", "day=i:101;time=i:102") modparam("avpops","avp_aliases","can_uri=i:34") modparam("avpops","avp_aliases","billing_party=i:1") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("mediaproxy", "natping_interval", 60) modparam("registrar", "nat_flag", 2) # ------------------------- 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; }; # setflag(ACCOUNTING_FLAG); # avp_write("SER_IP","$avp(s:sip-proxy)"); # avp_write("$ru", "$avp(can_uri)"); # 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"); if(is_method("BYE")) { # log it all the time acc_rad_request("200 ok"); acc_log_request("200 ok"); setflag(1); } route(1); }; if (src_ip==193.36.177.227) { fix_nated_sdp("2"); }; if(is_method("INVITE") && !has_totag()) { # set the acc flags setflag(1); setflag(2); }; if (method=="MESSAGE") { setflag(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); 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==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("sip.com", "subscriber")) { # www_challenge("sip.com", "0"); # exit; # }; if (!radius_www_authorize("")) { www_challenge("","1"); exit; } if (client_nat_test("3")) { setflag(2); force_rport(); fix_contact(); }; 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")) { # log to acc as missed call acc_rad_request("404 Not Found"); acc_log_request("404 Not Found"); sl_send_reply("404", "Not Found"); exit; }; }; if (method=="INVITE") { t_on_failure("1"); } else if (method == "BYE" || method == "CANCEL") { end_media_session(); }; if (loose_route()) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri==" sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; t_relay(); return; }; if (client_nat_test("3") && !search("^Record-Route:")) { # Mark as NAT'ed force_rport(); fix_contact(); }; if (method=="INVITE") { t_on_reply("1"); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; #if ((method=="INVITE" || method=="ACK") && !to_uri==" sip:frog1@xxx.xxx.10.12") { # use_media_proxy(); #}; if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; 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] { end_media_session(); } onreply_route[1] { if (status=~"(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); }; use_media_proxy(); }; }

1 0

[OpenSER-Users] ERROR: presence_xml:get_final_notify_body?
by KevinKinnan 17 Jan '08

17 Jan '08

The debug log in 9 level is attached. K, _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

Re: [OpenSER-Users] openSER Call Route Logic
by Patrick Baker 17 Jan '08

17 Jan '08

Sipwise.com might help ----- Original Message ----- From: users-bounces(a)lists.openser.org <users-bounces(a)lists.openser.org> To: users(a)lists.openser.org <users(a)lists.openser.org> Sent: Wed Jan 16 23:48:27 2008 Subject: [OpenSER-Users] openSER Call Route Logic Hi, I recently installed openser-1.3.0 and able to connect my cisco sip phones and everything is working. Now i need to integrate this with asterisk box for call forwarding to other locations & voicemails. i read the getting started guide and other information, but seems i need more help from the group. can any one give me a helping hand. i am sure i will do my home work before bug you again. I need a starting point to over come the complexity of the routing plan and logic. thanks -asterisk user ________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. <http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/c…>

3 3

Re: [OpenSER-Users] MySQL Alternative (help!)
by Douglas Garstang 17 Jan '08

17 Jan '08

Henning, Let me try asking my question another way. Is it possible to use avp_db_query() with UnixODBC? If so, how? Douglas. ----- Original Message ---- From: Henning Westerholt <henning.westerholt(a)1und1.de> To: users(a)lists.openser.org Cc: Douglas Garstang <dougmig33(a)yahoo.com> Sent: Thursday, January 17, 2008 1:15:09 AM Subject: Re: [OpenSER-Users] MySQL Alternative On Thursday 17 January 2008, Douglas Garstang wrote: > What other options do I have for generic database connectivity besides > MySQL/avp_db_query() in OpenSER? > > Can I use SQLite? Can I use unixodbc? (and I'm not talking about the > limited built in usrloc unixodbc stuff) Hi Douglas, don't know what you mean with the builtin ODBC stuff in usrloc.. I assume that you talking about connectivity options for OpenSER. Available database modules that can be used with most modules: MySQL PostgreSQL UnixODBC (should work with Oracle, MS-SQL, SQLite and other) dbtext db_berkeley (Berkeley DB) Interfaces roughly sorted in terms of maturity, stability.. Most people uses MySQL, so this is the best tested option. Cheers, Henning ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] Duplicated entries in "location" table when using "db_mode 3"
by Iñaki Baz Castillo 17 Jan '08

17 Jan '08

Hi, I use ("usrloc", "db_mode", 3) since some months and I've realized that many times I get duplicated entries in location table. Those entries have **exactly** the same values in all the columns. Note that is a testing serer which I restart 10 or 20 times each hour XD and that I've upgraded from 1.2 to 1.3 getting always this issue. If I send a new REGISTER (refresh) for an AoR with duplicated entries both entries are updated according to the new "expires" and CSEQ. Is it normal? -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] best response seems wrong.
by Aymeric Moizard 17 Jan '08

17 Jan '08

(sorry: just repost to follow correct thread/subjet) I just found another example from rfc3261 where an endpoint will use 5xx class answer: 12.2.2 UAS Behavior Page 76: If the remote sequence number is empty, it MUST be set to the value of the sequence number in the CSeq header field value in the request. If the remote sequence number was not empty, but the sequence number of the request is lower than the remote sequence number, the request is out of order and MUST be rejected with a 500 (Server Internal Error) response. Hope I convinced you! tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ On Wed, 16 Jan 2008, Iñaki Baz Castillo wrote: > On Wednesday 16 January 2008 13:44:10 A.smith wrote: > >> If I try and use the whole example cfg (not just the route) I am recieving >> an error like: >> >> 0(23448) find_cmd_export_t: <lookup> not found >> 0(23448) parse error (105,23-24): unknown command, missing loadmodule? > > "lookup()" is defined in "registrar" module that you are not loading: > http://www.openser.org/docs/modules/1.2.x/registrar.html#AEN342 > > -- > Iñaki Baz Castillo > ibc(a)in.ilimit.es > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

5 20

[OpenSER-Users] xlite with openser / hello world posts
by A.smith 17 Jan '08

17 Jan '08

1 0

[OpenSER-Users] Presence + XCAP issue
by Leonard Broman 17 Jan '08

17 Jan '08

Hi! I have an issue with the presence/xml/xcap modules. I'm trying to run the presence server without an "integrated" xcap server, since I have not managed to get openXCAP running on CentOS. However, for the moment I'd prefer not to be depending on it. I suppose this might be the root of my problem, as follows. Publish is working fine. But Subscribe is always answered with a 481 (which shouldn't occur normally) and I get this log: Jan 17 10:57:16 m2lin3 ./openser[7097]: ERROR:presence:get_stored_info: record not found in hash_table Jan 17 10:57:16 m2lin3 ./openser[7097]: ERROR:presence:handle_subscribe: getting stored info My primary question is wheather the this error is caused by the absence of documents in the xcap table? And if so can I do something to avoid this dependence? /Leo

1 0

[OpenSER-Users] openser 1.4 blocks
by Helmut Kuper 17 Jan '08

17 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, when I send the following SIP-Message on my openser, openser blocks and eats nearly all CPU power it can get: BYE sip:80.228.245.138:5070 SIP/2.0 Via: SIP/2.0/UDP 212.6.95.88:1028;branch=z9hG4bK-7rhl6qnns4ej;rport Route: <sip:212.6.123.118;lr=on;ftag=7w4sllx8cc> Route: <sip:212.6.123.118:5090;lr;ftag=7w4sllx8cc;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--> Route: <sip:Gk-int@80.228.245.176:5050;lr> From: "HK SoftSwitch" <sip:v101@sip0.en.ewetel.de>;tag=7w4sllx8cc To: <sip:12345@sip0.en.ewetel.de;user=phone>;tag=0082-00000054-0103i0 Call-ID: 3c26c9bec396-loo61fn1yu42 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:v101@212.6.95.88:1028;line=nopsgn9d>;flow-id=1 User-Agent: snom370/7.1.30 RTP-RxStat: Total_Rx_Pkts=120,Rx_Pkts=120,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=2263,Tx_Pkts=2263,Remote_Tx_Pkts=160 Content-Length: 0 What openser does is authentication, avp db loading an then a simple t_relay(). opensers debuglog (level 8) is this: Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:check_via_address: params 212.6.95.88, 212.6.95.88, 0 Jan 16 21:30:45 [8406] DBG:sl:run_sl_callbacks: callback id 0 entered Jan 16 21:30:45 [8406] DBG:siptrace:trace_sl_onreply_out: trace off... Jan 16 21:30:45 [8406] DBG:core:destroy_avp_list: destroying list 0xb5b00e50 Jan 16 21:30:45 [8406] DBG:core:receive_msg: cleaning up Jan 16 21:30:45 [8406] DBG:core:parse_msg: SIP Request: Jan 16 21:30:45 [8406] DBG:core:parse_msg: method: <BYE> Jan 16 21:30:45 [8406] DBG:core:parse_msg: uri: <sip:80.228.245.138:5080> Jan 16 21:30:45 [8406] DBG:core:parse_msg: version: <SIP/2.0> Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=2 Jan 16 21:30:45 [8406] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-goxekiv4ggfm>; state=6 Jan 16 21:30:45 [8406] DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Jan 16 21:30:45 [8406] DBG:core:parse_via: end of header reached, state=5 Jan 16 21:30:45 [8406] DBG:core:parse_headers: via found, flags=2 Jan 16 21:30:45 [8406] DBG:core:parse_headers: this is the first via Jan 16 21:30:45 [8406] DBG:core:receive_msg: After parse_msg... Jan 16 21:30:45 [8406] DBG:core:receive_msg: preparing to run routing scripts... BYE--------------------------------------------------------------------- [Wed Jan 16 21:30:45 2008] - NEW TRANSACTION! Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Jan 16 21:30:45 [8406] DBG:core:parse_to_param: tag=0082-0000004f-00e7i0 Jan 16 21:30:45 [8406] DBG:core:parse_to: end of header reached, state=29 Jan 16 21:30:45 [8406] DBG:core:parse_to: display={}, ruri={sip:044180002850@sip0.en.ewetel.de;user=phone} Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: <To> [74]; uri=[sip:044180002850@sip0.en.ewetel.de;user=phone] Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: to body [<sip:044180002850@sip0.en.ewetel.de;user=phone>] Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: cseq <CSeq>: <4> <BYE> Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: start searching: hash=36300, isACK=0 Jan 16 21:30:45 [8406] DBG:tm:matching_3261: RFC3261 transaction matching failed Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: no transaction found Jan 16 21:30:45 [8406] DBG:core:parse_to_param: tag=bpl4kg8f57 Jan 16 21:30:45 [8406] DBG:core:parse_to: end of header reached, state=29 Jan 16 21:30:45 [8406] DBG:core:parse_to: display={"HK SoftSwitch"}, ruri={sip:v101@sip0.en.ewetel.de} MAIN : New Message <BYE> From: sip:v101@sip0.en.ewetel.de, To: sip:80.228.245.138:5080 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=100 Jan 16 21:30:45 [8406] DBG:maxfwd:is_maxfwd_present: value = 70 Route 22: Check type of source ... Jan 16 21:30:45 [8406] DBG:permissions:allow_source_address: looking for <1, 585f06d4, 1028> Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=800000000 Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: content_length=0 Jan 16 21:30:45 [8406] DBG:core:get_hdr_field: found end of header Jan 16 21:30:45 [8406] DBG:core:pv_get_ppi_attr: no P-Preferred-Identity header Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 17==13 && [sip0.en.ewetel.de] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 17==13 && [sip0.en.ewetel.de] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:avpops:ops_dbquery_avps: query [SELECT uri_user FROM uri WHERE username='v101' AND domain='sip0.en.ewetel.de' LIMIT 1] Jan 16 21:30:45 [8406] DBG:avpops:db_query_avp: rows [1] Jan 16 21:30:45 [8406] DBG:avpops:db_query_avp: row [0] Jan 16 21:30:45 [8406] DBG:avpops:db_close_query: close avp query Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 1 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81dde58 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81dde58=pkg_free() RES_ROWS From-URI belongs to xxxxxxxxxxx Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:auth:check_nonce: comparing [478e6aa112f6919eb75ced3104f1774c59a1ce01] and [478e6aa112f6919eb75ced3104f1774c59a1ce01] Jan 16 21:30:45 [8406] DBG:auth_db:get_ha1: HA1 string calculated: 98eb7bda25d62322c1ff8261c216e8ad Jan 16 21:30:45 [8406] DBG:auth:check_response: our result = '43c9a3529f44aa0e6dda2f6f0894632c' Jan 16 21:30:45 [8406] DBG:auth:check_response: authorization is OK Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 1 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81dde88 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81dde88=pkg_free() RES_ROWS Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:comp_scriptvar: str 29 : xxxxxxxxx Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Route 04: <BYE>-Authentication OK Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=14000 Route 07: <BYE>: Loading A-Profile ... Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=2, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:dbrow2avp: db_flags=0, flags=3 Jan 16 21:30:45 [8406] DBG:avpops:db_close_query: close avp query Jan 16 21:30:45 [8406] DBG:core:db_free_rows: freeing 15 rows Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[0]=0x81de068 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[1]=0x81de070 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[2]=0x81de078 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[3]=0x81de080 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[4]=0x81de088 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[5]=0x81de090 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[6]=0x81de098 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[7]=0x81de0a0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[8]=0x81de0a8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[9]=0x81de0b0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[10]=0x81de0b8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[11]=0x81de0c0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[12]=0x81de0c8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[13]=0x81de0d0 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: row[14]=0x81de0d8 Jan 16 21:30:45 [8406] DBG:core:db_free_rows: 0x81de068=pkg_free() RES_ROWS Jan 16 21:30:45 [8406] DBG:avpops:ops_dbload_avps: loaded avps = 15 OK Jan 16 21:30:45 [8406] DBG:core:comp_scriptvar: int 20 : 1 / 1 MAIN : <BYE>: From-Data is: User='v101', Domain='sip0.en.ewetel.de', URI='sip:v101@sip0.en.ewetel.de' Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=200 Jan 16 21:30:45 [8406] DBG:rr:is_preloaded: is_preloaded: No Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:check_self: host != me Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 13==13 && [212.6.123.118] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:rr:after_loose: Topmost route URI: 'sip:212.6.123.118;lr=on;ftag=bpl4kg8f57' is me Jan 16 21:30:45 [8406] DBG:rr:after_loose: URI to be processed: 'sip:212.6.123.118:5090;lr;ftag=bpl4kg8f57;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--' Jan 16 21:30:45 [8406] DBG:rr:after_loose: Next URI is a loose router Jan 16 21:30:45 [8406] DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=bpl4kg8f57> DEBUG:uac:restore_from: getting 'vsf' Route param DEBUG:uac:restore_from: Route param 'vsf' not found Jan 16 21:30:45 [8406] DBG:avpops:ops_check_avp: check <17> against <8> as int /256 Jan 16 21:30:45 [8406] DBG:avpops:ops_check_avp: no match Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=200 Jan 16 21:30:45 [8406] DBG:rr:is_preloaded: is_preloaded: No Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 14==13 && [80.228.245.138] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5080 Jan 16 21:30:45 [8406] DBG:core:check_self: host != me Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if host==us: 13==13 && [212.6.123.118] == [212.6.123.118] Jan 16 21:30:45 [8406] DBG:core:grep_sock_info: checking if port 5060 matches port 5060 Jan 16 21:30:45 [8406] DBG:rr:after_loose: Topmost route URI: 'sip:212.6.123.118;lr=on;ftag=bpl4kg8f57' is me Jan 16 21:30:45 [8406] DBG:rr:after_loose: URI to be processed: 'sip:212.6.123.118:5090;lr;ftag=bpl4kg8f57;vsf=ZlRqaD8wOzEbMTUlTiwzCnRNJ2F0eDBzMTZ0HVdlUg--' Jan 16 21:30:45 [8406] DBG:rr:after_loose: Next URI is a loose router Jan 16 21:30:45 [8406] DBG:rr:run_rr_callbacks: callback id 0 entered with <lr=on;ftag=bpl4kg8f57> DEBUG:uac:restore_from: getting 'vsf' Route param DEBUG:uac:restore_from: Route param 'vsf' not found MAIN : Loose-Route Jan 16 21:30:45 [8406] DBG:uri:has_totag: totag found Route 08: <BYE>: Set Accounting for this call! Route 11: Removing secret Headers Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:registrar:lookup: '80.228.245.138' Not found in usrloc Route 01: Relaying BYE: RURI=sip:80.228.245.138:5080 F=sip:v101@sip0.en.ewetel.de IP=212.6.95.88; Calltimeout=<null> [Wed Jan 16 21:30:45 2008] Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Route 01: Next Hop DST-Set Jan 16 21:30:45 [8406] DBG:tm:t_newtran: transaction on entrance=(nil) Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=ffffffffffffffff Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: start searching: hash=36300, isACK=0 Jan 16 21:30:45 [8406] DBG:tm:matching_3261: RFC3261 transaction matching failed Jan 16 21:30:45 [8406] DBG:tm:t_lookup_request: no transaction found Jan 16 21:30:45 [8406] DBG:tm:run_reqin_callbacks: trans=0xb5b01048, callback type 1, id 1 entered Jan 16 21:30:45 [8406] DBG:siptrace:trace_onreq_in: trace off... Jan 16 21:30:45 [8406] DBG:tm:run_reqin_callbacks: trans=0xb5b01048, callback type 1, id 0 entered Jan 16 21:30:45 [8406] DBG:core:parse_headers: flags=78 Any ideas, why it hangs eating all my cpu power ? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHjmra4tZeNddg3dwRAvu4AJ4rK5jCbKF9UDGu8FP1suEyRHkwigCeM58p K5a16nleRI+zZujWTI0Ntfs= =XJqP -----END PGP SIGNATURE-----

2 3

[OpenSER-Users] Compilation Problem - please help me.
by Noble Antony T 17 Jan '08

17 Jan '08

Hi, I was trying to install Openser with RLS support, I got an error message, *In file included from notify.h:31, from hash.c:37: presentity.h:36:27: libxml/parser.h: No such file or directory make[1]: *** [hash.o] Error 1 make[1]: Leaving directory `/home/nthattil/openser-1.3.x/modules/presence' make: *** [modules] Error 2 * I had installed libxml2 already, * libxml2.so.2 libxml2.so.2.6.27* the above modules are present in folder /usr/lib/ I am using the machine, *Linux version 2.6.18 (gcc version 3.3.5 (Debian 1:3.3.5-13)) #1 SMP Thu Jul 26 01:27:32 EDT 2007 * I edited the Makefile and updated the line as follows, *exclude_modules?= jabber cpl-c postgres osp unixodbc \ avp_radius auth_radius group_radius uri_radius xmpp \ presence_xml presence_mwi pua_bla pua_mi \ pua_usrloc pua_xmpp mi_xmlrpc perl snmpstats perlvdb \ ldap carrierroute h350 xcap_client db_berkeley seas * I gave command prompt$ *make all* for compiling. I am getting the error message as shown above. Please mail me if I am missing something.. -- Noble Antony Thattil, Bangalore.

3 2

[OpenSER-Users] openSER Call Route Logic
by asterisk User 17 Jan '08

17 Jan '08

Hi, I recently installed openser-1.3.0 and able to connect my cisco sip phones and everything is working. Now i need to integrate this with asterisk box for call forwarding to other locations & voicemails. i read the getting started guide and other information, but seems i need more help from the group. can any one give me a helping hand. i am sure i will do my home work before bug you again. I need a starting point to over come the complexity of the routing plan and logic. thanks -asterisk user --------------------------------- Looking for last minute shopping deals? Find them fast with Yahoo! Search.

2 1

[OpenSER-Users] How to interacte X-lite and openser
by hammou moujane 17 Jan '08

17 Jan '08

Hello All, I have installed openser, is it possible to make voice call between two x-lite softphone using this SIP server? how can I start this operation if it's possible? Thankyou very much for your help. Regards -- Hammou MOUJANE Tel: +212 62 46 33 89 INPT

2 1

[OpenSER-Users] MySQL Alternative
by Douglas Garstang 17 Jan '08

17 Jan '08

What other options do I have for generic database connectivity besides MySQL/avp_db_query() in OpenSER? Can I use SQLite? Can I use unixodbc? (and I'm not talking about the limited built in usrloc unixodbc stuff) Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

Re: [OpenSER-Users] {Spam?} Hello World Config problem
by Andy Smith 16 Jan '08

16 Jan '08

Hi! Ok, thanks that problem solved and now I have another! I get the error "Registration error: 404 - Not Found" in Xlite. If I comment out this section of the cfg file I then get an error "Too Many Hops" :S :SMy laptop is directly (virtually on VPN) connected to the same LAN as the openser server.Any ideas? Or anyone have an already working hello world config??thanks in advance, Andy.On Wednesday 16 January 2008 15:51:58 Andy Smith wrote: > Thanks Iñaki, ok now my openser starts up ok, but I get the same "filtered > destination" Try adding: disable_dns_blacklist=yes at the top of your script.

1 0

[OpenSER-Users] Problems with unixodbc driver
by Alcindo Schleder 16 Jan '08

16 Jan '08

Hi.. I´m using unixodbc driver to connect with firebird SGDB. The problem is that the unixodbc driver not connect... ERROR:unixodbc:new_connection: failed to connect Jan 15 14:19:06 xsip openser[4577]: unixodbc:SQLDriverConnect=08004:1:-904:[unixODBC][ODBC Firebird Driver]Unable to connect to data source: library 'libfbclient.so' failed to load Jan 15 14:19:06 xsip openser[4577]: ERROR: group_db_bind: unable to connect to the database Jan 15 14:19:06 xsip openser[4577]: ERROR:group:mod_init: unable to open database connection

2 4

Re: [OpenSER-Users] {Spam?} Hello World Config problem
by Andy Smith 16 Jan '08

16 Jan '08

Thanks Iñaki, ok now my openser starts up ok, but I get the same "filtered destination" error from Xlite as I was getting with my original openser cfg with the hello world route. I was hoping the hello world config would allow me to connect/register whatever SIP device without any specific password :S thanks Andy.On Wednesday 16 January 2008 13:44:10 A.smith wrote: > If I try and use the whole example cfg (not just the route) I am recieving > an error like: > > 0(23448) find_cmd_export_t: <lookup> not found > 0(23448) parse error (105,23-24): unknown command, missing loadmodule? "lookup()" is defined in "registrar" module that you are not loading: http://www.openser.org/docs/modules/1.2.x/registrar.html#AEN342 -- Iñaki Baz Castillo ibc at in.ilimit.es

2 1

[OpenSER-Users] Hello World Config problem
by A.smith 16 Jan '08

16 Jan '08

HI, As I dont currently have a working PSTN I wanted to get the hello world config working so I can generate calls from one IP phone to another just across my LAN. However with the route config (put into my previous working registrar test cfg) from this example: http://openser.oralnet.co.uk/#AEN424 I am recieving a "filtered destination" error from XLite when it attempts to register. I can see the packets hitting my server from tcpdump... If I try and use the whole example cfg (not just the route) I am recieving an error like: 0(23448) find_cmd_export_t: <lookup> not found 0(23448) parse error (105,23-24): unknown command, missing loadmodule? Any help or tips on diagnosing this would be greatly appreciated! I am using OpenSER 1.2.2 on CentOS Linux... thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

3 2

[OpenSER-Users] Openser + Wesip Click2dial
by [ACTIVENDY] NOC 16 Jan '08

16 Jan '08

Hi, I've installed my Openser with Wesip Application Server. I've installed Wesip with Click2dial application from Wesip web site. The application is running, but when I try to call to a softphone sip address trough my browser I get an error code -2. In Wesip logs I get the following: CommonsLoggingOutput [HttpProcessor[7080][4]] - Exec: Click2Call.c2cDWR() CommonsLoggingOutput [HttpProcessor[7080][4]] - --Object created, not stored. id=0 MapFactory [HttpProcessor[7080][4]] - Unable to create class java.util.Hashtable returning HashTable instead StandardSipSession [HttpProcessor[7080][4]] - Transport: udp SipRequest [HttpProcessor[7080][4]] - ERROR sending message javax.sip.TransactionUnavailableException: no route! at com.voztele.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:264) at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:196) at com.voztele.wesip.apps.click2call.C2C.c2c(C2C.java:134) at com.voztele.wesip.apps.click2call.C2C.c2cDWR(C2C.java:99) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.directwebremoting.impl.ExecuteAjaxFilter.doFilter(ExecuteAjaxFilter.java:34) at org.directwebremoting.impl.DefaultRemoter$1.doFilter(DefaultRemoter.java:428) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:431) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:283) at org.directwebremoting.servlet.PlainCallHandler.handle(PlainCallHandler.java:52) at org.directwebremoting.servlet.UrlProcessor.handle(UrlProcessor.java:101) at org.directwebremoting.servlet.DwrServlet.doPost(DwrServlet.java:146) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedContextValve.invokeHTTP(ConvergedContextValve.java:144) at com.voztele.sipservlet.valves.ConvergedContextValve.invoke(ConvergedContextValve.java:64) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.core.StandardHttpContext.invoke(StandardHttpContext.java:2377) at com.voztele.sipservlet.core.ConvergedContext.invoke(ConvergedContext.java:98) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeHTTP(ConvergedHostValve.java:336) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.connector.http.HttpProcessor.process(HttpProcessor.java:994) at org.apache.catalina.connector.http.HttpProcessor.run(HttpProcessor.java:1105) at java.lang.Thread.run(Thread.java:619) java.lang.RuntimeException: javax.sip.TransactionUnavailableException: no route! at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:222) at com.voztele.wesip.apps.click2call.C2C.c2c(C2C.java:134) at com.voztele.wesip.apps.click2call.C2C.c2cDWR(C2C.java:99) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(NativeMethodAccessorImpl.java:39) at sun.reflect.DelegatingMethodAccessorImpl.invoke(DelegatingMethodAccessorImpl.java:25) at java.lang.reflect.Method.invoke(Method.java:597) at org.directwebremoting.impl.ExecuteAjaxFilter.doFilter(ExecuteAjaxFilter.java:34) at org.directwebremoting.impl.DefaultRemoter$1.doFilter(DefaultRemoter.java:428) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:431) at org.directwebremoting.impl.DefaultRemoter.execute(DefaultRemoter.java:283) at org.directwebremoting.servlet.PlainCallHandler.handle(PlainCallHandler.java:52) at org.directwebremoting.servlet.UrlProcessor.handle(UrlProcessor.java:101) at org.directwebremoting.servlet.DwrServlet.doPost(DwrServlet.java:146) at javax.servlet.http.HttpServlet.service(HttpServlet.java:760) at javax.servlet.http.HttpServlet.service(HttpServlet.java:853) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:200) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:146) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:206) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedContextValve.invokeHTTP(ConvergedContextValve.java:144) at com.voztele.sipservlet.valves.ConvergedContextValve.invoke(ConvergedContextValve.java:64) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.core.StandardHttpContext.invoke(StandardHttpContext.java:2377) at com.voztele.sipservlet.core.ConvergedContext.invoke(ConvergedContext.java:98) at com.voztele.sipservlet.valves.ConvergedHostValve.invokeHTTP(ConvergedHostValve.java:336) at com.voztele.sipservlet.valves.ConvergedHostValve.invoke(ConvergedHostValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.valves.ErrorDispatcherValve.invoke(ErrorDispatcherValve.java:124) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:118) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:594) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at com.voztele.sipservlet.valves.ConvergedEngineValve.invoke(ConvergedEngineValve.java:98) at org.apache.catalina.core.StandardPipeline$StandardPipelineValveContext.invokeNext(StandardPipeline.java:596) at org.apache.catalina.core.StandardPipeline.invoke(StandardPipeline.java:433) at org.apache.catalina.core.ContainerBase.invoke(ContainerBase.java:946) at org.apache.catalina.connector.http.HttpProcessor.process(HttpProcessor.java:994) at org.apache.catalina.connector.http.HttpProcessor.run(HttpProcessor.java:1105) at java.lang.Thread.run(Thread.java:619) Caused by: javax.sip.TransactionUnavailableException: no route! at com.voztele.javax.sip.SipProviderImpl.getNewClientTransaction(SipProviderImpl.java:264) at com.voztele.sipservlet.core.SipRequest.send(SipRequest.java:196) ... 44 more My configuration is: - openser.cfg debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 listen=MY_IP_ADDRESS mpath="/lib/openser/modules" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "seas.so" modparam("mi_fifo","fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) modparam("seas", "listen_sockets", "MY_IP_ADDRESS:5080") route{ 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; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if(!as_relay_t("wesip")){ t_reply("500","Application Server error"); } } - server.xml <Server port="5005" shutdown="SHUTDOWN" debug="0"> <Service name="WeSIP-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="7080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="217.172.74.8:5060" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="MY_IP_ADDRESS" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5080" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="javax.sip.STACK_NAME" value="wesip" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="217.172.74.8" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="MY_IP_ADDRESS" nameSip="MY_IP_ADDRESS_5060" debug="0" appBase="wesipapps" unpackWARs="true" autoDeploy="true" className="com.voztele.sipservlet.core.ConvergedHost">  <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> Thanks in advance, Sergi

2 2

[OpenSER-Users] How to convert 183 to 180 message
by Tung Tran 16 Jan '08

16 Jan '08

3 3

[OpenSER-Users] OpenSER config Wiz - What does it do? :P
by A.smith 16 Jan '08

16 Jan '08

Hi, Slightly stupid question I know, what I mean is that the config contains no rewrite host so Im just wondering whether the configs that the wizard generate are actually useable for something without modification. I ask because I want to understand what the wizard is trying to produce so I know if its appropriate for me to try and modify the config to match my requirements or Im better to start from scratch. And yes, I know I dont know much about OpenSER! :P Im trying to learn! thanks Andy. ________________________________________________ Message sent using UK Grid Webmail 2.7.9

1 0

[OpenSER-Users] Variable $rc
by Douglas Garstang 16 Jan '08

16 Jan '08

What's the point of the pseudo variable $rc? If I use avp_db_query() with: invalid SQL, then rc = -1. database down: then rc = -1. Everything fine... then rc = -1!!! What's the point? Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

4 3

[OpenSER-Users] Presence Problem on v1.3.0
by fengbin 16 Jan '08

16 Jan '08

Hi,All When I tried to start openser with presence module.I met an error of "ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence>" But I used the presence before with openser of v1.2.0 and it works fine. Can you take a look at it? THX BR The encolsed is my config file. -- Fengbin

2 1

[OpenSER-Users] Openser and RTPProxy
by Patrick Baker 16 Jan '08

16 Jan '08

Is is possible to have a SIP UA behind nat and communicate to an OpenSER/RTP Proxy server behind nat behind nat and have everything transverse appropriately? Or will I need something additional like STUN? Example: OpenSER + RTP Proxy + Asterisk will all reside on the same machine. SIP UA -> Cisco PIX ( Public IP to Localnet NAT ) -> OpenSER/RTP -> Asterisk Thank you for your time, Patrick

4 7

[OpenSER-Users] OpenSer 1.3 Presence Problem
by Ross Beer 15 Jan '08

15 Jan '08

Hi All, I am trying to setup an OpenSER server with support for registration, messaging & presence however I am having a problem with the presence server. I have had presence working correctly before, however for some reason it is no longer working. I have spent about two days trying everything! I would really appreciate some advice with this. I have included my config below. Thank you in advance, Ross ------------------------------------------------------------------------------ listen = udp:my.ip.address:5060 mpath = "/usr/local/lib64/openser/modules" children = 8 debug = 3 fork = yes disable_tcp = no #log_facility = LOG_DAEMON log_facility = LOG_LOCAL0 log_stderror = yes tcp_children = 4 mhomed = no server_signature = yes #sock_group = "openser" #sock_mode = 0600 #sock_user = "openser" #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 "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://username:password@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 1) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") #modparam("mi_fif0", "fifo_name", "FIFO") modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://username:password@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 0) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "") modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "textops.so" loadmodule "mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so" loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://username:password@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") #modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=id") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://username:password@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "mysql://username:password@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 0) modparam("avpops", "uuid_column", "id") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") loadmodule "enum.so" modparam("enum", "domain_suffix", "e164.org.") loadmodule "permissions.so" #modparam("permissions", "default_allow_file", "permissions.allow") #modparam("permissions", "default_deny_file", "permissions.deny") #modparam("permissions", "check_all_branches", 1) #modparam("permissions", "allow_suffix", ".allow") #modparam("permissions", "deny_suffix", ".deny") modparam("permissions", "db_url", "mysql://username:password@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") modparam("permissions", "tag_col", "tag") modparam("permissions", "peer_tag_avp", "$avp(i:707)") loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://username:password@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "dm_flag", 25) loadmodule "alias_db.so" modparam("alias_db", "db_url", "mysql://username:password@localhost/openser") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) # ----- presence params ----- loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so" loadmodule "pua.so" loadmodule "pua_mi.so" loadmodule "pua_usrloc.so" modparam("presence|presence_xml", "db_url","mysql://username:password@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:my.ip.address:5060") modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "max_expires", 3600) modparam("presence", "fallback2db", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) modparam("pua_usrloc", "default_domain", "my.domain.com") modparam("presence", "to_tag_pref", 'pres') ######################################################################## # 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|MESSAGE")) { 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(); setbflag(6); } route(7); } if(is_method("REGISTER")) { route(6); } if(is_method("INVITE")) { route(8); } if(is_method("PUBLISH|SUBSCRIBE")) { route(17); } if(is_method("CANCEL") || is_method("ACK")) { route(14); } ############################ # MESSAGES # ############################ if(is_method("MESSAGE")) { route(18); } route(15); } ######################################################################## # 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"); } ######################################################################## # 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"); } ######################################################################## # 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)", "*"); } ######################################################################## # 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)", "*"); } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[5] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[6] { 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[7] { xlog("L_INFO", "ROUTE 7 RUN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("2"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[8] { 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=$ci\n"); # sl_send_reply("403", "Spoofed From-URI Detected"); # exit; # } } } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(9); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[9] { if(alias_db_lookup("dbaliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(11); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select id from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(10); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[10] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[11] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); exit; } route(13); if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:[0-9]+@") { # only route numeric users to PSTN if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$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=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(7); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'normalize-e164' ######################################################################## route[12] { # European numbering plans look like this: # CC = country code (i.e. 43 for Austria) # NDC = national destination code (i.e. 1 for Vienna) # SN = subscriber number (i.e. 4001234) # # CC + NDC + SN, i.e. 4314001234 # # Within the same CC+NDC, it can be omitted, so if # +4314001234 wants to call +4315002345, one can dial # just 5002345. # # Within the same CC, CC can be ommitted and a "0" is prefixed # to NDC, so if +4314001234 wants to call +4326003456, # one can dial 026003456. # # For international calls, either "00" or + is prefixed, like # +49123456780 or 0049123456789. # avp_delete("$avp(s:orig_callee_user)/g"); $avp(s:orig_callee_user) = $rU; if(uri =~ "^sip:(\+[1-9])?[0-9]+@") { # looks like a PSTN number if(uri =~ "^sip:0[1-9][0-9]+@") { # we have format 0+NDC+SN strip(1); prefix("+44"); } else if(uri =~ "^sip:00[1-9]+@") { # we have format 00 + CC + NDC + SN strip(2); prefix("+"); } else if(!uri =~ "^sip:\+[1-9][0-9]+@") { # unknown format, maybe NDC wasn't added before? xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(-1); } # else we have "+" + CC + NDC + SN xlog("L_INFO", "Normalized callee '$avp(s:orig_callee_user)' to E.164 format '$rU' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(1); } else { xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); return(-1); } } ######################################################################## # Request route 'lookup-enum' ######################################################################## route[13] { route(12); if($rc == 1) { if(enum_query("e164.org.")) { # TODO: do GW fallback (load gws, set failure-route)? xlog("L_INFO", "ENUM query succeeded - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); exit; } else { xlog("L_INFO", "ENUM query failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # ENUM query failed, revert $rU avp_pushto("$ru/username", "$avp(s:orig_callee_user)"); } } } ######################################################################## # Request route 'base-route-local' ######################################################################## route[14] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$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=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[15] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[16] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); exit; } } ######################################################################### # Presence ######################################################################### route[17] { # if(nat_uac_test("19") || search("^Route:.*;nat=yes")) # { # fix_nated_contact(); # setbflag(6); # } if (!t_newtran()) { sl_reply_error(); exit; }; #if(is_method("PUBLISH")) #{ # xlog("L_INFO", "PUBLISH - Process Publish Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # handle_publish(); # t_release(); #} if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } if(is_method("SUBSCRIBE")) { handle_subscribe(); xlog("L_INFO", "SUBSCRIBE - Process Subscribe Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); t_release(); } exit; } ######################################################################## # SEND MESSAGES ######################################################################## route[18] { if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } if (!t_relay()) { sl_reply_error(); }; 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; } ######################################################################## # Failure route 'pstn-failover' ######################################################################## failure_route[1] { xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(16); if(!next_gw()) { xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); exit; } t_on_failure("1"); route(7); } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[2] { route(16); route(5); } _________________________________________________________________ Share what Santa brought you https://www.mycooluncool.com

3 5

[OpenSER-Users] OpenSER + OpenXCAP: NOTIFY response to unsubscription includes xml document?
by KevinKinnan 15 Jan '08

15 Jan '08

Openser: NOTIFY response to unsubscription doesn't have message body. Openser + OpenXCAP: NOTIFY response to unsubscription includes xml document which is the same as one in response to subscription, but the subscription_state in unsub NOTIFY message is "terminated". Can anyone give me a hint? Regards, Kevin _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

1 0

[OpenSER-Users] OpenSER: no NOTIFY when updating PUBLISH if integrating with OpenXCAP?
by KevinKinnan 15 Jan '08

15 Jan '08

1. Only Openser, no OpenXCAP, force_active=1 NOTIFY with new info is sent from Openser to client when updating PUBLISH message everytime. 2. Openser+OpenXCAP force_active=0 NO NOTIFY is sent from Openser if integrating with OpenXCAP. But PUBLISH message updates properly. I can't figure out why, any clue? Thank you! Kevin, _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

1 0

Re: [OpenSER-Users] avp_db_query() timeout
by Douglas Garstang 15 Jan '08

15 Jan '08

Inaki, I am doing updates with avp_db_query, so there is no avp I can check the value of . As I said in a subsequent post, I wrote a simple server that listened on port 3306, like mysql does, and pointed OpenSER at it. The avp_db_query commands() basically blocked forever and openser seemed to completely lose all ability to process new calls. That's really bad. Doug. ----- Original Message ---- From: Iñaki Baz Castillo <ibc(a)aliax.net> To: users(a)lists.openser.org Sent: Monday, January 14, 2008 1:46:00 PM Subject: Re: [OpenSER-Users] avp_db_query() timeout El Lunes, 14 de Enero de 2008, Douglas Garstang escribió: > I haven't had much luck getting replies to my questions lately, hopefully > I'll have more luck with this one. > > avp_db_query().... does it have a timeout? What is it? Can I set it? > > If I use it, and there is a MySQL database problem, I don't want > avp_db_query() to wait forever trying to connect to MySQL. If it can't > connect after a few seconds, I'd like it to just give up. I have multiple > calls to avp_db_query() in a typical call flow. It would be even better if > I could set a flag on the first timeout and not try again during that call. > > Can that be done? Hi, by the test I've done: avp_db_query("SQL_QUERY", "$avp(s:result)"); xlog("L_INFO", "retcode = $rc \n"); Cases: - If query doesn't return a row/value then $avp(s:result)=null and $rc=(-1). - If query returns a row/value the $avp(s_result) stores that value and $rc=1. - If there is a syntax error in query then $avp(s:result)=null and $rc=(-1). - If MySQL is not running then $avp(s:result)=null and $rc=(-1). I don't know what occurs if MySQL is running but because overload or any other reason it doesn't reply, but AFAIK the "avp_db_query" uses the same code of other OpenSer functions to access DB, so I'm sure it has a correct behaviour in case of failure. BTW you can "play" with the $rc code after "avp_db_query". -- Iñaki Baz Castillo _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

3 4

[OpenSER-Users] CPL and $var issue when different process
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hi, maybe someone could confirm this: At the begining of my script I set a variable: $var(gateway) = "123.123.123.123"; Later, I run CPL for calls to internal users and there could be forwardings and so. I set CPL "recurse" parameter so convert 302 from UAS into a new branch into CPL. At the end of CPL I run "route[20]" in which I use the variable $var(gateway), but sometimes it appears empty. I think this could occur because CPL can involve various OpenSer process. I think that when CPL receives a 302, maybe sometimes it's a new OpenSer process and since it's a reply the variable at the begining of my script is not read, so it appears empty. Am I right? A solution is to use AVP instead od variables since AVP's remains along al lthe transaction. Am I right? Thanks for any confirmation. Best regards. -- Iñaki Baz Castillo

4 5

[OpenSER-Users] ¿Se puede enviar DTMF durante el Early Media?
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hola, algo que fastidia a todo el mundo es el tema de que te cobren una llamada nada más salir el buzón de voz. Lo lógico y deseable (al margen de la actitud avariciosa de las grandes operadoras) es que al llamar a un número y salir el buzón de voz no se cobre la llamada y se ofrezca la posibilidad de dejar un mensaje pulsando algún dígito. El impedimento técnico que veo es que parece ser no se permite el envío de DTMF durante el EarlyMedia. Pero no encuentro documentación sobre ello. El caso es que durante el EarlyMedia el audio es bidireccional, o sea, el SDP del "183 Session Progress" dice: a=sendrecv (de hecho esto lo he verificado también a nivel de gateway PSTN: una llamada desde la PSTN a un gateway y éste la dirige a un SIP UAS que responde con 183, y el gateway envía y recibe audio desde el UAS). A todo esto una duda que tengo es precisamente esto de poder enviar y recibir audio durante un EarlyMedia, ¿se puede o no se puede? ¿se hace? Por otra parte, mi Twinkle no me permite enviar DTMF durante el EarlyMedia, pero vamos, que ni lo intenta por lo que no sé qué pasaría si sí me dejase. Y por las pruebas que he hecho llamando desde la PSTN (fijo y móvil) y respondiendo con un EarlyMedia, tampoco parece que se permita el envío de DTMF. En fin, que la única opción honesta que he encontrado es la de reproducir el mensaje del buzón en EarlyMedia, y sólo tras el mensaje responder la llamada para que el llamante deje su mensaje (y sólo en ese caso provocar que tenga que pagar su llamada). Con lo fácil que es... lo cab***as que son las compañías. Bueno, sin más, que no sé exactamente cuál es mi pregunta pero tal vez alguien me ayude a definirla XD Saludos. -- Iñaki Baz Castillo

4 6

[OpenSER-Users] Giving more power to CPL language: Addressbook Categories feature
by Iñaki Baz Castillo 15 Jan '08

15 Jan '08

Hi, in my opinion CPL language is not enough powerful, the absence of regular expression or any header reading makes it poor (IMHO). It would become really cool if it could match AVP's but it can :( I'm trying to give some power to CPL logic. I want to implement "addressbook categories" for each user, this is: each user has its own addressbook in which he can set a category for each contact: - Work - Friend - Family - Custom And of course what I need it a way to match that category **inside** the CPL logic. This is impossible with CPL itself so I'm thinking in a very very dirty trick: ** Addressbook Categories ** - In an incoming call for "bob(a)domain.org" OpenSer will match the callerid against Bob's addressbook. This will return a variable or AVP: $avp(s:caller_category) - CPL module just can match "Organization", "Subject", and "User-Agent", so OpenSer will do: append_hf("Bak_Organization: $hdr(Organization)\r\n"); # To preserve original Organization remove_hf("Organization"); # Remove original Organization append_hf("Organization: $avp(s:caller_category)\r\n"); # Set category as Organization - But since OpenSer appends/removes headers when the message leaves the proxy I can't run CPL now (it will not read the new "Organization" header), so I need to do a dirty loop in order to append the header. - When the message comes back into OpenSer the headers are updated so I do: remove_hf("Organization"); # Remove current Organization (that is a hack) append_hf("Organization: $hdr(Backup_Organization)\r\n"); # Restore original Organization (those changes will be applied again when message leaves OpenSer, not now). - And I run CPL: cpl_run_script("incoming","FORCE_STATEFUL"); - Into CPL I match "Organization" header by: <string-switch field="Organization"> <string is="Friends"> <reject status="reject" reason="I have no friends" /> </string> </string-switch> Is it enough dirty or could I make it more? Any suggestion? how do you people implement things like this? Thank a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[OpenSER-Users] OpenSER logo
by Reinhold Buchinger 15 Jan '08

15 Jan '08

Hi, Where can I find the OpenSER logo in a high resolution? Thanks for your help. Regards, Reinhold

3 2

[OpenSER-Users] CRL/OCSP in OpenSer
by Jan Ruzicka 15 Jan '08

15 Jan '08

Hi, is it possible to use CRL (OCSP)? I found an old message from Klaus (http://www.nabble.com/-Users--CRL-in-OpenSER-to12712663.html#a12712666). As far I don't see mentioned new cfg parameter, I'm afraid its not possible. Are there any plan to use these features? Thanks Jan

2 1

[OpenSER-Users] OpenSER Configuration via files?
by Douglas Garstang 15 Jan '08

15 Jan '08

Asked this question earlier in the week, and didn't get an answer. I'm using avp_db_query to log calls to MySQL. I'd like to be able to turn this and other functionality on and off in the openser.cfg file without having to comment lines. Obviously putting the settings into MySQL and reading them with avp_db_query() doesn't make much sense. Does OpenSER have some sort of simple key/value pair configuration it read from a file so that I can control the basic configuration with? The db_text and db_berkeley modules are unfortunately only designed to be used for userloc data. Why is that? Thanks, Doug. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] OpenXCAP: sphere and validity?
by KevinKinnan 15 Jan '08

15 Jan '08

I can't make it work when testing the effect of validity element in conditions element in Openser+OpenXCAP, also happen on sphere element and transformations. The presence document is based on rpid and it is quite simple. The only one I got it work is like below(without validity and sphere elements): <condition> <one id=""> </condition> <actions> <sub-handling>allow</sub-handling> </actions> The following two(adding validity and sphere) are not working: <condition> <one id=""> <validity> <from>...</from> <until>...</until> </validity> </condition> <actions> <sub-handling>allow</sub-handling> </actions> <condition> <shpere value="home"> <one id=""> </condition> <actions> <sub-handling>allow</sub-handling> </actions> Does OpenXCAP support these policies metioned above? Or, Openser is not supporting sphere and validity auth, it only knows sub-handling(xcap_auth.c in openser/modules/presence_xml)? Thank you! Kevin, _________________________________________________________________ Windows Live Photo gallery 数码相机的超级伴侣，轻松管理和编辑照片，还能制作全景美图！ http://get.live.cn/product/photo.html

1 0

[OpenSER-Users] avp_db_query() timeout
by Douglas Garstang 14 Jan '08

14 Jan '08

I haven't had much luck getting replies to my questions lately, hopefully I'll have more luck with this one. avp_db_query().... does it have a timeout? What is it? Can I set it? If I use it, and there is a MySQL database problem, I don't want avp_db_query() to wait forever trying to connect to MySQL. If it can't connect after a few seconds, I'd like it to just give up. I have multiple calls to avp_db_query() in a typical call flow. It would be even better if I could set a flag on the first timeout and not try again during that call. Can that be done? Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

[Serusers] xml rpc documentation
by Srinivas Kotamarti 14 Jan '08

14 Jan '08

Hi, Does anybody know where can I find more documentation on the new XML-RPC module of SER2. I want to write a management plugin for the application I am writing on top of SEMS using the XML-RPC interface. The documentation in the SER2.0 documentation page (http://www.iptel.org/rpc_xml_rpc) does not explain its full functionality and is not enough for writing a plugin. Thanks --Srinivas

2 1

[OpenSER-Users] strange error logs in syslog
by Cseke Tamas 14 Jan '08

14 Jan '08

Hello, We have a lots of error log like this, what fills syslog ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction ERROR:tm:w_t_relay: t_forward_nonack failed ERROR: t_check_status: cannot check status for a reply which has no T-state established Could you tell me how could i avoid this? I think these are not errors. Shall i just change the debug level for this logs? Thanks any help, Tamas

2 2

[OpenSER-Users] radius_does_uri_exist
by Marc LEURENT 14 Jan '08

14 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good afternoon, I'm trying to use the radius_does_uri_exist() function to check if the user exist before forwarding it to voicemail when it's not enable. I'm using radius AAA and the radius_does_uri_exist() seems to fail.. Any idea how to solve it? Thanks # -- uri_radius paramaters -- modparam("uri_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("uri_radius", "service_type", 11) # Radius service type used in radius_does_uri_exist check. ....... if (!lookup("location")) { if (radius_does_uri_exist()) { # Will Redirect to voicemail when fixed sl_send_reply("480", "Temporarily Unaivailable - Will Redirect To Voicemail"); exit; }; xlog("L_DBG", "ACC RADIUS: 404 NOT FOUND FOR $si \r\n"); sl_send_reply("404", "Not Found"); exit; ..... When I'm trying to call aeron which is not connected but exists, I have a 404 not found whereas I should have a 480... root@krusty:[~L/freeradius]# tail -f radius.log Mon Jan 14 16:08:18 2008 : Auth: Login incorrect: [aeron(a)sip.bob.net/<no User-Password attribute>] (from client private port 0) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHi311N4+o+2LtdFwRAgGNAKC4xB0R5mHjgZ0zNKCaEEcqCgCUaQCdG+p0 4KoP/SuAUHJ8u6GFnaOTLYw= =OXcY -----END PGP SIGNATURE-----

1 1

[OpenSER-Users] OpenXCAP problem
by Franz Edler 14 Jan '08

14 Jan '08

Hi OpenXCAP experts, I am trying to setup a demo-configuration for XCAP with Eyebeam- and Bria-client. OpenXCAP and OpenSER are running. When I start-up the Bria-client (data storage via XCAP enabled) I get the following errors from OpenXCAP - log (e.g. for resource-list): Jan 2 13:15:38 sip openxcap[10367]: [HTTPChannel,2,192.168.0.2] error: Invalid Document Selector xcap-root/resource-lists/users//resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') Jan 2 13:15:38 sip openxcap[10367]: [-] error: Invalid Document Selector xcap-root/resource-lists/users//resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') The HTTP dialog between the client and the XCAP server is the following: GET /xcap-root/resource-lists/users//resource-list.xml HTTP/1.1 Host: xcap.net1.test:8000 Keep-Alive: Connection: TE, Keep-Alive TE: trailers HTTP/1.1 404 Not Found Date: Wed, 02 Jan 2008 12:15:38 GMT Content-Length: 75 Server: OpenXCAP/0.9.7 the Document Selector context must be 'users' or 'global': 'resource-lists' PUT /xcap-root/resource-lists/users//resource-list.xml HTTP/1.1 Host: xcap.net1.test:8000 Connection: TE TE: trailers Content-Type: application/resource-lists+xml Content-Length: 323 <?xml version="1.0" encoding="UTF-8"?> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists" xmlns:cp="counterpath:properties"> <list name="Contact List"> <cp:prop name="category" value="Friends" /> <cp:prop name="category#2" value="Home" /> <cp:prop name="category#3" value="Work" /> </list></resource-lists> HTTP/1.1 404 Not Found Date: Wed, 02 Jan 2008 12:15:38 GMT Content-Length: 75 Server: OpenXCAP/0.9.7 the Document Selector context must be 'users' or 'global': 'resource-lists' Any hint highly appreciated! Franz

2 2

[Serusers] Registration expires timer
by Ranganath B 14 Jan '08

14 Jan '08

Hi, Need help in setting the registration expiry timer at SER. I would like to know the way we can enforce the registration expiry timer from server end. Please update me on the configuration details. Thanks in advance, Ranganath B

2 2

Re: [OpenSER-Users] TLS problem.
by Klaus Darilion 14 Jan '08

14 Jan '08

Hi Fengbin! Cc'ed to the openser list ... fengbin schrieb: > Hi,Klaus, > > How to use NULL cipher? Only setting in Openser is ok? I mean do I need > to set NULL cipher at client site? Usually the NULL cipher is not enabled (for security reasons). You have to enable it on both sides, the server and the client. But if you use the following approach you do not need it. > And where to put xlog("L_ERR","message buffer: $mb"); anywhere in > openser.cfg ? Put it just in the beginning of the route block. regards klaus > THX > BR > > > On 1/11/08, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at>> wrote: > > The capture file is not helpful, as it is encrypted. You could use NULL > cipher to have plaintext inside the TLS connection to inspect the > incoming SIP message, or add xlog("L_ERR","message buffer: $mb"); to see > the whole incoming SIP request. > > regards > klaus > > fengbin schrieb: > > Hi,Klaus > > Thank you for your reply. > > The enclosed is the config file ,the pcap between client and > server and > > the log on the openser 's console. > > Could you please take a look at them for me? > > > > THX > > BR > > > > > > On 1/10/08, *Klaus Darilion* <klaus.mailinglists(a)pernau.at > <mailto:klaus.mailinglists@pernau.at> > > <mailto:klaus.mailinglists@pernau.at > <mailto:klaus.mailinglists@pernau.at> >> wrote: > > > > Can you show us the REGISTER request? (both, port 5060 and > port 5061). > > > > Further show use your openser config > > > > regards > > klaus > > > > fengbin schrieb: > > > > > > Hi,all > > > I met a strange problem while I am testing TLS connection > between > > > minisip and openser. > > > The following is my openser.cfg (part of that) > > > > > > ......... > > > fork=no > > > log_stderror=yes > > > > > > # Uncomment this to prevent the blacklisting of > temporary not > > > available destinations > > > #disable_dns_blacklist=yes > > > > > > # # Uncomment this to prevent the IPv6 lookup after v4 > dns lookup > > > failures > > > #dns_try_ipv6=no > > > > > > # uncomment the following lines for TLS support > > > disable_tls = 0 > > > listen = tls: 10.11.57.197:5060 > <http://10.11.57.197:5060> <http://10.11.57.197:5060> > > <http://10.11.57.197:5060> > > > > > > > > > tls_verify_client = 1 > > > tls_method = TLSv1 > > > tls_certificate = "/usr/local/etc/openser//tls/user/user- > > cert.pem" > > > tls_private_key = > > "/usr/local/etc/openser//tls/user/user- privkey.pem" > > > tls_ca_list = "/usr/local/etc/openser//tls/user/user- > calist.pem" > > > tls_ciphers_list="NULL-SHA:NULL-MD5:AES256-SHA:AES128-SHA" > > > ...... > > > > > > When I set "tls:10.11.57.197:5061 > <http://10.11.57.197:5061> <http://10.11.57.197:5061> < > > http://10.11.57.197:5061>" the > > > registration never succeed. But if I set it to 5060 the > registration > > > over TLS is OK. > > > I compared the log of two scenarioes and found the TLS > session > > both are > > > OK,but the difference is that: > > > when the port is 5061 there is an error of forwarding. but the > > > forwarding is because openser think it's not the > destination of > > > the registration request. See bellow: > > > > > > Jan 10 16:46:56 [9199] DBG:rr:after_loose: No next URI > found > > > Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [ 10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197 <http://10.11.57.197>>] == > > > [10.11.57.197 <http://10.11.57.197> > <http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 16:46:56 [9199] DBG:core:grep_sock_info: > checking if port > > > 5061 matches port 5060 > > > Jan 10 16:46:56 [9199] DBG:core:check_self: host != me > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 16:46:56 [9199] DBG:tm:t_newtran: T on > > entrance=0xffffffff > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 16:46:56 [9199] DBG:core:parse_headers: flags=78 > > > Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: start > searching: > > > hash=58073, isACK=0 > > > Jan 10 16:46:56 [9199] DBG:tm:matching_3261: RFC3261 > transaction > > > matching failed > > > Jan 10 16:46:56 [9199] DBG:tm:t_lookup_request: no > > transaction found > > > Jan 10 16:46:56 [9199] DBG:core:mk_proxy: doing DNS > lookup... > > > Jan 10 16:46:56 [9199] ERROR:tm:update_uac_dst: failed > to fwd > > to af > > > 2, proto 1 (no corresponding listening socket) > > > Jan 10 16:46:56 [9199] ERROR:tm:t_forward_nonack: > failure to add > > > branches > > > > > > > > > > > > With comparition to that when the port is set to 5060 the > trace is : > > > > > > Jan 10 17:07:59 [9410] DBG:rr:find_next_route: No next > Route > > HF found > > > Jan 10 17:07:59 [9410] DBG:rr:after_loose: No next URI > found > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [ 10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197>] == > > > [ 10.11.57.197 <http://10.11.57.197> > <http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if port > > > 5060 matches port 5060 > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if > > > host==us: 12==12 && [10.11.57.197 > <http://10.11.57.197> <http://10.11.57.197> > > <http://10.11.57.197>] == > > > [10.11.57.197 <http://10.11.57.197> < > http://10.11.57.197> <http://10.11.57.197>] > > > Jan 10 17:07:59 [9410] DBG:core:grep_sock_info: > checking if port > > > 5060 matches port 5060 > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > flags=8000000 > > > Jan 10 17:07:59 [9410] DBG:core:parse_headers: > > flags=ffffffffffffffff > > > Jan 10 17:07:59 [9410] DBG:registrar:build_contact: > created > > Contact > > > HF: Contact: > > <sip:888@10.11.57.192:5061;transport=TLS>;expires=1000 > > > > > > > > > > > > And there is no fwd needed then.So the error didnt occur. > > > > > > Its a little bit strange that when I set the port to > 5061,why did > > > openser check the port 5060????? > > > Can anyone help me to figure it out? > > > THX > > > BR > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > Fengbin > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > <mailto:Users@lists.openser.org <mailto:Users@lists.openser.org>> > > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > -- > > Fengbin > > > > > > > -- > Fengbin

2 5

[OpenSER-Users] OpenSER 1.3 Presence Server
by Ross Beer 13 Jan '08

13 Jan '08

Hi All, I am trying to setup an OpenSER server with support for registration, messaging & presence however I am having a problem with the presence server. I have had presence working correctly before, however for some reason it is no longer working. I have spent about two days trying everything! I would really appreciate some advice with this. I have included my config below. Thank you in advance, Ross ------------------------------------------------------------------------------listen = udp:my.ip.address:5060mpath = "/usr/local/lib64/openser/modules"children = 8debug = 3fork = yesdisable_tcp = no#log_facility = LOG_DAEMONlog_facility = LOG_LOCAL0log_stderror = yestcp_children = 4mhomed = noserver_signature = yes#sock_group = "openser"#sock_mode = 0600#sock_user = "openser"#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 = yesloadmodule "usrloc.so"modparam("usrloc", "user_column", "username")modparam("usrloc", "domain_column", "domain")modparam("usrloc", "contact_column", "contact")modparam("usrloc", "expires_column", "expires")modparam("usrloc", "q_column", "q")modparam("usrloc", "callid_column", "callid")modparam("usrloc", "cseq_column", "cseq")modparam("usrloc", "methods_column", "methods")modparam("usrloc", "flags_column", "flags")modparam("usrloc", "user_agent_column", "user_agent")modparam("usrloc", "received_column", "received")modparam("usrloc", "socket_column", "socket")modparam("usrloc", "use_domain", 0)modparam("usrloc", "desc_time_order", 0)modparam("usrloc", "timer_interval", 60)modparam("usrloc", "db_url", "mysql://username:password@localhost/openser")modparam("usrloc", "db_mode", 1)modparam("usrloc", "matching_mode", 0)modparam("usrloc", "cseq_delay", 20)modparam("usrloc", "nat_bflag", 6)loadmodule "registrar.so"modparam("registrar", "default_expires", 3600)modparam("registrar", "min_expires", 60)modparam("registrar", "max_expires", 0)modparam("registrar", "default_q", 0)modparam("registrar", "append_branches", 1)modparam("registrar", "case_sensitive", 0)modparam("registrar", "received_param", "received")modparam("registrar", "max_contacts", 0)modparam("registrar", "retry_after", 0)modparam("registrar", "method_filtering", 0)modparam("registrar", "path_mode", 2)modparam("registrar", "path_use_received", 0)modparam("registrar", "received_avp", "$avp(i:801)")loadmodule "rr.so"modparam("rr", "enable_full_lr", 1)modparam("rr", "append_fromtag", 1)modparam("rr", "enable_double_rr", 1)modparam("rr", "add_username", 0)loadmodule "tm.so"modparam("tm", "fr_timer", 30)modparam("tm", "fr_inv_timer", 120)modparam("tm", "wt_timer", 5)modparam("tm", "delete_timer", 2)modparam("tm", "noisy_ctimer", 0)modparam("tm", "ruri_matching", 1)modparam("tm", "via1_matching", 1)modparam("tm", "unix_tx_timeout", 2)modparam("tm", "restart_fr_on_each_reply", 1)modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so"modparam("xlog", "buf_size", 4096)modparam("xlog", "force_color", 1)loadmodule "mi_fifo.so"modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")#modparam("mi_fif0", "fifo_name", "FIFO")modparam("mi_fifo", "fifo_mode", 0660)#modparam("mi_fifo", "fifo_group", "openser")#modparam("mi_fifo", "fifo_user", "openser")modparam("mi_fifo", "reply_dir", "/tmp/")modparam("mi_fifo", "reply_indent", "\t")loadmodule "domain.so"modparam("domain", "db_url", "mysql://username:password@localhost/openser")modparam("domain", "db_mode", 1)modparam("domain", "domain_table", "domain")modparam("domain", "domain_col", "domain")loadmodule "nathelper.so"modparam("nathelper", "natping_interval", 0)modparam("nathelper", "ping_nated_only", 1)modparam("nathelper", "rtpproxy_sock", "")modparam("nathelper", "rtpproxy_disable_tout", 60)modparam("nathelper", "rtpproxy_tout", 1)modparam("nathelper", "rtpproxy_retr", 5)modparam("nathelper", "sipping_method", "OPTIONS")modparam("nathelper", "received_avp", "$avp(i:801)")loadmodule "textops.so"loadmodule "mediaproxy.so"#modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock")modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock")modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients")modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients")modparam("mediaproxy", "natping_interval", 60)loadmodule "uri.so"loadmodule "sl.so"modparam("sl", "enable_stats", 1)loadmodule "maxfwd.so"modparam("maxfwd", "max_limit", 256)loadmodule "mysql.so"modparam("mysql", "ping_interval", 300)modparam("mysql", "auto_reconnect", 1)loadmodule "auth.so"modparam("auth", "nonce_expire", 300)modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes")modparam("auth", "rpid_avp", "$avp(s:rpid)")loadmodule "auth_db.so"modparam("auth_db", "db_url", "mysql://username:password@localhost/openser")modparam("auth_db", "user_column", "username")modparam("auth_db", "domain_column", "domain")modparam("auth_db", "password_column", "password")#modparam("auth_db", "password_column_2", "ha1b")modparam("auth_db", "calculate_ha1", 1)modparam("auth_db", "use_domain", 0)modparam("auth_db", "load_credentials", "$avp(s:caller_uuid)=id")loadmodule "uri_db.so"modparam("uri_db", "db_url", "mysql://username:password@localhost/openser")modparam("uri_db", "uri_table", "uri")modparam("uri_db", "uri_user_column", "username")modparam("uri_db", "uri_domain_column", "domain")modparam("uri_db", "uri_uriuser_column", "uri_user")modparam("uri_db", "subscriber_table", "subscriber")modparam("uri_db", "subscriber_user_column", "username")modparam("uri_db", "subscriber_domain_column", "domain")modparam("uri_db", "use_uri_table", 0)modparam("uri_db", "use_domain", 0)loadmodule "avpops.so"modparam("avpops", "db_url", "mysql://username:password@localhost/openser")modparam("avpops", "avp_table", "usr_preferences")modparam("avpops", "use_domain", 0)modparam("avpops", "uuid_column", "id")modparam("avpops", "username_column", "username")modparam("avpops", "domain_column", "domain")modparam("avpops", "attribute_column", "attribute")modparam("avpops", "value_column", "value")modparam("avpops", "type_column", "type")loadmodule "enum.so"modparam("enum", "domain_suffix", "e164.org.")loadmodule "permissions.so"#modparam("permissions", "default_allow_file", "permissions.allow")#modparam("permissions", "default_deny_file", "permissions.deny")#modparam("permissions", "check_all_branches", 1)#modparam("permissions", "allow_suffix", ".allow")#modparam("permissions", "deny_suffix", ".deny")modparam("permissions", "db_url", "mysql://username:password@localhost/openser")modparam("permissions", "db_mode", 1)modparam("permissions", "trusted_table", "trusted")modparam("permissions", "source_col", "src_ip")modparam("permissions", "proto_col", "proto")modparam("permissions", "from_col", "from_pattern")modparam("permissions", "tag_col", "tag")modparam("permissions", "peer_tag_avp", "$avp(i:707)")loadmodule "lcr.so"modparam("lcr", "db_url", "mysql://username:password@localhost/openser")modparam("lcr", "gw_table", "gw")modparam("lcr", "gw_name_column", "gw_name")modparam("lcr", "ip_addr_column", "ip_addr")modparam("lcr", "port_column", "port")modparam("lcr", "uri_scheme_column", "uri_scheme")modparam("lcr", "transport_column", "transport")modparam("lcr", "grp_id_column", "grp_id")modparam("lcr", "lcr_table", "lcr")modparam("lcr", "strip_column", "strip")modparam("lcr", "prefix_column", "prefix")modparam("lcr", "from_uri_column", "from_uri")modparam("lcr", "priority_column", "priority")modparam("lcr", "gw_uri_avp", "$avp(i:709)")modparam("lcr", "ruri_user_avp", "$avp(i:500)")modparam("lcr", "contact_avp", "$avp(i:711)")modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)")modparam("lcr", "fr_inv_timer", 90)modparam("lcr", "fr_inv_timer_next", 30)modparam("lcr", "rpid_avp", "$avp(i:302)")modparam("lcr", "dm_flag", 25) loadmodule "alias_db.so"modparam("alias_db", "db_url", "mysql://username:password@localhost/openser")modparam("alias_db", "user_column", "username")modparam("alias_db", "domain_column", "domain")modparam("alias_db", "alias_user_column", "alias_username")modparam("alias_db", "alias_domain_column", "alias_domain")modparam("alias_db", "use_domain", 0) # ----- presence params -----loadmodule "presence.so"loadmodule "presence_xml.so"loadmodule "presence_mwi.so"loadmodule "pua.so"loadmodule "pua_mi.so"loadmodule "pua_usrloc.so"modparam("presence|presence_xml", "db_url","mysql://username:password@localhost/openser")modparam("presence_xml", "force_active", 1)modparam("presence", "server_address", "sip:my.ip.address:5060")modparam("presence", "presentity_table", "presentity")modparam("presence", "active_watchers_table", "active_watchers")modparam("presence", "watchers_table", "watchers")modparam("presence", "clean_period", 100)modparam("presence", "max_expires", 3600)modparam("presence", "fallback2db", 1)modparam("presence_xml", "xcap_table", "xcap")modparam("presence_xml", "pidf_manipulation", 1)modparam("presence_xml", "integrated_xcap_server", 1)modparam("pua_usrloc", "default_domain", "my.domain.com")modparam("presence", "to_tag_pref", 'pres') ######################################################################### 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|MESSAGE")){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();setbflag(6);}route(7);}if(is_method("REGISTER")){route(6);}if(is_method("INVITE")){route(8);}if(is_method("PUBLISH|SUBSCRIBE")){route(17);}if(is_method("CANCEL") || is_method("ACK")){route(14);}############################# MESSAGES #############################if(is_method("MESSAGE")){route(18);}route(15);}######################################################################### 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");}######################################################################### 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");}######################################################################### 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=$ci\n");# load caller avpsavp_db_load("$avp(s:caller_uuid)", "*");}######################################################################### 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=$ci\n");# load callee avpsavp_db_load("$avp(s:callee_uuid)", "*");}######################################################################### Request route 'stop-media-proxy'########################################################################route[5]{if(isflagset(22)){end_media_session();}}######################################################################### Request route 'base-route-register'########################################################################route[6]{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[7]{xlog("L_INFO", "ROUTE 7 RUN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");if(isbflagset(6)){if(!isflagset(22) && !search("^Content-Length:[ ]*0")){setflag(22);use_media_proxy();}t_on_reply("2");}else{t_on_reply("1");}if(!isflagset(21)){t_on_failure("2");}if(isflagset(29)){append_branch();}if(is_present_hf("Proxy-Authorization")){consume_credentials();}xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# no 100 (we already sent it) and no DNS blacklistingif(!t_relay("0x05")){sl_reply_error();if(is_method("INVITE") && isbflagset(6)){end_media_session();}}exit;}######################################################################### Request route 'base-route-invite'########################################################################route[8]{sl_send_reply("100", "Trying");if(from_gw()){$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("$avp(s:peer_uuid)/s")){# use tag-column from trusted-table as uuid for this calleravp_copy("$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;# }}}if(nat_uac_test("19")) {fix_nated_contact();setbflag(6);}route(9);}######################################################################### Request route 'invite-find-callee'########################################################################route[9]{if(alias_db_lookup("dbaliases")){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(11);}avp_delete("$avp(s:callee_uuid)");avp_db_query("select id from subscriber where username = '$rU'", "$avp(s:callee_uuid)");if(is_avp_set("$avp(s:callee_uuid)")){xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(10);}else{$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(11);}exit;}######################################################################### Request route 'invite-to-internal'########################################################################route[10]{route(4);if(!lookup("location")) {xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");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(7);}exit;}######################################################################### Request route 'invite-to-external'########################################################################route[11]{if(isflagset(20)){xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(7);exit;}route(13);if(!isflagset(23)){# don't allow calls relaying from PSTN to PSTN, if not explicitely forwardedif(uri =~ "^sip:[0-9]+@"){# only route numeric users to PSTNif(!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(7);}}xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("404", "User Not Found");exit;}######################################################################### Request route 'normalize-e164'########################################################################route[12]{# European numbering plans look like this:# CC = country code (i.e. 43 for Austria)# NDC = national destination code (i.e. 1 for Vienna)# SN = subscriber number (i.e. 4001234)# # CC + NDC + SN, i.e. 4314001234## Within the same CC+NDC, it can be omitted, so if# +4314001234 wants to call +4315002345, one can dial# just 5002345.## Within the same CC, CC can be ommitted and a "0" is prefixed# to NDC, so if +4314001234 wants to call +4326003456, # one can dial 026003456.## For international calls, either "00" or + is prefixed, like# +49123456780 or 0049123456789.#avp_delete("$avp(s:orig_callee_user)/g");$avp(s:orig_callee_user) = $rU;if(uri =~ "^sip:(\+[1-9])?[0-9]+@"){# looks like a PSTN numberif(uri =~ "^sip:0[1-9][0-9]+@"){# we have format 0+NDC+SNstrip(1);prefix("+44");}else if(uri =~ "^sip:00[1-9]+@"){# we have format 00 + CC + NDC + SNstrip(2);prefix("+");}else if(!uri =~ "^sip:\+[1-9][0-9]+@"){# unknown format, maybe NDC wasn't added before?xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(-1);}# else we have "+" + CC + NDC + SNxlog("L_INFO", "Normalized callee '$avp(s:orig_callee_user)' to E.164 format '$rU' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(1);}else{xlog("L_INFO", "Not normalized callee '$avp(s:orig_callee_user)' to E.164 format - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");return(-1);}}######################################################################### Request route 'lookup-enum'########################################################################route[13]{route(12);if($rc == 1){if(enum_query("e164.org.")){# TODO: do GW fallback (load gws, set failure-route)?xlog("L_INFO", "ENUM query succeeded - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(7);exit;}else{xlog("L_INFO", "ENUM query failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# ENUM query failed, revert $rUavp_pushto("$ru/username", "$avp(s:orig_callee_user)");}}}######################################################################### Request route 'base-route-local'########################################################################route[14]{t_on_reply("1");if(t_check_trans()){xlog("L_INFO", "Request leaving server - M=$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=$ci\n");}exit;}######################################################################### Request route 'base-route-generic'########################################################################route[15]{xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");sl_send_reply("501", "Method Not Supported Here");exit;}######################################################################### Request route 'base-filter-failover'########################################################################route[16]{if(!t_check_status("408|500|503")){xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(5);exit;}}########################################################################## Presence#########################################################################route[17]{# if(nat_uac_test("19") || search("^Route:.*;nat=yes"))# {# fix_nated_contact();# setbflag(6);# } if (!t_newtran()){sl_reply_error();exit;};#if(is_method("PUBLISH"))#{# xlog("L_INFO", "PUBLISH - Process Publish Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");# handle_publish();# t_release();#}if(is_method("PUBLISH")){if($hdr(Sender)!= NULL)handle_publish("$hdr(Sender)");elsehandle_publish();t_release();}if(is_method("SUBSCRIBE")){handle_subscribe();xlog("L_INFO", "SUBSCRIBE - Process Subscribe Request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");t_release();}exit;}######################################################################### SEND MESSAGES########################################################################route[18]{if (!lookup("location")) {switch ($retcode) {case -1:case -3:t_newtran();t_reply("404", "Not Found");exit;case -2:sl_send_reply("405", "Method Not Allowed");exit;}}if (!t_relay()) {sl_reply_error();};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;}######################################################################### Failure route 'pstn-failover'########################################################################failure_route[1]{xlog("L_INFO", "Failure route for PSTN entered - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(16);if(!next_gw()){xlog("L_ERR", "Failed to select next PSTN gateway - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");route(5);exit;}t_on_failure("1");route(7);}######################################################################### Failure route 'base-standard-failure'########################################################################failure_route[2]{route(16);route(5);} _________________________________________________________________ Get Hotmail on your mobile, text MSN to 63463! http://mobile.uk.msn.com/pc/mail.aspx

1 0

[OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment
by Adrian Georgescu 13 Jan '08

13 Jan '08

>>> I would _really_ recommend to add this to the installations instructions on openxcap.org. Have a nice weekend! Sebastian Done. We will setup a read/write page for OpenXCAP deployment experiences. Regards, Adrian

1 0

[OpenSER-Users] Script variable duration is unexpected
by Iñaki Baz Castillo 13 Jan '08

13 Jan '08

Hi, I've been suffering an extrange issue and now I've realized why. It's related to script variables and their duration. I'm 90% sure that the behaviour is unexpected according to the doc so I've open a bug containing an easy example: https://sourceforge.net/tracker/index.php?func=detail&aid=1870301&group_id=… My conclusion is that using script variables instead of AVP's can be risky in certain cases. Opinions are welcome. Best regards. -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment
by Schumann Sebastian 12 Jan '08

12 Jan '08

Dear all I finally found the mistake: My VMWare ESX server for virtualization is not connected to the internet, my local PC was. It didn't come to my mind ever before. Although - I never saw it documented anywhere, that OpenXCAP needs internet access for downloading the namespace files. I would _really_ recommend to add this to the installations instructions on openxcap.org. Have a nice weekend! Sebastian ________________________________ From: Schumann Sebastian Sent: Friday, 11. January 2008 11:54 AM To: users(a)openser.org Subject: [OpenSER-Users] Strange OpenXCAP behaviour in virtualizedenvironment Dear all This week, I tried again to bring up finally the OpenXCAP server. I tried it first acc. the instruction on OpenXCAP.org on my local virtual VMWare Server 1.0 (before, I always used virtualized ESX server). It worked without major issues, meaning I could install lenny, all required files even via apt-get and the OpenXCAP was running. Surprising for me though, because it never worked yet. Now, when I take _the same_ virtual disk and export it to VMWare ESX server, I get the previously discussed errors: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema How can this be possible? I tried three times with three different machines. They work in VMware Server but not in ESX server. I don't find any answer at all neither can I imagine anyhow, how an application error like this occurs running the program on the very same, identical virtualized system with just two different virtualization platforms around. Anybody else has experiences in that, encountered somehow same strange behaviour with the application or has any other advice? I would really welcome every hint in somehow a direction to solve this. I didn't plan to extend my virtual OpenSER testing platform with a local PC, but it seems I have no other possibility so far... Thanks for your help! A nice weekend to all of you. Best regards Sebastian

2 1

[OpenSER-Users] DB aliases for outgoing calls
by Marc LEURENT 11 Jan '08

11 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Evening, I have a little question: I have users with usernames and 10 phone numbers from my SIP/PSTN provider. I have a dbaliases table for incoming calls, in order to redirect the 0xxxxxxxxx number to an user. 0xxxxxxxx1 => bob I would like for outgoing calls to replace the user by the telephone number in the From: Header..... bob => 0xxxxxxxx1 Any idea how to do it? Thanks, Marc Leurent -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHh6ssN4+o+2LtdFwRApxIAKCxqefV7OLejg1Xw6mwre5Yy3jYTQCeM66R bAjagfV/6aoN43hPQcMeKMM= =DQe8 -----END PGP SIGNATURE-----

2 2

[OpenSER-Users] Fix up source address.
by Alex Balashov 11 Jan '08

11 Jan '08

Greetings, I have a strange problem using OpenSER 1.3.x with nathelper. Two ethernet interfaces: eth0 = 192.168.0.0/24 eth1 = outside.ip/29 For some reason, no matter what I do to mangle the requests with nathelper's functions, the packet is *always* sent out of eth1 with the *source address* of the machine's eth0. Obviously, the response from the far-end SIP peer never gets back. The packet does physically go over eth1, I know that much from packet captures. I don't even see how this is possible; when OpenSER issues a packet, shouldn't it originate according to the machine's routing table, take the most specific route, and consequently, adopt the right source address? What gives? I have OpenSER 'listening' on both interfaces, and have tried both on and off with this setting. Thanks, -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

3 5

[OpenSER-Users] RTP-Proxy
by Helmut Kuper 11 Jan '08

11 Jan '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, does anybody know, if openser+rtpproxy are able to detect rtp-timeouts and react on them? It would be good for billing. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHhjWa4tZeNddg3dwRAmZcAJkBmJGh6b2qt/6NbFri0mBZU9BwPQCeNYNS Pb+u2IWB4Xp2K8G9fHkQKyo= =MxAh -----END PGP SIGNATURE-----

9 19

[Serusers] how to replace ruri host part with an avp value?
by Tomasz Zieleniewski 11 Jan '08

11 Jan '08

Hi, How can one replace the ruri host part with the usage of attr2uri method? One can change the domain part with the usage of "domain" param but this leaves the port not changed. I would like to replace the whole host part together with port value. Maybe this can be done other way round? Thanks in advance for any help. Bests -tomasz

2 2

[OpenSER-Users] OpenXCAP: rls-services 404 Not found,and no log?
by KevinKinnan 11 Jan '08

11 Jan '08

I tried example of rfc 4825. Creating resource-lists is ok, but not rls-services, 404 not found response is received. And, there is not any new log appearing in the syslog, I can't debug where I am wrong. HTTP URI is: http://xcap-root/rls-services/users/sip:alice@open-ims.test/index XML is: <?xml version=\"1.0\" encoding=\"UTF-8\"?> <rls-services xmlns=\"urn:ietf:params:xml:ns:rls-services\"> <service uri=\"sip:myfriends@open-ims.test\"> <resource-list>http://xcap-root/resource-lists/users/sip:alice@open-ims.test/index/~~/reso…</resource-list> <packages> <package>presence</package> </packages> </service> </rls-services> Regards, Kevin _________________________________________________________________ 手机也能上 MSN 聊天了，快来试试吧！ http://mobile.msn.com.cn/

2 1

[OpenSER-Users] Strange OpenXCAP behaviour in virtualized environment
by Schumann Sebastian 11 Jan '08

11 Jan '08

Dear all This week, I tried again to bring up finally the OpenXCAP server. I tried it first acc. the instruction on OpenXCAP.org on my local virtual VMWare Server 1.0 (before, I always used virtualized ESX server). It worked without major issues, meaning I could install lenny, all required files even via apt-get and the OpenXCAP was running. Surprising for me though, because it never worked yet. Now, when I take _the same_ virtual disk and export it to VMWare ESX server, I get the previously discussed errors: Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent call last): Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/bin/openxcap", line 56, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server import XCAPServer Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import authentication Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage import getApplicationForURI Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, in ? Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = {'xcap-caps': XCAPCapabilitiesApplication(), Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, in __init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = etree.XMLSchema(xml_schema_doc) Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", line 67, in etree.XMLSchema.__init__ Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: Document is not valid XML Schema How can this be possible? I tried three times with three different machines. They work in VMware Server but not in ESX server. I don't find any answer at all neither can I imagine anyhow, how an application error like this occurs running the program on the very same, identical virtualized system with just two different virtualization platforms around. Anybody else has experiences in that, encountered somehow same strange behaviour with the application or has any other advice? I would really welcome every hint in somehow a direction to solve this. I didn't plan to extend my virtual OpenSER testing platform with a local PC, but it seems I have no other possibility so far... Thanks for your help! A nice weekend to all of you. Best regards Sebastian

1 0

[OpenSER-Users] Testing.... List Working?
by Douglas Garstang 11 Jan '08

11 Jan '08

Is the list working? Haven't seen much traffic on list today. First post was ok. Second went to list and archive, but I never received it via email. Third has gone nowhere! This is my 4th... ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users