- sr-users - kamailio.org

[OpenSER-Users] AVP Setup
by Ali Jawad 06 Jun '08

06 Jun '08

Dear All I am trying to do the following: I want to enable/disable users, to do that I added a column named status to the subscribers table. It contains a string saying enabled or disabled. I did add the following info to the config file. # ----- AVP params -------- modparam("avpops", "avp_url", "mysql://openser:0o0o0o0o0@xx.xx.xx.xx/openser") modparam("avpops","avp_table","subscriber") modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=status;value_type=string;table= subscriber") And in the route statement I added "This is a sample provided by MPWizard..thanks for that" however it apparently needs some changes to work for me. if (avp_db_load("$tU", "$avp(s:disabled)")) { xlog("L_INFO", "User is local but disabled, denying registration") } The error I get in the log file is as follows: Jun 6 11:20:38 [21919] ERROR:mysql:db_mysql_submit_query: driver error: Unknown column 'value' in 'field list' Jun 6 11:20:38 [21919] ERROR:mysql:db_mysql_query: error while submitting query Jun 6 11:20:38 [21919] ERROR:avpops:ops_dbload_avps: db_load failed Any suggestions please. Thanks to all of you guys. For all the support now and before ..you rock.

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Re: [OpenSER-Users] Openser Multi-home
by Bogdan-Andrei Iancu 06 Jun '08

06 Jun '08

Note that there are two way to configure openser in multi-home envs. : - either enable mhome param and openser will autodetect the outbound interface based on the destination - quite slow - either use force_send_socket() function if the msg flow is clear and you know from the script what interface should be used - very fast Regards, Bogdan jvelev(a)avangardsolutions.com wrote: > That fixed the problem. 10x > > Regards, > J.Velev > > On Thu, 2008-06-05 at 20:11 +0300, Bogdan-Andrei Iancu wrote: > >> Hi, >> >> have you enabled the mhomed param? >> http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#mhomed >> >> Regards, >> Bogdan >> >> jvelev(a)avangardsolutions.com wrote: >> >>> Hello all >>> >>> I'm trying to run OpenSER on a multi-home environment. OpenSER listens >>> on the real IP and on the private one. The main idea is to proxy traffic >>> coming from real IPs to the internal network. >>> >>> OpenSER tries to reach the internal IPs talking to them with it's REAL >>> IP despite it's directly connected to the internal network and can use >>> it's private IP. >>> >>> Comments & Ideas appreciated. >>> >>> >>> >>> >>> >>> >> > > >

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[OpenSER-Users] CDRTool: How to avoid invoices for answered calls of duration 0
by Iñaki Baz Castillo 06 Jun '08

06 Jun '08

Hi, RFC3261 says that a UAC should send inmediately a BYE after a 2XX response if the offer in the 2xx response is not acceptable (when the INVITE has no body). So this will generate an answered call of duration 0 (probably). But if the CDRTool rate has "connect" fixed amount then this invalid call will have cost. I think this shouldn't have cost since it's a SIP protocol issue (accepting a call and sending a BYE inmediately) but not an human decision. Opinions? 13.2.2.4 2xx Responses -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] [OpenSER-Devel] Memory dump for OpenSER 1.3.1 which slowly leaks memory
by Julian Yap 06 Jun '08

06 Jun '08

Bogdan, I'll need to check out the shmem statistics. I mean, this is the memory output for the OpenSER processes having run for about 24 hours. Does it look right?: # ps -C openser -o vsz,rss VSZ RSS 234124 12636 369940 159392 369940 158840 369940 158860 370340 159220 369804 158696 370068 158932 370204 159048 370464 159344 234124 20904 234124 9484 234128 7952 234124 7948 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 8332 Here are the modules I'm using. Not sure if any have a dependency on libxml: loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "usrloc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "maxfwd.so" loadmodule "mi_fifo.so" loadmodule "nathelper.so" loadmodule "perl.so" loadmodule "registrar.so" loadmodule "rr.so" loadmodule "textops.so" loadmodule "uri.so" Thanks, Julian On Wed, May 7, 2008 at 5:22 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi Julian, > > Have you tried to use the the shmem statistics to monitor the usage of SHM > memory? > > Both for PKG and SHM memory, openser allocs at startup the whole memory > (depending of how much you configured). Any leak will not be vizible for any > external tools. > > Also, are you using any module that have dependency of libxml library? > > The dump you posted shows more or less nothing more than init phase - > before sending the signal, be sure you select a worker process (use fifo > command "ps" to get the list and types of processes). > > Regards, > Bogdan > > Julian Yap wrote: > > > > > > > > > On Mon, May 5, 2008 at 10:25 PM, Henning Westerholt > > <henning.westerholt(a)1und1.de> wrote: > > > > > > > On Saturday 03 May 2008, Julian Yap wrote: > > > > I'm currently running OpenSER 1.3.1. > > > > > > > > It seems to slowly leaks memory. > > > > > > > > I've recompiled it as per for debug support in the memory manager: > > > > http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > > > > > > > > Here is the memory manager dump of the memory status: > > > > http://julianyap.com/download/openser1.3.1_memory_dump20080502.txt > > > > > > > > Any suggestions would be very welcome. > > > > > > Hi Julian, > > > > > > can you give a little more details about your problem? Why do you think > it > > > leaks memory? Do you get memory allocation errors in the logs? > > > > > > Cheers, > > > > > > Henning > > > > > > > > > > Henning, > > > > I don't get any memory allocation errors. > > > > I've recently updated to OpenSER 1.3 branch SVN r4107 but the 'leak' > > still remains. > > > > On a 4GB box, free memory just seems to deplete. > > > > I have a cron job which restarts OpenSER every night but it seems like > > not all the memory is free'd up. Eventually the box needs to be > > rebooted. In the past, running out of memory has led to oom-killer > > killing OpenSER. > > > > For example, I have this cron job: > > */5 * * * * root echo "`date` `free | grep 'buffers/cache'`" >> > > /var/log/free_memory.log > > > > Memory just depletes. Here's where OpenSER was restarted around midnight: > > Sun May 4 23:10:01 HST 2008 -/+ buffers/cache: 2014972 2095204 > > Sun May 4 23:15:01 HST 2008 -/+ buffers/cache: 2027184 2082992 > > Sun May 4 23:20:01 HST 2008 -/+ buffers/cache: 2027604 2082572 > > Sun May 4 23:25:01 HST 2008 -/+ buffers/cache: 2038064 2072112 > > Sun May 4 23:30:01 HST 2008 -/+ buffers/cache: 2037868 2072308 > > Sun May 4 23:35:01 HST 2008 -/+ buffers/cache: 2053184 2056992 > > Sun May 4 23:40:01 HST 2008 -/+ buffers/cache: 2055216 2054960 > > Sun May 4 23:45:01 HST 2008 -/+ buffers/cache: 2075612 2034564 > > Sun May 4 23:50:01 HST 2008 -/+ buffers/cache: 2103232 2006944 > > Sun May 4 23:55:01 HST 2008 -/+ buffers/cache: 2114384 1995792 > > Mon May 5 00:00:01 HST 2008 -/+ buffers/cache: 2117320 1992856 > > Mon May 5 00:05:01 HST 2008 -/+ buffers/cache: 860304 3249872 > > Mon May 5 00:10:01 HST 2008 -/+ buffers/cache: 861208 3248968 > > Mon May 5 00:15:01 HST 2008 -/+ buffers/cache: 868968 3241208 > > > > Here's where I just restarted OpenSER: > > Mon May 5 23:30:01 HST 2008 -/+ buffers/cache: 2041740 2068436 > > Mon May 5 23:35:01 HST 2008 -/+ buffers/cache: 2049572 2060604 > > Mon May 5 23:40:01 HST 2008 -/+ buffers/cache: 2054856 2055320 > > Mon May 5 23:45:01 HST 2008 -/+ buffers/cache: 2065316 2044860 > > Mon May 5 23:50:01 HST 2008 -/+ buffers/cache: 2062248 2047928 > > Mon May 5 23:55:01 HST 2008 -/+ buffers/cache: 2076148 2034028 > > Tue May 6 00:00:01 HST 2008 -/+ buffers/cache: 2087256 2022920 > > Tue May 6 00:05:01 HST 2008 -/+ buffers/cache: 2091300 2018876 > > Tue May 6 00:10:01 HST 2008 -/+ buffers/cache: 2093244 2016932 > > Tue May 6 00:15:01 HST 2008 -/+ buffers/cache: 1382632 2727544 > > > > Thanks, > > Julian > > > > _______________________________________________ > > Devel mailing list > > Devel(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/devel > > > > > > > >

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Re: [OpenSER-Users] Terminate a call from script
by Bogdan-Andrei Iancu 06 Jun '08

06 Jun '08

Hi David, But this is not from script. Is is from outside openser. Regards, Bogdan David Villasmil wrote: > I've done that... > > Using the dialog module, and getting the hash_id and hash_entry, you can: > > :dlg_end_dlg:_reply_fifo_file_ > 342 > 56 > _empty_line_ > > and that ends the call. > > On Thu, Jun 5, 2008 at 3:17 PM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > No, there is no way to terminate the call directly from the script > - at > least not so far :). > > Regards, > Bogdan > > kionez wrote: > > Hi all, > > I'm trying to add a simple priority mechanism in my openser 1.3 > script. > > > > Reading docs i found that dialog module gives a MI command to > terminate > > a call, dlg_end_dlg. Is there any way to terminate a call within a > > script (or a perl script)? I can't find anything on the web, the > only > > way i found is to exec something like "openserctl fifo > dlg_end_dlg $hash > > $id", but it's slower than a direct function inside the main > script.. > > > > thanks in advance > > > > k. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

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Re: [OpenSER-Users] Openser-Perl module in onreply_route
by Mik Cheez 06 Jun '08

06 Jun '08

The 'perl' module has been a godsend for me, but the issue I'm having still has to do with the memory leak in the perl DBI module; my question is, has anyone considered using the 'perl' framework for other scripting languages? Python and PHP would be terrific, in my opinion. As I have things now, I can lose up to 1.5MB/minute using the DBI module (during heavy usage). This is in reference to a posting made back in October 2007: http://www.mail-archive.com/users@openser.org/msg13387.html Best regards, Mik

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[OpenSER-Users] New OpenSER-mi-proxy release 1.0.0
by Adrian Georgescu 06 Jun '08

06 Jun '08

Hello, There is a new component useful for implementing the presence server solution using OpenSER and OpenXCAP. OpenSER MI Proxy ---------------- OpenSER MI Proxy is designed to allow remote XML-RPC clients to issue MI commands to OpenSER. This proxy replaces the need for OpenSER built-in xml-rpc server, which has no access list and is prone to crash the server. One of its purposes is to allow a remote OpenXCAP server to safely call the refreshWatchers() command on the management interface of the SIP Presence server when the presence policy document has changed. The proxy can listen on custom IP addresses and provides an access list for the clients allowed to connect. The software can be downloaded from: http://download.ag-projects.com/OpenXCAP/ Kind regards, Adrian Georgescu

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[OpenSER-Users] how to reload openser.cfg
by Qi Fan 06 Jun '08

06 Jun '08

Hello OperSER users, This is my first post here. ;-) Can anyone tell me know how to load the modified openser.cfg on the fly? Is there any command like "extensions reload" in Asterisk? Many thanks for your help. Regards, Qi Fan

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[OpenSER-Users] Openser Multi-home
by jvelev＠avangardsolutions.com 05 Jun '08

05 Jun '08

Hello all I'm trying to run OpenSER on a multi-home environment. OpenSER listens on the real IP and on the private one. The main idea is to proxy traffic coming from real IPs to the internal network. OpenSER tries to reach the internal IPs talking to them with it's REAL IP despite it's directly connected to the internal network and can use it's private IP. Comments & Ideas appreciated. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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[OpenSER-Users] OT: TLS mandatory for proxies?
by Klaus Darilion 05 Jun '08

05 Jun '08

Hi! The TLS docs say that TLS is mandatory for SIP proxies. I tried to find the relevant text in the RFC but could not find it. Does somebody knows if it is really mandatory and where it is described? thanks klaus

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[OpenSER-Users] Doesn't "proxy_authorize()" allow credentials if "From" == "Anonymous" <sip:anonymous@anonymous.invalid> ?
by Iñaki Baz Castillo 05 Jun '08

05 Jun '08

Hi, when I do a call with hiddn callerid from my Twinkle it sends an INVITE with: ----------- INVITE sip:1414@my_domain.net SIP/2.0 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=emirn Privacy: id P-Preferred-Identity: "IBC" <sip:ibc@my_domain.net> ----------- and of course my Openser asks Twinkle for authentication: ----------- SIP/2.0 407 Proxy Authentication Required Proxy-Authenticate: Digest realm="my_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", qop="auth" ----------- and I see the $rc code of "proxy-authentication()" is -4 (no credentials). So my Twinkle sends a correct INVITE (CSeq + 1 and "Proxy-Authorization" header): ----------- INVITE sip:1414@my_domain.net SIP/2.0 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=emirn Privacy: id P-Preferred-Identity: "IBC" <sip:ibc@my_domain.net> Proxy-Authorization: Digest username="ibc", realm="mt_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", uri="sip:1414@my_domain.net", response="061d82967778669a01e53d168f2da93b", algorithm=MD5, cnonce="9n7ln49l46", qop=auth, nc=00000001 ----------- But "proxy-authentication()" replies again with -4 (no credentials). ----------- SIP/2.0 407 Proxy Authentication Required Proxy-Authenticate: Digest realm="my_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", qop="auth" ----------- Why? Obviously the INVITE **does include** Proxy-Authorization. Maybe RC3261 says that a server must not challenge for authentication a user with From "Anonymous" <sip:anonymous@anonymous.invalid> ? Of course authentication works properly when not hidding callerid. Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] t_replicate function`s doubt??
by Marcio 05 Jun '08

05 Jun '08

Hi Zahid, Thanks for help, maybe you can help me....I have an environment of test with more than 2 proxies....A UAc of a domain visiting another domain.....wanted him to succeed in registering domain in place and had home domain pointing to the local domain(visited), the case reached a "request" to this UAc....would be a distributed a sketch of records that follow where the UAc is. The idea to use t_replicate() function, i did think that could resolve this question, but now i think not.....because the home proxy of the UAc not know the location of your UAc, would have to be informed at the time of replication, such as by the validation credentials into table "subscriber", could insert the location into table "location"....??? It would have some idea that could help me? best regards, Marcio 2008/6/5, Zahid Mehmood <zm23(a)columbia.edu>: > > Marcio, > the way we use t_replicate is as follows. > > UAC -- register ---> proxy1 > > proxy1 code > > { > > if (src_ip == ip_of_proxy_2 ) { > # do not challenge because we assume that proxy2 did the > authenticate already > # save location and any other thing you want to do > } else { > # do challenge / authenticate here > # save location > # t_replicate to proxy2 > } > > proxy2 code > > { > > if (src_ip == ip_of_proxy_1 ) { > # do not challenge because we assume that proxy2 did the > authenticate already > # save location and any other thing you want to do > } else { > # do challenge / authenticate here > # save location > # t_replicate to proxy2 > } > > > Hope this helps. > > -- > Zahid > > > > On Jun 4, 2008, at 4:10 PM, Marcio wrote: > > Hi, >> >> Someone would know if t_replicate() function....perform a t_relay() of the >> REGISTER....and carry the "credentials" of the UAc to slave or remote UAs? >> >> And the reply of the UAs(remote or slvae) is "200, ok" to UAs (fork`s >> origin) or not a reply?? >> >> best regards, >> Marcio >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > >

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[OpenSER-Users] smart stund
by Pezhman Lali 05 Jun '08

05 Jun '08

Dear, some sip-phones behind the symmetric nat, can not be registered, if the ip of stun server was filled ,in the setting of sip-phone. I am looking for a way,that all of sip-phones, with the same setting , work properly. may be a solution, the stun server doesn't reply to the request behind the symmetric request, what is your idea? best Mani

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[OpenSER-Users] 6xx handling - new options
by Bogdan-Andrei Iancu 05 Jun '08

05 Jun '08

Hi, Following a several discussions on devel + users and sip-implementers mailing lists, I just added a new module parameter in TM to control how the 6xx replies are handled: -(0) the 6xx replies will block any further serial forking (adding new branches). This is the RFC3261 behaviour. -(1) the 6xx replies will be handled as any other negative reply - serial forking will be allowed. Logically, you need to break RFC3261 if you want to do redirects to announcement and voicemail services. Again, this is more a logical debate (about the 6xx replies).See: https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-June/019413.h… and this new param was added to let you choose between: - following RFC, but not being able to implement some scenarios - breaking RFC, but being more practical ;) Best regards, Bogdan

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[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 05 Jun '08

05 Jun '08

I just modified the relevnt source files such as notify.c and subscribe.c. It works now :) Kevin _________________________________________________________________ 用手机MSN聊天写邮件看空间，无限沟通，分享精彩！ http://mobile.msn.com.cn/

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[OpenSER-Users] Terminate a call from script
by kionez 05 Jun '08

05 Jun '08

Hi all, I'm trying to add a simple priority mechanism in my openser 1.3 script. Reading docs i found that dialog module gives a MI command to terminate a call, dlg_end_dlg. Is there any way to terminate a call within a script (or a perl script)? I can't find anything on the web, the only way i found is to exec something like "openserctl fifo dlg_end_dlg $hash $id", but it's slower than a direct function inside the main script.. thanks in advance k.

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Re: [OpenSER-Users] srv load-balancing
by Stanisław Pitucha 05 Jun '08

05 Jun '08

----- "Sajith T S" <sajith(a)gmail.com> wrote: > stanis(a)zimbra-1.gradwell.net wrote: > > > and got 6 calls on asterisk-2 and none on asterisk-1. If I change > > the order, all calls go to the first host on the list. Shouldn't > > openser do load-balancing in this scenario? I don't use any options > > in t_relay(). > > You should use openser dispatcher module for this. Yes... I've just noticed in http://www.openser.org/pipermail/users/2007-October/013660.html that openser basically decided to go against SRV RFC and ignores weights. (RFC says that clients should sort and select from list of servers, not servers as B.A.I. suggested if I understood that correctly) Anyways - about dispatcher: I couldn't find any description about it's behavior with call transfers / taking over. I understand that while I'm using callid-based hash there will be no problems with REFER, because call-id stays the same and each leg will go to the same server. But what about INVITE + Replaces: ? I don't really see a way for it to work just with hashing. Has anyone tested this / has solution? PS. sorry Sajith, I replied to you before by mistake Thanks, Stan

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[OpenSER-Users] openser 1.3.0 config
by Mark Quitoriano 05 Jun '08

05 Jun '08

HI All, Im new to openser and trying to config it following this howto http://www.openser.org/dokuwiki/doku.php/install:openser-from-svn i get errors when running the database setup script. # /usr/local/sbin/openserdbctl create ERROR: could not load the script in /usr/local/lib64/openser/openserctl/openserdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' I'm usng Centos 5 btw.

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[OpenSER-Users] need help on Free Radius & CDR Tool
by Do Nguyen Ha 05 Jun '08

05 Jun '08

hi i manage the Cisco BTS 10200 system, and now i am assigned a task to work with the Radius i choose the FreeRadius to work with Cisco BTS system the attached file are message flow & message's structure of account start modify some config in radiusd.conf & clients.conf first, BTS send message Access Request to FreeRadius and then Freeradius send message Access Accept, they working properly when BTS send message Account Start to Radius, there are a lot attribute and freeradius save them in the radacct table but in radacct , there some columm doesn't match with the message Account Start, i want to change the name of some columm to match the message Should i need to edit the sql.conf for this issue On the CDR Tool, should i need to edit to working properly Thanks Ha`

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[OpenSER-Users] need help on Free Radius & CDR Tool
by Do Nguyen Ha 05 Jun '08

05 Jun '08

hi i manage the Cisco BTS 10200 system, and now i am assigned a task to work with the Radius i choose the FreeRadius to work with Cisco BTS system AReq : Radius Access-Request AAcc: Radius Access-Accept AcctStart: Radius Accounting-Start AcctStop: Radius Accounting-Stop AcctRes: Radius Accounting-Response modify some config in radiusd.conf & clients.conf first, BTS send message Access Request to FreeRadius and then Freeradius send message Access Accept, they working properly when BTS send message Account Start to Radius, there are a lot attribute and freeradius save them in the radacct table but in radacct , there some columm doesn't match with the message Account Start, i want to change the name of some columm to match the message this is the account start message structure: Should i need to edit the sql.conf for this issue On the CDR Tool, should i need to edit to working properly Thanks Ha`

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[OpenSER-Users] Openser working with REFER (Transfer)
by Thiago Maluf 05 Jun '08

05 Jun '08

Hi List, I am research in Brazil University and in our system we have several User Agents in SIP. We have the openser working as Sip Proxy and Asterisk or Cisco Routers as Gateway with PSTN or PBX. We have too Hardphone Cisco IP Phone and Polycom IP Pphone and Softphone X-Lite, Eyebeam and Ekiga. Just now our system is working to all users, but I found one problem. When the user, work with BlindTransfer* the call is transfered no problem. But when the user (no matter if is Hardphone or Softphone) try transfer** the call, it don't work. In my OpenSER, when I receive one REFER without LOOSE_ROUTE I just end the request. While if this request have LOOSE_ROUTE I send this request to route defined in dialog. I would like if you can I help me, if I can insert one logical in OpenSER to do it work. is it possible? Very thanks for everybody, Thiago! (!) * BLINDTRANSFER: UA1 call to UA2 and when the call is established, UA1 for example send one request REFER to UA2 to this user call UA3. Then UA2 hangup call with UA1, and make one call to UA3. ** TRANSFER: UA1 call to UA2 and when the call is established, UA1 for example put this call in hold on, and make one call to UA3. When this last call is established, UA1 send one request REFER to UA2 to get the call with UA3. Then UA2 will talk with UA3, and UA1 is hangup its calls. -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

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[OpenSER-Users] t_replicate function`s doubt??
by Marcio 04 Jun '08

04 Jun '08

Hi, Someone would know if t_replicate() function....perform a t_relay() of the REGISTER....and carry the "credentials" of the UAc to slave or remote UAs? And the reply of the UAs(remote or slvae) is "200, ok" to UAs (fork`s origin) or not a reply?? best regards, Marcio

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[Serusers] SER 0.10.x AVP syntax documentation
by Steven C. Blair 04 Jun '08

04 Jun '08

I'm trying to convert my SER 0.9.x config file to a 0.10.x file. I need some help converting AVPs. I use to use rules such as " if (avp_db_load("$ruri/username", "s:callfwdall"))..." however this now generates a syntax error. Is there a how-to which describes how to convert these types of statements to the new format? Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[Serusers] SER 0.10 AVPs
by Steven C. Blair 04 Jun '08

04 Jun '08

Hello: I saw your post on the iptel.org web site about SER 0.10 and AVPs. I'm trying to convert a SER 0.9.x system to SER 0.10.x and the AVPOPS are giving me the most trouble. Did you find any documentation on how to implement the new avpops to do MySQL queries ? Thanks,Steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 04 Jun '08

04 Jun '08

I am using N95 to test SUBSCRIBE request with openser that dedicates as presence server. According to nokia notes, in nokia java sip stack, the accept-contact header is mandatory in all requests. Any request without this header will not be recognized by sip stack in N95. The NOTIFY request from openser seems not add this header in itself. Or maybe I omit some configurations. Please give me some clue, thank you. Regards, Kevin _________________________________________________________________ 新年换新颜，快来妆扮自己的MSN给心仪的TA一个惊喜！ http://im.live.cn/emoticons/?ID=18

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[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 04 Jun '08

04 Jun '08

I will be out of the office starting Mon 06/02/2008 and will not return until Mon 06/09/2008. I will have very limited access to mail, so please contact the CSC at 888-899-4227 if you need immediate assistance.

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[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 04 Jun '08

04 Jun '08

It is not the matter of rfc3261, the nokia sip stack force any request adding this header. Without it, the test will not be going on. Regards, Kevin _________________________________________________________________ 新年换新颜，快来妆扮自己的MSN给心仪的TA一个惊喜！ http://im.live.cn/emoticons/?ID=18

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Re: [OpenSER-Users] [OpenSER-Devel] OpenSER usrloc timeouts
by Henning Westerholt 04 Jun '08

04 Jun '08

On Wednesday 04 June 2008, Jason Penton wrote: > Hi Henning > > would you mind elaborating a little? Hi Jason, please always CC to the user list. First of all you need to define a failure_route in your config. After the lookup() to your location database you arm this with t_on_failure. In the failure route you check with t_check_status("408") the error code and forward then to your voicemail. Cheers, Henning

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Re: [OpenSER-Users] [OpenSER-Devel] OpenSER usrloc timeouts
by Henning Westerholt 04 Jun '08

04 Jun '08

On Wednesday 04 June 2008, Jason Penton wrote: > yes, but maybe instead of deregistering the user, it could foeard to > voicemail or similar instead of the callee just hearing 'nothing' and not > knowing what is going on????? (moved to the user list) Hi Jason, normally you just setup a prober INVITE timeout, and forward on the 408 that is generated to voicemail. Cheers, Henning

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[OpenSER-Users] T_replicate Bad Request Issue
by Ali Jawad 04 Jun '08

04 Jun '08

Hi All I have the following setup Openser Server A --> Uses Localhost MySQL Server Openser Server B --> Uses ServerA MySQL server Each server is in a different location. Both servers share the same database. I.e. in other words a user can successfully logon to either server. Both servers have a similar configuration file. My problem at hand is the following User A and User C logon to Server A User B logs on to Server B I need user A or user C to be able to call user B. I am using t_replicate for this setup. The relevant config section is found below: if(src_ip == OTHERSERVER) { xlog("L_INFO","PACKET FROM 9 PEER SERVER"); save("location"); }else{ xlog("L_INFO","NORMAL Packet"); if(is_method("REGISTER")){ if(!www_authorize("sero.test.net", "subscriber")){ www_challenge("sero.ahwar.net", "1"); return; } save("location"); t_replicate("sip:OTHERSERVER:5060"); return; } I have looked at traces and it seems to me that if a user logs on to a server t_replcicate forwards those register packets to the second server. However the second server is not reponding to those register packets it is replying with Bad Request replies U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bKe47f.4bd99ca4.0. Via: SIP/2.0/UDP 192.168.0.176:45374;received=193.227.186.146;branch=z9hG4bK-d87543-00509 f6c6f4a047f-1--d87543-;rport=45374. Max-Forwards: 69. Contact: <sip:9613041705@193.227.186.146:45374;rinstance=1367153d6d22ca20>;expire s=0. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7d242230. Call-ID: MmE3MTlmZDhiYWI0NjNmZjI5NWU5Njg1MmNhNDUwMDU.. CSeq: 6 REGISTER. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653b8b642971aef30 31dfa8d894fa90a4faf3",uri="sip:OriginatingServer",response="fd7943d66556 832aff9abcba071a9d85",cnonce="4f6aa73e9cba374f",nc=00000001,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bKe47f.4bd99ca4.0. Via: SIP/2.0/UDP 192.168.0.176:45374;received=193.227.186.146;branch=z9hG4bK-d87543-00509 f6c6f4a047f-1--d87543-;rport=45374. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.502f. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7d242230. Call-ID: MmE3MTlmZDhiYWI0NjNmZjI5NWU5Njg1MmNhNDUwMDU.. CSeq: 6 REGISTER. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK4cef.3d1bfc.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-221288 3e6a0dc710-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 2 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="fc7ee7ed4e3c 68f85de2a298f7579421",cnonce="42f3c2c5b71b4623",nc=00000001,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK4cef.3d1bfc.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-221288 3e6a0dc710-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.7860. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 2 REGISTER. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>;expires=5 67. P-Registrar-Error: Invalid CSeq number. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK5cef.dfe76626.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-022b94 01307f967e-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>;expires=0 . To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 3 REGISTER. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="9ded822006fa 51735838d3fdea96800b",cnonce="68a12fcb5c00aeaf",nc=00000002,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK5cef.dfe76626.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-022b94 01307f967e-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.93c8. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 3 REGISTER. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK2cef.bec40741.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-ea14ff 1ac524e10c-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@193.227.186.146:7988;rinstance=7d4ba8842a9e4a7e>. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 4 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="f126d934be4b 12c32a8ec783097a2ff6",cnonce="e3d03c6f36863764",nc=00000003,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK2cef.bec40741.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-ea14ff 1ac524e10c-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.5d32. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 4 REGISTER. Contact: <sip:9613041705@193.227.186.146:7988;rinstance=7d4ba8842a9e4a7e>;expires =566. P-Registrar-Error: Invalid CSeq number. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. .

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Re: [OpenSER-Users] compilation on solaris 10
by Bogdan-Andrei Iancu 04 Jun '08

04 Jun '08

Hi Eric, To be honest, no, I haven;t, because I do not have access to such a machine. Could you post the errors you get now? Regards, Bogdan wliangy(a)gmail.com wrote: > Thanks! > But there is still problem. > May I ask whether you have compiled successfully on Solaris 8 or 10? > > 2008/5/26 Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: > >> Hi Eric, >> >> use "gmake" and not "make" on Solaris. >> >> Regards, >> Bogdan >> >> wliangy(a)gmail.com wrote: >> >>> Hi experts, >>> I am start user of openser and need use openser to support sctp. >>> now I have download the source codes of openser-1.3.1-tls. >>> then I compiled it, there have many errors that need your supports. >>> >>> e.g. >>> openser-1.3.1-tls> make all >>> Makefile.defs:717: *** missing separator. Stop. >>> >>> then I commented Makefile.defs line717 and 718. >>> # $(warning Old gcc detected ($(CC_SHORTVER)), use >>> gcc >= 3.1 \ >>> # for better results) >>> >>> Errors were still there. >>> >>> openser-1.3.1-tls> make all >>> Makefile:402: warning: overriding commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Makefile:393: warning: ignoring old commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> yacc -d -b cfg cfg.y >>> "cfg.y", line 391: warning: redeclaration of precedence of SLASH. >>> >>> conflicts: 1 shift/reduce >>> flex cfg.lex >>> statistics.c:58: warning: #warning STATISTICS: Architecture with no >>> support for atomic operations. Using Locks!! >>> Makefile:402: warning: overriding commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Makefile:393: warning: ignoring old commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Compiling action.c >>> gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' >>> -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' >>> -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no >>> -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM >>> -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >>> -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC >>> -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 >>> -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD >>> -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS >>> -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o >>> parser/msg_parser.h: In function `char_msg_val': >>> In file included from action.h:27, >>> from action.c:44: >>> parser/msg_parser.h:300: parse error before `)' >>> forward.h: In function `msg_send': >>> In file included from action.c:49: >>> forward.h:95: parse error before `)' >>> forward.h:103: parse error before `)' >>> forward.h:107: parse error before `)' >>> mem/../lock_alloc.h: In function `lock_set_alloc': >>> In file included from mem/../locking.h:66, >>> from mem/../statistics.h:115, >>> from mem/shm_mem.h:33, >>> from ut.h:52, >>> from action.c:54: >>> mem/../lock_alloc.h:68: parse error before `)' >>> ut.h: In function `int2bstr': >>> In file included from action.c:54: >>> ut.h:172: parse error before `)' >>> ut.h: In function `un_escape': >>> ut.h:365: parse error before `)' >>> ut.h: In function `shm_str_dup': >>> ut.h:498: parse error before `)' >>> ut.h: In function `pkg_str_dup': >>> ut.h:515: parse error before `)' >>> ut.h: In function `str_strcmp': >>> ut.h:532: parse error before `)' >>> ut.h:535: parse error before `int' >>> ut.h:539: `minlen' undeclared (first use in this function) >>> ut.h:539: (Each undeclared identifier is reported only once >>> ut.h:539: for each function it appears in.) >>> ut.h:547: `alen' undeclared (first use in this function) >>> ut.h:547: `blen' undeclared (first use in this function) >>> ut.h: In function `str_strcasecmp': >>> ut.h:563: parse error before `)' >>> ut.h:566: parse error before `int' >>> ut.h:570: `minlen' undeclared (first use in this function) >>> ut.h:578: `alen' undeclared (first use in this function) >>> ut.h:578: `blen' undeclared (first use in this function) >>> action.c: In function `run_action_list': >>> action.c:140: parse error before `)' >>> action.c: In function `do_assign': >>> action.c:193: parse error before `)' >>> action.c:202: parse error before `)' >>> action.c:222: parse error before `)' >>> action.c:228: parse error before `)' >>> action.c: In function `do_action': >>> action.c:302: parse error before `)' >>> action.c:309: parse error before `)' >>> action.c:335: parse error before `)' >>> action.c:498: parse error before `)' >>> action.c:524: parse error before `)' >>> action.c:682: parse error before `)' >>> action.c:710: parse error before `)' >>> action.c:723: parse error before `)' >>> action.c:757: parse error before `)' >>> action.c:763: parse error before `)' >>> action.c:807: parse error before `)' >>> action.c:819: parse error before `)' >>> action.c:870: parse error before `)' >>> action.c:896: parse error before `)' >>> action.c:904: parse error before `)' >>> action.c:930: parse error before `)' >>> action.c:939: parse error before `)' >>> action.c:945: parse error before `)' >>> action.c:966: parse error before `)' >>> make: *** [action.o] Error 1 >>> >>> Your any help is much appreciated! >>> >>> BR, >>> Eric >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > >

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[OpenSER-Users] Why can't I "append_branch()" in FAILURE_ROUTE if reply is 6XX ??
by Zappasodi Daniele 03 Jun '08

03 Jun '08

Hello, If I want to have the ability to add a new branch for every negative response, included the 6xx (and despite what is written in the RFC3261), where should I change? It is enough to remove the following code in t_should_relay_response function? 783 if (new_code>=600) { 784 /* this is a winner and close all branches */ 785 which_cancel( Trans, cancel_bitmap ); 786 picked_branch=branch; 787 /* no more new branches should be added to this transaction */ 788 Trans->flags |= T_NO_NEW_BRANCHES_FLAG; ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

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[OpenSER-Users] Problem using path module
by Thomas Mangin 03 Jun '08

03 Jun '08

Hello, I am trying to configure a number of openser servers. I have the following testbed: Phone1 + Proxy - Registrar - out proxy Phone2/ The Proxy acts as outbound proxy for customers and uses ienum to route messages to the right registrar (using user@domain), the registrar then figures out where the destination is and sends the packet out to ENUM/PSTN or in my case back to the same proxy. The proxy is running nathelper (and redirects rtp to an rtp proxy) and performs route-recording. The proxy configuration has: modparam("path", "use_received", 0) and the registrar configuration contains: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) When registering all is fine and well and the packet flows like: Phone<X> -> Proxy -> Registrar (storing in DB) -> Proxy -> Phone<X> and the DB contains in path: <sip:ProxyIP;lr;received=sip:PhoneIP:PhonePort> However, when performing an invite the ACK packet (INVITE/OK/ACK) can not be routed properly as it is not part of the transaction, which causes the call to be cancelled after a few seconds (as it should be given the constance). The ACK packet starts as URI: <sip:user@<Registrar IP>:5060;rinstance=xxxxxx> Contact: <sip:user@PhoneIP:PhonePort> with the IP and Port correctly translated from the internal to the firewall external. It then travels like this: Phone1 -> Proxy -> Registrar > Proxy <stop here> When leaving Registrar the ACK URI has been changed to the one stored in the DB path field ie: <sip:ProxyIP;lr;ftag=yyyyyy> at that point the Proxy is not able to figure out where to forward the packet even if the Contact field of the packet is still correctly set to the IP:Port of the phone to be reached. I was wondering if I am missing some understanding of how RFC3327 is working. I can see a few possible solutions: - generate a first Path header with the Phone information on the proxy before the current one. - try to catch those 'out of transaction ACK' and set the URI to the contact But before doing anything I would appreciate if someone could let me know if I am attempting something idiotic from a design point of view and tell me why what I expected to 'just work' does not behave as expected. Thank you very much your your time and help. Thomas

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[OpenSER-Users] NEW FUNCTIONALITY: secure nonce in auth module
by Bogdan-Andrei Iancu 03 Jun '08

03 Jun '08

Hi, Thanks to Anca Vamanu < anca at voice-system dot ro> , the auth module has a new mechanism for protecting the nonce against third party attacks. So far, the auth module provided only nonce lifetime limitation to prevent re-usage of the nonce. The new mechanism offers protection against sniffing intrusion. The module generates and verifies the nonces so that they can be used only once (in an auth response). This is done by having a lifetime value and an index associated with every nonce. Using only an expiration value is not good enough because,as this value has to be of few tens of seconds, it is possible for someone to sniff on the network, get the credentials and then reuse them in another packet with which to register a different contact or make calls using the other's account. The index ensures that this will never be possible since it is generated as unique through the lifetime of the nonce. What problem is solved: ======================== Typical intrusion scenario is: 1) UAC sends INVITE with no credentials 2) openser sends challenge to UAC 3) UAC sends INVITE with credentials (lifetime 1 minute) 4) the credentials (auth response) are exposed on the net, so anybody can grab them (lets assume by person X) 5) openser accepts the credentials and call is establish 6) person X injects (by sending to openser) an re-INVITE (to change the media destination, for example) by re-using the credentials that he grabbed from the net. 7) credentials are accepted by openser if the lifetime is not expired. So, third parties can modify (or even initiate) dialogs by re-using the credentials they captures from a previous authentication. The only limitation is the nonce lifetime which is about 30 - 60 seconds, long enough for allowing such an atack. The new mechanism, by blocking the re-usage of the same nonce (even if the lifetime is still valid), will block step 7) as the same nonce was used in step 5) (by UAC). Technical details: ================== The auth module keeps state for each nonce - to validate it only on the first usage. A binary array (which can by default accomodate 100K nonces) is used to keep the state. An index in this array is allocated when the challenge is generated; this index in kept for the whole life duration of the nonce. After the first auth result (for the nonce), the following auth results for that nonce are discarded and re-challenged. As nonce can be generated and later checked from different processes, the nonce state (binary array) is kept in shm memory and the access to it is synchronized via locking. The see what is the overhead added by the computation time (keeping the nonce state) and locking (sharing between processes), Anca run several of tests : - openser single process - openser multiple processes, single processor machine - openser multiple processes, multi processor machine For tests we were using authentication via PV (for password and username) to avoid using any slow data backend (radius or DB) that might eclipse the overhead. The measured overhead is 4% - 7% (from the original version). Regards, Bogdan

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[OpenSER-Users] how to configure to support PRACK
by Steven Wu 03 Jun '08

03 Jun '08

Hi All, Could someone tell me how to config openser 1.3.x route to support PRACK? I had installed openser 1.3.x and didn't change any route configuration. My scenario is: A openser B | --> INVITE (with 100rel) -> | --> INVITE (with 100rel) --> | | | | | <--183 <-- | <-- 183 <-- | | | --> PRACK --> | | <-- 404 Not here <--- | For the default the openser reply the "404 Not here" to A. I found the '404' was replied here: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* !loose_route */ /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } ## These code will cause openser reply 483 response to PRACK if ( is_method("PRACK") ) { t_relay(); exit; } } ## Here reply me the 404 sl_send_reply("404","Not here"); } exit; } Than I added the green code to route the PRACK, But I received "483","Too Many Hops" from openser. And I found the Max-Forward is 70 not a wrong value. Does anyone can help me? Best regards, Steven Wu

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Re: [OpenSER-Users] Trouble with RADIUS Accounting
by Dmitry Lyubimkov 03 Jun '08

03 Jun '08

But we did it!!! I don't need to remove this attributes. We need User-Name, Called-Station-Id, Calling-Station-Id, etc for billing purpose. But our RADIUS server does not receive them from OpenSER! Look at our Start/Stop messages. There are not such important attributes as User-Name, etc. Is this bug of OpenSER or bug of our script? If it is bug of script, then what commands, what modules, what modparam can lead to such result? Dmitry -----Original Message----- From: Mike O'Connor [mailto:mike@pineview.net] Sent: Monday, June 02, 2008 1:07 PM To: Dmitry Lyubimkov Subject: Re: [OpenSER-Users] Trouble with RADIUS Accounting Hi Dmitry You not can remove them from OpenSER with out adjusting the code, instead you need to put the radius packets thought something like a radius proxy which removes them. Mike Dmitry Lyubimkov wrote: > Hi. > We have new installation of OpenSER V.1.3.1 with new script. All work fine > except RADIUS accounting. Accounting Start and Stop records come to RADIUS > server, but there are no some attributes in them - User-Name, > Called-Station-Id, Calling-Station-Id,... > Example of Start and Stop records (FreeRADIUS log): > Acct-Status-Type = Start > Service-Type = IAPP-Register > Sip-Response-Code = 200 > Sip-Method = 1 > Event-Timestamp = "Jun 2 2008 10:34:05 MSD" > Sip-From-Tag = "5d12c92c" > Sip-To-Tag = "6845627C-1EF5" > Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." > NAS-Port = 5060 > Acct-Delay-Time = 0 > NAS-IP-Address = 172.16.2.10 > Client-IP-Address = 172.16.2.10 > Timestamp = 1212388445 > > Acct-Status-Type = Stop > Service-Type = IAPP-Register > Sip-Response-Code = 200 > Sip-Method = 8 > Event-Timestamp = "Jun 2 2008 10:34:12 MSD" > Sip-From-Tag = "5d12c92c" > Sip-To-Tag = "6845627C-1EF5" > Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." > NAS-Port = 5060 > Acct-Delay-Time = 0 > NAS-IP-Address = 172.16.2.10 > Client-IP-Address = 172.16.2.10 > Timestamp = 1212388452 > > Where is trouble? Required attributes are described in dictionary file. If I > remove it then OpenSER says > Jun 2 11:02:23 [4064] ERROR:auth_radius:mod_init: auth_radius: can't get > code for the User-Name attribute > Jun 2 11:02:23 [4064] ERROR:core:init_mod: failed to initialize module > auth_radius > Jun 2 11:02:23 [4064] ERROR:core:main: error while initializing modules > > How and where in script we can prohibit send of some attributes? > > Dmitry > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] Can't get sipping working in nathelper module
by Pablo Guijarro Enríquez 03 Jun '08

03 Jun '08

Hi, Im using OpenSER v1.3.1 and sipping does not work with my configuration. Here is a sample: loadmodule "usrloc.so" loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 10) #modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@cmovil.net") modparam("nathelper", "received_avp", "$avp(i:801)") modparam("registrar", "received_avp", "$avp(i:801)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # for clients behind nats if ( nat_uac_test("1") ) { if (method=="REGISTER") { fix_nated_register(); setbflag(6); setbflag(7); } else if ( method=="SUBSCRIBE" || method=="PUBLISH ) { fix_nated_contact(); setbflag(6); setbflag(7); }; }; ... } ... TCP clients register without problems and NAT users are identified properly. Routing for them works fine and so on, but I cannot see any ping packet leaving OpenSER to keep the TCP connection alive. Is there something missing? Thank in advance. Pablo

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[OpenSER-Users] Register Distributed
by Marcio 02 Jun '08

02 Jun '08

Hi, Someone could have tips on how to register in local domain a UAc mobility of another domain (federation) using the OpenSER's config....when one of the INVITE's UAc have to be verified as valid in the home domain and the answer would be allowed its domain registration in place, with a pointer/indicator in the home domain...where he was as that moment?? UAs(local) (accept register after validation on UAs(home) \ \ (Invite/register) \ UAc (mobility)-- -- -- -- -- UAs(home Thx for any help... Marcio

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[OpenSER-Users] setting up radius accounting
by Pete Kay 02 Jun '08

02 Jun '08

Hi, I am trying to set up using Radius for accounting. As a newbie, I am simply following the standard configuration from the menu. Here is where I put my flags in openser.cfg: if(method=="BYE") { setflag(2); }; if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; The cdr is successfully logged to the database via radius, but the starttime and endtime are incorrect. The starting time and ending time being recorded is the same. Also, I noticed that Openser only sends the message (both INSERT and UPDATE cdr) to radius when the BYE method is received. I was expecting Openser to instruct Radius to create a new record ( INSERT ) during the INVITE method and update the stoptime during the BYE method. Here is the SQL Query log i extracted from mysql: 80603 4:38:06 6481 Query UPDATE radacct SET acctstoptime = '2008-06-03 04:38:06', acctsessiontime = '', acctinputoctets = '0' << 32 | '0', acctoutputoctets = '0' << 32 | '0', acctterminatecause = '', acctstopdelay = '0', connectinfo_stop = '' WHERE acctsessionid = 'NWQyZGI1ZTkzNGM5ZjMzZTg5ZGQ2OGVmZDY5MjYxMGE.' AND username = '1005' AND nasipaddress = '127.0.0.1' 6481 Query INSERT INTO radacct (acctsessionid, acctuniqueid, username, realm, nasipaddress, nasportid, nasporttype, acctstarttime, acctstoptime, acctsessiontime, acctauthentic, connectinfo_start, connectinfo_stop, acctinputoctets, acctoutputoctets, calledstationid, callingstationid, acctterminatecause, servicetype, framedprotocol, framedipaddress, acctstartdelay, acctstopdelay) VALUES ('NWQyZGI1ZTkzNGM5ZjMzZTg5ZGQ2OGVmZDY5MjYxMGE.', '1412cd5bf3dd4a31', '1005', '192.168.1.104', ' 127.0.0.1', '5060', '', DATE_SUB('2008-06-03 04:38:06', INTERVAL (0 + 0) SECOND), '2008-06-03 04:38:06', '', '', '', '', '0' << 32 | '0', '0' << 32 | '0', 'sip:1006@192.168.1.104<sip%3A1006(a)192.168.1.104>', 'sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>', '', 'IAPP-Register', '', '', '0', '0') I would appreciate any help on getting the radius accounting to work. Thanks alot in advance. Pete

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[OpenSER-Users] Question about proxy_authorization
by mark morreny 02 Jun '08

02 Jun '08

Hi, I am not able to proxy_authorize a INVITE request from UA if the US connects to Opnerser using Openser's public IP. I am wonering if there is any additional setting I need to enable that? From the log, Openser does recognize '88,251.97.47' as a local domain. I did not hard-code the private ip anywhere in openser.cfg. Tue Jun 3 03:20:05 2008-58.251.97.17::method <INVITE> r-uri < sip:1005@58.251.97.17 <sip%3A1005(a)58.251.97.17>> to-tag <<null>> 2nd via <<null>> to-header <sip:1005@58.251.97.17 <sip%3A1005(a)58.251.97.17>> from-header <sip:1006@58.251.97.17 <sip%3A1006(a)58.251.97.17>> auth-user <1006> Jun 3 03:20:05 [27566] DBG:maxfwd:is_maxfwd_present: value = 70 before fix nated contact Jun 3 03:20:05 [27566] DBG:core:parse_headers: flags=80 checking to see if setflag should do or not Jun 3 03:20:05 [27566] DBG:uri:has_totag: no totag Jun 3 03:20:05 [27566] DBG:core:parse_headers: flags=78 Jun 3 03:20:05 [27566] DBG:tm:t_lookup_request: start searching: hash=29544, isACK=0 Jun 3 03:20:05 [27566] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 3 03:20:05 [27566] DBG:tm:t_lookup_request: no transaction found it is not a REGISTER, so going to route 3 Jun 3 03:20:05 [27566] DBG:domain:is_domain_local: Realm '88,251.97.47' is local Jun 3 03:20:05 [27566] DBG:core:db_free_rows: freeing 1 rows Jun 3 03:20:05 [27566] DBG:core:db_free_rows: row[0]=0x81a1e18 Jun 3 03:20:05 [27566] DBG:core:db_free_rows: 0x81a1e18=pkg_free() RES_ROWS request is from a non-trusted domain Jun 3 03:20:05 [27566] DBG:auth:check_nonce: comparing [4844491107980b084e3d5d3a283446ad02324d49] and [4844491107980b084e3d5d3a283446ad02324d49] Jun 3 03:20:05 [27566] DBG:auth:check_response: our result = '578a6ec1af28eed79329ed5414313f43' Jun 3 03:20:05 [27566] DBG:auth:check_response: authorization failed Thanks in advance for your help. Regards, Mark

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[OpenSER-Users] Mediaproxy + XLite RTCP timeout
by Dmitry Lyubimkov 02 Jun '08

02 Jun '08

OpenSER v.1.3.1 We use Mediaproxy for NAT traversal purpose. All work fine except one feature. XLite has option to hangup call after RTCP inactivity. When call is not proxied this option work right. But when call is proxied by Nediaproxy it cannot be longer than 54 second + RTCP timeout in XLite. Xlite cannot see RTCP activity and hangup the call. Does anybody meet with this trouble? Dmitry

1 0

[OpenSER-Users] Problem authenticating Xlite or Zoiper in Openser
by voipanywhere 02 Jun '08

02 Jun '08

Hi All, After doing ngrep teste(a)xxx.xxx.xxx.xxx -lqtt -W byline port 5060 xlite somehow is sending wrong information. my subscriber is teste(a)xxx.xxx.xxx.xxx, i put it on xlite but on sip messages i get teste%40xxx.xxx.xxx.xxx and then of course the 401 error. something is changing the @ in the username to %40. i've been trying to change locales in debian, still didn't figure it out what the problem really is. i had the console with wrong code support for portuguese language, and could be it (i tought) now it's ok. tried to check /etc/environment and /etc/locales.gen to see if everything is okay, and it is now. Can be some strange stuff because of utf, unicode etc? Thanks Regards from Portugal, Helder -- View this message in context: http://www.nabble.com/Problem-authenticating-Xlite-or-Zoiper-in-Openser-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] Trouble with RADIUS Accounting
by Dmitry Lyubimkov 02 Jun '08

02 Jun '08

Hi. We have new installation of OpenSER V.1.3.1 with new script. All work fine except RADIUS accounting. Accounting Start and Stop records come to RADIUS server, but there are no some attributes in them - User-Name, Called-Station-Id, Calling-Station-Id,... Example of Start and Stop records (FreeRADIUS log): Acct-Status-Type = Start Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 1 Event-Timestamp = "Jun 2 2008 10:34:05 MSD" Sip-From-Tag = "5d12c92c" Sip-To-Tag = "6845627C-1EF5" Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 172.16.2.10 Client-IP-Address = 172.16.2.10 Timestamp = 1212388445 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 8 Event-Timestamp = "Jun 2 2008 10:34:12 MSD" Sip-From-Tag = "5d12c92c" Sip-To-Tag = "6845627C-1EF5" Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 172.16.2.10 Client-IP-Address = 172.16.2.10 Timestamp = 1212388452 Where is trouble? Required attributes are described in dictionary file. If I remove it then OpenSER says Jun 2 11:02:23 [4064] ERROR:auth_radius:mod_init: auth_radius: can't get code for the User-Name attribute Jun 2 11:02:23 [4064] ERROR:core:init_mod: failed to initialize module auth_radius Jun 2 11:02:23 [4064] ERROR:core:main: error while initializing modules How and where in script we can prohibit send of some attributes? Dmitry

1 0

[OpenSER-Users] rtpproxy
by Pezhman Lali 02 Jun '08

02 Jun '08

Dear, I found using rtpproxy on different machine is really more useful, for solving nat problems, but I don't know the reseon. good help for setting rtpproxy on different mach : http://www.iptel.org/rtpproxy_on_different_machine Best Mani

1 0

[OpenSER-Users] how to drop sip call when i receive ring
by abo 02 Jun '08

02 Jun '08

Dear sir: i am trying openser 1.2.0 version.and now i want to drop call when i got 180 or 183.this is my openser.cfg.can you help me. thanks a lot! # ----------- global configuration parameters ------------------------ debug=9 # 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) port=5060 children=4 #fifo="/tmp/openser_fifo" listen=192.168.1.254 alias=openser.org # Uncomment this to prevent the blacklisting of temporary not available destinations disable_dns_blacklist=yes # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database # - MySQL loaded for accounting as well loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/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) # -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "db_flag", 7) modparam("acc", "db_missed_flag", 8) modparam("acc", "log_level", 3) modparam("acc", "log_flag", 2) modparam("acc", "log_missed_flag", 8) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "early_media", 1) modparam("acc", "db_extra", "src_ip=$si;dst_user=$tU;dst_domain=$du;src_user=$hdr(User-Agent)") # -- avpops params -- # avp aliases - define c_uri (canonical uri) as AVP ID 888 and s_ip as ID 999 avp_aliases="c_uri=i:888;s_ip=i:999" #tm timeout for voicemail params modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer", 35) modparam("tm", "noisy_ctimer", 1) # ------------------------- request routing logic ------------------- # main routing logic #Routing Script route { #check for old messages: could mean a problem with the DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; }; #check for extremely large messages; we don't need a sip dos attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, >= 12048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; #do not send to voicemail if BYE or CANCEL #is used to end call before user pickup or timeout if(method=="CANCEL" || method=="BYE") { setflag(7); setflag(8); }; #grant route if route headers already present if (loose_route()) { route(1); return; }; #Always require authentication, which could result in a PSTN, ie $$$ if (method=="REGISTER") { # if(!www_authorize("openser.org", "subscriber")) # { # www_challenge("openser.org", "0"); # return; # } # else # { # if (!check_to()) # { # sl_send_reply("401", "Unauthorized"); # return; # }; #Save into user database, used below when checking if user is available xlog("L_INFO", "REGISTER: User Authenticated Correctly\n"); save("location"); return; # }; }; if (method=="INVITE") { setflag(2); setflag(7); setflag(8); route(1); } } route[1] { #send the call outward if(uri=~"sip:[0-9]+@.*") { rewritehostport("192.168.1.100:5070"); #record_route(); t_on_reply("1"); # t_relay(); } if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } onreply_route[1] { xlog("----------aaaaaaaaaa-------\n"); if (t_check_status("183")||t_check_status("180")){ exit; xlog("------------bbbbbbb------- \n"); } }

1 0

[OpenSER-Users] Please help with "auth:pre_auth: credentials with given realm not found" error
by mark morreny 31 May '08

31 May '08

Hi, I have found a few posts on this error from Google but none of the solutions works. I am able to use radius_proxy_authorize for REGISTER request without problem. But when I use radius_proxy_authorize for INVITE, Opensers can't authroize it. I have tried: radius_proxy_authorize("192.168.1.104")) proxy_authorize("192.168.1.104","subscriber") radius_proxy_authorize("")) proxy_authorize("","subscriber") I have been stuck with this for days already and I truely hope someone can tell me what could be wrong with my setup. I am pretty much in a stage of desperation right now. Thanks in advance for all your help. Here are the relevant lines in openser.cfg: #if (!radius_proxy_authorize("192.168.1.104")) { if (!proxy_authorize("192.168.1.104","subscriber")) { www_challenge("192.168.1.104","1"); exit; } Here is the openser debug log, I have highlighted the part that proxy_authorize throws error: Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <INVITE> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-f072f593>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=10 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [33]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> ] Jun 1 05:37:16 [29028] DBG:core:pv_get_to_attr: no To tag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: content_length=393 Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: found end of header Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=39a9124ca59076d9o0 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1005"}, ruri={ sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=14000 Jun 1 05:37:16 [29028] DBG:core:pv_get_authattr: no [Proxy-]Authorization header Sun Jun 1 05:37:16 2008-192.168.1.101::method <INVITE> r-uri < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> to-tag <<null>> 2nd via <<null>> to-header <sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> from-header <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>> auth-user <<null>> Jun 1 05:37:16 [29028] DBG:maxfwd:is_maxfwd_present: value = 70 Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 before fix nated contact Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:uri:has_totag: no totag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=78 Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: start searching: hash=59293, isACK=0 Jun 1 05:37:16 [29028] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: no transaction found Jun 1 05:37:16 [29028] DBG:domain:is_domain_local: Realm '192.168.1.104' is local Jun 1 05:37:16 [29028] DBG:core:db_free_rows: freeing 1 rows Jun 1 05:37:16 [29028] DBG:core:db_free_rows: row[0]=0x81a1c28 Jun 1 05:37:16 [29028] DBG:core:db_free_rows: 0x81a1c28=pkg_free() RES_ROWS Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=10000 Jun 1 05:37:16 [29028] DBG:auth:pre_auth: credentials with given realm not found Jun 1 05:37:16 [29028] DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841c6380fa8573a4e9794b1a0a2df9f3e53cb49", qop="auth" ' Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list 0xb5bf9be0 Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <ACK> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-f072f593>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=026ab2c49274b4e0186dd72188f7d50a.939a Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [75]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>>] Jun 1 05:37:16 [29028] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list (nil) Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <INVITE> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-dc12d56c>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=10 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [33]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> ] Jun 1 05:37:16 [29028] DBG:core:pv_get_to_attr: no To tag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: content_length=393 Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: found end of header Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=39a9124ca59076d9o0 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1005"}, ruri={ sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=14000 Sun Jun 1 05:37:16 2008-192.168.1.101::method <INVITE> r-uri < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> to-tag <<null>> 2nd via <<null>> to-header <sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> from-header <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>> auth-user <1005> Jun 1 05:37:16 [29028] DBG:maxfwd:is_maxfwd_present: value = 70 Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:uri:has_totag: no totag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=78 Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: start searching: hash=59290, isACK=0 Jun 1 05:37:16 [29028] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: no transaction found Jun 1 05:37:16 [29028] DBG:domain:is_domain_local: Realm '192.168.1.104' is local Jun 1 05:37:16 [29028] DBG:core:db_free_rows: freeing 1 rows Jun 1 05:37:16 [29028] DBG:core:db_free_rows: row[0]=0x81a1c28 Jun 1 05:37:16 [29028] DBG:core:db_free_rows: 0x81a1c28=pkg_free() RES_ROWS Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=10000 Jun 1 05:37:16 [29028] DBG:auth:pre_auth: credentials with given realm not found Jun 1 05:37:16 [29028] DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841c6380fa8573a4e9794b1a0a2df9f3e53cb49", qop="auth" ' Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list 0xb5bf9be0 Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <ACK> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-dc12d56c>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=026ab2c49274b4e0186dd72188f7d50a.6388 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [75]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>>] Jun 1 05:37:16 [29028] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list (nil) Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Here is the SIP trace: U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 INVITE. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.f608. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.f608. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 ACK. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 INVITE. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="e39bd9dc46a72ea191fca87e98cf9136",qop=auth,nc=00000001,cnonce="9d027a79". Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.218e. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.218e. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 ACK. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="dc66b568ed8ec48f0f405aa063156eaa",qop=auth,nc=00000001,cnonce="9d027a79". Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 INVITE. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="8d2b52278a72326a57361ec1ca2601c3",qop=auth,nc=00000001,cnonce="f4b6f3a0". Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.8d8a. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.8d8a. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 ACK. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="8ebde837f7c7c49f721d9f2c44b3797d",qop=auth,nc=00000001,cnonce="f4b6f3a0". Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0.

1 0

[Serusers] (no subject)
by Bin Zhang 31 May '08

31 May '08

-- Bin Zhang

1 0

[OpenSER-Users] Hints about to capture codes negative www_authorize??
by Marcio 30 May '08

30 May '08

Hi, Who would have tips on how to capture the return codes negative of the function "www_authorize" ??? I read somewhere that said to use $retval........but ...How can I to use it ("$retval") ?? -1 - Non existent user; -2 - invalid passwd; -3 - stale nonce; -4 - no credentials; -5 - error; Any help will be greatly appreciated. thx, Marcio

3 5

[OpenSER-Users] Fwd: Re: Installing perl module
by Henning Westerholt 30 May '08

30 May '08

---------- Forward message ---------- Subject: Re: [OpenSER-Users] Installing perl module Date: Freitag, 30. Mai 2008 From: "Yazid Hadj Said" <yazid.hadjsaid(a)gmail.com> Tp: "Henning Westerholt" <henning.westerholt(a)1und1.de> Hi Henning, I think that the problem is in openser ebuild, so I do this modifications: # diff openser-1.3.1.ebuild openser-1.3.1.ebuild.old 7c7 < IUSE="ipv6 mysql radius postgres jabber ssl odbc perl perlvdb" --- 58,63d57 < use perl && < modules="${modules} perl" < < use perlvdb && < modules="${modules} perlvdb" < --------------------------------------- Hi Yazid, please always CC at the list, so that others can read the resolution too. Great that it works now for you. I'm not sure if the ebuild that is included in the openser distribution is also used in gentoo. Perhaps you can file a bug against the gentoo package (on gentoo.org), and attach your patch, so that the maintainer could fix this? Cheers, Henning

2 1

[OpenSER-Users] Installing perl module
by Yazid Hadj Said 30 May '08

30 May '08

Hi, I use 1.3.1 openser version, and i don't find perl module (I don't know Why). My question is if I can add perl module without recompiling openser. (I use Gentoo distribution) Regards Yazid

2 1

[OpenSER-Users] variables in regexp
by Yazid Hadj Said 30 May '08

30 May '08

Hi, please I want to know: If and How I can use variables in "regexp" . example: avp_subst("avp(s:var1)/$avp(s:var2)","/(.*)$avp(s:var3)(.*)/\1/") In this example, simply the "$avp(s:var3)" string is matched, not the $avp(s:var3)'s content. I am very need this function. Thanks in advance, Yazid

3 2

[OpenSER-Users] Correction in documentation of tm module(version 1.2.X)
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi, The following sentence in section 1.4.8. t_flush_flags() of tm module documentation(version 1.2.X) has several spelling mistakes. Incorrect Flush the flags from current request into the already created transaction. It make sens only in routing block if the trnasaction was created via t_newtran() and the flags have been altered since. Correct Flush the flags from current request into the already created transaction. It make sense only in routing block if the transaction was created via t_newtran() and the flags have been altered since. Thanks, Ruchir

3 3

[OpenSER-Users] Openser not processing requests at rate being sent to it
by Brian Peters 30 May '08

30 May '08

I am doing what I thought was a simple volume test that OpenSER should be able to handle, but the results I am getting so far show it is not handling the load I'm sending it. Here is my configuration: - OpenSER 1.3.1 - Simple script that uses OpenSER as a proxy to forward requests to a carrier. All it does is rewrite the host and port of the R-URI and relays to the carrier. - children=16 - shared memory set to 1 G. (-m 1000) - using transaction module to handle the requests. - Sending 10 INVITE requests per second to OpenSER using Asterisk. When I do a "tcpdump" trace of the network traffic to/from OpenSER, it is clear that a large percentage of the INVITE requests sent to OpenSER were not handled. Asterisk needed to resend the INVITE requests several times before OpenSER would handle them, and in about 20% of the cases Asterisk terminated the INVITES because it reached the timeout of 30 seconds without any response. If I go directly from asterisk to the carrier, this test works without any issues. I'm not sure what the limitations are of OpenSER, but this seems like something it should be able to handle. Any help or information on how to configure OpenSER so it will work would be appreciated. Thanks, Brian.

6 17

[OpenSER-Users] openser trunk performance
by jh＠tutpro.com 30 May '08

30 May '08

also i did some performance tests on openser trunk in order to clear doubt that has been expressed recently. with only 4 child processes and several radius/mysql queries per invite, i was able to achieve 50 new calls per second using sipp including radius accounting to database of each call. this was on my two years old laptop and everything was running on the same host. i didn't try yet how many calls i could have achieved before retransmissions. i'm satisfied with the result and don't think that there are any performance issues with openser trunk. -- juha

3 2

[OpenSER-Users] Corrections in rr module documentation version 1.2.X
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi, I've noticed some mistakes in documentation of rr module in version 1.2.X. 1.5.6. is_direction(dir): Incorrect sentence: As for checking it's used the "ftag" Route header parameter, the append_fromtag (see Section 1.4.2 module parameter must be enables. Also this must be call only after loose_route() (see Section 1.5.1). Correct sentence: As for checking it's used the "ftag" Route header parameter, the append_fromtag (see Section 1.4.2 module parameter must be enabled. Also this must be called only after loose_route() (see Section 1.5.1). Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd.

2 1

[OpenSER-Users] Scrolling through resultset
by mark morreny 30 May '08

30 May '08

Hi, If I want to walk through a resultset of numbers, how can I do that? I have tried: $var(index) = 1 while (is_avp_set($var(i:$index) ) $var(index) = $var(index) +1; However $var(i:$index) won't work. I am wondering if someone can tell me how I can correct it. The other question I have is about psuedo-variable. Is there any psuedo-variable can give me the IP of the Openser server? Thanks, Mark

2 1

[OpenSER-Users] Explanation of 0x02 in "t_relay()" ?
by Iñaki Baz Castillo 30 May '08

30 May '08

Hi. I read in "t_relay" doc: -------- 0x02 - do not internally send a negative reply in case of failure. It applies only when the transaction is created. By default one is sent. Useful if you want to implement a serial forking in case of failure. -------- Could you please explain it a little more? I use failure_route with append_branch and before it OpenSer doesn't send back a negative reply to caller (in fct it would end the transaction). What the purpose of this bit option? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 3

[OpenSER-Users] Need help with proxy_authorize for NATed UA
by mark morreny 30 May '08

30 May '08

Hi, I am using the same UA and Openser to do two tests. For the first case, UA is pointing to the external IP of openser.; the other test points the UA to openser's local ip. Registering is ok for both cases, but when I try to do an INVITE, the UA pointing to the local IP works whereas the one pointing to the external IP fails during proxy_authorize. 88.251.45.4 -> external ip of the entire network 192.168.1.104 -> openser's local ip 192.168.1.100 -> UA's local ip Here is the INVITE with authentication digest from the UA pointing to the external IP address of openser: U 88.251.45.4:59864 -> 192.168.1.104:5060 INVITE sip:1005@88.251.45.4 <sip%3A1005(a)88.251.45.4> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:17480 ;branch=z9hG4bK-d87543-ea31ca6af52bd14c-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1006@88.251.45.4:59864>. To: "1005"<sip:1005@88.251.45.4 <sip%3A1005(a)88.251.45.4>>. From: "1006"<sip:1006@88.251.45.4 <sip%3A1006(a)88.251.45.4>>;tag=d21e4745. Call-ID: NGFlOTRmNTg3ZDFlYzJlYzdhNTlhNzZmMDRlOTJmOGY.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1006",realm="88.251.45.4 ",nonce="483edcaf82beda80a0c91fc75f486b8fd3f7bbc3",uri="sip:1005@88.251.45.4<sip%3A1005(a)88.251.45.4> ",response="4423a99e44bb7266912de3a4af752b3f",cnonce="ece582bf4b1708b67bd1e86952313a63",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 423. . v=0. o=- 1 2 IN IP4 192.168.1.100. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.100. t=0 0. m=audio 34842 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : bIn/n4cp vm9zXwFh 192.168.1.100 34842. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. Here is the INVITE resend from the UA pointing to openser's local IP: U 192.168.1.100:50314 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:50314 ;branch=z9hG4bK-d87543-c34be44e3704d673-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1006@192.168.1.100:50314>. To: "1005"<sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. From: "1006"<sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104> >;tag=d4048226. Call-ID: YTc4OTE2NmIyMWRkMjk3ZGNmYmY4OTg3NTQ5NWRhZDI.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1006",realm="192.168.1.104 ",nonce="483ede954fef336afd24141e99d516b0b7d9091a",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",response="1b77bf191fef631049bd6d6b7dcc9094",cnonce="e96a7e7d9677c4c41952bce7ac2a36e2",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 423. . v=0. o=- 7 2 IN IP4 192.168.1.100. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.100. t=0 0. m=audio 14816 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : SP8QJ5e9 96jlnBkg 192.168.1.100 14816. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. The openser script associated wit proxy_authorize is : if (!proxy_authorize("","subscriber")) { xlog("not authorize need to do proxy challenge\n"); proxy_challenge("","1"); exit; } What could be wrong with my setup? Thanks in advance for your help. Thanks, Mark

2 1

[OpenSER-Users] Problems with (RE-)REGISTER
by Lucio Maciel 30 May '08

30 May '08

Hi, I am having a situation where the client sends each RE-REGISTER with a new Call-ID, causing it to create a new entry in the location table. The problem is that this client sends the Re-REGISTER in a Expires/2 time, and as the RFC says, the 200ok is sent with the 2 contacts, with each expire, then the client receive this 2 contacts with 2 expires and "thinks" that it should renew the REGISTER for the old contact and sends a new REGISTER, with a new Call-ID, and adds another entry in the location table, which causes the response returns 3 contacts with 3 expires, and so on... There is some way to handle this issue? Thanks Lucio

2 1

[OpenSER-Users] how to use AVP variables in regular expressions
by Yazid Hadj Said 30 May '08

30 May '08

Hi, Please, I want to know how to use AVP variables in regular expressions with avp_subst? Thanks Yazid

2 1

[OpenSER-Users] Question about proxy_authenticate
by mark morreny 30 May '08

30 May '08

Hi, I am wondering if someone could help me to understand a bit more about the internal mechanism of proxy_challenge? If the UA is connecting to the server using the external ip, proxy_challenge fails, but if the UA is connecting to the server using the lan ip, proxy_challenge succeeds. What is the reason why this is happening? How does proxy_challenge work exactly? Thanks in advance for your help. Regards, Mark

2 1

[OpenSER-Users] Unable to t_relay (ERROR:tm:update_uac_dst)
by Pete Kay 30 May '08

30 May '08

Hi, After next_gw(), openser appends "64.231.23.234:5060;transport=tcp" as the domain of th ruri. When doing a t_relay with that ruri, I am getting ERROR:tm:update_uac_dst. before relay ruri = sip:86143455048@64.231.23.234:5060;transport=tcp starting to relay May 29 21:18:04 [12107] ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 2 (no corresponding listening socket) May 29 21:18:04 [12107] ERROR:tm:t_forward_nonack: failure to add branches However, if I do a t_relay with ruri= "sip:86143455048@64.231.23.234<sip%3A86143455048(a)64.231.23.234>", then the call can go through. Could anyone tell me why? A workaround that I can think of is to reconstruct the ruri to get rid of the port and the "transport=tcp" with something like: rewriteuri("sip:$rU@$rd"); But, the above line gives "sip:$rU@$rd" instead of the desired " sip:86143455048@64.231.23.234 <sip%3A86143455048(a)64.231.23.234>". Is there anyway of using psuedo variable in rewriteuri? Thanks alot in advance for your help. Regards, Pete

5 4

[OpenSER-Users] Managing MSILO responses
by Saúl Ibarra 30 May '08

30 May '08

Hi all, I'm having a little trouble setting up msilo: When I send a MESSAGE to an offline user, MSILO does it's job and stores it, and when the other user gets registered, he sees the MESSAGES. But if I enable onification, by putting: modparam("msilo", "registrar", "sip:softswitch@OPENSER_IP") MSILO repplies me sayng the user is offline, but hten I can't continue sending MESSAGEs, because they will be directed to the URI indicated by the registyrar parameter instead of the original user I wanted to send the MESSAGE. I'm using Twinkle 1.2. Thanks in advance!! -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/

2 2

[OpenSER-Users] BYE not relaying properly
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi All, I have OpenSER 1.2.3(previously it was 1.2.1 but installed latest version to make sure that it is not an OpenSER bug) configured and running for following features: * Registrar * Proxy * Media proxy for NAT traversal * LCR module for PSTN termination * SEMS for voicemail * Multi-leg accounting with Radiator radius * Call forwarding using user_preferences I'm having one really strange issue. My OpenSER is configured to listen on particular IP address say 1.2.3.4. I've added alias for the domain name say voip.domain.com. User agent register to OpenSER using voip.domain.com as a proxy. They can call each other & pstn. I'm using Media Proxy for handling media in case of clients behind NAT. Audio & signaling seems to be working fine in case of internal and pstn calling. I have configured call forwarding unconditional, no answer & busy. In case of unconditional call forwarding, audio and signaling works fine. But when I test call forwarding on no answer or busy, audio is proper but signaling doesn't work properly. Specially BYE. Here is my setup: * 3 users configured; 1000, 2000 & 3000. * 2000 has no answer forwarding to pstn number. * 3000 has busy forwarding to 2000 If 1000 calls 2000 and 2000 doesn't pickup the phone in 30 seconds(configured using fr_inv_timer param of tm module), the pstn number is pushed in request uri and request is sent to gateway for pstn termination. Audio works fine. If caller(1000) hangs up, the callee is hung up properly. However if callee (pstn number) hangs up the call, the caller doesn't receive the BYE message and hence stays connected. Media Proxy session is closed properly but in OpenSER, BYE request is not sent to caller but keeps looping in OpenSER instead. I have several doubts about this issue. One thing is that the pstn gateway is sending BYE to OpenSER IP address instead of domain name. I know this kind of problem can occur if we haven't configured alias parameter properly. Reference: http://www.openser.org/mos/view/-OpenSER-Installation-Notes/ (Section 4. Troubleshooting) I have added ip and domain in domain table as well but still no change. I've been facing this issue since several days and still not able to solve. I've done many OpenSER setups but having such issue for the first time. I've compared the OpenSER config file to my older setups but it seems proper. Can anyone please guide me in correct direction? Thanks, Ruchir Brahmbhatt

1 2

[OpenSER-Users] Can openser auto-retry if a call fails?
by Pete Kay 29 May '08

29 May '08

Hi, I know this kind of function exists in Asterisk, but I want to know if Openser can do it as well. Let's say there are two fail-over numbers(DID-b,DID-c) that associates with a DID(DID-a), if the original number(DID-a) that get dailed can't rearch the destination due to BUSY, NO RESPONSE, etc, is there anyway of having Openser to auto-retry with the next DID(DID-b)? If the next one fails, try the next next one(DID-c). One way I can think of is to do a lookup from DB for the alias in the fail_on_route section, but if I have multiple aliases, then I will have no way of *remembering* which alias has been tried unless I store it in the DB which is not a good solution. I am wondering if there any other way of doing it? Thanks in advance for all your kind suggestion. Regards, Pete

4 6

[OpenSER-Users] OPENSER Dinner Berlin, May 28
by Daniel-Constantin Mierla 29 May '08

29 May '08

Hello, bound to Asterisk Tag (http://www.asterisk-tag.org) and Linux Tag (http://www.linuxtag.org) events, Henning Westerholt and I will take care of an OPENSER group meeting in Berlin, Germany, May 28 at 20:00. The place is a small brewery next to Chartlottenburg Castle in Berlin, named Lusinen Brau, the website: http://www.luisenbraeu.com/ Organized in same idea of casual community meeting, everybody is welcome to join at any time. It is recommended to contact me or Henning to announce your participation so you got your seat reserved. Cheers, Daniel -- http://www.asipto.com

2 1

[OpenSER-Users] t_relay() issue when bad host name
by Iñaki Baz Castillo 29 May '08

29 May '08

Hi, perhaps I missing something but AFAIK if I do: rewritehostport("kk.kkkkkkkkkkaaa.bad"); append_branch(); xlog("L_INFO", "1111111111\n"); if (!t_relay()) { xlog("L_ERROR","222222222222\n"); } xlog("L_ERROR","3333333333\n"); Then, since "kk.kkkkkkkkkkaaa.bad" is a non existing domain I expect to see "222222222" because t_relay should reply a negative value: -4 bad destination (unresolvable address) But the fact is that I don't see that "2222222222", neither "3333333333" but this error: ---------- May 29 10:38:32 [16795] CRITICAL:core:mk_proxy: could not resolve hostname: "kk.kkkkkkkkkkaaa.bad" May 29 10:38:32 [16795] ERROR:tm:uri2proxy: bad host name in URI <sip:number_doesnt_exist@kk.kkkkkkkkkkaaa.new;transport=UDP;user=phone> May 29 10:38:32 [16795] ERROR:tm:t_forward_nonack: failure to add branches ----------- Why? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] Variables in regular expression
by Yazid Hadj Said 29 May '08

29 May '08

Hi Can I use AVP variables in regular expression? regards Yazid.

2 1

[OpenSER-Users] presence module - force subscription termination
by Sigrid Thijs 29 May '08

29 May '08

Hi, is it possible to terminate all presence subscriptions for a certain presentity through the current config script (1.3.x branch)? Or is there another easy way to accomplish this, without modifying the authorization rules? kind regards, Sigrid

2 1

[Serusers] Presence-rule handling in SER2.0
by pravin.swami＠wipro.com 29 May '08

29 May '08

Dear All, We are having problems with presence-rules handling of SER 2.0 Presence Server. +------+ +----------+ +------+ | | | Presence | | | | UE1 |--------------| Server |--------------| UE2 | | | | (PS) | | | +------+ +----------+ +------+ 1. UE1 & UE2 subscribe to "watcher-info". 2. UE1 SUBSCRIBEs to UE2's "presence" information. 3. PS sends info winfo-NOTIFY to UE2 (with UE1 status as pending). 4. UE2 fetches its presence-rules.xml document(from path /var/simulated-xcap-root/pres-rules/users/UE2/presence-rules.xml) 5. UE2 updates presence-rules.xml by adding UE1 into ALLOWLIST section & PUTs the document back to PS. 6. User changes presence status at UE2. UE2 sends PUBLISH request to PS. At this stage, PS should trigger presence NOTIFY to UE1. This isn't happening. (Another thing we observe is: Everytime watcher subscription is refreshed, UE2 receives NOTIFY with UE1 status as pending) Are we missing something in the above callflow? Should UE2 send any SIP/HTTP request after HTTP PUT presence-rules.xml ? Are we missing something when building presence-rules.xml ? We have modified the ser.cfg for presence authorization as per ser presence Handbook. modparam("pa", "auth", "xcap") modparam("xcap", "xcap_root", "http://localhost/xcap-root") Should I make any more changes to ser.cfg We have attached the ser.cfg & presence-rules.xml file that we are using. Please help us in solving this problem. Regards, -Pravin Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] how to force free-radius to use my own database
by Do Nguyen Ha 29 May '08

29 May '08

hi i have question on freeRadius i create my own database and now i want to use my own database instead of Openser database if yes, please show me which file i should edit Thanks Ha`

2 1

[OpenSER-Users] multiples rows in avp_db_query
by Yazid Hadj Said 29 May '08

29 May '08

Hi, When I use avp_db_query(), I only can access to to the first row by using simply the name of my variable ($avp(s:example)), but if I try to use for example ($avp(s:example[1],$avp(s:example[2]) , it indicate me that this variables isn't set. Regards Yazid

2 1

Re: [OpenSER-Users] Problem with Openser-freeradius communication [SOLVED]
by Pete Kay 29 May '08

29 May '08

Hi, I just figured it out. In the radgroupcheck, I had Local as my Value and then I changed it to "Digest", it is working fine now. Thanks a million for all of your help. Regards, Pete

2 1

Re: [OpenSER-Users] Problem with Openser-freeradius communication
by Dan-Cristian Bogos 29 May '08

29 May '08

Hi Pete, the Service-Type does matter only if u use it in your auth part. The reason you see "IAPP-Register" is due to radius dictionaries, responsible for translating between numbers and attributes (radius only transports numbers). Regarding the problem of FreeRADIUS not authorizing the requests I think it is a matter of which module do u want to perform that and configure that properly. I would recomed you should running freeradius in debug mode (-X ) and post more info here. Cheers, DanB On Wed, May 28, 2008 at 10:02 AM, Pete Kay <petedao(a)gmail.com> wrote: > Hi, > I am having problem with configuring FreeRadius to authenticate suername > and password. I have tried to test freeradius with a test client and it > works by sending the following message: > Ready to process requests. > User-Name = "1005(a)192.168.1.104" > User-Password = "1234" > Service-Type = Authenticate-Only > NAS-Port = 0 > NAS-IP-Address = 127.0.0.1 > > The message that is sent by Openser is as follows: > Waking up in 1.9 seconds. > User-Name = "1005(a)192.168.1.104" > Digest-Attributes = 0x0a0631303036 > Digest-Attributes = 0x010f3139322e3136382e312e313034 > Digest-Attributes = > 0x022a34383364376265396630623738663734363935343562386366373865303561316161623032633366 > Digest-Attributes = 0x04137369703a3139322e3136382e312e313034 > Digest-Attributes = 0x030a5245474953544552 > Digest-Response = "eb5f24a5563e9c1096855ef8b2c72d2b" > Service-Type = IAPP-Register > X-Ascend-PW-Lifetime = 825241654 > NAS-Port = 5060 > NAS-IP-Address = 127.0.0.1 > > Freeradius can't find the password and is reject users as a result of > that. Also, is the Service-Type supposed to be Authenticate-Only instead? > Is this something to do with my openser.cfg setup that is causing the > problem? > > Thanks in advance for your help. > > Regards, > Pete > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

3 9

[OpenSER-Users] Problem with t_relay after next_gw()
by Pete Kay 29 May '08

29 May '08

Hi, After running next_gw(), I can see that the ruri is being modified with the gateway ip as the domain. However, when doing a t_relay, I am getting the following error. From ngrep, I don't see any going package to the gateway ip address, so I am suspecting that the error is internal. before relay ruri = sip:86143455048@64.231.23.234:5060;transport=tcp starting to relay May 29 21:18:04 [12107] ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 2 (no corresponding listening socket) May 29 21:18:04 [12107] ERROR:tm:t_forward_nonack: failure to add branches Could someone tell me what could be the problem? Thenks in advance for all your help. Regards, Pete

1 0

[OpenSER-Users] How to differentiate auth level for gateway
by mark morreny 28 May '08

28 May '08

Hi, Assuming I have 3 gateways, GW-a, GW-b, and GW-c. All of them can REGISTER without username and password by having an entry in the trusted table. Is it possible to config Opernser so that GW-a and GW-b can route to GW-c (using INVITE) without username and password, but any INVITEs from GW-c will not be accepted? Thanks in advance for your suggestion. Regards, Mark

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Adrian Georgescu 28 May '08

28 May '08

Dan, you are right and I found the problem. The fix will be available with the next CDRTool release due next week together with Mediaproxy 2.0 Adrian If you need a fix now for your 6.4.1: --- rating_lib-6.4.1.phtml 2008-05-28 18:25:18.000000000 +0200 +++ rating_lib.phtml 2008-05-28 18:27:00.000000000 +0200 @@ -6078,26 +6078,28 @@ $timestamp=time(); } - $application="audio"; - if ($NetFields['application']) $application=$NetFields ['application']; + $CDRStructure=array ( + $this->CDRS->CDRFields ['callId'] => $NetFields['callid'], + $this->CDRS->CDRFields ['aNumber'] => $NetFields['from'], + $this->CDRS->CDRFields ['CanonicalURI'] => $NetFields['to'], + $this->CDRS->CDRFields ['duration'] => $NetFields['duration'], + $this->CDRS->CDRFields ['timestamp'] => time() + ); + + $CDR = new $this->CDRS->CDR_class(&$this->CDRS, & $CDRStructure); + $CDR->normalize(); $Rate = new Rate($this->settings, $this->db); $RateDictionary=array( - 'callId' => $NetFields ['callid'], - 'timestamp' => $timestamp, - 'duration' => $NetFields ['duration'], - 'inputTraffic' => $NetFields ['inputtraffic'], - 'outputTraffic' => $NetFields ['outputtraffic'], - 'DestinationId' => $NetFields ['destinationid'], - 'BillingPartyId' => $NetFields ['from'], - 'domain' => $NetFields ['domain'], - 'gateway' => $NetFields ['gateway'], - 'aNumber' => $NetFields ['from'], - 'cNumber' => $NetFields ['to'], - 'ENUMtld' => $NetFields ['enumtld'], - 'applicationType' => $application, - 'RatingTables' => &$this->CDRS- >RatingTables, + 'duration' => $CDR->duration, + 'callId' => $CDR->callId, + 'timestamp' => $CDR->timestamp, + 'DestinationId' => $CDR- >DestinationId, + 'domain' => $CDR->domain, + 'BillingPartyId' => $CDR- >BillingPartyId, + 'ENUMtld' => $CDR->ENUMtld, + 'RatingTables' => &$this->CDRS- >RatingTables ); $Rate->calculate($RateDictionary);

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Adrian Georgescu 28 May '08

28 May '08

Dan I have no changed anything related to this. Usually I write in the changelog about such dramatic changes. What did you notice and between which versions? Adrian >>>>>>>>> I have noticed that recently ShowPrice behavior has changed, and since did not find it in Changelog I decided to try my shoot over here . There are no more queries based on destination number but DestinationId. Was the support for destination number lost for good, or only temporarily? If that was done, why is "To=" parameter still mandatory? The presence or absence of it does not make any different in queries, as far as I can see. My version: version CDRTool version 6.4.1 Ta, DanB

2 1

[Serusers] ser_ctl question
by Steven C. Blair 28 May '08

28 May '08

Hello: I'm just starting to make a serious attempt to build a SER v2 system. SER is built, ser_mysql has been executed and the MySQL tables defined. With mysql -user -p I can access the database with ser_ctl add domain I cannot. I get the following error. DB_URI in serctl.conf was changed to reflect the new password for the ser user. Yet I still get the following error. Any ideas why? ser_ctl domain add myschool.edu ser_ctl: (1045, "Access denied for user 'ser'@'localhost' (using password: YES)"): OperationalError Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

1 1

[OpenSER-Users] Save Contact List in Server
by Habib, Yeasin 28 May '08

28 May '08

Hi Everybody, Is there any plug-in available for SIP Communicator to save contact list in the server side. I mean when the SIP Communicator register itself to server the server will save to contact list to its own persistence storage. So the client doesn't need a local copy of contact list. I am using OpenIMSCore and OpenSER. I am just wondering whether this functionality is already available in SIP Communicator or not? If not then how much work has to be done to implement in properly? I am just looking for some idea about this. So that I can make an estimation about this implementation. I am looking forward to hearing from all of you. Best regards Yeasin Habib

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Dan-Cristian Bogos 28 May '08

28 May '08

Guys, I have noticed that recently ShowPrice behavior has changed, and since did not find it in Changelog I decided to try my shoot over here . There are no more queries based on destination number but DestinationId. Was the support for destination number lost for good, or only temporarily? If that was done, why is "To=" parameter still mandatory? The presence or absence of it does not make any different in queries, as far as I can see. My version: version CDRTool version 6.4.1 Ta, DanB

2 1

[OpenSER-Users] New tables in openser DB
by Yazid Hadj Said 28 May '08

28 May '08

Hi, If I add tables to openser database with an other tool (e.g phpMyAdmin), is it possible to create me some problems? Regards, Yazid

2 1

[Serusers] presence status - issue when Authorization is enabled in SER 2.0.
by pravin.swami＠wipro.com 28 May '08

28 May '08

Hi... All , I am facing problem with SER presence status, when authorization is enabled. B elow is the scenario of my test. 1) PUA1 & PUA2 are two user agents subscribed to SER (2.0) presence server. 2) PUA1 adds PUA2 into its buddly list . PUA2 gets Watcher-info notification from PUA1 with subscription - state pending. 3) PUA2 accepts PUA1 and presence-rules.xml file is updated for PUA2 . In whitelist (Allow List)," <id PUA1(a)ser.org <mailto:PUA1@ser.org> <id> " is added successfully. problem statement : PUA2 status is offline in PUA1 's buddy list. I f PUA2 changes its status (sends PUBLISH with away/on phone/busy activity ) , SER presence sends 200 OK success response , but no notifcation will be generated from server to watcher PUA1. - also when PUA2 logs out and logs in again Wathcer-info subscription will be generated with pending state though presence-rules.xml document was already updated for this. I have modified the ser.cfg for presence authorization below as per ser presence Handbook . modparam("pa", "auth", "xcap") modparam("xcap", "xcap_root", "http://localhost/xcap-root <http://localhost/xcap-root> ") I have attached the ser.cfg file. which i tried ... but doesn't work out. let me know any of your comments. regards, Pravin . Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] Problem with Openser-freeradius communication
by Pete Kay 28 May '08

28 May '08

Hi, I am having problem with configuring FreeRadius to authenticate suername and password. I have tried to test freeradius with a test client and it works by sending the following message: Ready to process requests. User-Name = "1005(a)192.168.1.104" User-Password = "1234" Service-Type = Authenticate-Only NAS-Port = 0 NAS-IP-Address = 127.0.0.1 The message that is sent by Openser is as follows: Waking up in 1.9 seconds. User-Name = "1005(a)192.168.1.104" Digest-Attributes = 0x0a0631303036 Digest-Attributes = 0x010f3139322e3136382e312e313034 Digest-Attributes = 0x022a34383364376265396630623738663734363935343562386366373865303561316161623032633366 Digest-Attributes = 0x04137369703a3139322e3136382e312e313034 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "eb5f24a5563e9c1096855ef8b2c72d2b" Service-Type = IAPP-Register X-Ascend-PW-Lifetime = 825241654 NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Freeradius can't find the password and is reject users as a result of that. Also, is the Service-Type supposed to be Authenticate-Only instead? Is this something to do with my openser.cfg setup that is causing the problem? Thanks in advance for your help. Regards, Pete

1 0

[OpenSER-Users] Question on modification A-number & B-number
by Do Nguyen Ha 28 May '08

28 May '08

hi all i read several docs of Openser, i didnt find the information to modify the A-number & B-number does openser have the ability to modify A-number & B-number if yes, please send me the link to this Thanks Ha`

2 1

[OpenSER-Users] SIP Text Message to SMSC
by Dome Charoenyost 27 May '08

27 May '08

Dear All, I use Openser for test SIP Text Message from IP phone to IP phone. it's work perfect :) Look like only openser support this feature light now. Asterisk not support and Freeswitch fail also. (I use yealink ip phone). So now my customer want to relay SIP text Message to PSTN via SMSC Can someone tell me is posible to use openser SMS module send SIP text Message to SMSC ? Best Regards. Dome C.

1 0

[OpenSER-Users] Me too: OpenSER 1.3.2 is unusually slow
by Peter Lemenkov 27 May '08

27 May '08

Hello All! After upgrading to 1.3.2 from 1.3.1 I'm encountered (similar to described in that list already) issue, e.g. openser answers unreliably slow to all messages (~40 seconds delay). DNS lookup related fix doesn't helps. I'll try to find what exactly caused this delay (if it's openser-related issue and not something else). -- With best regards!

3 4

[OpenSER-Users] OpenSER corrects the "Contact" field only one way.
by Höskuldur 27 May '08

27 May '08

Hi I have two user clients behind separate NAT networks. My OpenSER is directly connected to the Internet and has it's own external IP address. Both clients are using STUN to get their external IP address and correct port for SIP messages. They successfully register with Openser, which is using Nathelper module and Mediaproxy. In this case, since the client is behind NAT, it puts the wrong port in the Contact field of a SIP message. As a result, SOME SIP messages from OpenSER are never received by the client. But with the help of Nathelper, OpenSER should correct the port in Contact field, by replacing the NAT port with the src port of a SIP message (UDP pack). The problem is, that it looks like OpenSER only corrects the port in Contact field for client A (The one who initiates the call) but not Client B(The one who's called). For a better explaination, please check out my logs from ngrep. CLIENT A to OPENSER U 2008/05/26 16:53:26.285586 157.157.78.173:54182 -> 212.50.206.33:5060 INVITE sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> SIP/2.0. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33>>. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1006<sip:1006@157.157.78.173:62356>. !!!NB!!! <-- This is the WRONG reply port. OPENSER RELAYS TO CLIENT B U 2008/05/26 16:53:26.301864 212.50.206.33:5060 -> 130.208.183.251:49211 INVITE sip:1001@130.208.183.251:49208 SIP/2.0. Record-Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33>>. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.0;rport. Via: SIP/2.0/UDP 157.157.78.173:62356 ;rport=54182;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1006<sip:1006@157.157.78.173:54182>. !!!NB!!! <-- OpenSER detects it and changes it to correct reply port. Works great. CLIENT B ANSWERS TO OPENSER U 2008/05/26 16:53:27.301368 130.208.183.251:49211 -> 212.50.206.33:5060 SIP/2.0 200 OK. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.0. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1001<sip:1001@130.208.183.251:49208>. !!!NB!!! <-- Now lets try Client B. Wrong reply port. OPENSER TO CLIENT A U 2008/05/26 16:53:27.305241 212.50.206.33:5060 -> 157.157.78.173:54182 SIP/2.0 200 OK. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1001<sip:1001@130.208.183.251:49208>. !!!NB!!! <-- OpenSER didn't change the port like before, so now Client B never receives ACK from Client A. CLIENT A TO OPENSER U 2008/05/26 16:53:27.518621 157.157.78.173:54182 -> 212.50.206.33:5060 ACK sip:1001@130.208.183.251:49208 SIP/2.0. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261505. Contact: 1006<sip:1006@157.157.78.173:62356>. Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. OPENSER TO CLIENT B U 2008/05/26 16:53:27.519192 212.50.206.33:5060 -> 130.208.183.251:49208 !!!NB!!! <-- OpenSER sends to the WRONG port! ACK sip:1001@130.208.183.251:49208 SIP/2.0. Record-Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. CSeq: 101 ACK. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.2;rport. Via: SIP/2.0/UDP 157.157.78.173:62356 ;rport=54182;branch=z9hG4bKc0a8150a10007e0026261505. This last ACK message is never received by Client B because the client isn't listening on port 49208. If I manually use rewriteport("49211") it works, because obviously this last package is then forced to port 49211 which the client listening on. Also, the first line in my main route is force_rport(); but that still doesn't do it. I've been struggling with this problem for a few weeks now so all comments and suggestions are very welcome. My OpenSER config file can be found here: http://hoski.public.is/openser.cfg Best regards, Höskuldur

2 4

[OpenSER-Users] need help on Radius --
by Do Nguyen Ha 27 May '08

27 May '08

Hi all i am intent to use Radius for prepaid but i need more information of RADIUS, how to limit the call, how the RADIUS work together with Openser could you suggest where i got it Thanks Ha

3 2

[OpenSER-Users] How to set the "radius.seq" file (posible bug) ?
by Iñaki Baz Castillo 27 May '08

27 May '08

Hi, I use FRadius accounting. OpenSer tries to use a file /var/run/radius.seq but it fails when restarting computer since OpenSer init script creates this file as: /var/run/openser/openser_radius.seq create_radius_seqfile () { # Create a radius sequence file to be used by the radius client if # radius accounting is enabled. This is needed to avoid any issue # with the file not being writable if openser first starts as user # root because DUMP_CORE is enabled and creates this file as user # root and then later it switches back to user openser and cannot # write to the file. If the file exists before openser starts, it # won't change it's ownership and will be writable for both root # and openser, no matter what options are chosen at install time RADIUS_SEQ_FILE=/var/run/openser/openser_radius.seq I don't find where to change this file in OpenSer script. Isn't a bug if default value is: /var/run/radius.seq but OpenSer init script (Debian Etch) creates it in: /var/run/openser/openser_radius.seq ? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] bridging two existing channels
by mark morreny 27 May '08

27 May '08

Hi, In my scenario, I have UA1 -> Openser -> B2BUA -> Openser -> UA2. The reason why I need B2BUA to originate the channel to UA2 is because it is going to play a wav file when UA2 answers the call. After the call has been answered by UA2 and dialog is in progress, is it possible for B2BUA to notify Openser that it wants to be out of the look, so the scenario can before: UA1 -> Openser -> UA2 without UA2 hanging up? How would I be able to config that? Any help will be greatly appreciated. Thanks, Mark

2 1

[OpenSER-Users] obtain content of header dynamically
by Yazid Hadj Said 27 May '08

27 May '08

Hi, I try to obtain content of header dynamically, but with $hdr(...) i can't do this because it accept only static parameters. For example I try to use: $avp(s:body)=$mb; $avp(s:HdName)="To"; avp_subst("$avp(s:corps)/$avp(s:HdContent)","/^$avp(s:HdName):(.*)\r\n(.*)/\1/"); With this manner, i can change header name ($avp(s:HdName) )dynamically from data source (e.g database) and use his content in my script. But until now i don't obtain what i want. Regards, Yazid

2 1

[OpenSER-Users] Fixing wrong SDP with "textutils" module (LF -> CRLF)
by Iñaki Baz Castillo 27 May '08

27 May '08

Hi, I'm handling with a Nortel CS2K softswitch that sends me incorrect SDP's since it uses LF at the end of each line in SDP instead of CRLF (as it's mandatory according to the SDP grammar). AFAIK some phones can fail when receiving this incorrect SDP. Unfortunatelly I can't try a workaround now but I'm thinkink about using "textmodules" in this way: if ($si == NORTEL_CS2K_IP) replace_body("\n$", "\r\n"); By your experience, should it work? I assume that "Content-Length" is automatically updated before the message leaves OpenSer. So just it, should it work? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 6

[OpenSER-Users] lcr problem - openser 1.3
by Ion Minzu 27 May '08

27 May '08

hello i use openser 1.3 But I have trouble NOTICE:core:main: version: openser 1.3.2-notls (i386/freebsd) ERROR:lcr:reload_gws: Unknown or unsupported transport <0> CRITICAL:lcr:mod_init: Failed to reload gateways and routes ERROR:core:init_mod: failed to initialize module lcr ERROR:core:main: error while initializing modules ####config openser.cfg##### loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://sqluser:ser@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", "$avp(i:711)") modparam("lcr", "fr_inv_timer_avp", "$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) ################## What is problem? Thanks

2 1

[OpenSER-Users] problem nat, openser 1.3 + mediaproxy +gateway
by Ion Minzu 27 May '08

27 May '08

Hello, I use openser 1.3 + mediaproxy and all work ok. But I have trouble calling to PSTN gateway, where is no voice both ways. route[7] is the route to gateway Thanks ############################################## listen = udp:my.openser.server:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes #group = "openser" #user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no 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 = no 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://sqluser:ser@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) 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", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) #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", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "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", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) #for accounting loadmodule "acc.so" modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("acc", "report_ack", 0) ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[1] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$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=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("1"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$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[4] { sl_send_reply("100", "Trying"); 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(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$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=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } #go to pstn if (uri=~"sip:[2-8][0-9][0-9][0-9][0-9][0-9]+@.*") { prefix("400"); t_relay("my.pstn.gateway:5060"); end_media_session(); exit; } #finish xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$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=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$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 'base-standard-failure' ######################################################################## failure_route[1] { route(10); route(1); }

1 0

[OpenSER-Users] Res: Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Hi Andel Yes, it would be possible. But I'd like to hide the code even the root user. Like Bogdan said, I want "way to run OpenSER with script that it is not plain text or accessible for humans". Your suggest works as well when the another users do not have the root password. Anyway, thanks for your comment, Andel. ----- Mensagem original ---- De: Ladislav Andel <ladia6(a)centrum.cz> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)lists.openser.org Enviadas: Segunda-feira, 26 de Maio de 2008 7:18:00 Assunto: Re: [OpenSER-Users] Hiding the config code How about setting the user rights in your unix-based system ? Lada Bruno Machado wrote: > Hi all > > I have been wondering if is possible to hide the Openser's config file > or do anything else to avoid reading by another people. Possible? > > Tks > > Bruno > > ------------------------------------------------------------------------ > Abra sua conta no Yahoo! Mail > <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, > o único sem limite de espaço para armazenamento! > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

[OpenSER-Users] Res: Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Exactly, Bogdan. It would be a good feature for OpenSER considering security purposes. Sometimes, you need to hide the script from some people, even from the same company. I think that it would be a good feature. Thanks Bogdan ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Ladislav Andel <ladia6(a)centrum.cz> Cc: Bruno Machado <bsmufu(a)yahoo.com.br>; users(a)lists.openser.org Enviadas: Segunda-feira, 26 de Maio de 2008 7:48:41 Assunto: Re: [OpenSER-Users] Hiding the config code Well, that's a limited hiding - I understood the question as "if it a way to run OpenSER with script that it is not plain text or accessible for humans. Regards, Bogdan Ladislav Andel wrote: > How about setting the user rights in your unix-based system ? > Lada > > Bruno Machado wrote: > >> Hi all >> >> I have been wondering if is possible to hide the Openser's config file >> or do anything else to avoid reading by another people. Possible? >> >> Tks >> >> Bruno >> >> ------------------------------------------------------------------------ >> Abra sua conta no Yahoo! Mail >> <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, >> o único sem limite de espaço para armazenamento! >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

Re: [OpenSER-Users] Possible problem with script presence-create.sql
by Florent Le Saout 26 May '08

26 May '08

Hello all, I also found the reason reading precisely the documentation. I use the line modparam("presence_xml", "xcap_table", "xcaps") in my configuration as Anca Vamanu said. All is clear now. Thank for your help. Florent. > Message du 26/05/08 12:20 > De : "Henning Westerholt" <henning.westerholt(a)1und1.de> > A : "Anca Vamanu" <anca(a)voice-system.ro> > Copie à : "Florent Le Saout" <le-saout.florent(a)laposte.net>, users(a)lists.openser.org > Objet : Re: [OpenSER-Users] Possible problem with script presence-create.sql > > On Monday 26 May 2008, Anca Vamanu wrote: > > Indeed the name of the script was wrong in the documentation and I have > > fixed it. > > The name of the xcap table was not misspelled as 'xcaps'. > > /The module requires 1 table in OpenSER database: xcap. / > > You probably refer to the line with the example of how to set the xcap > > table name module parameter: > > / 'modparam("presence_xml", "xcap_table", "xcaps")'. / > > This should be used if you what the table to have a different name, and > > of course, you should use the same when creating the database table. > > Thank you Anca for the clarification, and the fix. > > Henning > >

1 0

[OpenSER-Users] openSER with freeRADIUS
by Patrick Pichler 26 May '08

26 May '08

Hi everybody, does anyone has got a tested documention on setting up freeRADIUS to work together with openSER. I'm currently having problems with the authentication via a MySQL DB. I didn't get it how to tell freeRADIUS to take the users credentials from the "subscribers" table. Could anybody help me dealing eith this problem? My environment: openSER with radiusclient-ng, mysql and freeradius on a single machine, all are up and running so far. Without RADIUS it's prossible for a UAC to register...but with RADIUS authentication via MySQL the UAC just gets a "401 unauthorized". Thanks in advance. Patrick

2 3

[OpenSER-Users] [OT] Why the project name in SourceForge is "Scalable SIP server"?
by Iñaki Baz Castillo 26 May '08

26 May '08

Hi, if you get into: https://sourceforge.net/projects/openser/ you won't read the word "OpenSER" but "Scalable SIP server". Also, in the "location" line we read: SF.net > Projects > Scalable SIP server > Summary Scalable SIP server I wonder: why not name it as OpenSER? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 4

[OpenSER-Users] Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Hi all I have been wondering if is possible to hide the Openser's config file or do anything else to avoid reading by another people. Possible? Tks Bruno Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

3 3

[OpenSER-Users] serweb-0.9.6 on openser-1.3.2
by Zen Kato 26 May '08

26 May '08

I installed serweb-0.9.6 on openser-1.3.2, but when I enter username and password, the following Warning comes out. How should I solve this problem? -------------------------------------------- http://localhost/openserweb/user/index.php username:0103@192.168.0.3 password:xxxxx --------------------------------------------- ----------------------------------------------------------------------------- http://localhost/openserweb/user/my_account.php?kvrk=483a8e78876b5 ( ! ) Warning: fopen(/tmp/openser_fifo) [function.fopen]: failed to open stream: Permission denied in /var/www/openserweb/serweb-0.9.6/html/functions.php on line 345 ----------------------------------------------------------------------------- /var/www/openserweb/serweb-0.9.6/html/functions.php (L:344-348) /* open fifo now */ $fifo_handle=fopen( $config->fifo_server, "w" ); if (!$fifo_handle) { $errors[]="sorry -- cannot open write fifo"; return false; } -- Zen Kato

1 0

[OpenSER-Users] Possible problem with script presence-create.sql
by Florent Le Saout 26 May '08

26 May '08

Hello, I'm a new user of openser. (I use openser 1.3.1-3+b1 from package of debian testing) I want to use the module of presence. If I refer to documentation I read that I have to use presence_xml-create.sql to create the tables. But I can't find it. I use the script presence-create.sql which seems to replace it. I create a table named xcap but the module need xcaps, if I create the same table renamed xcaps that work well. I think there is a mistake of naming of the table xcap(s). I take a look to 1.3.x of your repository and I see the same thing in the sql script. Thank you for your job. Best regards. Florent le Saout. Créez votre adresse électronique prénom.nom(a)laposte.net 1 Go d'espace de stockage, anti-spam et anti-virus intégrés.

3 3

[OpenSER-Users] Acceptor peer failed establishing state[3003], closing
by bhargav.b＠renovau.com 26 May '08

26 May '08

actually i am authenticating a user using openser and opendiameters NASREQ application as the client and server.at present the openser is sending the all needed avp's to opendiameter client.i can see all the avp code and avp data on the client side. my problem is that the on the diameter server side the avp,s are not transferred by the client. the client is giving something like (6372|1698151312) Acceptor peer failed establishing state [3003], closing i am confused about it.by reading some drafts i came to know that the error is due to realm mismatch.but i am sending the same realm as the opendiameter's realm.in both the cases it is [bhargav.s]. before i sent the avp's from openser side there was no such problem showing on the client side.if anybody know the solution please help me regards bhargav.b(a)renovau.com

2 1

[OpenSER-Users] How to determine that a reply is a retransmission?
by Jon Steer 26 May '08

26 May '08

Environment: Openser 1.3.1-TLS Fedora 7 I have a module that performs a heavyweight operation on Invites and their respective Replys. I would prefer not to do the operation again if there is a retranmission of the Reply. How can I determine in a REPLY-ROUTE if a reply has already been seen? $T_reply_code would work, if on entry to the first reply route it was zero, but I suspect it is the value of the current reply. thanks, jon

2 1

[OpenSER-Users] Centos 5.1 or Debian4.0.r0 for Openser server
by fatmeh fatah 26 May '08

26 May '08

Hello: May you comparison Centos 5.1 and Debian4.0.r0 for Openser server? Thank you.

3 2

[OpenSER-Users] Click to Call functionality
by Darren Sessions 26 May '08

26 May '08

Quick question . . >From what I've been able to gather, the fifo db functionality has been removed since 1.2.x. What is another easy way to do click-to-call functionality now that it is gone? Thanks, - Darren

2 1

Re: [OpenSER-Users] Problems with sl module in openser1.3.1
by Arturo Ochoa 26 May '08

26 May '08

On Friday 23 May 2008, arturo arturo wrote: > Could some help me with this problem, I'm trying to run openser 1.3.1, but > it dies complaining about the sl module. I issued just 'make all' command > before this > > root@zeus:/home/aochoa/openser/openser-1.3.1-notls > # ./openser > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/mysql.so > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/sl.so > May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags > mismatch for* /usr/local/lib/openser/modules/sl.so > core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Hi Arturo, the version of the sl module installed on your machine don't match the one of your server in the home directory. Do a make distclean; make all in the source directory, and install again. Cheers, Henning Thanks for your response Henning, I've already do what you mention, but the problem persists. I tried also with openser 1.3.2, and the same thing. Also I noted that there are different linked libs on the same module but compiled on different machines: Fedora: [root@apolo] /usr/local/lib/openser/modules # ldd sl.so linux-gate.so.1 => (0x0028b000) libc.so.6 => /lib/libc.so.6 (0x00111000) /lib/ld-linux.so.2 (0x0028c000) Centos: (this is the one with the problem) root@zeus:/home/aochoa/openser # ldd /usr/local/lib/openser/modules/sl.so libc.so.6 => /lib/tls/libc.so.6 (0x00149000) /lib/ld-linux.so.2 (0x003f7000) Any comments?

2 1

[OpenSER-Users] CANCEL not always beeing relayed
by Tobias Lindgren 26 May '08

26 May '08

Hi all, I have some problems with CANCELs not always beeing relayed from A to B via an OpenSER 1.3.1. When I try to make a call and CANCEL it with my own equipment, everything works just fine. OpenSER receives CANCEL and passes it on to the PBX. However, some customer calls that are beeing CANCELed are not always relayed by OpenSER and I cannot find out why. They are received by OpenSER but never sent forward. I have sip-logs and OpenSER debugs, are there someone with more understanding of the OpenSER-code that can take a look at them? Br, /Tobias

2 1

Re: [OpenSER-Users] call failed 403 forbiden
by luzango mfupe 26 May '08

26 May '08

Hi mates, I still need your pointers regarding my problem in this post, today i have attached the routes suspected to be involved in this saga. From my config file plz see below. # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local()) { # We prevent unauthorised relays "clever guys we got u" append_hf("P-hint: outbound\r\n"); route(10); # need to be authenticated } else { sl_send_reply("403", "Forbidden"); }; return; }; route[10] { #------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds");# if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; # consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Busted!!!, you are not allowed this route"); # }; return; } With Best Regards, LU. > > Message: 3 > Date: Tue, 20 May 2008 17:38:50 +0200 > From: "luzango mfupe" <luzango.mfupe(a)gmail.com> > Subject: [OpenSER-Users] Call failed 403 Forbiden > To: users(a)lists.openser.org > Message-ID: > <9cdd611a0805200838oc11cfedg9762b7451bf543c4(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi mates, > Everytime i do try to make my 2 xlite clients talk (which i correctly added > them into the database), i encountered with the same problm, Openser > perfoms > authentication and return call failed 403 forbiden signal. > > My setup comprise of the first box with openser 1.3 and mediaproxy the > second box with Mysql and two Xlite clients in two other boxes all are in > an > internal network, as far as am concerned my NetAdmin have already opened > ports 5060 and 3306 for me. I need your right direction on this probm. > below is my ngrep snapshot > > mzee:/# ngrep -d eth1 -W byline port 5060 > interface: eth1 (168.172.200.0/255.255.255.0) > filter: (ip or ip6) and ( port 5060 ) > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > INVITE sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87> > >SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > Max-Forwards: 70. > Contact: <sip:dreamteam@168.172.200.70:1824>. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>>>. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Session-Expires: 95. > Min-SE: 90. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, > INFO. > Content-Type: application/sdp. > Supported: timer. > User-Agent: X-Lite release 1011s stamp 41150. > Content-Length: 426. > . > v=0. > o=- 9 2 IN IP4 168.172.200.70. > s=CounterPath X-Lite 3.0. > c=IN IP4 168.172.200.70. > t=0 0. > m=audio 52166 RTP/AVP 107 119 100 106 0 105 98 8 101. > a=alt:1 1 : uZB2dYm+ NKBRK8Ep 168.172.200.70 52166. > a=fmtp:101 0-15. > a=rtpmap:107 BV32/16000. > a=rtpmap:119 BV32-FEC/16000. > a=rtpmap:100 SPEEX/16000. > a=rtpmap:106 SPEEX-FEC/16000. > a=rtpmap:105 SPEEX-FEC/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > # > U 168.172.200.87:5060 -> 168.172.200.70:1824 > SIP/2.0 403 Forbidden. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Server: OpenSER (1.3.0-notls (i386/linux)). > Content-Length: 0. > . > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > ACK sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 ACK. > Content-Length: 0. > > WBR, > LU. > >

2 1

[OpenSER-Users] Problem with BYE signaling during no answer call forward to PSTN
by Krunal Patel 26 May '08

26 May '08

Hi, Here is the scenario of the call. 12345 ---> 56565656 ----> PSTN I have registered user 12345 using sjphone. Dialing 56565656 which is registered to the same openser. Now i let the call ring & after timeout call is transffered to PSTN number. Which is terminated by PSTN gateway. I am using mediaproxy for audio signaling & getting audio properly both side. Now the problem is when PSTN number hangsup the call,it should send BYE signal to the domain name. But instead of sending BYE signal to domain name it sends it to the IP. & because of that the UA can't acknowledge the BYE signal & stay connected. & BYE keeps looping in openser. As the pstn number sends BYE to IP , I have tried by setting alias=xxx.xxx.xxx.xxx . Eventhough it was not solved. Thanks in advance -- Regards, Krunal Patel

2 1

[OpenSER-Users] compilation on solaris 10
by wliangy＠gmail.com 26 May '08

26 May '08

Hi experts, I am start user of openser and need use openser to support sctp. now I have download the source codes of openser-1.3.1-tls. then I compiled it, there have many errors that need your supports. e.g. openser-1.3.1-tls> make all Makefile.defs:717: *** missing separator. Stop. then I commented Makefile.defs line717 and 718. # $(warning Old gcc detected ($(CC_SHORTVER)), use gcc >= 3.1 \ # for better results) Errors were still there. openser-1.3.1-tls> make all Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' yacc -d -b cfg cfg.y "cfg.y", line 391: warning: redeclaration of precedence of SLASH. conflicts: 1 shift/reduce flex cfg.lex statistics.c:58: warning: #warning STATISTICS: Architecture with no support for atomic operations. Using Locks!! Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' Compiling action.c gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o parser/msg_parser.h: In function `char_msg_val': In file included from action.h:27, from action.c:44: parser/msg_parser.h:300: parse error before `)' forward.h: In function `msg_send': In file included from action.c:49: forward.h:95: parse error before `)' forward.h:103: parse error before `)' forward.h:107: parse error before `)' mem/../lock_alloc.h: In function `lock_set_alloc': In file included from mem/../locking.h:66, from mem/../statistics.h:115, from mem/shm_mem.h:33, from ut.h:52, from action.c:54: mem/../lock_alloc.h:68: parse error before `)' ut.h: In function `int2bstr': In file included from action.c:54: ut.h:172: parse error before `)' ut.h: In function `un_escape': ut.h:365: parse error before `)' ut.h: In function `shm_str_dup': ut.h:498: parse error before `)' ut.h: In function `pkg_str_dup': ut.h:515: parse error before `)' ut.h: In function `str_strcmp': ut.h:532: parse error before `)' ut.h:535: parse error before `int' ut.h:539: `minlen' undeclared (first use in this function) ut.h:539: (Each undeclared identifier is reported only once ut.h:539: for each function it appears in.) ut.h:547: `alen' undeclared (first use in this function) ut.h:547: `blen' undeclared (first use in this function) ut.h: In function `str_strcasecmp': ut.h:563: parse error before `)' ut.h:566: parse error before `int' ut.h:570: `minlen' undeclared (first use in this function) ut.h:578: `alen' undeclared (first use in this function) ut.h:578: `blen' undeclared (first use in this function) action.c: In function `run_action_list': action.c:140: parse error before `)' action.c: In function `do_assign': action.c:193: parse error before `)' action.c:202: parse error before `)' action.c:222: parse error before `)' action.c:228: parse error before `)' action.c: In function `do_action': action.c:302: parse error before `)' action.c:309: parse error before `)' action.c:335: parse error before `)' action.c:498: parse error before `)' action.c:524: parse error before `)' action.c:682: parse error before `)' action.c:710: parse error before `)' action.c:723: parse error before `)' action.c:757: parse error before `)' action.c:763: parse error before `)' action.c:807: parse error before `)' action.c:819: parse error before `)' action.c:870: parse error before `)' action.c:896: parse error before `)' action.c:904: parse error before `)' action.c:930: parse error before `)' action.c:939: parse error before `)' action.c:945: parse error before `)' action.c:966: parse error before `)' make: *** [action.o] Error 1 Your any help is much appreciated! BR, Eric

2 1

[OpenSER-Users] Need help to solve BYE signalling problem
by Krunal Patel 26 May '08

26 May '08

Hi, ++ I have registered 3 UACs to the same openser server which are behind NAT. UAC 1 -- 111111 UAC 2 -- 222222 UAC 3 -- 333333 ++ So in location table , I am getting : ------------------------------------------------+-----------------------+-------------------------- Contact | Received | ------------------------------------------------+-----------------------+-------------------------- 111111@<private-IP>:5060 | sip:public-IP : 5060| 222222@<private-IP>:5060 | sip:public-IP : 5060| 333333@<private-IP>:5060 | sip:public-IP : 5060| ++ I have set user preferences : +------+------------------+-------------------------+-------------+------+------------------------------------------------------+ | uuid | username | domain | attribute | type | value | +------+------------------+-------------------------+-------------+------+------------------------------------------------------+ | | 222222 | domainname.com | fwdbusy | 0 | fwdbusy:sip:333333@domainname.com <fwdbusy%3Asip%3A333333(a)domainname.com> | | 222222 | domainname.com | noanscfwd | 0 | noanscfwd:sip:333333@domainname.com<noanscfwd%3Asip%3A333333(a)domainname.com> +------+------------------+-------------------------+--------------+------+------------------------------------------------------+ ++ I am using mediaproxy for audio signaling. ++ Now I am dialing 222222 using 111111 & keeps 222222 ringing until it stops automatically. So call is being forwarded to 333333. 111111 ---------------> 222222---------------> 333333 ++ I pick up the call . So 111111 & 333333 is connected. ++ Now I hangup 333333 , so it sends BYE request to 111111. + Now the problem is 111111 do not accept the BYE request. So it stays connected. & the BYE request keeps looping infinitly in openser. Here is the BYE request : Message Body: BYE sip:111111@<server-IP>:5060 SIP/2.0 Via: SIP/2.0/UDP <111111's Private IP>:5060;branch=z9hG4bK-ed33673a;rport From: <sip:222222@domainname.com <sip%3A222222(a)domainname.com> >;tag=632a4773b79ebfa8i1 To: Anonymous <sip:111111@domainname.com <sip%3A111111(a)domainname.com> >;tag=26190b20d13c8197o0 Call-ID: 4d753c94-f366deb3@localhost CSeq: 101 BYE Max-Forwards: 70 Route: <sip:[SERVER'sIP]:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on>, <sip:[SERVER's IP]:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on> User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 0(5877) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5877) #############################record_route############################### 0(5877) #######################After record_route########################3 0(5877) ------ ------ bye ---call tear down --- 0(5877) parse_headers: flags=78 0(5877) ##################################Before loose_route########################## 0(5877) parse_headers: flags=200 0(5877) is_preloaded: No 0(5877) grep_sock_info - checking if host==us: 13==13 && [SERVER IP] == [SERVER IP] 0(5877) grep_sock_info - checking if port 5060 matches port 5060 0(5877) after_strict: Next hop: 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' is loose router 0(5877) parse_headers: flags=ffffffffffffffff 0(5877) rewrite_uri: Rewriting Request-URI with 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' 0(5877) after_strict: The last route URI: 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' 0(5877) ##################################into loose_route########################## 0(5877) DEBUG: has_totag: totag found 0(5877) DEBUG: has_totag: totag found 0(5877) check_via_address(PUBLIC IP of 111111, PRIVATE IP of 111111, 0) 0(5877) ##############################client_nat_test or search #########BYE############################## 0(5877) ###########################setbflag6################################# 0(5877) ------ ------ bye - loose route ----- 0(5877) DEBUG: t_newtran: T on entrance=0xffffffff 0(5877) parse_headers: flags=ffffffffffffffff 0(5877) parse_headers: flags=78 0(5877) t_lookup_request: start searching: hash=16425, isACK=0 0(5877) DEBUG: RFC3261 transaction matching failed 0(5877) DEBUG: t_lookup_request: no transaction found 0(5877) DBG: trans=0xb612d7f8, callback type 1, id 0 entered 0(5877) DEBUG: mk_proxy: doing DNS lookup... 0(5877) check_via_address(PUBLIC IP of 111111, PRIVATE IP of 111111, 0) 0(5877) DBG:check_against_rule_list: using list dns 0(5877) DEBUG:tm:set_timer: relative timeout is 500000 0(5877) DEBUG: add_to_tail_of_timer[4]: 0xb612d944 (401700000) 0(5877) DEBUG:tm:set_timer: relative timeout is 20 0(5877) DEBUG: add_to_tail_of_timer[0]: 0xb612d960 (421) 0(5877) DEBUG:tm:t_relay_to: new transaction fwd'ed 0(5877) -------------- cheked t reply --- 0(5877) DEBUG:tm:UNREF_UNSAFE: after is 0 0(5877) DEBUG:destroy_avp_list: destroying list (nil) 0(5877) receive_msg: cleaning up ++ From the parsing of BYE request , I found evenif I am doing loose route in cfg , It is doing strict route. Can you please help me to solve this. Thanks in advance. -- Regards, Krunal Patel

1 1

[OpenSER-Users] Need help with buffer overflow error
by Pete Kay 25 May '08

25 May '08

Hi, When I run a *long* sql query in openser.cfg using avp_db_query, I am getting the following error: May 26 02:52:15 [13784] ERROR:core:pv_printf: no more space for text [584] May 26 02:52:15 [13784] ERROR:core:pv_printf: buffer overflow -- increase the buffer size... May 26 02:52:15 [13784] ERROR:avpops:ops_dbquery_avps: cannot print the query How can I increase the buffer size? Thanks in advance for your help. Regards, Pete

2 1

[OpenSER-Users] DISC
by Patrick Pichler 25 May '08

25 May '08

Hi everybody, does anyone have a tested documentation (on installing, configuring or setting up DISC to work together with openSER) ... the DISC (DIameter Server Client) ? On http://developer.berlios.de/projects/disc/ and after searching the web I couldn't find anything except the tarball... thanks in advance regards Patrick

1 0

[OpenSER-Users] MediaProxy or RTPProxy
by MagoDaViola 25 May '08

25 May '08

Untitled Document Hi... What´s the best system in performance question, to use NAT MediaProxy or RTPProxy? -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

4 4

[OpenSER-Users] serMyAdmin-0.5-BETA.war on FC5.tomcat5
by Zen Kato 24 May '08

24 May '08

Hi, I tried to install serMyAdmin-0.5-BETA.war on FC5, tomcat5-5.5.15-1jpp_6fc. I did the following, but http://localhost:8080/serMyAdmin does not work. Please let me know how to install sermyadmin on fedora. (1)# cp mysql-connector-java-5.1.6-bin.jar /usr/share/tomcat5/common/lib (2)/usr/share/tomcat5/conf/context.xml <?xml version="1.0" encoding="UTF-8"?> <Context path="/serMyAdmin"> <Resource auth="Container" driverClassName="com.mysql.jdbc.Driver" maxActive="20" maxIdle="10" maxWait="-1" name="jdbc/openser_MySQL" type="javax.sql.DataSource" url="jdbc:mysql://localhost:3306/openser" username="sermyadmin" password="secret"/> </Context> (3)# mysql>grant all privileges on openser.* to sermyadmin@'%' identified by 'sercret'; (4)# cp serMyAdmin-0.5.war /usr/share/tomcat5/webapps/serMyAdmin.war /var/log/tomcat5/catalina.out ....(snip)......... 2008/05/24 7:10:39 org.apache.catalina.startup.HostConfig deployWAR 情報: Webアプリケーションアーカイブ serMyAdmin.war を配備します log4j:ERROR setFile(null,true) call failed. java.io.FileNotFoundException: stacktrace.log (Permission denied) at java.io.FileOutputStream.openAppend(Native Method) at java.io.FileOutputStream.<init>(FileOutputStream.java:177) at java.io.FileOutputStream.<init>(FileOutputStream.java:102) at org.apache.log4j.FileAppender.setFile(FileAppender.java:290) at org.apache.log4j.FileAppender.activateOptions(FileAppender.java:164) at org.apache.log4j.config.PropertySetter.activate(PropertySetter.java:257) at org.apache.log4j.config.PropertySetter.setProperties(PropertySetter.java:133) at org.apache.log4j.config.PropertySetter.setProperties(PropertySetter.java:97) at org.apache.log4j.PropertyConfigurator.parseAppender(PropertyConfigurator.java:689) at org.apache.log4j.PropertyConfigurator.parseCategory(PropertyConfigurator.java:647) at org.apache.log4j.PropertyConfigurator.parseCatsAndRenderers(PropertyConfigurator.java:568) at org.apache.log4j.PropertyConfigurator.doConfigure(PropertyConfigurator.java:442) .......(snip)........ -- Zen Kato

1 0

[OpenSER-Users] sermyadmin: problems with tomcat6 automatic initialization
by Robson Souza 23 May '08

23 May '08

Hello, I got a openser 1.3.2 working with sermyadmin 0.5 installed in a Debian 4.0 OS. After some tests, everything is working fine except for one problem. By the time I start or restart the server, the tomcat spends approximately 15 min to start as follows (catalina.out file): May 23, 2008 4:10:47 PM org.apache.catalina.core.AprLifecycleListener init INFO: The APR based Apache Tomcat Native library which allows optimal performance in production environments was not found on the java.library.path: /usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/lib/i386/client:/usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/lib/i386:/usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/../lib/i386 May 23, 2008 4:10:48 PM org.apache.coyote.http11.Http11Protocol init INFO: Initializing Coyote HTTP/1.1 on http-8080 May 23, 2008 4:10:48 PM org.apache.catalina.startup.Catalina load INFO: Initialization processed in 3111 ms May 23, 2008 4:10:48 PM org.apache.catalina.core.StandardService start INFO: Starting service Catalina May 23, 2008 4:10:48 PM org.apache.catalina.core.StandardEngine start INFO: Starting Servlet Engine: Apache Tomcat/6.0.16 May 23, 2008 4:10:48 PM org.apache.catalina.startup.HostConfig deployWAR INFO: Deploying web application archive serMyAdmin.war May 23, 2008 4:10:54 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "org.springframework.web.util.Log4jConfigListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:10:54 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "org.codehaus.groovy.grails.web.context.GrailsContextLoaderListener" is already configured for this context. The duplicate definition has been ignored. [0] spring.GrailsWebApplicationContext Refreshing org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336: display name [org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336]; startup date [Fri May 23 16:26:43 BRT 2008]; parent: org.springframework.web.context.support.XmlWebApplicationContext@e61a35 [0] spring.GrailsWebApplicationContext Bean factory for application context [org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336]: org.springframework.beans.factory.support.DefaultListableBeanFactory@baa573 May 23, 2008 4:26:49 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "listeners.ContextListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:26:49 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "listeners.SessionListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:26:49 PM org.apache.coyote.http11.Http11Protocol start INFO: Starting Coyote HTTP/1.1 on http-8080 May 23, 2008 4:26:49 PM org.apache.jk.common.ChannelSocket init INFO: JK: ajp13 listening on /0.0.0.0:8009 May 23, 2008 4:26:49 PM org.apache.jk.server.JkMain start INFO: Jk running ID=0 time=0/40 config=null May 23, 2008 4:26:49 PM org.apache.catalina.startup.Catalina start INFO: Server startup in 960673 ms In order to start faster (10 sec) after the server restart, I have to kill the java process (in CPU 99%) and start tomcat manually by performing "/etc/init.d/tomcat6 start". I'm using the tomcat6 init file provided in www.sermyadmin.org such as its instructions. How can I solve that? Robson --------------------------------- Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!

2 2

Re: [OpenSER-Users] Problems with sl module in openser1.3.1
by Arturo Ochoa 23 May '08

23 May '08

-----Mensaje original----- De: Henning Westerholt [mailto:henning.westerholt@1und1.de] Enviado el: Friday, May 23, 2008 8:52 AM Para: users(a)lists.openser.org CC: arturo arturo Asunto: Re: [OpenSER-Users] Problems with sl module in openser1.3.1 On Friday 23 May 2008, arturo arturo wrote: > Could some help me with this problem, I'm trying to run openser 1.3.1, but > it dies complaining about the sl module. I issued just 'make all' command > before this > > root@zeus:/home/aochoa/openser/openser-1.3.1-notls > # ./openser > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/mysql.so > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/sl.so > May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags > mismatch for* /usr/local/lib/openser/modules/sl.so > core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Hi Arturo, the version of the sl module installed on your machine don't match the one of your server in the home directory. Do a make distclean; make all in the source directory, and install again. Cheers, Henning Thanks for your response Henning, I've already do what you mention, but the problem persists. I tried also with openser 1.3.2, and the same thing. Also I noted that there are different linked libs on the same module but compiled on different machines: Fedora: [root@apolo] /usr/local/lib/openser/modules # ldd sl.so linux-gate.so.1 => (0x0028b000) libc.so.6 => /lib/libc.so.6 (0x00111000) /lib/ld-linux.so.2 (0x0028c000) Centos: (this is the one with the problem) root@zeus:/home/aochoa/openser # ldd /usr/local/lib/openser/modules/sl.so libc.so.6 => /lib/tls/libc.so.6 (0x00149000) /lib/ld-linux.so.2 (0x003f7000) Any comments?

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[OpenSER-Users] Help - As I picked up codes negative www_authorize()
by Marcio 23 May '08

23 May '08

Hi... Anybody could me help, I'm doing some tests and would like to pick up negative codes of the return www_authorize.... I trying with ($retcode) ($?) I've not had success..... Does anyone know how...? Any hints.....?? thx...

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[OpenSER-Users] Problems with sl module in openser1.3.1
by arturo arturo 23 May '08

23 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Open Source VoIP in Porto, Portugal
by Johansson Olle E 23 May '08

23 May '08

Friends, I will be spending a few days in Porto, Portugal in the beginning of June. Any Asterisk and/or OpenSER users there that wants to go out and have dinner and Open Source Voip talk? Respond off list, and we'll see if we can meet. Have a nice weekend! /Olle --- * Olle E. Johansson - oej(a)edvina.net * Asterisk Training http://edvina.net/training/

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[OpenSER-Users] [OT] NGREP showing \r and \n symbols
by Iñaki Baz Castillo 23 May '08

23 May '08

Hi, does somebody know how to use NGREP command to show \r and \n symbols instead of a dot "." replacing them? By default NGREP prints a dot "." relacing any non-printable caracter. This symbol can be changed with "-p" option but waht I want is to show \r and \n literally. Thanks for any suggestion. PD: Any other recomendation (program) to get SIP traces showing \r and \n? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Authentication problem in openSER (Henning Westerholt)
by Ahmed Huraimel 23 May '08

23 May '08

hello Henning, i would like to thank you very much. you save alot of work because i decided to change my database from LDAP to MySQL. just to make it clear to anyone who may face this problem. the solution to keep you passwords hashed in the LDAP server is to save it as HA1 = MD5(username:realm:password). after you build the HA1 save as clear in the LDAP server (i am using openLADP). if you save it as MD5 you will retrieve it as {MD5} blablabla which may cause a problem. so hash it then save it as clear. thanks for every one helps me in this issue regards, Ahmed Huraimel ALALI On Fri, May 23, 2008 at 12:00 PM, <users-request(a)lists.openser.org> wrote: > 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. Re: [OT] NGREP showing \r and \n symbols (Andreas Heise) > 2. Re: [OT] NGREP showing \r and \n symbols (I?aki Baz Castillo) > 3. Re: Problem with audio when either or both UAs are not behind > NAT (I?aki Baz Castillo) > 4. Re: Authentication problem in openSER (Henning Westerholt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 22 May 2008 23:55:56 +0200 > From: Andreas Heise <aheise(a)gmx.de> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: users(a)lists.openser.org > Message-ID: <4835EBEC.7070800(a)gmx.de> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi I?aki, > > I had the same issue with a Nortel CS1k switch, the easiest way to check > this > problem is to use the Wireshark feature "Display raw text for SIP message" > which can be enabled under preferences / protocols / SIP > > regards, > Andreas > > > Ovidiu Sas schrieb: > > Hello Inaki, > > > > Perform a raw capture using tcpdump and open the file with wireshark. > > Check the lower window in wireshark where you have the message > > displayed in hex format. > > There you will be able to spot '\n' and '\r'. To find the hex codes, > > perform a 'man ascii' on your linux box. > > > > > > Regards, > > Ovidiu Sas > > > > On Tue, May 20, 2008 at 10:53 AM, I?aki Baz Castillo <ibc(a)in.ilimit.es> > wrote: > > > >> El Tuesday 20 May 2008 16:37:03 Bogdan-Andrei Iancu escribi?: > >> > >>> Hi I?aki, > >>> > >>> have you tried the "-W byline" ? - it will format you the lines. > >>> > >> Hi Bogdan. Yes, in fact I use this option "-W byline", but this option > just > >> splits the trace into lines (macthing "\n"). This is not enough for me > since > >> \r is not showed. > >> > >> I need it since I send and receive traffic from a carrier using a > painful > >> Nortel CS2K softswitch that separes SDP lines with \r instead of the > >> mandatory \r\n adn they are applying dirty and not working patches > during > >> production state so I need a way to monitorize it. > >> > >> Thanks a lot. > >> > >> > >> > >> -- > >> I?aki Baz Castillo > >> ibc(a)in.ilimit.es > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.openser.org > >> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 23 May 2008 09:31:31 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: users(a)lists.openser.org > Message-ID: <200805230931.31959.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="iso-8859-1" > > El Thursday 22 May 2008 23:55:56 Andreas Heise escribi?: > > Hi I?aki, > > > > I had the same issue with a Nortel CS1k switch > > It's nice to know that I'm not theonly sending/receiving traffic from a > Nortel > CS[12]K. ;) > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 3 > Date: Fri, 23 May 2008 09:32:21 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] Problem with audio when either or both > UAs are not behind NAT > To: users(a)lists.openser.org > Message-ID: <200805230932.21444.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="ISO-8859-1" > > El Thursday 22 May 2008 18:54:53 Krunal Patel escribi?: > > Hi, > > > > I have registered an UA 12345 , which is public. > > In location table, I found NULL value for received. > > I dial any UA , irrespective to their NAT status. > > Call has been established but can't get audio both side. > > It does not use mediaproxy for audio signaling. > > > > How to resolve this problem? > > Do a full sip flow capture and analyze trhe address and port of media in > SDP > of INVITE and 200 OK. > > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 4 > Date: Fri, 23 May 2008 10:57:56 +0200 > From: Henning Westerholt <henning.westerholt(a)1und1.de> > Subject: Re: [OpenSER-Users] Authentication problem in openSER > To: users(a)lists.openser.org > Cc: Ahmed Huraimel <huraimel(a)gmail.com> > Message-ID: <200805231057.56974.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Thursday 22 May 2008, Ahmed Huraimel wrote: > > i am investigating the authentication on openSER. I search for a proper > > explanations but unfortunately i did not find how it is exactly done so i > > did some experiments. i assumed that the response is generated as the > > following: note that i set the username and password with the same string > > "alali" > > > > Response = MD5( username + MD5(password) + realm + nonce) > > [..] > > could anyone tell me how exactly the authentication is done in openSER? > is > > the response generated is like the one i assumed? what + means in the > > response? does is mean concatenation or exoring? > > Hi Ahmed, > > the authentification in OpenSER/ SIP is based on HTTP auth. You find a > detailed explanation for the construction of the response for example at: > http://en.wikipedia.org/wiki/Digest_access_authentication , some further > informations at: http://www.voip-info.org/wiki/view/SIP+Authentication > > For the exact logic inside OpenSER just take a look at the auth module > source > code, it should be not that hard to understand. ;-) > > Cheers, > > Henning > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 36, Issue 48 > ************************************* >

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[OpenSER-Users] Authentication problem in openSER
by Ahmed Huraimel 23 May '08

23 May '08

hello all, i am investigating the authentication on openSER. I search for a proper explanations but unfortunately i did not find how it is exactly done so i did some experiments. i assumed that the response is generated as the following: note that i set the username and password with the same string "alali" Response = MD5( username + MD5(password) + realm + nonce) and in the openser.cfg i tell the server to take the username from the header and set the password from the username since the username and the password are the same (only for testing). TEST1: setting modparam("auth", "calculate_ha1", 1) and telling the server to expect a clear password $avp(s:username) = $fU; $avp(s:password) = $fU; IT WORKS TEST2:setting modparam("auth", "calculate_ha1", 0) and telling the server to expect a hashed password $avp(s:username) = $fU; $avp(s:password) = $(fU{s.md5}); IT DOES NOT WORK could anyone tell me how exactly the authentication is done in openSER? is the response generated is like the one i assumed? what + means in the response? does is mean concatenation or exoring? if(!pv_www_authorize("")) { www_challenge(""/*realm*/,"0"/*qop*/); exit; } if (!save("location")) sl_reply_error(); sl_send_reply("200", "ok"); exit;

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[OpenSER-Users] Problem with audio when either or both UAs are not behind NAT
by Krunal Patel 23 May '08

23 May '08

Hi, I have registered an UA 12345 , which is public. In location table, I found NULL value for received. I dial any UA , irrespective to their NAT status. Call has been established but can't get audio both side. It does not use mediaproxy for audio signaling. How to resolve this problem? Thanks in advance. -- Regards, Krunal Patel

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[Serusers] Mysql Problem
by vijay tiwari 22 May '08

22 May '08

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[OpenSER-Users] OPENSER Dinner, Vienna, Austria, May 24
by Daniel-Constantin Mierla 22 May '08

22 May '08

Hello, just to announce the next OPENSER Social Networking Event in Vienna, Austria, May 24, 2008, 19:30 at Plutzer Brau: http://www.plutzerbraeu.at/plutzerbraeu.html <http://www.voipuser.org/ship_to.php?url=http://www.plutzerbraeu.at/plutzerb…> Among the participants that confirmed the presence already: - Elena-Ramona Modroiu, co-founder OpenSER, COO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> - Andreas Granig, developer OpenSER, CEO at http://www.sipwise.com <http://www.voipuser.org/ship_to.php?url=http://www.sipwise.com> - Daniel-Constantin Mierla, co-founder OpenSER, CEO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> The primary goal of the event is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to the event is free, everybody pays for its expenses. Anyone interested in OpenSER, SIP or VoIP in general is welcome to join. You should contact us to reserve your seat and get notifications in case of changes of time or place. Previous similar event was in Barcelona, May 6 -- remarks and photos: http://www.asipto.com/photos/BARCELONA2008-ALBUM/ http://openser.blogspot.com/2008/05/openser-dinner-barcelona-remarks.html Next one is in Berlin, May 28 (more details soon). Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] [Fwd: Re: facing problem]
by Amit Vijayvargiya 22 May '08

22 May '08

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Re: [Serusers] ser/sems question
by Stefan Sayer 22 May '08

22 May '08

o Steven C. Blair [05/22/08 13:52]: > Thanks Stefan. Is the README for sems-1.0.0 correct? I'm questioning step #4. If I am running this version of SEMS on a Linux box without a proxy server, our proxy resides elsewhere, do I need to perform step #4? no, that is optional, and you only need to do that if you want SER as frontend to SEMS. That's why it says 'Optional' in in http://ftp.iptel.org/pub/sems/doc/current/Readme.html Stefan > > Thanks,Steve > >> -----Original Message----- >> From: Stefan Sayer [mailto:stefan.sayer@iptego.com] >> Sent: Wednesday, May 21, 2008 4:45 PM >> To: Steven C. Blair >> Cc: serusers(a)iptel.org >> Subject: Re: [Serusers] ser/sems question >> >> Hello, >> >> o Steven C. Blair [05/20/08 20:13]: >>> I've been away from SEMS for a few years. II hear the new version of >>> SEMS makes an excellent conference server. Does anyone happen to know >> if >>> I can install SEMS on a standalone Linux machine and interface it, >> via >>> SIP, to a SER 0.9.7-pre3 proxy server? >> yes, you can use it now with any SIP proxy thanks to its own SIP stack. ... -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] Problem PUA_USRLOC, Presence acc. REGISTER/INVITE
by Schumann Sebastian 22 May '08

22 May '08

Dear all I have a problem, which I thought I can solve with PUA_USRLOC. I have a OpenSER SIP Proxy and want him to create presence information about the messages that pass him. Some examples: - REGISTER: Publish as online/available - REGISTER+Expires=0: Publish as offline - INVITE/200 OK: Publish as busy - BYE: Publish as available again. PUA_USRLOC seems to me can handle only simple on/offline according registration state, that is saved on the server. As I have only a proxy server and messages just pass by, this solution seems not the best. What would you recommend me to solve the problem? Change the module, add a module? As I don't have that much of experience in that area I was thinking about using perl module and trigger some perl functions that parse the messages that pass the proxy and make perl send the PUBLISHes to the presence server. Maybe some of you have other ideas or suggestions about that.... I would be happy if you share them with me. Best regards Sebastian

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[OpenSER-Users] Does somebody use $(rU{s.unescape.user}) to be RFC3261 compliant?
by Iñaki Baz Castillo 22 May '08

22 May '08

Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded username in any URI, this is: sip:alice@domain == sip:%61l%69ce@domain For allowing this we must use, explicitely, the transformation $(rU{s.unescape.user}). Do OpenSer administrators allow this hex encoding? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Small documentation mistake pua_usrloc
by Schumann Sebastian 22 May '08

22 May '08

Dear all I noticed a small mistake in the documentation: I had some errors in config file using the pua_usrloc modparam parameters with copy & paste. The problem were the semicolons ; behind the modparam definitions. As I had the error some time ago already I easily figured that out. Although, it might be good to remove them from the doc for other users in the future to not have the same problem as well. Best regards Sebastian

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[Serusers] Creating SIP trunk from OpenSER to Cisco CallManager
by G. Ross 22 May '08

22 May '08

Hi, I've been tasked with getting an OpenSER system to talk to a Cisco CallManager system.I've tried searching the docs, the mailing lists and google, but all my searches end up with references to the "trunk" in the source code repository. Can someone point me somewhere which could help me in this task ? Thanks, GTG

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[OpenSER-Users] facing problem
by Amit Vijayvargiya 22 May '08

22 May '08

Hi, I am facing a problem when user is online more then one machines, openser not able to send request to all machines. Thanks in advance Amit

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[Serusers] ser/sems question
by Steven C. Blair 21 May '08

21 May '08

I've been away from SEMS for a few years. II hear the new version of SEMS makes an excellent conference server. Does anyone happen to know if I can install SEMS on a standalone Linux machine and interface it, via SIP, to a SER 0.9.7-pre3 proxy server? Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[OpenSER-Users] v1.3.2 , presence hash table error
by Jeremy George 21 May '08

21 May '08

Hello All, BLA was working nicely for me with v1.3.0 and Polycom 3.0.0 phones. I downloaded 1.3.2 and I'm now getting this error on the presence server. I tried adding the hash table modparams explicitly, but with got the same error message. modparam("presence", "subs_htable_size", 11) modparam("presence", "pres_htable_size", 11) Thoughts? And thanks. May 20 11:44:35 [12319] ERROR:presence:get_stored_info: record not found in hash_table May 20 11:44:35 [12319] ERROR:presence:handle_subscribe: getting stored info - Jeremy

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[OpenSER-Users] Unable to do transformation on expires param if display name present
by Robert Dyck 21 May '08

21 May '08

I am currently using 1.3.2 but this problem exists in older versions as well. My config script includes a transformation on the expires parameter in a Contact header. I have found that a Linksys SPA3000 will include the display name in the Contact header and that there is white space between the display name and the URI. The function parse_param_name in parse_param.c will stop parsing when it encounters the white space. As a workaround I commented the line in parse_param_name that checks for white space. This works for me but I do not know if there are side effects. This form of Contact is not common but it appears to be legal according to RFC 3261. 20.10 Contact A Contact header field value provides a URI whose meaning depends on the type of request or response it is in. A Contact header field value can contain a display name, a URI with URI parameters, and header parameters. This document defines the Contact parameters "q" and "expires". These parameters are only used when the Contact is present in a REGISTER request or response, or in a 3xx response. Additional parameters may be defined in other specifications. When the header field value contains a display name, the URI including all URI parameters is enclosed in "<" and ">". If no "<" and ">" are present, all parameters after the URI are header parameters, not URI parameters. The display name can be tokens, or a quoted string, if a larger character set is desired. Even if the "display-name" is empty, the "name-addr" form MUST be used if the "addr-spec" contains a comma, semicolon, or question mark. There may or may not be LWS between the display-name and the "<". Here is an excerpt from my script -- if ((is_present_hf("Expires") && $(hdr(Expires) {s.int}) == 0)\ ||($(ct{param.value,expires}) == '0')) { log(1, "looks like deregistration\n"); I added some debug messages to illustrate the problem -- May 19 22:16:32 [14447] DBG:core:parse_param_name: string passed to parse_param_name is RJD <sip:17476316719@192.168.1.4:5061>;expires=3600 May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is R May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is J May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is D May 19 22:16:32 [14447] DBG:core:parse_param_name: next char is May 19 22:16:32 [14447] ERROR:core:parse_params: invalid character, ; expected May 19 22:16:32 [14447] DBG:core:run_transformations: return val is -1 May 19 22:16:32 [14447] CRITICAL:core:comp_scriptvar: cannot get left var value May 19 22:16:32 [14447] WARNING:core:do_action: error in expression

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Re: [OpenSER-Users] ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes
by Henning Westerholt 21 May '08

21 May '08

On Wednesday 21 May 2008, Yazid Hadj Said wrote: > Hi Henning, > I configured mysql with latin1 (in my.cnf), but i don't know when can i > specify to openser to use latin1. In other word how can I verify if openser > use UTF-8? Hi Yazid, its not possible to use the UTF-8 charset at the moment with OpenSER and mysql. I think this is a limitation from mysql, look for example to this bug: http://bugs.mysql.com/bug.php?id=4541 The openserdbctl script failed to detect that your server uses UTF-8 as default charset, because there was an error present in the tool. This is fixed now, it should now work for you regardless if you use latin1 or utf8 as default charset. Cheers, Henning

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[OpenSER-Users] ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes
by Yazid Hadj Said 21 May '08

21 May '08

Hi, When I try to create openser database (openserdbctl create) on Gentoo, I have the message: INFO: creating database openser ... ERROR 1071 (42000) at line 2: Specified key was too long; max key length is 1000 bytes ERROR: Creating core tables failed! I search when i can modify TABLE_TYPE but i not found. Regards, Yazid

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[OpenSER-Users] nat problems
by Pezhman Lali 21 May '08

21 May '08

Dear I am testing a lot of configs, for ser, openser and asterisk, but till now, I can not find a total solution for solving nat, do you think ser/openser+ (defined modules) can support all of nat services, around the world ? best,

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[OpenSER-Users] NAT problem in bridging mode
by Christian Koch 21 May '08

21 May '08

Hi, I have a problem with openser and rtpproxy. I'm trying to use them as a gateway between the public internet and a LAN. Clients in the internet may be natted, so I'm using nathelper. Calls are only made from LAN to outside or vice versa, but not from LAN to LAN or from outside to outside. The following should illustrate my configuation: ----- ------------------ UAC1 --- | NAT | ------------- | OpenSER/rtpproxy | ----------------UAC2 ----- ------------------ | | | | | dynamic public IP 2.3.4.5 192.168.103.121 192.168.103.189 (e.g. 1.2.3.4) UAC1 and UAC2 are both registered at OpenSER. Now I'm making a call from UAC1 to UAC2. SIP messages are passed just fine, but the RTP traffic from UAC2 to UAC1 is dropped at the NAT. I used tcpdump on the OpenSER/rtpproxy machine to figure out what happens to RTP and it shows the following (ports and IPs are just examples): stream1: 1.2.3.4:10000 -> 2.3.4.5:35000 ->RTP is forwarded by rtpproxy-> 192.168.103.121:35000 -> 192.168.103.189:11000 stream2: 1.2.3.4:20000 <- 2.3.4.5:35000 <-RTP is forwarded by rtpproxy<- 192.168.103.121:35000 <- 192.168.103.189:11000 Port 20000 in stream2 is the RTP-port used internally by UAC1 behind the NAT (this port is found in the INVITE from UAC1 to OpenSER). I understand, that rtpproxy sends the first packets to port 20000. But, after receiving some packets from port 10000, shouldn't it change the destination port to 20000 so they can pass the NAT? rtpproxy is started like this: "./rtpproxy -l 192.168.103.121/2.3.4.5 -f ". It produces the following output: [root@192 rtpproxy]# /usr/local/bin/rtpproxy -l 192.168.103.121/2.3.4.5 -f rtpproxy started, pid 22125 received command "UIE 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189 192.168.103.189 49156 207860870326595;1" new session 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189, tag 207860870326595;1 requested, type strong new session on a port 35000 created, tag 207860870326595;1 pre-filling caller's address with 192.168.103.189:49156 sending reply "35000 2.3.4.5 " received command "L 9D740CB7-18A4-40B2-A96D-13FC3C5B27D3(a)192.168.103.189 1.2.3.4 49154 207860870326595;1 5364140283;1" lookup on ports 35000/35000, session timer restarted pre-filling callee's address with 1.2.3.4:49154 sending reply "35000 192.168.103.121 In my openser.cfg I'm not really checking wheter a client is really natted, but I think it shouldn't be a problem to assume, that all clients are behind a NAT? I attached the openser.cfg to this mail (real public IP is changed to 2.3.4.5). Do you have any ideas how to fix this problem? Any help would be greatly appreciated! Thanks, Christian # $Id: nat-rtpproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ #/usr/local/bin/rtpproxy -l 192.168.103.121/2.3.4.5 -f debug=6 fork=yes log_stderror=no log_facility=LOG_LOCAL1 listen=192.168.103.121 # Internal IP listen=2.3.4.5 # External IP port=5060 children=4 dns=no rev_dns=no #set module path 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 "auth.so" loadmodule "auth_db.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "nathelper.so" loadmodule "textops.so" loadmodule "mi_fifo.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://xxx:yyy@192.168.103.122/openser") 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", "unix:/home/sys/bin/rtpproxy.sock") modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("registrar", "received_avp", "$avp(s:rcv)") modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("rr", "enable_full_lr", 1) route { setflag(1); # log this msg # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # 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"); return; }; if (method=="INVITE") { setflag(6); force_rport(); fix_nated_contact(); if (dst_ip == 192.168.103.121){ force_rtp_proxy("faie"); } else { force_rtp_proxy("faei"); } }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; 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 # ---------------------------------------------------------------- setflag(6); fix_nated_register(); force_rport(); sl_send_reply("100", "Trying"); if (!save("location")) { log(1,"Error :-(.\n"); sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- setflag(6); lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- force_rport(); fix_nated_contact(); if (dst_ip == 192.168.103.121){ log(1,"INVITE 4 intern extern\n"); force_rtp_proxy("fie"); } else { log(1,"INVITE 4 extern intern\n"); force_rtp_proxy("fei"); } } onreply_route[1] { force_rtp_proxy("f"); fix_nated_contact(); }

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[OpenSER-Users] question with acl using openserctl and sermyadmin
by mark morreny 21 May '08

21 May '08

Hi, I am hoping someone can help me our with associating user with group in openser. If I use the command line: Question 1: debian:/home/ser# openserctl acl grant 1006 LD WARNING: Invalid privilege: acl 'LD' ignored debian:/home/ser# What is wrong with the syntax? Question 2: If I use sermyadmin to associate user to groups, I get the following error: org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [UserGroup]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at UserController$_closure6_closure11.doCall(UserController.groovy:111) at UserController$_closure6.doCall(UserController.groovy:101) at UserController$_closure6.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [UserGroup] at $Proxy3.saveOrUpdate(Unknown Source) ... 32 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '1005-192.168.1.104-' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:209) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 33 more What is wrong with my setup of sermyadmin? Thank you in advance for your help. Regards, Mark

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[OpenSER-Users] Help, for configure my client (Xlite).
by ZaidKa 21 May '08

21 May '08

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[OpenSER-Users] UDP Buffer Size / OpenSER / Linux
by Carsten Bock 21 May '08

21 May '08

Hi, short question: Can anybody give me a hint regarding the Linux UDP-Buffer size? Are there any known good values for this or is the default value (128k) of Linux sufficient? I've read, regarding other SIP-Systems (namely IBM SIP Servlets and Communigate pro), the recommendation to increase the UDP-Buffer-size to 3 or even 4 MByte.... Are there any recommended / known best values for heavy load OpenSER Installations? Thanks in advance, Carsten Reference: http://publib.boulder.ibm.com/infocenter/wasinfo/v6r1/index.jsp?topic=/com.…

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[OpenSER-Users] Why can't I "append_branch()" in FAILURE_ROUTE if reply is 6XX ??
by Iñaki Baz Castillo 21 May '08

21 May '08

Hi, when I receive a 480 or 603 from a local user I run "on_failure_route" and create new branch to forward the INVITE to a media server in which I reply an early-media "The number you are calling is not available now...". It works perfectly if the error is 4XX but it doesn't work if it's a 6XX. I get this error: --------------------- ## onreply_route[ON_REPLY_TO_USER]: 603: Decline ## failure_route[ON_FAILURE_TO_USER] --- TO_MEDIA_SERVER: Destino user_not_available May 21 12:25:25 [8502] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction May 21 12:25:25 [8502] ERROR:tm:w_t_relay: t_forward_nonack failed ---------------------- My code is very simple: ---------------------- failure_route[ON_FAILURE_TO_USER] { if (t_check_status("480|603")) { $rU = "user_not_available"; route(ROUTE_TO_MEDIA_SERVER); exit; } } route[ROUTE_TO_MEDIA_SERVER] { rewritehostport("MEDIA_SERVER_IP:MEDIA_SERVER_PORT"); append_branch(); t_relay(); } --------------------- It seems that if the reply is a 6XX then OpenSer can't generate a new_branch in failure_route. May 21 12:25:25 [8502] ERROR:tm:t_forward_nonack: discarding fwd for a cancelled/6xx transaction Why? doesn't it make sense to forward to some destination even if the negative reply is a 4XX or 6XX? Thanks a lot for any help. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Call failed 403 Forbiden
by luzango mfupe 20 May '08

20 May '08

Hi mates, Everytime i do try to make my 2 xlite clients talk (which i correctly added them into the database), i encountered with the same problm, Openser perfoms authentication and return call failed 403 forbiden signal. My setup comprise of the first box with openser 1.3 and mediaproxy the second box with Mysql and two Xlite clients in two other boxes all are in an internal network, as far as am concerned my NetAdmin have already opened ports 5060 and 3306 for me. I need your right direction on this probm. below is my ngrep snapshot mzee:/# ngrep -d eth1 -W byline port 5060 interface: eth1 (168.172.200.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 168.172.200.70:1824 -> 168.172.200.87:5060 INVITE sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>SIP/2.0. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. Max-Forwards: 70. Contact: <sip:dreamteam@168.172.200.70:1824>. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>>. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 INVITE. Session-Expires: 95. Min-SE: 90. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: timer. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 426. . v=0. o=- 9 2 IN IP4 168.172.200.70. s=CounterPath X-Lite 3.0. c=IN IP4 168.172.200.70. t=0 0. m=audio 52166 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : uZB2dYm+ NKBRK8Ep 168.172.200.70 52166. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. # U 168.172.200.87:5060 -> 168.172.200.70:1824 SIP/2.0 403 Forbidden. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 INVITE. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0. . # U 168.172.200.70:1824 -> 168.172.200.87:5060 ACK sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> SIP/2.0. Via: SIP/2.0/UDP 168.172.200.70:1824 ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. From: "dream"<sip:dreamteam@168.172.200.87 <sip%3Adreamteam(a)168.172.200.87> >;tag=af4bd714. Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. CSeq: 1 ACK. Content-Length: 0. WBR, LU. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA

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[OpenSER-Users] Wesip question
by mark morreny 20 May '08

20 May '08

Hi, Is it possible to implement a media proxy using wesip? If so, is there a reference info available? Other than wesip, is there any other tool that can be used to extend Openser possibility through a scripting language? Thanks, Mark

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[OpenSER-Users] db_extra: the avp value does not correspond to the real value
by Bruno Machado 20 May '08

20 May '08

Hi friends. OK. People... I'm almost getting crazy. I don't understand what is happening. I already looked for an answer, but I didn't find nothing yet. I am recording 8 extra values in ACC, but the SQL command is using a weird value for one of them. So, I removed 6 of them to facilitate the analyze. Look at the code: . . modparam("acc", "db_extra", "valueA=$avp(i:15); valueB=$avp(i:16)") . . . avp_write("i:15","$avp(i:15)"); avp_write("i:16","$avp(i:16)"); if(!t_relay()) { sl_reply_error(); exit; }; avp_print(); exit; OK. Looking at these lines, I would say: "the sql INSERT command will use the value 15 for avp:15 and 16 for avp:16". Now, look at the result of "avp_print": 2(11112) INFO:avpops:print_avp: p=0xb3b4a900, flags=0x0000 2(11112) INFO: id=<16> 2(11112) INFO: val_int=<16> 2(11112) INFO:avpops:print_avp: p=0xb3b4a8e8, flags=0x0000 2(11112) INFO: id=<15> 2(11112) INFO: val_int=<15> Nice. AVP 15 has the value 15 and the AVP 16 has 16. But something is happening.... It is the insert command: insert into acc (from_uri,to_uri,sip_method...............valueA,valueB,time) values .........,'16','16','2008-05-19 12:48:38'). 16 twice!! Why? This question is running in loop inside my head. It should be "....'15','16','2008-05-19 12:48:38'.....". The openser code above is executed just once by the INVITE. I can bet this error is the like when we sum "2 and 3" and the result is "6". You look at the equation, but you don't see it at all. And, when you ask your younger brother to see, he says: "you are stupid... 2+3=5". I never had this problem. I use several avps to log in the database and it is the first time that it happens. You can see that I'M NOT CHANGING the values of the avps. Am I doing a stupid thing? Or, maybe, am I being blind? By the way, I'm using version 1.1.1. Tks people. OK. Apparently, there's nothing wrong. Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

2 1

[OpenSER-Users] DNS query
by Yazid Hadj Said 20 May '08

20 May '08

Hi, I want to perform a DNS query with the Record-Route header's content, and replace this content by the response (replacing "domain.com" by xxx.xxx.xxx.xxx). someone knows how? Regards, Yazid

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[OpenSER-Users] OPENSER Workshop, Berlin, May 26-27
by Daniel-Constantin Mierla 20 May '08

20 May '08

Hello, for those interested, there will be two OPENSER sessions hosted by Asterisk Tag Conference, Berlin, Germany, May 26-27, 2008. The event gathers an important group of people from VoIP area, including the inventors of Asterisk, Adhearsion, PGP, Yate, ..., for more details visit: http://asterisk-tag.org In the first day, the OPENSER presentation will focus on how to use the proxy to increase the capacity of Asterisk-based platforms, the second day will host 2 hours workshop targeting to reveal where OPENSER fits in IP communication services, Voice over IP and beyond, and to give the starting point to use it. FYI: the following days is the Linux Tag conference/exhibition, also in Berlin: http://www.linuxtag.org . A preview notice -- on the 28th of May will be an OPENSER dinner (more beer than food, as it happens in Germany) -- place and time will be announced very soon. Cheers, Daniel -- CEO ASIPTO http://www.asipto.com

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[OpenSER-Users] Authentication with LDAP databse
by Ahmed Huraimel 20 May '08

20 May '08

Hello, i am successfully integrated an openLDAP server with my openSER SIP proxy server. however i am facing a security problem. let me explain it briefly. **** Successful Registration with password save as clear in openLDAP DB **** * # Configration # - user name was stored in clear in openSER database - modparam("auth", "calculate_ha1", 1) which means the server will assume that the "password_spec" pseudo-variable contains plaintext passwords and it will calculate HA1 strings on the fly. # Senario # - after the UAC receives Authentication request he will build the response = MD5(username + MD5(passowrd) + realm + nonce) - then the server will build the challenge by searching the the user in the database and retrieving the password in clear then hash the password with MD5 build the challenge such that challenge=MD5(username + MD5(passowrd) + realm + nonce) . . - by comparing the the response the with the challenge the user will be authenticated. - *it works * **** Successful Registration with password save as MD5 in openLDAP DB **** * # Configration # - user name was stored in MD5 in openSER database - modparam("auth", "calculate_ha1", 0) which means the server assumes the pseudo-variable contains the HA1 strings directly and will not calculate them. # Senario # - after the UAC receives Authentication request he will build the response = MD5(username + MD5(password) + realm + nonce) - then the server will build the challenge by searching the the user in the database and retrieving the password in MD5 then challenge such that challenge=MD5(username + MD5(password) + realm + nonce) . - by comparing the the response the with the challenge the user will be authenticated. - *401 unauthorized !!!! ***** CONCLUSION **** *there for possible scenarios: 1- password clear + calculate_ha1= 0 ==> 401 unauthorized !!!! 2- password clear + calculate_ha1= 1 ==> Authorized 3- password MD5 + calculate_ha1= 0 ==> 401 unauthorized !!!! 4- password MD5 + calculate_ha1= 1 ==> 401 unauthorized !!!! * * ----------------------------------------------------------------------------------------- * asuumptions: *1- the password might be not hashed. if so then why modparam("auth", "calculate_ha1", ) used? does it mean that the password might be received hashed or not? *2- in scenario(2) the sip server hash the password by setting calculate_ha1= 1. if the password is already hashed in the database then scenario(3) should work unless there is a conflict with the hash. is this might be related to hash type or size?* or something else that i do not know!!! *question*: 1- why scenario(3) does not work? where might be the problem? 2- what to do if i want to change the hash algorithm used? for example i need to SSH1 instead of MD5 because nowadays MD5 is proved to be weak algorithm regards, Ahmed ALALI

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[OpenSER-Users] How to install OpenSER on VMware?
by Habib, Yeasin 20 May '08

20 May '08

Hi, I am successful till the addition of voip overlay. After that I have also include the line source /usr/portage/local/layman/make.conf in /etc/make.conf. Now when I try to install openser using the command ACCEPT_KEYWORDS=!x86 emerge openser, every time I got the following reply back from the console emerge: there are no ebuilds to satisfy "openser" Does anybody know how I can solve this problem and install and configure OpenSER on top my OpenIMSCore Best regards Yeasin Habib You can install layman, add voip overlay and install OpenSER using standard gentoo way (emerge openser) 1) start OpenIMS image 2) Login 3) emerge --sync. On this step emerge will download information about all packages. Maybe about 200 mb 4) emerge layman 5) edit make.conf 6) add voip overlay (layman -a voip) 7) install openser (ACCEPT_KEYWORDS=~x86 emerge openser) On Tue, Apr 22, 2008 at 9:53 AM, Rob <otrebor79 at alice.it<https://lists.berlios.de/mailman/listinfo/openimscore-users>> wrote: > Hi, > I've downloaded this image of OpenIMSCore for VMware > http://www.digilo.com/openims/files/OpenIMS.2008-03-30.r545.rar > I've installed and runned it. > Now I want install on it a Presence Server so I've downloaded the > packages of the OpenSER 1.2.0 version 1.20. > Now, how can I install it onto che VMware image? > > Rob > > _______________________________________________ > OpenIMSCore-Users mailing list > OpenIMSCore-Users at lists.berlios.de<https://lists.berlios.de/mailman/listinfo/openimscore-users> > https://lists.berlios.de/mailman/listinfo/openimscore-users > -- Vladimir Romanov

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[OpenSER-Users] Newbie quesion on configuring Openser as SIP Registrar
by Pete Kay 20 May '08

20 May '08

Hi, I followed instruction from http://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-regi… setup Openser. #modparam("nathelper", "rtpproxy_disable", 1) #modparam("registrar", "nat_flag", 6) #modparam("registrar", "use_domain", 0) The above three lines will cause the server to give error: May 20 23:29:51 [2784] CRITICAL:core:yyerror: parse error in config file, line 93, column 19-20: Can't set module parameter I saw from the document that those are valid arguments, and I don't understand why Openser complain about those parameters. Then, I comment out those three lines, and get another errors: May 20 23:48:25 [3218] CRITICAL:mediaproxy:mod_init: bad config - nat ping enabled, but no nat bflag set in the usrloc module May 20 23:48:25 [3218] ERROR:core:init_mod: failed to initialize module mediaproxy May 20 23:48:25 [3218] ERROR:core:main: error while initializing modules I am very new to openser. Any help or suggestion will be greatly appreciated. Here is my openser.cfg ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/speeddial.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- #modparam("usrloc|auth_db|domain|speeddial|acc", "db_url", "mysql://localhost:openserrw@localhost/openser") modparam("auth_db", "db_url","mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("domain", "db_mode", 1) #modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 60) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) #modparam("registrar", "nat_flag", 6) #modparam("registrar", "use_domain", 0) modparam("rr", "enable_full_lr", 1) #modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") #modparam("auth", "rpid_avp", "s:rpid")

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[OpenSER-Users] problem in presence module
by Amit Vijayvargiya 20 May '08

20 May '08

Hi, i m add presence module in openser 1.3.0 it's working fine.... but some time it given error in stoped.... see the logs... May 20 14:58:09 [6432] INFO:pua_usrloc:ul_publish: should not send ul publish May 20 14:58:09 [6435] ERROR:presence:handle_subscribe: Missing or unsupported event header field value May 20 14:58:09 [6435] ERROR:presence:handle_subscribe: event= message-summary May 20 14:58:09 [6433] ERROR:mysql:db_mysql_submit_query: driver error: Duplicate entry 'sip:userA@domain.com-userB-domain.com-presence' for key 2 May 20 14:58:09 [6433] ERROR:mysql:db_mysql_insert: error while submitting query May 20 14:58:09 [6433] ERROR:presence:insert_db_subs_auth: in sql insert May 20 14:58:09 [6433] ERROR:presence:handle_subscribe: while inserting record in watchers table May 20 14:58:09 [6433] ERROR:mysql:db_mysql_submit_query: driver error: Malformed packet May 20 14:58:09 [6433] ERROR:mysql:db_mysql_query: error while submitting query May 20 14:58:09 [6433] ERROR:presence:get_subs_db: while querying database May 20 14:58:09 [6433] ERROR:presence:get_subs_dialog: getting dialogs from database May 20 14:58:09 [6443] CRITICAL:core:receive_fd: EOF on 10 May 20 14:58:09 [6429] INFO:core:handle_sigs: child process 6435 exited by a signal 11 May 20 14:58:09 [6429] INFO:core:handle_sigs: core was generated May 20 14:58:09 [6429] INFO:core:handle_sigs: terminating due to SIGCHLD May 20 14:58:09 [6442] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6432] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6437] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6433] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6434] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6436] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6439] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6440] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6441] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6443] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6438] INFO:core:sig_usr: signal 15 received May 20 14:58:09 [6429] NOTICE:presence:destroy: destroy module ... plz help me wat is the problem???? Thanks & Regards, Amit Vijayvargiya

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[OpenSER-Users] problem in audio when using media proxy
by Krunal Patel 20 May '08

20 May '08

Hi, We are using openser with media proxy and radiator. We're having audio problems when we send calls to mediatrix gateway for termination. Internal calls work fine. We hear audio both side and we can see the session and bytes transfer properly in media proxy monitoring webpage. When we send calls to mediatrix gateway for termination, calls connect properly but we don't hear audio. We see inactive session in mediaproxy status page with no bytes transferring. One doubt I have is it is not entering reply route when it receives replies like 183, 180, etc. Has anyone encountered similar issue? How can we fix it? Thanks in advance! -- regards, Krunal Patel

3 4

[OpenSER-Users] openser.vim
by Daniel-Constantin Mierla 19 May '08

19 May '08

Hello, just found vim syntax specs for openser configuration file. vim fans might find it interesting: http://www.vim.org/scripts/script.php?script_id=2242 All thanks to Stanislaw Pitucha. Cheers, Daniel -- http://www.asipto.com

2 1

[Serusers] Release schedule for 2.0?
by Marcus Better 18 May '08

18 May '08

Hi, I'm working on packaging ser for Debian. We currently ship 0.9.7, but have also packaged a 2.0 snapshot which seems to be working well. We now need to decide which version to ship in the upcoming "lenny" release of Debian. Do you have any information that you can share about the release plans for SER 2.0? This would help us a lot in making this decision. Thanks, Marcus

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Re: [OpenSER-Users] null socket list
by Daniel-Constantin Mierla 17 May '08

17 May '08

On 05/17/08 00:53, Michael Barinek wrote: > mysql works as well. Thanks again. perfect, thanks for testing. Daniel > > > > Michael Barinek wrote: >> I think that worked, Thank you! I'm setting up mysql and will have >> more info soon. Thanks again. >> >> >> Daniel-Constantin Mierla wrote: >>> Hello, >>> >>> found a similar case and hopefully the fix I just committed works. >>> You have to install branch 1.3 from SVN and then test. Let me know >>> if it is working for you. >>> >>> Cheers, >>> Daniel >>> >>> >>> On 03/26/08 22:18, Michael Barinek wrote: >>>> I have openser-1.3.1-notls installed although am unable to start >>>> openser. Any suggestions/ideas regarding the below error? Any >>>> existing docs/links would be great. >>>> >>>> Thanks >>>> >>>> /// >>>> >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.86.56 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.87.100 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.37.129.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.211.55.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.86.56 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.138.87.100 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.37.129.2 >>>> Mar 20 08:59:14 [18124] WARNING:core:fix_socket_list: could not >>>> rev. resolve 10.211.55.2 >>>> Listening on >>>> udp: 127.0.0.1 [127.0.0.1]:5060 >>>> udp: 10.138.86.56 [10.138.86.56]:5060 >>>> udp: 10.138.87.100 [10.138.87.100]:5060 >>>> udp: 10.37.129.2 [10.37.129.2]:5060 >>>> udp: 10.211.55.2 [10.211.55.2]:5060 >>>> tcp: 127.0.0.1 [127.0.0.1]:5060 >>>> tcp: 10.138.86.56 [10.138.86.56]:5060 >>>> tcp: 10.138.87.100 [10.138.87.100]:5060 >>>> tcp: 10.37.129.2 [10.37.129.2]:5060 >>>> tcp: 10.211.55.2 [10.211.55.2]:5060 >>>> Aliases: >>>> tcp: my.machine.name:5060 >>>> udp: my.machine.name:5060 >>>> >>>> Mar 20 08:59:14 [18124] WARNING:core:main: no fork mode and more >>>> than one listen address found(will use only the first one) >>>> Mar 20 08:59:14 [18124] INFO:core:init_tcp: using kqueue as the TCP >>>> io watch method (auto detected) >>>> Mar 20 08:59:14 [18124] NOTICE:core:main: version: openser >>>> 1.3.1-notls (i386/darwin) >>>> Mar 20 08:59:14 [18124] INFO:core:main: using 32 Mb shared memory >>>> Mar 20 08:59:14 [18124] INFO:core:main: using 1 Mb private memory >>>> per process >>>> Mar 20 08:59:14 [18124] INFO:sl:mod_init: Initializing StateLess >>>> engine >>>> Mar 20 08:59:14 [18124] INFO:tm:mod_init: TM - initializing... >>>> Mar 20 08:59:14 [18124] CRITICAL:tm:uac_init: null socket list >>>> Mar 20 08:59:14 [18124] ERROR:tm:mod_init: uac_init failed >>>> Mar 20 08:59:14 [18124] ERROR:core:init_mod: failed to initialize >>>> module tm >>>> Mar 20 08:59:14 [18124] ERROR:core:main: error while initializing >>>> modules >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.openser.org >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >> -- http://www.asipto.com

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[OpenSER-Users] auth_radius:radius_authorize_sterman: rc_auth failed
by Simon J Xuereb 16 May '08

16 May '08

Hi, I am trying to configure openser + freeradius for authentication & accounting. I have my freeradius configured with openldap and it's working successfully. tested with radclient -f digest 127.0.0.1:1812 auth testing123 results are successfull. However I cannot get it to work with openser. I see nothing happening in my "/usr/sbin/radius -X" output coming from openser when i try to register. My enviroment was tested on Fedora 8 and Fedora 9 both with rpm's and same results. auth_radius:radius_authorize_sterman: rc_auth failed no output is seen in my /usr/sbin/radius -X Below please find my openser.cfg & radiusclient.conf & servers ######## radiusclient.conf # General settings # specify which authentication comes first respectively which # authentication is used. possible values are: "radius" and "local". # if you specify "radius,local" then the RADIUS server is asked # first then the local one. if only one keyword is specified only # this server is asked. auth_order radius,local # maximum login tries a user has login_tries 4 # timeout for all login tries # if this time is exceeded the user is kicked out login_timeout 60 # name of the nologin file which when it exists disables logins. # it may be extended by the ttyname which will result in # a terminal specific lock (e.g. /etc/nologin.ttyS2 will disable # logins on /dev/ttyS2) nologin /etc/nologin # name of the issue file. it's only display when no username is passed # on the radlogin command line issue /etc/radiusclient-ng/issue # RADIUS settings # RADIUS server to use for authentication requests. this config # item can appear more then one time. if multiple servers are # defined they are tried in a round robin fashion if one # server is not answering. # optionally you can specify a the port number on which is remote # RADIUS listens separated by a colon from the hostname. if # no port is specified /etc/services is consulted of the radius # service. if this fails also a compiled in default is used. authserver 127.0.0.1 authserver 10.0.0.10 authserver localhost # RADIUS server to use for accouting requests. All that I # said for authserver applies, too. # acctserver 127.0.0.1:1813 # file holding shared secrets used for the communication # between the RADIUS client and server servers /etc/radiusclient-ng/servers # dictionary of allowed attributes and values # just like in the normal RADIUS distributions #dictionary /usr/share/radiusclient-ng/dictionary dictionary /etc/radiusclient-ng/dictionary.openser # program to call for a RADIUS authenticated login login_radius /usr/sbin/login.radius # file which holds sequence number for communication with the # RADIUS server seqfile /var/run/radius.seq # file which specifies mapping between ttyname and NAS-Port attribute mapfile /etc/radiusclient-ng/port-id-map # default authentication realm to append to all usernames if no # realm was explicitly specified by the user # the radiusd directly form Livingston doesnt use any realms, so leave # it blank then default_realm # time to wait for a reply from the RADIUS server radius_timeout 10 # resend request this many times before trying the next server radius_retries 3 # local address from which radius packets have to be sent bindaddr * # LOCAL settings # program to execute for local login # it must support the -f flag for preauthenticated login login_local /bin/login ######################################################## #/etc/radiusclient-ng/servers #Server Name or Client/Server pair Key #---------------- --------------- #portmaster.elemental.net hardlyasecret #portmaster2.elemental.net donttellanyone localhost testing123 127.0.0.1 testing123 10.0.0.10 testing123 ####################################################### ##########openser.cfg # -- rr params -- debug=3 log_stderror=yes log_facility=LOG_LOCAL0 fork=yes children=4 port=5060 # ------------------ module loading ---------------------------------- mpath="/usr/lib/openser/modules/" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "avpops.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "uri.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc","db_url","mysql://openser:openser@localhost/openser") modparam("usrloc", "db_mode", 2) # -- acc params -- modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.conf") # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # authenticate registers if (method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); exit; }; save("location"); exit; }; route(1); } # generic forward route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } # Thanks for your help SJX

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Re: [OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER
by Kaeman Chris 16 May '08

16 May '08

Like i already took this as a benefit of doubt. They used G711u for a California calls and it was successful and used G711u NSE to the NewYork and the call failed with 484. so i asked them to use the G711u for NewYork and not G711u NSE still the call failed with G711u (excluding NSE). I enquired with the provider already that they support G711u but not G711u NSE. but still with G711u they failed and that too with 484. thats really wierd and i am stuck up in the middle. KC LetMeKnow <sunkara.raviprakash.feb14(a)gmail.com> wrote: Hi Chris, This Problem is happened to me, when i used EyeBeam---> OpenSER--> Asterisk as PSTN Gateway, Asterisk is seems to send 484, this is due to the imcompartable of the Codecs in Eyebean and Asterisk gateway........................................... Check out.... logs from these customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 supplier have producing this t SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _ d E pl@ 7 ! @ V)ÇèNÃ Ä Ä |MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% E èÌ@ @ £ ÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H ; ® ® 0H% E èÍ@ @ ¢ÿÇèNÃ@ V) Ä Ä ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E =èÎ@ @ G;ÇèNÃ@®N Ä Ä ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _ d E º** 2 Ws@®NÇèNÃ Ä Ä ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _ d E Æ+ 2 V@®NÇèNÃ Ä Ä ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% E fè×@ @ K ÇèNÃ@®N Ä Ä R úACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download. _______________________________________________ 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 91-9999882776 --------------------------------- Check out the all-new face of Yahoo! India. Click here. --------------------------------- Check out the all-new face of Yahoo! India. Click here.

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[OpenSER-Users] Implementing international destination allow/deny system with PDT module (patch included for 1.3.X)
by Iñaki Baz Castillo 16 May '08

16 May '08

Hi, I've implemented a system to control international destinations (allow/deny) using "PDT" module (based on prefix). In OpenSer 1.4 there will be a new module "userblacklist" which can handle this issue in a more ellegant way, but for OpenSer 1.3.X we can use "pdt" module (with a little patch to pdt.c) to implement it. I've submitted the patch and the how-to: https://sourceforge.net/tracker/index.php?func=detail&aid=1965211&group_id=… Hope it could be useful. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Res: sermyadmin almost works
by Flavio Goncalves 16 May '08

16 May '08

Please, check the updated instructions to install at: www.sermyadmin.org Flavio ----- Mensagem original ---- De: whudson05 <whudson05(a)gmail.com> Para: users(a)lists.openser.org Enviadas: Quarta-feira, 14 de Maio de 2008 11:29:26 Assunto: Re: [OpenSER-Users] sermyadmin almost works I'm also at the same stage. I've looked at the mySQL queries and they appear to be correct. Have you added the user admin to subscriber? Lucas Catranis wrote: > > Anyone Get this setup correctly? I've been at it for a while and after > getting it to insert new users I cannot login as an admin. I'm not sure > where the problem is but Any help would be appreciated. > -- View this message in context: http://www.nabble.com/sermyadmin-almost-works-tp17219053p17232461.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[OpenSER-Users] problem - 484 address incomplete for the correct TO NUMBER
by Kaeman Chris 15 May '08

15 May '08

Hello, I am having the following problem. please help me to figure out the issue. I have my client calling me and getting 484 Address incomplete. I checked all the SIP headers and found no issues. Also, when i use the same called number and make the calls through the same provider it works from my end with 200 OK. its really interesting and seems to be some issues with the headers. I am using openser and following are the logs. sorry - its too long, but i need someones help to figure out whats going on really. Also, following is the definition for the ips and telephone numbers. FROM number - 17592034561 TO number - 19846572302 customer signaling ip-addr - 23.21.76.23 customer media ip-addr - 231.43.25.1 supplier media ip-addr - 43.25.32.85 supplier signaling ip-addr - 13.45.213.245 my signaling ip-addr - 123.123.1.234 --------------------------------------------------------------------- logs as below; Y#HE& _d E pl@ 7!@V)ÇèNÃÄÄ|MàINVITE sip:+17592034561@123.123.1.234 SIP/2.0 Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#H¯' ¨ ¨ 0H% EèÌ@ @£ÇèNÃ@V)ÄÄ¯SIP/2.0 100 Trying To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; ® ® 0H% EèÍ@ @¢ÿÇèNÃ@V)ÄÄ¯ SIP/2.0 100 Giving a try To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 ÄY#H; M M 0H% E=èÎ@ @G;ÇèNÃ@®NÄÄ) ÑINVITE sip:17592034561@13.45.213.245 SIP/2.0 Record-Route: <sip:123.123.1.234;lr=on;ftag=5d411c09;osp-o=t5198095097179410177_s23.21.76.23_a1210276292_c1> Record-Route: <sip:23.21.76.23;lr;ftag=5d411c09;did=3b7.c57bab42> To: <sip:+17592034561@23.21.76.23> From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Contact: <sip:+19846572302@43.25.32.85:8394> Max-Forwards: 68 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username="19846572302",realm="23.21.76.23",nonce="48235aef53d38bbdccb25be97f6ac3599ce752eb",uri="sip:+17592034561@23.21.76.23",response="2993367e31be586485909086d44a328d",algorithm=MD5 Content-Length: 270 v=0 o=- 10556151 10556316 IN IP4 43.25.32.85 s=eyeBeam c=IN IP4 231.43.25.1 t=0 0 m=audio 54864 RTP/AVP 18 101 a=alt:1 1 : 7DBA1F5A 00000092 10.0.0.1 8360 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=direction:active a=nortpproxy:yes ÄY#HÖº Ê Ê _d E º** 2Ws@®NÇèNÃÄÄ¦ SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#HY Ö Ö _d E Æ+ 2V@®NÇèNÃÄÄ² SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 Via: SIP/2.0/UDP 23.21.76.23;branch=z9hG4bKa52d.81f76261.0 Via: SIP/2.0/UDP 43.25.32.85:50006;branch=z9hG4bK-d87543-918910683-1--d87543-;rport=8394 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a Call-ID: 5d1bb15abd63917a CSeq: 2 INVITE Content-Length: 0 ÄY#H v v 0H% Efè×@ @K ÇèNÃ@®NÄÄRúACK sip:17592034561@13.45.213.245 SIP/2.0 Via: SIP/2.0/UDP 123.123.1.234;branch=z9hG4bKa52d.b2b867d1.0 From: 19846572302<sip:+19846572302@23.21.76.23>;tag=5d411c09 Call-ID: 5d1bb15abd63917a To: <sip:+17592034561@23.21.76.23>;tag=gK0cf56d6a CSeq: 2 ACK User-Agent: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 --------------------------------- Download prohibited? No problem. CHAT from any browser, without download.

1 0

[OpenSER-Users] OpenSER 1.3.2 is released
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi, I'm happy to announce that OpenSER 1.3.2 is out - after almost 6 months from the first 1.3 release, 1.3.2 is available and it includes valuable experiences and feedback from operating 1.3.x in live platform. This means more stability, more reliability. This is a minor release, so it includes only fixes. That means that configuration and DB structure was not changes and it is fully compatible with all the other releases from 1.3 branch. A comprehensive compilation of fixes (thanks to Henning Westerholt) is available via the Changelog - see http://www.openser.org/pub/openser/1.3.2/ChangeLog For more about how to download this version, see http://www.openser.org/mos/view/News/NewsItem/OpenSER-1.3.2-is-released/ If people want to contribute with packages generated on different OS, please let me know - I will be glad to upload them on the repository. Regards, Bogdan

1 0

[OpenSER-Users] [RFC] - siptrace change
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi, I would like to get your opinion/comments on some siptrace issue. In 1.3, a new capability was added in siptrace - to trace the ACK resulted from sending stateless replies. The problem I discovered with this is that it cannot be controlled by selecting the messages or transactions to be be traced. Usually you use siptrace to trace only certain transactions / messages - tracing all traffic is usually not a realistic option. So, with the stateless ACK, the siptrace module cannot apply the selection from script and trace them all, with no filtering at all. So, if you want to trace a traffic for a single user, all ACK going through the platform will be traced. This reduces the usability of the module and you get a lot of garbage tracing. My suggestion will be to disable stateless ACK tracing until a way to control/filter it is found. I'm asking this considering the upcoming 1.3.2 release from tomorrow. Regards, Bogdan

3 8

Re: [OpenSER-Users] problem in audio when using media proxy
by Krunal Patel 15 May '08

15 May '08

Hi, Actually I am new to openser. Here i am sending trace from openser. So please check out SDP. *************************************************************************************************************************************** Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05 From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28079 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48297, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60e0960, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) parse_headers: flags=10000 0(14396) pre_auth(): Credentials with given realm not found 0(14396) build_auth_hf(): 'Proxy-Authenticate: Digest realm=" voip.domain.com", nonce="482c8213b23f69ac2b268e593620cb2367dc643b" ' 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) WARNING: script writer didn't release transaction 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60e09a8 (10704) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=b802e73ab92fec121a4f880263d541a3.3a34 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [83]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <INVITE> 0(14396) uri: <sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: INVITE sip:5555555555@voip.domain.com:5061 -> sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> : 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:28:55 2008]: Request: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG:parse_to:end of header reached, state=10 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [41]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> > ] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=<null> 0(14396) SIP message-id: 201 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: INVITE sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com>SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>> Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE Proxy-Authorization: Digest username="5555555555",realm="voip.domain.com ",nonce="482c8213b23f69ac2b268e593620cb2367dc643b",response="7a7d3cf46a5b7cf90cc623664ca52e4d",uri=" sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>" Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 291 v=0 o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35 s=X-Lite c=IN IP4 122.xxx.xxx.35 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=291 0(14396) found end of header 0(14396) find_first_route: No Route headers found 0(14396) loose_route: There is no Route HF 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=0 0(14396) DEBUG: RFC3261 transaction matching failed 0(14396) DEBUG: t_lookup_request: no transaction found 0(14396) DBG: trans=0xb60df1c8, callback type 1, id 0 entered 0(14396) ##########################method=INVITE################################## 0(14396) ##############################client_nat_test(7)############################## 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) DEBUG:check_content_type: type <application/sdp> found valid 0(14396) parse_headers: flags=80 0(14396) check_nonce(): comparing [482c8213b23f69ac2b268e593620cb2367dc643b] and [482c8213b23f69ac2b268e593620cb2367dc643b] 0(14396) DEBUG:auth_radius:radius_authorize_sterman: Success 0(14396) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 0(14396) ERROR:avp_radius:avp_load_user: rc_auth failed 0(14396) DEBUG:usrloc:db_load_urecord: aor not found in DB 0(14396) lookup(): '999999999(a)voip.domain.com' Not found in usrloc 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) alias_db_lookup: no alias found for R-URI 0(14396) does_uri_exit(): User in request uri does not exist 0(14396) parse_headers: flags=78 0(14396) ------[sip:5555555555@voip.domain.com:5061] ---checkeding---- 0(14396) radius_is_user_in(): Success 0(14396) -------- media route 5 ---- 0(14396) ########################## isbflagset(6) isbflagset(7)#################################### 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=8000000 0(14396) DEBUG:lcr:load_gws: add matched_gws[0]=[0,0] 0(14396) load_gws(): DEBUG: Added gw_uri_avp <sip:|0@213.XXX.XXX.28 :5060;transport=udp> 0(14396) -----------------lcr is loaded into the ------ 0(14396) load_gws(): DEBUG: Added ruri_user_avp <999999999> 0(14396) ------------- Gateway is loaded --------- 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) parse_headers: flags=2000 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) clen_builder: content-length: 313 (313) 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:tm:set_timer: relative timeout is 500000 0(14396) DEBUG: add_to_tail_of_timer[4]: 0xb60df314 (10701100000) 0(14396) DEBUG:tm:set_timer: relative timeout is 20 0(14396) DEBUG: add_to_tail_of_timer[0]: 0xb60df330 (10720) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <100> 0(14396) reason: <Trying> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=100 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=-1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <183> 0(14396) reason: <Session Progress> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=0, uac[0]=100 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=0, new_code=183 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 884, new size: 822 0(14396) build_res_from_sip_res: copied size: orig:318, new: 256, rest: 566 msg= SIP/2.0 183 Session Progress Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) WARNING:vqm_resize: resize(0) called 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 1..., shmem=0xb60e8218: SIP/2.0 1 0(14396) DEBUG:tm:set_timer: relative timeout is 50 0(14396) DEBUG: add_to_tail_of_timer[1]: 0xb60df330 (10750) 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:4,tl=0xb60df314 next=(nil), timeout=10701100000 1(14397) DEBUG: timer routine:2,tl=0xb60e09a8 next=(nil), timeout=10704 1(14397) DEBUG: wait_handler : removing 0xb60e0960 from table 1(14397) DEBUG: delete transaction 0xb60e0960 1(14397) DEBUG: wait_handler : done 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=183, uac[0]=183 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=183, new_code=200 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG: add_to_tail_of_timer[2]: 0xb60df210 (10710) 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Reply (status): 0(14396) version: <SIP/2.0> 0(14396) status: <200> 0(14396) reason: <OK> 0(14396) parse_headers: flags=2 0(14396) get_hdr_field: cseq <CSeq>: <28080> <INVITE> 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) Found param type 232, <branch> = <z9hG4bKfacb.bd19b862.0>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) grep_sock_info - checking if host==us: 13==23 && [195.XXX.XXX.51] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:forward_reply: found module nathelper, passing reply to it 0(14396) DEBUG:forward_reply: found module tm, passing reply to it 0(14396) DEBUG: t_check: start=0xffffffff 0(14396) parse_headers: flags=22 0(14396) parse_headers: flags=8 0(14396) DEBUG: t_reply_matching: hash 48303 label 646681051 branch 0 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_reply_matching: reply matched (T=0xb60df1c8)! 0(14396) DEBUG: t_check: end=0xb60df1c8 0(14396) DEBUG:tm:reply_received: org. status uas=200, uac[0]=200 local=0 is_invite=1) 0(14396) DEBUG:tm:t_should_relay_response: T_code=200, new_code=200 0(14396) DEBUG:tm:t_should_relay_response: 200 OK for INVITE after final sent 0(14396) DEBUG:tm:relay_reply: branch=0, save=0, relay=0 0(14396) old size: 891, new size: 829 0(14396) build_res_from_sip_res: copied size: orig:304, new: 242, rest: 587 msg= SIP/2.0 200 OK Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 INVITE From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Content-Length: 209 Content-Type: application/sdp Supported: replaces Contact: sip:999999999@213.XXX.XXX.28:5060 User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45 00a0ba009b05 v=0 o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28 s=SIP Call c=IN IP4 213.XXX.XXX.28 t=0 0 m=audio 19462 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv 0(14396) DEBUG:tm:relay_reply: sent buf=0x819a510: SIP/2.0 2..., shmem=0xb60e05e0: SIP/2.0 2 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list (nil) 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 206 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 0(14396) SIP Request: 0(14396) method: <ACK> 0(14396) uri: <sip:999999999@213.XXX.XXX.28:5060> 0(14396) version: <SIP/2.0> 0(14396) parse_headers: flags=2 0(14396) Found param type 235, <rport> = <n/a>; state=6 0(14396) Found param type 232, <branch> = <z9hG4bK384293560A7C9508E28212EC0D72CDFA>; state=16 0(14396) end of header reached, state=5 0(14396) parse_headers: Via found, flags=2 0(14396) parse_headers: this is the first via 0(14396) After parse_msg... 0(14396) preparing to run routing scripts... 0(14396) DEBUG : sl_filter_ACK: to late to be a local ACK! 0(14396) In request log route 0(14396) 0(14396) ############################## NEW REQUEST RECEIVED ############################## 0(14396) parse_headers: flags=10 0(14396) DEBUG: add_param: tag=2027736222 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={5555555555}, ruri={ sip:5555555555@voip.domain.com:5061} 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: ACK sip:5555555555@voip.domain.com:5061 -> sip:999999999@213.XXX.XXX.28:5060: 0(14396) request from 122.xxx.xxx.35 [Thu May 15 20:29:02 2008]: Request: ACK sip:999999999@213.XXX.XXX.28:5060 0(14396) From: sip:5555555555@voip.domain.com:5061;tag=2027736222 0(14396) parse_headers: flags=8 0(14396) DEBUG: add_param: tag=5430ebc2c5d6575 0(14396) DEBUG:parse_to:end of header reached, state=29 0(14396) DBUG:parse_to: display={}, ruri={sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> } 0(14396) DEBUG: get_hdr_field: <To> [61]; uri=[ sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>] 0(14396) DEBUG: to body [<sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> >] 0(14396) To: sip:999999999@voip.domain.com<sip%3A999999999(a)voip.domain.com> ;tag=5430ebc2c5d6575 0(14396) SIP message-id: 207 0(14396) parse_headers: flags=40 0(14396) Call-id: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 0(14396) parse_headers: flags=20 0(14396) get_hdr_field: cseq <CSeq>: <28080> <ACK> 0(14396) CSeq: 28080 0(14396) Contact: <sip:5555555555@122.xxx.xxx.35:5061> 0(14396) Message Body: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 70 Content-Length: 0 0(14396) parse_headers: flags=100 0(14396) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(14396) #############################record_route############################### 0(14396) #######################After record_route########################3 0(14396) ##################################Before loose_route########################## 0(14396) parse_headers: flags=200 0(14396) is_preloaded: No 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) grep_sock_info - checking if host==us: 14==23 && [213.XXX.XXX.28] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) DEBUG:check_self: host != me 0(14396) grep_sock_info - checking if host==us: 23==23 && [voip.domain.com] == [voip.domain.com] 0(14396) grep_sock_info - checking if port 5060 matches port 5060 0(14396) after_loose: Topmost route URI: 'sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on' is me 0(14396) parse_headers: flags=200 0(14396) DEBUG: get_hdr_body : content_length=0 0(14396) found end of header 0(14396) find_next_route: No next Route HF found 0(14396) after_loose: No next URI found 0(14396) #################IN Loose_route############################ 0(14396) DEBUG: has_totag: totag found 0(14396) ############################has_totag() method==INVITE method==ACK################################ 0(14396) parse_headers: flags=80 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) parse_headers: flags=8000000 0(14396) ###########################setbflag6################################# 0(14396) ###############################force_rport############################# 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) #############################use_media_proxy############################### 0(14396) parse_headers: flags=78 0(14396) DEBUG: t_newtran: T on entrance=0xffffffff 0(14396) parse_headers: flags=ffffffffffffffff 0(14396) parse_headers: flags=78 0(14396) t_lookup_request: start searching: hash=48303, isACK=1 0(14396) parse_headers: flags=38 0(14396) DEBUG:tm:REF_UNSAFE: after is 1 0(14396) DEBUG: t_lookup_request: e2e proxy ACK found 0(14396) DEBUG:tm:t_newtran: building branch for end2end ACK 0(14396) DEBUG:tm:t_relay: forwarding ACK 0(14396) DEBUG: mk_proxy: doing DNS lookup... 0(14396) check_via_address(122.xxx.xxx.35, 122.xxx.xxx.35, 0) 0(14396) DBG:check_against_rule_list: using list dns 0(14396) DEBUG:forward_request: sending: ACK sip:999999999@213.XXX.XXX.28:5060 SIP/2.0 Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on> Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2 Via: SIP/2.0/UDP 122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222 To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com> >;tag=5430ebc2c5d6575 Contact: <sip:5555555555@122.xxx.xxx.35:5061> Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10 CSeq: 28080 ACK Max-Forwards: 69 Content-Length: 0 . 0(14396) DEBUG:forward_request: orig. len=519, new_len=593, proto=1 0(14396) -------------- cheked t reply --- 0(14396) DEBUG:tm:UNREF_UNSAFE: after is 0 0(14396) DEBUG:destroy_avp_list: destroying list 0xb60e04c8 0(14396) receive_msg: cleaning up 1(14397) DEBUG: timer routine:2,tl=0xb60df210 next=(nil), timeout=10710 1(14397) DEBUG: wait_handler : removing 0xb60df1c8 from table 1(14397) DEBUG: delete transaction 0xb60df1c8 1(14397) DEBUG: wait_handler : done *************************************************************************************************************************************** -- Thanks & regards, Krunal Patel On Thu, May 15, 2008 at 12:03 AM, LetMeKnow < sunkara.raviprakash.feb14(a)gmail.com> wrote: > hey krunal, > i hopes ur fine and also all members of Ecosmob are fine, > > i have read ur mail, > yours problem is mention properly in mailing list. > > In ur openser script file some problem in there and u not handling the nat > in ur script. > make sure u have use both modules nathelper and mediaproxy, > > Before doing on this checkout is openser system is located inside the nat > or not. > > while test use ngrep tool on openser system, see whether u have got the > properly signaling happening not. > > check out after 200 ok of invite msg > > > On Wed, May 14, 2008 at 5:40 PM, Krunal Patel <krunal.patel(a)ecosmob.com> > wrote: > >> Hi, >> >> We are using openser with media proxy and radiator. We're having audio >> problems when we send calls to mediatrix gateway for termination. >> Internal calls work fine. We hear audio both side and we can see the >> session and bytes transfer properly in media proxy monitoring webpage. >> When we send calls to mediatrix gateway for termination, calls connect >> properly but we don't hear audio. We see inactive session in mediaproxy >> status page with no bytes transferring. >> One doubt I have is it is not entering reply route when it receives >> replies like 183, 180, etc. >> Has anyone encountered similar issue? How can we fix it? >> Thanks in advance! >> >> -- >> regards, >> Krunal Patel >> >> >> _______________________________________________ >> 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 > 91-9999882776 >

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Re: [OpenSER-Users] [OpenSER-Devel] [RFC] - siptrace change
by Bogdan-Andrei Iancu 15 May '08

15 May '08

Hi Norman, yup, that is exactly my problem - I use it only for doing siptrace per user and on platform with high traffic, the stateless ACK tracing produces a lot of residual data in DB. Regards, Bogdan Norman Brandinger wrote: > This change would be good. We have reduced our use of siptrace for > this very reason. > > Norm > > Bogdan-Andrei Iancu wrote: >> Hi, >> >> I would like to get your opinion/comments on some siptrace issue. >> >> In 1.3, a new capability was added in siptrace - to trace the ACK >> resulted from sending stateless replies. >> >> The problem I discovered with this is that it cannot be controlled by >> selecting the messages or transactions to be be traced. Usually you >> use siptrace to trace only certain transactions / messages - tracing >> all traffic is usually not a realistic option. >> >> So, with the stateless ACK, the siptrace module cannot apply the >> selection from script and trace them all, with no filtering at all. >> So, if you want to trace a traffic for a single user, all ACK going >> through the platform will be traced. >> >> This reduces the usability of the module and you get a lot of garbage >> tracing. >> >> My suggestion will be to disable stateless ACK tracing until a way to >> control/filter it is found. I'm asking this considering the upcoming >> 1.3.2 release from tomorrow. >> >> Regards, >> Bogdan >> >> _______________________________________________ >> Devel mailing list >> Devel(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/devel >> >> >> > >

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[OpenSER-Users] parallel forking
by Allan Chao ( 趙廷睿 ) 15 May '08

15 May '08

Dear All: I have found there are some devices can't process two 180 ringing after use OPENSER to do parallel forking, so, i have a idea that is used OPENSER to delete To tag in the To header before relay it , and it works well with eyebeam. But i can't find any spec to support my idea because it is against "early dialog" prograss and my question is how does UE(eyebeam or some client) to create dialog if there is no any To tag in 180 ringing but 200 OK has in the latter ? i think the UA will generate local ring tone but no any dialog after receives 180 ringing with no To tag? is it right ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

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[OpenSER-Users] cdrtool error: "cannot figure out the destination id ..."
by Adrian Georgescu 15 May '08

15 May '08

You must provision in the destinations table with all prefixes for PSTN destinations (at least all country codes). The software will find the longest match from that table. Adrian >>>>>> Trying cdrtool prepaid rating, got stuck with something. When a request like "MaxSessionTime From=sip:me at example.com To=sip:0031650222 at example.com Duration=7200 Lock=0" is given, cdrtool always return "0", no matter what amount of balance is available in prepaid table. This is what I see in syslog: "CDRTool rating[5759]: Error: cannot figure out the destination id for sip:0031650222 at example.com" Cdrtool version is 6.3.2. Apparently rating engine needs the variable $CDR->DestinationID set (rating_lib.phtml, line #5926), but this is not happening. It isn't clear to me how this is initialised (In weak defense, I'm totally clueless about php), and it doesn't work even if DestinationID parameter is added in the above stated request.

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[OpenSER-Users] username mangling
by David Terry 14 May '08

14 May '08

Hi Openser, My question is regarding case-handling of SIP usernames with Openser 1.3.1 presence modules. I'm using a stand-alone presence server with pua/pua_usrloc modules. This is the test case: 1. user "sip:1234@10.10.90.51" sends SUBSCRIBE for user "sip:Jane@10.10.90.51" 2. user "sip:Jane@10.10.90.51" sends REGISTER After 2, watcher does not receive NOTIFY from Openser/presence. DB contents is as follows: active_watchers/presentity_uri contains "sip:Jane@10.10.90.51" pua/pres_uri contains "sip:jane@10.10.90.51" presentity/username contains "jane". location/username contains "jane" location/contact contains "sip:Jane@10.10.90.51:10001" I'm not sure why case is different in these tables, and whether this causes the missed NOTIFY. Can anyone suggest what might be wrong? Here's the Openser route block: route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (is_method("SUBSCRIBE|PUBLISH")) route(2); if (method == "REGISTER") { pua_set_publish(); save("location"); exit; } } route[2] { # presence handling if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); }; exit; }

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[OpenSER-Users] SerMyAdmin - Cannot log in as admin
by Wyndham Hudson 14 May '08

14 May '08

Hi, I've set up serMyAdmin on the tomcat server and everything appears to be working, however I don't seem to be able to log in as the admin. I've added a user admin to subscriber with password etc. but it still does not allow me in. MySQL is recieveing the queries and processing them as I have checked in the general-query-log. I'm currently running openser 1.2.x with serMyAdmin 0.5. Any help would be appreciated. Thanks

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[OpenSER-Users] remove message body
by Andreas Granig 14 May '08

14 May '08

Hi, Is there any possibility to remove the whole message body of a SIP message? replace_body_all and subst_body seem to operate per line, leaving a bunch of "\n" behind. There's a two year old feature request for that on the tracker (https://sourceforge.net/tracker/?func=detail&atid=743023&aid=1546501&group_…). Any news on that? Any other possibilities? Andreas

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[OpenSER-Users] Segmentation Fault in Openser 1.3.1 - 64 bits.
by Sergio Gutierrez 14 May '08

14 May '08

Hello everybody. Today we got a crash in our production deployment of OpenSER, which is running version 1.3.1 on Solaris SPARC. OpenSER was compiled on 64 bits mode.file is: The backtrace of the core file is: Program terminated with signal 11, Segmentation fault. #0 0xffffffff79719774 in insert_dlg_timer_unsafe (tl=0xffffffff61ea8ad0) at dlg_timer.c:91 91 if ( ptr->timeout <= tl->timeout ) (gdb) bt #0 0xffffffff79719774 in insert_dlg_timer_unsafe (tl=0xffffffff61ea8ad0) at dlg_timer.c:91 #1 0xffffffff79719a8c in update_dlg_timer (tl=0xffffffff61ea8ad0, timeout=300) at dlg_timer.c:163 #2 0xffffffff7970fe9c in dlg_onroute (req=0x100341fb0, route_params=0xffffffff7fffe2b0, param=0x0) at dlg_handlers.c:612 #3 0xffffffff7d80c278 in run_rr_callbacks (req=0x100341fb0, rr_params=0xffffffff7d910468) at rr_cb.c:83 #4 0xffffffff7d807b0c in after_loose (_m=0x100341fb0, preloaded=0) at loose.c:865 #5 0xffffffff7d803744 in loose_route (_m=0x100341fb0, _s1=0x0, _s2=0x0) at loose.c:901 #6 0x000000010000f088 in do_action (a=0x1002e3e30, msg=0x100341fb0) at action.c:816 #7 0x0000000100009e48 in run_action_list (a=0x1002e3e30, msg=0x100341fb0) at action.c:132 #8 0x0000000100086a08 in eval_elem (e=0x1002e3ed0, msg=0x100341fb0, val=0x0) at route.c:1069 #9 0x000000010008db44 in eval_expr (e=0x1002e3ed0, msg=0x100341fb0, val=0x0) at route.c:1360 #10 0x000000010008dd00 in eval_expr (e=0x1002e3f18, msg=0x100341fb0, val=0x0) at route.c:1381 #11 0x000000010000e0b8 in do_action (a=0x1002e53c8, msg=0x100341fb0) at action.c:672 #12 0x0000000100009e48 in run_action_list (a=0x1002d8878, msg=0x100341fb0) at action.c:132 #13 0x000000010000a4b0 in run_actions (a=0x1002d8878, msg=0x100341fb0) at action.c:112 #14 0x000000010000a6fc in run_top_route (a=0x1002d8878, msg=0x100341fb0) at action.c:172 #15 0x00000001000781ec in receive_msg ( buf=0x1002aaf68 "INVITE sip:76907295@200.114.42.182<sip%3A76907295(a)200.114.42.182>;user=phone SIP/2.0\r\nVia: SIP/2.0/UDP 200.30.79.17:5066;branch=z9hG4bKa8fae90df\r\nRoute: <sip:200.13.225.243;lr;ftag=885a2b1ed6b8a7d8;did=7cf.1721>\r\nCall-ID: f6a12831f"..., len=747, rcv_info=0xffffffff7ffff860) at receive.c:156 #16 0x00000001000db620 in udp_rcv_loop () at udp_server.c:438 #17 0x00000001000321b8 in main_loop () at main.c:834 #18 0x00000001000373bc in main (argc=7, argv=0xffffffff7ffffbe8) at main.c:1373 I would thank any hint you could provide about it. Best regards. Sergio Gutiérrez.

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[OpenSER-Users] sermyadmin almost works
by Lucas Catranis 14 May '08

14 May '08

Anyone Get this setup correctly? I've been at it for a while and after getting it to insert new users I cannot login as an admin. I'm not sure where the problem is but Any help would be appreciated. -- View this message in context: http://www.nabble.com/sermyadmin-almost-works-tp17219053p17219053.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[OpenSER-Users] Where exactly are " sip-clients" and "rtp-clients" supposed to be?
by luzango mfupe 14 May '08

14 May '08

Hi mates, In my O'SER 1.3 config file from svn i have this modules parameters which shows the location of rtp and sip clients but in reality these files doesnt exist in the said location or anywhere in my box, what must be wrong with my config?? below are the two lines: modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clients") -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA website:http://www.tuune.mobi

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[OpenSER-Users] cdrtool error: "cannot figure out the destination id ..."
by Sajith T S 14 May '08

14 May '08

Hi! Trying cdrtool prepaid rating, got stuck with something. When a request like "MaxSessionTime From=sip:me@example.com To=sip:0031650222@example.com Duration=7200 Lock=0" is given, cdrtool always return "0", no matter what amount of balance is available in prepaid table. This is what I see in syslog: "CDRTool rating[5759]: Error: cannot figure out the destination id for sip:0031650222@example.com" Cdrtool version is 6.3.2. Apparently rating engine needs the variable $CDR->DestinationID set (rating_lib.phtml, line #5926), but this is not happening. It isn't clear to me how this is initialised (In weak defense, I'm totally clueless about php), and it doesn't work even if DestinationID parameter is added in the above stated request. Since there seems to be other people using cdrtool with success, I guess might be doing something really stupid. I haven't trudged too far away from a standard cdrtool install. Can anyone tell me what possibly could have gone wrong? Or even better: teach a man how to fish (ie, how to trace out this), so that he might shut up when there's no fish... :) Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06

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[OpenSER-Users] Authentication challenge failure
by Bogdan-Andrei Iancu 14 May '08

14 May '08

Hi, I just found and fix a really strange bug in authentication module, when building the auth challenge header. I say strange, because I found it while using a UAC that implements a very strict view on the auth process. Also this UAC tries to reuse the nonces. This bug had as effect the UACs stopping to re-register with openser after an openser restart. Quite unpleasant effect to have all the UACs dropping out if you do a server restart :( . More technically, the bug consists in openser's failure to append the stale parameter in the challenge request if the nonce is not recognize as local - this can happen after a restart, when openser uses a new schema to generate nonces. Scenario: 1) start openser -> it will set SCHEMA1 for generating nonces 2) UAC registers with authentication and receives during challenge the nonce NONCE1 (based on SCHEMA1) 3) OpenSER restarts and sets a new SCHEMA2 for generating nonces 4) UAC tries to re-register using the previous nonce it received - NONCE1. 5) OpenSER rejects the auth as received NONCE1 does not follow current SCHEMA2. 6) OpenSER sends a new challenge to the UAC, but so far, the stale parameter was not added to indicate that the nonce is invalid 7) UAC simply drops any registration attempts as it thinks that the password it has is wrong -> it authentication was rejected and no stale indication was received. With the fix, openser now adds the stale parameter in the challenge and to indicate to UAC a nonce issue if the nonce is not recognized. The script auth functions were already reporting (as return code) NONCE_STALE indication in this case, but the challange was not properly computed. Have anybody experience such problem also? or I was the first coming across a such UAC :) . Regards, Bogdan

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[OpenSER-Users] compilation on solaris 10/Spar
by Liangyou Wang (liangwan) 14 May '08

14 May '08

Hi experts, Anyone has experience on compilation on solaris 10/Spar? I am start user of openser and need use openser to support sctp. now I have download the source codes of openser-1.3.1-tls. then I compiled it, there have many errors that need your supports. e.g. openser-1.3.1-tls> make all Makefile.defs:717: *** missing separator. Stop. then I commented Makefile.defs line717 and 718. # $(warning Old gcc detected ($(CC_SHORTVER)), use gcc >= 3.1 \ # for better results) Errors were still there. openser-1.3.1-tls> make all Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' yacc -d -b cfg cfg.y "cfg.y", line 391: warning: redeclaration of precedence of SLASH. conflicts: 1 shift/reduce flex cfg.lex statistics.c:58: warning: #warning STATISTICS: Architecture with no support for atomic operations. Using Locks!! Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' Compiling action.c gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o parser/msg_parser.h: In function `char_msg_val': In file included from action.h:27, from action.c:44: parser/msg_parser.h:300: parse error before `)' forward.h: In function `msg_send': In file included from action.c:49: forward.h:95: parse error before `)' forward.h:103: parse error before `)' forward.h:107: parse error before `)' mem/../lock_alloc.h: In function `lock_set_alloc': In file included from mem/../locking.h:66, from mem/../statistics.h:115, from mem/shm_mem.h:33, from ut.h:52, from action.c:54: mem/../lock_alloc.h:68: parse error before `)' ut.h: In function `int2bstr': In file included from action.c:54: ut.h:172: parse error before `)' ut.h: In function `un_escape': ut.h:365: parse error before `)' ut.h: In function `shm_str_dup': ut.h:498: parse error before `)' ut.h: In function `pkg_str_dup': ut.h:515: parse error before `)' ut.h: In function `str_strcmp': ut.h:532: parse error before `)' ut.h:535: parse error before `int' ut.h:539: `minlen' undeclared (first use in this function) ut.h:539: (Each undeclared identifier is reported only once ut.h:539: for each function it appears in.) ut.h:547: `alen' undeclared (first use in this function) ut.h:547: `blen' undeclared (first use in this function) ut.h: In function `str_strcasecmp': ut.h:563: parse error before `)' ut.h:566: parse error before `int' ut.h:570: `minlen' undeclared (first use in this function) ut.h:578: `alen' undeclared (first use in this function) ut.h:578: `blen' undeclared (first use in this function) action.c: In function `run_action_list': action.c:140: parse error before `)' action.c: In function `do_assign': action.c:193: parse error before `)' action.c:202: parse error before `)' action.c:222: parse error before `)' action.c:228: parse error before `)' action.c: In function `do_action': action.c:302: parse error before `)' action.c:309: parse error before `)' action.c:335: parse error before `)' action.c:498: parse error before `)' action.c:524: parse error before `)' action.c:682: parse error before `)' action.c:710: parse error before `)' action.c:723: parse error before `)' action.c:757: parse error before `)' action.c:763: parse error before `)' action.c:807: parse error before `)' action.c:819: parse error before `)' action.c:870: parse error before `)' action.c:896: parse error before `)' action.c:904: parse error before `)' action.c:930: parse error before `)' action.c:939: parse error before `)' action.c:945: parse error before `)' action.c:966: parse error before `)' make: *** [action.o] Error 1 Your any help is much appreciated! BR, Eric

2 1

[OpenSER-Users] [OT?] voip-to-voip rating using cdrtool
by Adrian Georgescu 14 May '08

14 May '08

You can rate sip calls to specific SIP URIs, which are not telephone numbers in CDRTool. The SIP URIs must be fully qualified, that is you can set a price for user(a)example.com but you cannot do a wildcard matching for *(a)example.com Just add the SIP URI in the destination table as any other PSTN prefix. Regards, Adrian >>>>>>>>> Apologies if this is off-topic here, but this seemed like a good friendly place to ask. Apologies for being so naïve also... Question is, how can I get cdrtool to rate voip-to-voip calls when destination is non-numeric? As in, say, sip:holmes at example.com calls sip:watson at example.com. How do I get cdrtool to match *(a)example.com or *(a)example2.com? (FWIW, I'm just trying to see how this works... no plan of rolling out this Evil Scheme into a real world deployment.) While cdrtool has decent documentation, I wish this was a little more easy on me. Has anyone read/reviewed this book yet? http://www.packtpub.com/building-telephony-systems-with-openser/book I'm specifically interested in the "Billing with Freeradius and CDRTool" part. Thanks, Sajith.

2 1

[OpenSER-Users] presence not working with X-lite
by Amit Vijayvargiya 14 May '08

14 May '08

Hi, we are done all the configuration of presence module in openser.cfg file set all the parameters..........i.e. modparam("presence|presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:domain:5060") modparam("presence", "fallback2db", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "to_tag_pref", 'pres') modparam("presence", "expires_offset", 10) modparam("xcap_client", "db_url", "mysql://openser:openser@localhost/openser") modparam("xcap_client", "xcap_table","xcap") modparam("presence_xml", "db_url", "mysql://openser:openser@localhost/openser") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 1) modparam("presence_xml", "integrated_xcap_server", 1) and route the PUBLISH and SUBSCRIBE massages if( is_method("PUBLISH|SUBSCRIBE")) route(2); route[2] is::::: route[2] { sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } when we are run X-lite is asking you want to allow or deny the user-B........... but not showing online even we are allow the user-B.......... when we are allow the user x-lite not send any type of message...... we see in Notify openser sending a notify with watcher-list but with the status pending inside XML format. please tell me where we are wrong???????????? there is a problem with X-lite or our configuration????? Thanks in advance Amit Vijayvargiya

1 0

[Serusers] Presence Server
by Allan Chao ( 趙廷睿 ) 14 May '08

14 May '08

Dear All: I want to use SER's Presence module to do presence server , but someone say it's not stable , so , 1. Does anyone could tell me why it doesn't stable ? because it looks work after i do some testing with eyebeam. 2. how does openser's presence ? is it better than SER's presence ? thanks, allan DISCLAIMER: Sample Disclaimer added in a VBScript.

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[Serusers] TLS problem ser 2_0_0 - ERROR: tls_init.c:366: Unable to set the memory allocation functions
by .:: Francesco la Torre ::. 13 May '08

13 May '08

Have you found any solution for this issue ? I'm facing with a similar problem (not the same because the source of same error is line 520). Thanks in advance Francesco la Torre IIT/CNR-Pisa

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[OpenSER-Users] Presence, multiple status info handling
by Dan-Cristian Bogos 13 May '08

13 May '08

Guys, while dealing with presence/presence_xml modules, I came into one situation when one user crashes simultaneously and each time displaying different presence info (online, away, offline). Have noticed that the presence server is smart enough to inform all the other watchers about three different status information for the same user. My question is, how can the watcher know the most recent status information. Have looked on both order of info in the tuple as well as on ids but did not find any logic. Can u please guide me about possibilities I have? Thank you in advance, DanB PS: FYI, I am pasting such a tuple example. <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t9062"> <status xmlns="urn:ietf:params:xml:ns:pidf"> <basic>closed</basic> </status> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p8838"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5358"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4226"><rpid:activities><rpid:away/></rpid:activities></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t7139"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Away</note> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="p4606"><rpid:activities/></dm:person> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="t5632"> <status><basic>open</basic></status><contact>sip:user@domain.com<sip%3Auser(a)domain.com> </contact><note>Online</note> </tuple></presence>

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[OpenSER-Users] Routing Logic
by VoIP Forums www.Go4Calls.com 13 May '08

13 May '08

Hi Friends, I have the following route plan in my openser.cfg all blocks are working well instead of 800800. if (uri=~"sip:800800[1-9][0-9]+@.*") { rewritehostport("officePBX-IP:5060"); route(1); exit; } else if ($(rU{s.len})>=8) { rewritehostport("MyPSTNprovider-IP:5061"); route(1); exit; }else{ lookup("location"); route(1); exit; }; When i am trying to call 800800 the error message appearing is "513 Message too big" 800800 is routed to the asterisk for our internal office's extensions. Pls guide us how i can solve this issue. Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[OpenSER-Users] Authentcation using DISC is successful
by Dilip 13 May '08

13 May '08

Hello Sir, I have been able to make the DISC code compatible with openser-1.1.1-notls. I have eliminated some of the bugs which were there in the source code of DISC.But the thing is that the disc code is not able to work with x86_64 architecture. But it is successfully running with i386 arch. Could u please suggest me the neccessary task,i have to perform in order to make DISC compiled on x86_64 arch. I tried taking the successfully running code from i386 arch and copying it on x86_64 arch and then to compile it, but it is giving some architecture compilation problem which is given below. In file included from ../mem/shm_mem.h:53, from avp.c:33: ../mem/../lock_ops.h:182:2: error: #error "no locking method selected" ../mem/../lock_ops.h:265:2: error: #error "no lock set method selected" In file included from avp.c:33: ../mem/shm_mem.h:79: error: expected '=', ',', ';', 'asm' or '__attribute__' before '*' token ../mem/shm_mem.h: In function '_shm_malloc': ../mem/shm_mem.h:111: warning: implicit declaration of function 'lock_get' ../mem/shm_mem.h:111: error: 'mem_lock' undeclared (first use in this function) ../mem/shm_mem.h:111: error: (Each undeclared identifier is reported only once ../mem/shm_mem.h:111: error: for each function it appears in.) ../mem/shm_mem.h:113: warning: implicit declaration of function 'lock_release' avp.c: In function 'AAACreateAVP': avp.c:124: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:140: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c: In function 'AAAFreeAVP': avp.c:321: error: 'mem_lock' undeclared (first use in this function) avp.c:321: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c:323: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type avp.c: In function 'AAACloneAVP': avp.c:438: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:448: warning: passing argument 3 of '_shm_malloc' discards qualifiers from pointer target type avp.c:451: error: 'mem_lock' undeclared (first use in this function) avp.c:451: warning: passing argument 4 of 'qm_free' discards qualifiers from pointer target type make[2]: *** [avp.lo] Error 1 Please Do help me ,to sort out the problem. regards, Dilip

1 0

[Serusers] Asterisk->SER rtp problems
by Thorsten 13 May '08

13 May '08

Hi, I have an asterisk server running with an private IP. This asterisk forwards all calls to a SER server with a public IP. The SER server then forwards its calls to a public SIP provider. The problem now is that SER tries to stay in the loop which it shouldn't because there is no media proxy running. I don't get any audio because of this issue. But if I register the asterisk box directly to the SIP provider it works. Does anybody know how to fix this. My ser.cfg debug=4 # debug level (cmd line: -dddddddddd) #debug=0 fork=no log_stderror=yes # (cmd line: -E) check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #listen=0.0.0.0 #listen=82.98.89.140 port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/opt/ser/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser/lib/ser/modules/sl.so" loadmodule "/usr/local/ser/lib/ser/modules/tm.so" loadmodule "/usr/local/ser/lib/ser/modules/rr.so" loadmodule "/usr/local/ser/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser/lib/ser/modules/textops.so" loadmodule "/usr/local/ser/lib/ser/modules/avpops.so" #loadmodule "/usr/local/ser/lib/ser/modules/group.so" loadmodule "/usr/local/ser/lib/ser/modules/xlog.so" loadmodule "/usr/local/ser/lib/ser/modules/auth.so" loadmodule "/usr/local/ser/lib/ser/modules/nathelper.so" loadmodule "/usr/local/ser/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/ser/lib/ser/modules/auth.so" #loadmodule "/opt/ser/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) #modparam("registrar", "nat_flag", 6) #modparam("nathelper", "natping_interval", 30) # Ping interval 30 s #modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- 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"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } if (method=="REGISTER") { #record_route(); save("location"); break; }; if (method=="INVITE") { #if (uri =~ "sip:[0-9]@*") { # if (nat_uac_test("19")) { # fix_nated_contact(); # fix_nated_sdp("3"); # } # route(3); # break; #} if (uri =~ "sip:[0-9]@*") { # record_route(); route(3); break; } }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3] { if (uri =~ "sip:[0-9]@*") { log(1, "Forwarding to mg3.net-m.de \n"); #rewritehostport("192.168.13.102:5060"); rewritehostport("62.214.145.199:5060"); #forward(62.214.145.199, 5060); route(1); break; } } My extensions.conf [toser] exten => _X.,1,Dial(sip/${EXTEN}(a)82.98.89.139) Thanks for any help Ciao Thorsten

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[OpenSER-Users] how to set FROM to a different uri
by sye 13 May '08

13 May '08

HI, I use aliases table set sip forwarding. During the forwarding I want to do is set FROM uri to a different sip address. ( I want to hide the original FROM uri). How do I do it? I check the cook book and don't see any function that I can use to do that work. Any hint? Thanks, Stanley

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[OpenSER-Users] regarding presence module facing problem
by Amit Vijayvargiya 13 May '08

13 May '08

hi, i m add presence module in openser 1.3. but it not working properly. when x-lite send a subscribe to client-B, it receive notify but with different body. i see in the table presentity it shows right body. where is the problem????? my configuration like this for subscribe and publish...... route[2] { sl_send_reply("100","trying"); append_to_reply("Contact: <sip:sip.pyrogroup.com:5060>\r\n"); if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) handle_publish("$hdr(Sender)"); else handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } Thanks & Regards, Amit

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[Serusers] SER 2.0 Question
by Santiago Gimenez Ocano 12 May '08

12 May '08

Hi, I'm using SER 2.0 rc1. Thing is I have found excelent documentation (like tm module) on iptel pages, but I couldn't find information for other modules like acc_radius. There's a web site or something where I can find this kind of help? Thanks, Santiago _________________________________________________________________ Do more with your photos with Windows Live Photo Gallery. http://www.windowslive.com/share.html?ocid=TXT_TAGLM_Wave2_photos_022008

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Re: [OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 09 May '08

09 May '08

1 0

Re: [OpenSER-Users] 200 OK retransmissions on missing ACK can cause subsequent calls to fail
by Sean O'Donnell 09 May '08

09 May '08

2 1

[Serusers] SER-Server-ID, Radius Dictionary Issue
by Santiago Gimenez Ocano 09 May '08

09 May '08

Hi all, I'm trying to use acc_radius module in SER 2.0.0. But when it's loading I get this error: 0(9009) ERROR: acc: can't get code for the SER-Server-ID attribute 0(9009) init_mod(): Error while initializing module acc_radius ERROR: error while initializing modules I think this is a dictionary error, when I add the attribute I get the following message: 0(9031) ERROR: sr_module.c:887: Error while fixing parameter, AVP, SELECT, and int conversions failed I assume I'm making some mistake, but I don't know where. I don't think that is in ser.cfg and maybe is a module or dictionary problem. Can anybody help me? Thanks, Santiago _________________________________________________________________ Get your fix of news, sports, entertainment and more on MSN Mobile http://www.msnmobilefix.com/Default.aspx

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Re: [Serusers] SER - Strange Calling Issues
by Arun Kumar 09 May '08

09 May '08

> > Hi ALL, > > I'm facing some strange issue with my VoIP Setup. I've configure 5 servers > : > > 1. 192.168.20.1 = SER / MediaProxy > 2. 192.168.20.2 = MySQL > 3. 192.168.20.3 = Presence > 4. 192.168.20.4 = Asterisk for PSTN with A2Billing > 5. 192.168.20.5 = Voicemail > > *First Problem:* > I've created few users on my SER and they are able to call each other > (Users are behind NAT) *but if User A calls User B and User A disconnect > call from his side the User B phone keeps on ringing until User B > disconnects the call. > > Second Problem(Only some time): > *when user dial PSTN number they hear ringing on there side but other > side has picked up the phone and they started to talk also but calling side > is still hearing the ring. > > *Third: > *Some time users complain that there calls are getting disconnect (PSTN > Calls) after 20-30 second, I feel its more related to my PSTN Provider but > how can I prove so I can tell my provider about this problem. > > Please provide your feedback. thanks in advance. > > I've attached my ser.cfg file b'coz if I'll paste message will be big so. > =================== > >

1 0

[Serusers] CANCEL doubts
by Nuno Ribeiro 08 May '08

08 May '08

Hi all, I have a doubt about the CANCEL messages and I would like to hear your opinion. The routes that should be inserted in the CANCEL message should be based int the ones that are part of the received INVITE or in the ones that are part of the transmitted INVITE? From my point of view the CANCELs should be created based in the routes of the transmitted INVITE, in order to the stateless proxies be able to route the CANCEL requests properly. The RFC says the following: 9.1 Client Behavior A CANCEL request SHOULD NOT be sent to cancel a request other than INVITE. Since requests other than INVITE are responded to immediately, sending a CANCEL for a non-INVITE request would always create a race condition. The following procedures are used to construct a CANCEL request. The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. A CANCEL constructed by a client MUST have only a single Via header field value matching the top Via value in the request being cancelled. Using the same values for these header fields allows the CANCEL to be matched with the request it cancels (Section 9.2 indicates how such matching occurs). However, the method part of the CSeq header field MUST have a value of CANCEL. This allows it to be identified and processed as a transaction in its own right (See Section 17). If the request being cancelled contains a Route header field, the CANCEL request MUST include that Route header field's values. This is needed so that stateless proxies are able to route CANCEL requests properly. Thanks in advance, Nuno Ribeiro

2 1

[OpenSER-Users] recover from previous strict routing triggered incorrectly in ALG scenario
by Andrew Pogrebennyk 08 May '08

08 May '08

Hi, Im'm cross-posting this from OpenSER-Devel because I am not 100% sure this is a bug. I am trying to create openser configuration for proxy that connects private (VPN) network with public network. I started with a config similar to http://voipembedded.com/resources/openser_cr.cfg, added a call to force_send_socket() to send from public IP and rtpproxy/nathelper in bridge mode. As per config loose_route() is called from openser.cfg to route a sequential request within a dialog through record-routing. Now when a BYE from callee comes in, it contains two Route headers: Route: <a.b.c.d;r2=on;lr;ftag=...> where a.b.c.d is server's public IP, Route: <10.0.0.100;r2=on;lr;ftag=...> where 10.0.0.100 is a private IP and username(a)a.b.c.d in Request-URI. Take a look at loose_route(): /* * Do loose routing as defined in RFC3261 */ int loose_route(struct sip_msg* _m, char* _s1, char* _s2) { int ret; if (find_first_route(_m) != 0) { LM_DBG("There is no Route HF\n"); return -1; } if (parse_sip_msg_uri(_m)<0) { LM_ERR("failed to parse Request URI\n"); return -1; } ret = is_preloaded(_m); if (ret < 0) { return -1; } else if (ret == 1) { return after_loose(_m, 1); } else { #ifdef ENABLE_USER_CHECK if (is_myself(&_m->parsed_uri.user, &_m->parsed_uri.host, _m->parsed_uri.port_no)) { #else if (is_myself(&_m->parsed_uri.host, _m->parsed_uri.port_no)) { #endif return after_strict(_m); } else { return after_loose(_m, 0); } } } Obviously, is_myself() check passes and after_strict() is called. when after_strict() is called, it looks if the URI given in Route contains lr parameter which marks loose routers, if so this block of code is executed: { LM_DBG("Next hop: '%.*s' is loose router\n", uri.len, ZSW(uri.s)); if(get_maddr_uri(&uri, &puri)!=0) { LM_ERR("failed to check maddr\n"); return RR_ERROR; } if (set_dst_uri(_m, &uri) < 0) { LM_ERR("failed to set dst_uri\n"); return RR_ERROR; } /* Next hop is a loose router - Which means that is is not endpoint yet * In This case we have to recover from previous strict routing, that * means we have to find the last Route URI and put in in R-URI and * remove the last Route URI. */ if (rt != hdr->parsed) { /* There is a previous route uri which was 2nd uri of mine * and must be removed here */ rem_off = hdr->body.s; rem_len = rt->nameaddr.name.s - hdr->body.s; if (!del_lump(_m, rem_off - _m->buf, rem_len, 0)) { LM_ERR("failed to remove Route HF\n"); return RR_ERROR; } } res = find_rem_target(_m, &hdr, &rt, &prev); if (res < 0) { LM_ERR("searching for last Route URI failed\n"); return RR_ERROR; } else if (res > 0) { /* No remote target is an error */ return RR_ERROR; } uri = rt->nameaddr.uri; if(get_maddr_uri(&uri, 0)!=0) { LM_ERR("checking maddr failed\n"); return RR_ERROR; } if (rewrite_uri(_m, &uri) < 0) { LM_ERR("failed to rewrite R-URI\n"); return RR_ERROR; } /* The first character if uri will be either '<' when it is the * only URI in a Route header field or ',' if there is more than * one URI in the header field */ LM_DBG("The last route URI: '%.*s'\n", rt->nameaddr.uri.len, ZSW(rt->nameaddr.uri.s)); if (prev) { rem_off = prev->nameaddr.name.s + prev->len; rem_len = rt->nameaddr.name.s + rt->len - rem_off; } else { rem_off = hdr->name.s; rem_len = hdr->len; } if (!del_lump(_m, rem_off - _m->buf, rem_len, 0)) { LM_ERR("failed to remove Route HF\n"); return RR_ERROR; } } Therefore recover from previous strict routing is triggered: contents of request line is destroyed and 10.0.0.100 is put instead: Apr 15 16:16:24 vpn-proxy oser[11027]: DBG:core:rewrite_uri: rewriting Request-URI with 'sip:10.0.0.100;r2=on;lr;ftag=aa40c2a5-54fb-1810-970d-0015c5bf2da5' Apr 15 16:16:24 vpn-proxy oser[11032]: [1] Got in dialog request not catched by loose_route(): BYE sip:10.0.0.100;r2=on;lr;ftag=aa40c2a5-54fb-1810-970d-0015c5bf2da5 SIP/2.0^M Via:... And we respond with 404 to in-dialog requests not caught by loose_route(). Previous hop is SER and I don't know why OpenSER considered it a strict router. Looks like a peculiarity of ALG mode. Please advice what should I do to solve this problem. Is it a configuration issue or perhaps loose_route() should not call after_strict() in case of double record-routing? -- Thank you, Andrew Pogrebennyk

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[OpenSER-Users] segfault with general protection error
by Marcello Lupo 08 May '08

08 May '08

Hi to all, today i got a segfault with general protection error openser 1.2.2 notls under Linux 2.6.16.27-0.9-smp #1 SMP Tue Feb 13 09:35:18 UTC 2007 x86_64 x86_64 x86_64 GNU/Linux. It is due to: May 8 18:29:25 voip /data/openser/sbin/openser[19569]: val2str: Destination buffer too short May 8 18:29:25 voip /data/openser/sbin/openser[19569]: print_values: Error while converting value to string May 8 18:29:25 voip /data/openser/sbin/openser[19569]: ERROR:acc:acc_db_request: Error while inserting to database May 8 18:29:26 voip /data/openser/sbin/openser[19560]: child process 19573 exited by a signal 11 May 8 18:29:27 voip /data/openser/sbin/openser[19560]: core was generated I examined the core dump and it has been caused from the same thing that happened to me on October 2007 on another installation. I already posted the question to the list without getting any response. http://lists.openser.org/pipermail/users/2007-October/013796.html I post now the new gdb trace. I'm a very beginner using gdb but like very much to know why this is happening. Core was generated by `/data/openser/sbin/openser'. Program terminated with signal 11, Segmentation fault. #0 0x00002b83e4be4150 in escape_string_for_mysql () from /usr/lib64/ libmysqlclient.so.15 (gdb) bt #0 0x00002b83e4be4150 in escape_string_for_mysql () from /usr/lib64/ libmysqlclient.so.15 #1 0x00002b83e49465d1 in val2str (_c=0x2b83e4de05a0, _v=<value optimized out>, _s=0x2b83e4a48f8d "'33301','','469','','4','','','','102 INVITE','','12.13.14.15','')", _len=0x7fffc67d7804) at val.c:316 #2 0x00002b83e4942952 in db_mysql_print_values (_c=0x6b8768, _b=0x2b83e4a48e0e "'INVITE','5ec34ebc6ad9825','as54a9f138','647f8f41f2ed74cfb28b83633bb8efdb(a)voip.test.biz ','200','OK','2008-05-08 18:29:25','sip:035123456@voip.test.biz;tag=5ec34ebc6ad9825','sip:031454647@voip "..., _l=65170, _v=<value optimized out>, _n=28) at dbase.c:145 #3 0x00002b83e4943967 in db_mysql_insert (_h=0x6b9018, _k=<value optimized out>, _v=0x2b83e65aa5c0, _n=28) at dbase.c:584 #4 0x00002b83e64956d3 in acc_db_request (rq=0x2b83e6c9dc28) at acc.c: 378 #5 0x00002b83e6497244 in tmcb_func (t=<value optimized out>, type=<value optimized out>, ps=<value optimized out>) at acc_logic.c:376 #6 0x00002b83e52958f1 in run_trans_callbacks (type=128, trans=0x2b83e7673690, req=<value optimized out>, rpl=<value optimized out>, code=<value optimized out>) at t_hooks.c:205 #7 0x00002b83e52a3e76 in relay_reply (t=0x2b83e7673690, p_msg=<value optimized out>, branch=0, msg_status=200, cancel_bitmap=0x7fffc67d7ac8) at t_reply.c:1155 #8 0x00002b83e52a5b0d in reply_received (p_msg=0x6b9138) at t_reply.c: 1388 #9 0x000000000041929e in forward_reply (msg=0x6b9138) at forward.c:489 #10 0x000000000043cbd2 in receive_msg ( buf=0x60e700 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP 80.122.12.3;branch=z9hG4bKdd28.b33b4137.0;received=80.122.12.3\r\nVia: SIP/2.0/UDP 20.21.22.23:5060;rport=5060;branch=z9hG4bK31ffac575\r \nRecord-Route: <sip:031454647"..., len=850, rcv_info=0x7fffc67d7b90) at receive.c:195 #11 0x000000000046d45f in udp_rcv_loop () at udp_server.c:451 #12 0x0000000000428dd0 in main_loop () at main.c:834 #13 0x000000000042a899 in main (argc=<value optimized out>, argv=<value optimized out>) at main.c:1393 Thanks in advance to anyone can help me to debug this thing. Bye, Marcello

1 0

[OpenSER-Users] [OT?] voip-to-voip rating using cdrtool
by Sajith T S 08 May '08

08 May '08

Apologies if this is off-topic here, but this seemed like a good friendly place to ask. Apologies for being so naïve also... Question is, how can I get cdrtool to rate voip-to-voip calls when destination is non-numeric? As in, say, sip:holmes@example.com calls sip:watson@example.com. How do I get cdrtool to match *(a)example.com or *(a)example2.com? (FWIW, I'm just trying to see how this works... no plan of rolling out this Evil Scheme into a real world deployment.) While cdrtool has decent documentation, I wish this was a little more easy on me. Has anyone read/reviewed this book yet? http://www.packtpub.com/building-telephony-systems-with-openser/book I'm specifically interested in the "Billing with Freeradius and CDRTool" part. Thanks, Sajith. -- Don't you wish you had more energy... or less ambition? 9DB8FF06 : CB80 0BA6 7D13 B10A 6FBB D43E B4D2 28AD 9DB8 FF06

1 0

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