Re: [SR-Users] Kamailio [No Audio]
Hello, On 5/6/12 9:40 PM, Fred Flintsone wrote: I am attempting to route local registered users to local registered users without going to the media server. I have a media server [*] for PSTN. But it doesn't support all the CODECS i wan't to use. Signalling seems fine, but i get NO AUDIO on either call leg. I have the RTP captures from the local side making the call and the media packets seem to be going exactly where the SDP dictates. So i'm a little confused. Is this where an RTP Proxy would come in handy? I haven't been able to get the audio to work without going to the media server. If an RTP Proxy is the answer, how much overhead does the proxy add to the Kamailio server? Is it something that you don't run on the same machine and use a distributed environment of RTP proxies on other servers? Or is this something that should be fine and working without a media proxy? if users are behind nat, then you have to do rtp relay. Either the media servers support COMEDIA extension and you have to enforce that in the sdp or you have to use rtpproxy. Cheers, Daniel I really appreciate any help! Thank you, Fred ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Unexpectedly Terminating
Hello,
I see no reason why it crashes at line 104 when realm_prefix.len==0.
Might be a memory problem (hardware). Can you try with use_domain
parameter set to 0 for usrloc module?
Cheers,
Daniel
On 5/5/12 5:29 PM, Akan wrote:
I made the change and still have the problem. I have included the
output from stepping thru via gdb and the block of code where I made
the change and that was executed.
98 if (reg_use_domain) {
99 if (user_len)
100 aor_buf[_a-len++] = '@';
101 /* strip prefix (if defined) */
102 realm_prefix.s = cfg_get(registrar,
registrar_cfg, realm_pref).s;
103 realm_prefix.len = cfg_get(registrar,
registrar_cfg, realm_pref).len;
104 if (realm_prefix.len0
realm_prefix.lenpuri.host.len
105 (memcmp(realm_prefix.s, puri.host.s,
realm_prefix.len)==0) ) {
106 memcpy(aor_buf + _a-len, puri.host.s
+ realm_prefix.len,
107 puri.host.len -
realm_prefix.len);
108 _a-len += puri.host.len -
realm_prefix.len;
109 } else {
110 memcpy(aor_buf + _a-len, puri.host.s,
puri.host.len);
111 _a-len += puri.host.len;
112 }
113 }
Thanks
Nathaniel
On 5/4/2012 2:55 AM, Daniel-Constantin Mierla wrote:
Hello,
interesting, it seems to crash at the evaluation of
'realm_prefix.len' - because it is 0, the IF condition should stop
evaluation of the rest of the expression.
Can you try to change the first part of the condition at line 104 to:
if (realm_prefix.len0 ...
Cheers,
Daniel
On 5/4/12 4:42 AM, Akan wrote:
I was able to step thru via gdb to the point where Kamailio took a
segment fault. I have included a backtrace as well as the output
from me stepping thru common.c to the point where it failed. Hope
this will help.
Thanks
Nathaniel
On 4/29/2012 4:37 AM, Daniel-Constantin Mierla wrote:
Hello,
the issue was a sig bus this time, which more enforced in
sparc/solaris, not common in linux. I was looked at the structure
and seems aligned ok, mis-alignment being the most often used to
rise sigbus. What I could think as next reason was the empty string
default value which may make solaris think is not accessible
anymore at runtime via the cfg structure, so I changed the field
for prefix to str and initialized to null to force allocation in
any case.
Can you try with the attached patch and tell if works fine?
Cheers,
Daniel
On 4/28/12 6:49 PM, Akan wrote:
I tried adding the realm_prefix and still got the same problem. I
ran kamailio thru gdb to try and step thru and get more
information and have included the output in the attached text
file. Hope this helps.
Thanks
Nathaniel L Keeling
On 4/26/2012 2:42 PM, Akan wrote:
No, but I do have an alias defined.
alias=mydomain.com:5080
Thanks
Nathaniel
On 4/26/2012 3:41 AM, Daniel-Constantin Mierla wrote:
Hello,
do you set the realm_prefix parameter of registrar module?
http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2495082
If yes, can you paste it here?
Cheers,
Daniel
On 4/25/12 9:42 PM, Akan wrote:
I have 2 servers running Solaris and Kamailio 3.2.3 where on
one Kamailio is terminating when it tries to save the location
for a register request and the other is producing a core dump
when processing an Option request. I have one server handling
Register request while the other sip server forwards the
register requests and handles the other requests. I have
included the backtraces from the core dumps and the output from
the log for the registrar server as well as the command that is
causing kamailio to terminate:
if (!save(location))
sl_reply_error();
4(3364) ERROR: *** cfgtrace:
c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=714 a=17 n=if
4(3364) ERROR: *** cfgtrace:
c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=711 a=26 n=save
14(3374) : core [pass_fd.c:293]: ERROR: receive_fd: EOF on 15
14(3374) DEBUG: core [tcp_main.c:3555]: DBG:
handle_ser_child: dead child 4, pid 3364 (shutting down?)
14(3374) DEBUG: core [io_wait.h:617]: DBG: io_watch_del
(1003743d8, 15, 0, 0x0) fd_no=18 called
0(3360) ALERT: core [main.c:751]: child process 3364 exited
by a signal 10
0(3360) ALERT: core [main.c:754]: core was not generated
0(3360) INFO: core [main.c:766]: INFO: terminating due to
SIGCHLD
6(3366) INFO: core [main.c:817]: INFO: signal 15 received
1(3361) INFO: core [main.c:817]: INFO: signal 15 received
2(3362) INFO: core [main.c:817]: INFO: signal 15 received
3(3363) INFO: core [main.c:817]: INFO: signal 15 received
5(3365) INFO: core [main.c:817]: INFO: signal 15 received
7(3367) INFO: core [main.c:817]: INFO: signal 15 received
8(3368) INFO: core [main.c:817]: INFO: signal 15 received
Re: [SR-Users] [sr-dev] kamdbctl error
hi Daniel, your trick workednow I have mysql library files in the library folder... Thanks.I have added `mysql` and `db_mysql` in include_module variable in module.lst... Regards, Vineet Menon On 8 May 2012 12:28, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, one option - edit modules.lst in the root folder with kamailio sources and add db_mysql to the include_modules variable. Then recompile and reinstall. Cheers, Daniel On 5/8/12 6:45 AM, Vineet Menon wrote: Ya, you were right...mysql.so or db_mysql.so is not present in the lib folder... Is der any remedy for this? $ ls -R /usr/lib64/kamailio/ /usr/lib64/kamailio/: kamctl libkmi.so.1libsrdb1.so libsrdb2.so.1.0 modules_k libkcore.so libkmi.so.1.0 libsrdb1.so.1libtrie.so modules_s libkcore.so.1libser_cds.so libsrdb1.so.1.0 libtrie.so.1 libkcore.so.1.0 libser_cds.so.0libsrdb2.so libtrie.so.1.0 libkmi.solibser_cds.so.0.1 libsrdb2.so.1modules /usr/lib64/kamailio/kamctl: dbtextdbkamctl.dbtext kamctl.ser_mikamdbctl.base kamctl.base kamctl.fifokamctl.sqlbase kamdbctl.dbtext kamctl.ctlbase kamctl.ser kamctl.unixsock /usr/lib64/kamailio/kamctl/dbtextdb: dbtextdb.py /usr/lib64/kamailio/modules: async.so ctl.so matrix.so prefix_route.so sms.so auth.so db_flatstore.so mediaproxy.so ratelimit.so textopsx.so avpops.sodebugger.so mi_rpc.so rtpproxy.so tm.so blst.so dialplan.so mqueue.so sanity.sotopoh.so cfg_db.soenum.so mtree.so sdpops.soxhttp.so cfg_rpc.so ipops.so pdb.so sipcapture.so counters.so lcr.so pipelimit.so sl.so /usr/lib64/kamailio/modules_k: acc.so exec.so permissions.so sqlops.so alias_db.so group.so pike.so sst.so auth_db.so htable.so p_usrloc.so statistics.so benchmark.so imc.sopv.so textops.so call_control.so kex.soqos.so tmx.so cfgutils.so maxfwd.so regex.souac_redirect.so db_text.so mi_datagram.soregistrar.souac.so dialog.somi_fifo.sorr.so uri_db.so dispatcher.somsilo.so rtimer.so userblacklist.so diversion.so nathelper.so seas.so usrloc.so dmq.so nat_traversal.so siptrace.so xlog.so domain.sopath.so siputils.so drouting.so pdt.sospeeddial.so /usr/lib64/kamailio/modules_s: auth_db.so domain.so nathelper.soregistrar.so uac.so dialog.so exec.sopdt.so rr.so uri_db.so dispatcher.so maxfwd.so permissions.so speeddial.so usrloc.so diversion.so msilo.so pike.so textops.soxlog.so Regards, Vineet Menon On 7 May 2012 20:51, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, On 5/7/12 5:07 PM, Vineet Menon wrote: yes, flavour is set to kamailio and isn't db_mysql module complied the way i just mentioned?? it should be, but typos/errors in command or bugs may prevent that. So it is better to check in the folder with the modules. Cheers, Daniel Regards, Vineet Menon On 7 May 2012 18:09, Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, is the flavour set to kamailio and db_mysql module installed? Some guidelines are presented at: http://www.kamailio.org/wiki/install/3.2.x/git Cheers, Daniel On 5/7/12 11:16 AM, Vineet Menon wrote: Hi, I am compiling sources of kamailio 3.2.3 on ubuntu 12.04. I have used make command `make group_include=standard mysql all` to make the sources. and then `make install`. Now when i do kamdbctl it throws an error, $ kamdbctl create ERROR: could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' Any guess what is the problem??? Regards, Vineet Menon ___ sr-dev mailing listsr-...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamdbctl
On 08.05.2012 12:19, Henning Westerholt wrote: Am Dienstag, 8. Mai 2012, 12:06:46 schrieb Klaus Darilion: I wonder what is the status of kamdbctl tool for 'migrate' command - should it work? The description says: kamdbctl migrateold_db new_db .(migrates DB from 1.2 to 1.3, not implemented yet!) This seems a bit outdated. Hi Klaus! Its indeed outdated, and probably should be removed. What should be removed - the comment or the command? :-) Maybe someone else can do this, I am not familiar with the internals of kamdbctl. thanks Klaus ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] detect retransmission messages...
Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] detect retransmission messages...
t_check_trans() generally does the trick. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com Vineet Menon mvineetme...@gmail.com wrote: Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] detect retransmission messages...
See t_newtran() and t_check_trans() to create/check transaction state for certain requests: http://www.kamailio.org/docs/modules/3.2.x/modules/tm.html#t_newtran http://www.kamailio.org/docs/modules/3.2.x/modules/tm.html#t_check_trans There is no method to check if responses are retransmissions - this is handled internally in tm module. regards Klaus On 09.05.2012 11:54, Vineet Menon wrote: Hi, How do we detect a message as retransmitted message, in a transaction oriented proxy? Regards, Vineet Menon ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] [SIREMIS] Multiple kamailio management commands
Hello list, I'm trying to have my Siremis interface send MI commands to multiple kamailio servers i.e reload dispatcher of all the kamailio servers when I reload from Siremis interface. The issue Im facing is that the commands dont get executed on any other server except localhost/ as mentioned in URL: http://kb.asipto.com/siremis:install32x:mi-commands I've edited the Remote name=remote address=127.0.0.1 port=8033/ to : Remote name=remote address=192.168.2.156 port=8033/ Please guide on how to have multiple kamailio receive command from single Siremis interface. Regards, Sammy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Accounting '180 Ringing' without SDP
Hello, We are running Kamailio 3.1.5 and using the acc module to generate CDRs and provide reporting information. We have a requirement to report ring time and we do this by accounting on early media for 183 provisional responses. Unfortunately we have a couple of customer PBX's that cannot provide a SDP with the ringing response. i.e they just respond with a '180 ringing'. My understanding of the module is that the early_media parameter is looking for a SDP in either a 180 or 183 response before accounting. Although it may be possible to account the 180 without SDP using the acc module I cannot find anything obvious in the documentation. If it is possible could someone point me in the right direction or suggest a creative solution, otherwise can I ask that this functionality be introduced. Many Thanks. Steve Dodge. Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Alias configuration with multidomain
Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesn't work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I don't have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: arjan.kui...@intercityzakelijk.nlmailto:arjan.kui...@intercityzakelijk.nl I: www.intercityzakelijk.nlhttp://www.intercityzakelijk.nl/ inline: image001.png kamailio.cfg Description: kamailio.cfg sip2.log Description: sip2.log ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Rtpproxy issue with connection information
Hi, i'm currently working with kamailio 3.2 and rtpproxy 1.2.1. Both are set up on the same computer. When rtpproxy adds an SDP to an Invite, it adds two IPv4 addresses in owner/creator session and connection information field with an error, i.e: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 * 172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 the correct ones should be : *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 172.27.170.98 *Connection Information (c)*: IN IP4 172.27.170.98 Thanks for help!! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Rtpproxy issue with connection information
Hi, On 05/09/2012 02:40 PM, Openser Kamailio wrote: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 *172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 Could it be possible that you're calling rtpproxy_offer() twice? Andreas signature.asc Description: OpenPGP digital signature ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Accounting '180 Ringing' without SDP
Hello, I don't remember by hart all the parameter options for acc module, but for accounting an event is always a backup route - use acc_db_request(...) when that event happens -- in this case should be an onreply_route with a condition on status code 180. Cheers, Daniel On 5/9/12 1:32 PM, Stephen Dodge (Bistech) wrote: Hello, We are running Kamailio 3.1.5 and using the acc module to generate CDRs and provide reporting information. We have a requirement to report ring time and we do this by accounting on early media for 183 provisional responses. Unfortunately we have a couple of customer PBX's that cannot provide a SDP with the ringing response. i.e they just respond with a '180 ringing'. My understanding of the module is that the early_media parameter is looking for a SDP in either a 180 or 183 response before accounting. Although it may be possible to account the 180 without SDP using the acc module I cannot find anything obvious in the documentation. If it is possible could someone point me in the right direction or suggest a creative solution, otherwise can I ask that this functionality be introduced. Many Thanks. Steve Dodge. Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Alias configuration with multidomain
Hello, can you send here on the mailing lust the error log messages you get? They should give some hints about what goes wrong. Cheers, Daniel On 5/9/12 2:01 PM, Arjan Kuiken wrote: Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesnt work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I dont have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: arjan.kui...@intercityzakelijk.nl I: www.intercityzakelijk.nl ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Alias configuration with multidomain
Hi Daniel, Here are the error messages: INVITE sip:0235630155@a1 SIP/2.0 Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK63637d58;rport From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1 Contact: sip:0235630111@172.20.30.45 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 102 INVITE User-Agent: Xelion Phone System Max-Forwards: 70 Date: Wed, 09 May 2012 11:42:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-Xelion-Id: c02895a5c5a3f02e X-Xelion-CallType: internal Content-Type: application/sdp Content-Length: 285 v=0 o=root 1683 1683 IN IP4 172.20.30.45 s=session c=IN IP4 172.20.30.45 t=0 0 m=audio 15044 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK63637d58;rport=5060 From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1;tag=4ba51407f309cf9a12154e6d0edeacd7.4a70 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 102 INVITE Proxy-Authenticate: Digest realm=a1, nonce=T6pZXU+qWDH7zWMqqCxZvI6lEuyj1cK1, qop=auth Server: kamailio (3.2.3 (i386/linux)) Content-Length: 0 ACK sip:0235630155@a1 SIP/2.0 Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK63637d58;rport From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1;tag=4ba51407f309cf9a12154e6d0edeacd7.4a70 Contact: sip:0235630111@172.20.30.45 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 102 ACK User-Agent: Xelion Phone System Max-Forwards: 70 Content-Length: 0 INVITE sip:0235630155@a1 SIP/2.0 Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK29c960c1;rport From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1 Contact: sip:0235630111@172.20.30.45 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 103 INVITE User-Agent: Xelion Phone System Max-Forwards: 70 Proxy-Authorization: Digest username=0235630111@a1, realm=a1, algorithm=MD5, uri=sip:0235630155@a1, nonce=T6pZXU+qWDH7zWMqqCxZvI6lEuyj1cK1, response=750f82134d8140ad475f201e3fd56949, qop=auth, cnonce=5a11a0d8, nc=0001 Date: Wed, 09 May 2012 11:42:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces X-Xelion-Id: c02895a5c5a3f02e X-Xelion-CallType: internal Content-Type: application/sdp Content-Length: 285 v=0 o=root 1683 1684 IN IP4 172.20.30.45 s=session c=IN IP4 172.20.30.45 t=0 0 m=audio 15044 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv SIP/2.0 404 Not Found Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK29c960c1;rport=5060 From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1;tag=d7c7cbe55800d02daedc6e4bd4c35c65-cb60 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 103 INVITE Server: kamailio (3.2.3 (i386/linux)) Content-Length: 0 ACK sip:0235630155@a1 SIP/2.0 Via: SIP/2.0/UDP 172.20.30.45:5060;branch=z9hG4bK29c960c1;rport From: 0235630111 sip:0235630111@a1;tag=as0c12287c To: sip:0235630155@a1;tag=d7c7cbe55800d02daedc6e4bd4c35c65-cb60 Contact: sip:0235630111@172.20.30.45 Call-ID: 0c29c63e38dd5d2f1f707b5e11751a5e@99a1 CSeq: 103 ACK User-Agent: Xelion Phone System Max-Forwards: 70 Content-Length: 0 Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: arjan.kui...@intercityzakelijk.nlmailto:arjan.kui...@intercityzakelijk.nl I: www.intercityzakelijk.nlhttp://www.intercityzakelijk.nl/ Van: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] Namens Arjan Kuiken Verzonden: woensdag 9 mei 2012 14:02 Aan: 'sr-users@lists.sip-router.org' Onderwerp: [SR-Users] Alias configuration with multidomain Hello SR-users, I want to configure a kamailio 3.2.3. We want to use it to connect multiple SIP servers as users and let the Kamailio work as a Trunk/ Proxy connected to a PSTN gateway. We have multiple domains and aliases per SIP servers. When I register a SIP server to Kamailio everything goes well, but when I try to call an dbalias it doesn't work. It gives 2 errors (trace included). I think something is done wrong in my kamailio.cfg but I don't have a clue anymore. I included the cfg file as attachement. Can you please have a look? Thanks in advance! Kind regards, Arjan Kuiken Technical Specialist [intercity-zakelijk-logo-handetekening (2)] Kruisweg 659, 2132 NC, Hoofddorp T: 020 - 655 3000 F: 020 - 6531534 E: arjan.kui...@intercityzakelijk.nlmailto:arjan.kui...@intercityzakelijk.nl I: www.intercityzakelijk.nlhttp://www.intercityzakelijk.nl/ inline: image001.png___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
[SR-Users] Problem with parallel forking of aliases
Greetings, I'm having trouble getting parallel forking to work with aliasdb. I'm running kamailio 3.2 with the standard kamailio.cfg script. I have found that if an alias points to a set of addresses that all reference local devices that are registered with the server, kamailio sends an invitation to the first device in the set (the one that the aliasdb lookup function sets to the ruri, but does *not* invite any of the other devices in the set, which aliasdb adds as branches. However, if one of the other aliases points to a non-local address, such as a PSTN address, kamailio does correctly invite the non-local address in parallel with the first alias address, which is a local device. It seems as if kamailio is ignoring invitations that it is in effect sending to itself via the additional parallel branches. I would expect that to call a branch in parallel, kamailio would need to do a lookup on the branch address and rewrite it to send the invitation to the registered device. But none of that seems to be happening. There must be some additional configuration change required to make this work. Any suggestions? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio ignores some ACK
Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A - INVITE - kamailio - INVITE - B [omitting 100 and 180] A - 200 OK - kamailio - 200 OK - B A - ACK - kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. tcpdump -A -s0 -i any -n port 5060 receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060;rport=60442;x-route-tag=tgrp:A;branch=z9hG4bK1634E6A88 Record-Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Contact: sip:80.80.80.80:65002;transport=udp To: test_accountsip:b...@server.com;tag=cb7dd641 From: sip:alice@111.111.11.50;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag=tgrp:A;branch=z9hG4bK1634E7DE8 From: sip:alice@111.111.11.50;tag=599248D4-260 To: test_accountsip:b...@server.com;tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Rtpproxy issue with connection information
I call rtpproxy_offer() once, but i use also rtpproxy_manage(). When i disable rttproxy_mange(), it works well. Thanks! On Wed, May 9, 2012 at 2:57 PM, Andreas Granig agra...@sipwise.com wrote: Hi, On 05/09/2012 02:40 PM, Openser Kamailio wrote: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 *172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 Could it be possible that you're calling rtpproxy_offer() twice? Andreas ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Rtpproxy issue with connection information
I had the same problem when calling mediaproxy twice by mistake. rtpproxy_manage( ) calls implicitely rtpproxy_offer( ). This is the problem. Either you use only rtpproxy_manage once on the INVITE and let it start and terminate the session, or you use rtpproxy_offer, rtpproxy_answer and rtpproxy_destroy and if finer control is needed. Best is to call rtpproxy_manage and let it do its magic unless you have specific reason to manually control the session, such as terminating the rtpproxy session when transferring the call to a pstn gateway that handles rtp for example. You can find more details in the rttpproxy module documentation. Reda On Wed, May 9, 2012 at 4:16 PM, Openser Kamailio kamailioopen...@gmail.comwrote: I call rtpproxy_offer() once, but i use also rtpproxy_manage(). When i disable rttproxy_mange(), it works well. Thanks! On Wed, May 9, 2012 at 2:57 PM, Andreas Granig agra...@sipwise.comwrote: Hi, On 05/09/2012 02:40 PM, Openser Kamailio wrote: *Owner/Connection Information (o)*: doubango 1983 678901 IN IP4 *172.27.170.984* 172.27.170.98 *Connection Information (c)*: IN IP4 *172.27.170.984* 172.27.170.98 Could it be possible that you're calling rtpproxy_offer() twice? Andreas ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio ignores some ACK
You can use something like wireshark on Kamailio server to see if ACK packets go in right direction. I had problem with ACK and BYE, and I saw that in some cases ACK and BYE packets looped back in kamailio. May be I used wrong client. On Wed, May 9, 2012 at 5:15 PM, Efelin Novak efelin.no...@gmail.com wrote: Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A - INVITE - kamailio - INVITE - B [omitting 100 and 180] A - 200 OK - kamailio - 200 OK - B A - ACK - kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. tcpdump -A -s0 -i any -n port 5060 receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060 ;rport=60442;x-route-tag=tgrp:A;branch=z9hG4bK1634E6A88 Record-Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Contact: sip:80.80.80.80:65002;transport=udp To: test_accountsip:b...@server.com;tag=cb7dd641 From: sip:alice@111.111.11.50;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag=tgrp:A;branch=z9hG4bK1634E7DE8 From: sip:alice@111.111.11.50;tag=599248D4-260 To: test_accountsip:b...@server.com;tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio ignores some ACK
Seems like a loose routing issue. Are you loose routing in your config file? On Wed, May 9, 2012 at 4:34 PM, Stoyan Mihaylov stoyan.v.mihay...@gmail.com wrote: You can use something like wireshark on Kamailio server to see if ACK packets go in right direction. I had problem with ACK and BYE, and I saw that in some cases ACK and BYE packets looped back in kamailio. May be I used wrong client. On Wed, May 9, 2012 at 5:15 PM, Efelin Novak efelin.no...@gmail.comwrote: Hi folks, I have a strange problem when Kamailio ignores ACKs in a specific scenario. The call flow is as follows: A - INVITE - kamailio - INVITE - B [omitting 100 and 180] A - 200 OK - kamailio - 200 OK - B A - ACK - kamailio There are INVITE Xlogs, Reply ROUTE xlogs and media-proxy logs in the syslog. However there is no information about these ACKs. No XLOGs are printed even if there is one on the top of the main route. tcpdump -A -s0 -i any -n port 5060 receives this message correctly: 14:47:01.246153 IP 111.111.11.11.5060 80.80.80.80.60442: SIP, length: 915 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.111.11.11:5060 ;rport=60442;x-route-tag=tgrp:A;branch=z9hG4bK1634E6A88 Record-Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Contact: sip:80.80.80.80:65002;transport=udp To: test_accountsip:b...@server.com;tag=cb7dd641 From: sip:alice@111.111.11.50;tag=599248D4-260 Call-ID: 9AFCFC51.11.50 CSeq: 101 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE Content-Type: application/sdp Content-Length:263 v=0 o=- 492575093 492575093 IN IP4 111.111.11.60 s=test_device i=(o=IN IP4 192.168.1.10) c=IN IP4 111.111.11.71 t=0 0 m=audio 16416 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 14:47:01.254511 IP 111.111.11.50.60442 111.111.11.11.5060: SIP, length: 521 ACK sip:80.80.80.80:65002;transport=udp SIP/2.0 Via: SIP/2.0/UDP 111.111.11.50:5060;x-route-tag=tgrp:A;branch=z9hG4bK1634E7DE8 From: sip:alice@111.111.11.50;tag=599248D4-260 To: test_accountsip:b...@server.com;tag=cb7dd641 Call-ID: 9AFCFC51.11.50 Route: sip:111.111.11.11;lr;ftag=599248D4-260;vsf=W0FVT0ZWHF1aNy4xGzA-;nat=yes;did=3bb.327c47e6 Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 My Kamailio version is kamailio 3.1.0 (i386/linux) 1e204f. Does anybody knows where can be a problem? How can I check whether Kamailio receives something? ... Jan ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users This email is subject to the disclaimer of Smile Communications (PTY) Ltd. at http://www.smilecoms.com/disclaimer ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] [SIREMIS] Multiple kamailio management commands
Hi, The Kamailio server you wish to control remotely should have the mi_datagram module listening on the correct interface and not on the loopback one (127.0.0.1). loadmodule mi_datagram.so *modparam(mi_datagram, socket_name, udp:192.168.2.156:8033)* Reda On Wed, May 9, 2012 at 1:02 PM, SamyGo govoi...@gmail.com wrote: Hello list, I'm trying to have my Siremis interface send MI commands to multiple kamailio servers i.e reload dispatcher of all the kamailio servers when I reload from Siremis interface. The issue Im facing is that the commands dont get executed on any other server except localhost/ as mentioned in URL: http://kb.asipto.com/siremis:install32x:mi-commands I've edited the Remote name=remote address=127.0.0.1 port=8033/ to : Remote name=remote address=192.168.2.156 port=8033/ Please guide on how to have multiple kamailio receive command from single Siremis interface. Regards, Sammy. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Config include
Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include path/to/some/config.conf). Is it possible to implement a such feature ? Thanks! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio SIP server and Cisco 7971G-GE
Hello everyone, I am trying to get a new SIP firmware version on our Cisco 7971G-GE. Now the problem is that I don't really know which configuration files have to be included to get everything working. For now on I have the new SIP firmware and SEPmaccnf.xml and the XMLDefault.cnf.xml files. Does anybody have a template for the .xml files to get a new firmware version working on our Cisco and also to get it work with Kamailio? With kind regards, Koen. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] NAT fixups not applied for voicemail
Greetings, Here's another problem I'm having with kamailio 3.2 and the standard kamailio.cfg script. If the calling device is NATed, everything works fine if the original call gets connected. That is, the INVITE sent to the called device has the correct NAT fixups applied. But if the called device fails to answer and the script runs route[TOVOICEMAIL], the call connects, but the INVITE sent to the voicemail server doesn't have the NAT fixup applied. The result is that the audio is connected in only one direction. It would appear that some rtpproxy function needs to get called to apply the fixups prior to sending the INVITE to the voicemail server. I've tried adding calls to route(NATMANAGE) at various places, but to no avail. Any ideas? -- Mark Sidell Partner Forte, Inc. 919-942-7068 fax 919-969-2844 www.forteinc.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Config include
Hello, It is already there, see http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x On 05/09/2012 06:04 PM, Konstantin M. wrote: Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include path/to/some/config.conf). Is it possible to implement a such feature ? Thanks! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Config include
Hi Andrew, I have missed that. Thank you very much! 2012/5/9 Andrew Pogrebennyk apogreben...@sipwise.com Hello, It is already there, see http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x On 05/09/2012 06:04 PM, Konstantin M. wrote: Hi, I would like (and a many people here I believe) to have a functional of including a multiple config files like (foe example asterisk's #include path/to/some/config.conf). Is it possible to implement a such feature ? Thanks! ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Config include
After including a part of main config to included file -- I got a several
errors like:
0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command
is_method
0(1582) : core [cfg.y:3532]: parse error in config file
/opt/kamailio/etc/kamailio/debug.cfg, line 4, column 55: unknown command,
missing loadmodule?
0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command
xlog
0(1582) : core [cfg.y:3532]: parse error in config file
/opt/kamailio/etc/kamailio/debug.cfg, line 9, column 101: unknown command,
missing loadmodule?
...
A part of included file /opt/kamailio/etc/kamailio/debug.cfg:
--
route[DEBUG]
{
if (is_method(PUBLISH|SUBSCRIBE|REGISTER|OPTIONS))
{
return;
}
xlog(L_INFO, *** UNHANDLED *** SIP Request: method [$rm], status
[$rs] from [$fu] to [$tu]\n);
}
route[DEBUG_FROM]
{
xlog(L_NOTICE, [$mi] Received SIP Message (method [$rm]) ($ml[$cl]
bytes) from $Ri:$Rp ($si:$sp):\n$mb\n);
}
...
Do I have to duplicate the section of modules to that included file or I'm
wrong ?
2012/5/9 Konstantin M. evilz...@gmail.com
Hi Andrew,
I have missed that. Thank you very much!
2012/5/9 Andrew Pogrebennyk apogreben...@sipwise.com
Hello,
It is already there, see
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x
On 05/09/2012 06:04 PM, Konstantin M. wrote:
Hi,
I would like (and a many people here I believe) to have a functional of
including a multiple config files like (foe example asterisk's
#include path/to/some/config.conf).
Is it possible to implement a such feature ?
Thanks!
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Question about using LCR wit multiple gateways
Kamilio 3.2.0 I'd like to use one gateway as primary gateway and the another gateway as backup for failover. I could not make it to work. Here is my table entries: Lcr_gw: +++-+-+--+--+++---+---++--+---+-+ | id | lcr_id | gw_name | ip_addr | hostname | port | params | uri_scheme | transport | strip | prefix | tag | flags | defunct | +++-+-+--+--+++---+---++--+---+-+ | 1 | 2 | gateway1 | 10.10.1.1 | NULL | 5060 | NULL | 1 | 0 | NULL | NULL | NULL | 0 |NULL | | 2 | 2 | gateway2 | 10.10.1.2 | NULL | 5060 | NULL | 1 | 0 | NULL | NULL | NULL | 0 |NULL | Lcr_rule: +---+++--+-+- | id| lcr_id | prefix | from_uri | stopper | enabled +---+++--+-+ | 1 | 2 | 1 | NULL | 0 | 1 | 2 | 2 | 011| NULL | 0 | 1 +---+++--+-+ Lcr_rule_target: +---++-+---+--++ | id| lcr_id | rule_id | gw_id | priority | weight | +---++-+---+--++ | 1 | 2 | 1| 1 |9 | 1 | | 2 | 2 | 1 | 2 |8 | 1 | | 3 | 2 | 2 | 1 |9 | 1 | | 4 | 2 | 2 | 2 |8 | 1 | When making call , it only uses the first gateway. If first gateway failed, it could not find second gateway. What is the correct table entry for this to work? Thanks. Gary ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Question about using LCR wit multiple gateways
Gary Chen writes: When making call , it only uses the first gateway. If first gateway failed, it could not find second gateway. What is the correct table entry for this to work? there is not necessarily anything wrong with your tables. put some xlog statements to your script to find out what is going on when you make load_gws and next_gw calls. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Config include
Konstantin, You should put the include_file directive after loadmodule and modparam directives. So it can be either before main route block or at the bottom of your main kamailio.cfg. On 05/09/2012 06:48 PM, Konstantin M. wrote: After including a part of main config to included file -- I got a several errors like: 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command is_method 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 4, column 55: unknown command, missing loadmodule? 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command xlog 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 9, column 101: unknown command, missing loadmodule? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Config include
Thank you, I found a logical error in order. Also would be good if lex parser can understand a wildmasks, like: include_file modules/*.cfg... 2012/5/9 Andrew Pogrebennyk apogreben...@sipwise.com Konstantin, You should put the include_file directive after loadmodule and modparam directives. So it can be either before main route block or at the bottom of your main kamailio.cfg. On 05/09/2012 06:48 PM, Konstantin M. wrote: After including a part of main config to included file -- I got a several errors like: 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command is_method 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 4, column 55: unknown command, missing loadmodule? 0(1582) ERROR: core [cfg.y:3393]: cfg. parser: failed to find command xlog 0(1582) : core [cfg.y:3532]: parse error in config file /opt/kamailio/etc/kamailio/debug.cfg, line 9, column 101: unknown command, missing loadmodule? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio Unexpectedly Terminating
I changed the use_domain parameter for usrloc to 0 and retried. Kamailio
did not crash and I was able to get registered. I have another server
with the same configurations as the one having the problem. I will apply
the patch there and see if the problem still occurs. Also, setting the
use_domain to 0 for usrloc, does this limit me to not being able to use
other domains within the SIP message?
Thanks
Nathaniel
On 5/9/2012 3:05 AM, Daniel-Constantin Mierla wrote:
Hello,
I see no reason why it crashes at line 104 when realm_prefix.len==0.
Might be a memory problem (hardware). Can you try with use_domain
parameter set to 0 for usrloc module?
Cheers,
Daniel
On 5/5/12 5:29 PM, Akan wrote:
I made the change and still have the problem. I have included the
output from stepping thru via gdb and the block of code where I made
the change and that was executed.
98 if (reg_use_domain) {
99 if (user_len)
100 aor_buf[_a-len++] = '@';
101 /* strip prefix (if defined) */
102 realm_prefix.s = cfg_get(registrar,
registrar_cfg, realm_pref).s;
103 realm_prefix.len = cfg_get(registrar,
registrar_cfg, realm_pref).len;
104 if (realm_prefix.len0
realm_prefix.lenpuri.host.len
105 (memcmp(realm_prefix.s, puri.host.s,
realm_prefix.len)==0) ) {
106 memcpy(aor_buf + _a-len, puri.host.s
+ realm_prefix.len,
107 puri.host.len -
realm_prefix.len);
108 _a-len += puri.host.len -
realm_prefix.len;
109 } else {
110 memcpy(aor_buf + _a-len,
puri.host.s, puri.host.len);
111 _a-len += puri.host.len;
112 }
113 }
Thanks
Nathaniel
On 5/4/2012 2:55 AM, Daniel-Constantin Mierla wrote:
Hello,
interesting, it seems to crash at the evaluation of
'realm_prefix.len' - because it is 0, the IF condition should stop
evaluation of the rest of the expression.
Can you try to change the first part of the condition at line 104 to:
if (realm_prefix.len0 ...
Cheers,
Daniel
On 5/4/12 4:42 AM, Akan wrote:
I was able to step thru via gdb to the point where Kamailio took a
segment fault. I have included a backtrace as well as the output
from me stepping thru common.c to the point where it failed. Hope
this will help.
Thanks
Nathaniel
On 4/29/2012 4:37 AM, Daniel-Constantin Mierla wrote:
Hello,
the issue was a sig bus this time, which more enforced in
sparc/solaris, not common in linux. I was looked at the structure
and seems aligned ok, mis-alignment being the most often used to
rise sigbus. What I could think as next reason was the empty
string default value which may make solaris think is not
accessible anymore at runtime via the cfg structure, so I changed
the field for prefix to str and initialized to null to force
allocation in any case.
Can you try with the attached patch and tell if works fine?
Cheers,
Daniel
On 4/28/12 6:49 PM, Akan wrote:
I tried adding the realm_prefix and still got the same problem. I
ran kamailio thru gdb to try and step thru and get more
information and have included the output in the attached text
file. Hope this helps.
Thanks
Nathaniel L Keeling
On 4/26/2012 2:42 PM, Akan wrote:
No, but I do have an alias defined.
alias=mydomain.com:5080
Thanks
Nathaniel
On 4/26/2012 3:41 AM, Daniel-Constantin Mierla wrote:
Hello,
do you set the realm_prefix parameter of registrar module?
http://kamailio.org/docs/modules/stable/modules_k/registrar.html#id2495082
If yes, can you paste it here?
Cheers,
Daniel
On 4/25/12 9:42 PM, Akan wrote:
I have 2 servers running Solaris and Kamailio 3.2.3 where on
one Kamailio is terminating when it tries to save the location
for a register request and the other is producing a core dump
when processing an Option request. I have one server handling
Register request while the other sip server forwards the
register requests and handles the other requests. I have
included the backtraces from the core dumps and the output
from the log for the registrar server as well as the command
that is causing kamailio to terminate:
if (!save(location))
sl_reply_error();
4(3364) ERROR: *** cfgtrace:
c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=714 a=17 n=if
4(3364) ERROR: *** cfgtrace:
c=[/opt/kamailio-3.2/etc/kamailio/kamailio.cfg] l=711 a=26 n=save
14(3374) : core [pass_fd.c:293]: ERROR: receive_fd: EOF on 15
14(3374) DEBUG: core [tcp_main.c:3555]: DBG:
handle_ser_child: dead child 4, pid 3364 (shutting down?)
14(3374) DEBUG: core [io_wait.h:617]: DBG: io_watch_del
(1003743d8, 15, 0, 0x0) fd_no=18 called
0(3360) ALERT: core [main.c:751]: child process 3364 exited
by a signal 10
0(3360) ALERT: core [main.c:754]: core was not generated
Re: [SR-Users] Problem with radius_www_authorize
Thanks Juha.! Regards, -Mensaje original- De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Juha Heinanen Enviado el: miércoles, 09 de mayo de 2012 0:39 Para: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Asunto: [SR-Users] Problem with radius_www_authorize Ricardo Martinez writes: Despite of this, for BadPassword or UserNotExist from the radius answer the radius_www_authorize command return “-2” as a retcode. So, is the documentation missing something? looks like bug in readme file. in those cases AUTH_INVALID_PASSWORD is returned, which is -2. i'll fix it. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Question about dialog module (state 1)
Hello. I’m using the dialog module to keep control of simultaneous calls. In some cases, and I’m still trying to find why this happens, the dialog stays in “STATE:: 1”, which according to the docs is a dialog which no provisional response has been sent yet. Is there a way to eliminate this kind of dialogs??? When I list the dialogs I can see : dialog:: hash=2523:2040584107 state:: 1 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: 25b8e5354ca0850a4d0bd9c43eeb6...@pxext.redvoiss.net from_uri:: sip:557100052...@pxext.redvoiss.net from_tag:: as10dc7c85 caller_contact:: sip:557100052213@200.68.19.44:5060 caller_cseq:: 103 caller_route_set:: caller_bind_addr:: udp:64.76.154.110:5060 callee_bind_addr:: to_uri:: sip:0981985...@pxext.redvoiss.net to_tag:: callee_contact:: callee_cseq:: callee_route_set:: when I try to eliminate this dialog : kamctl fifo dlg_terminate_dlg 25b8e5354ca0850a4d0bd9c43eeb6...@pxext.redvoiss.net 500 command 'dlg_terminate_dlg' failed Thanks in advance, Regards, Ricardo Martinez.- ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
