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37471 discussions

[Kamailio-Users] new bunch of PVs
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

Hello, I added a bunch of new variables to deal easier with next hop address and be able to retrieve micro-seconds in the config file. Check wiki at: http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#timeval http://sip-router.org/wiki/cookbooks/pseudo-variables/devel#next_hop_address Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] Problems with snmpstats in Kamailio-3.0.0
by Santiago Gimeno 28 Jan '10

28 Jan '10

Hi, I'm trying to migrate from K-1.5 to K-3.0.0 and I'm having problems loading the snmpstats module in kamailio-3.0.0. The error is: ERROR: <core> [sr_module.c:390]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/snmpstats.so>: /usr/local/lib/kamailio/modules_k/snmpstats.so: undefined symbol: module_loaded By checking the code it looks like module_loaded function is not implemented in sr_module.c anymore. Could this be the problem? Thanks. Best regards, Santi

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

On 28 January 2010 13:41, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > > On 1/28/10 2:36 PM, Jon Farmer wrote: >>> >>> is the call going via route[2]? >>> >>> Try to print the whole sip message $mb before and after executing your >>> php >>> script. >>> >> >> Yes route[2] >> >> Here is a link to the $mb before and after http://pastebin.com/d4b7ca03 >> > > k, it is good, however $mb is always received message so it is the same > before and after -- just wanted to check if got corrupted. > > I want to see now the r-uri before and after: $ru > > Daniel >> Aha found it... my PHP is returning error when URI has ";user=phone". Daniel you are a star thank you for your help and patience on this one. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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[Kamailio-Users] Request for a special parallel serial forking scenario
by Klaus Feichtinger 28 Jan '10

28 Jan '10

Hi Kamailio users, I have a special question regarding a mixture of serial and parallel forking, operated by Kamailio-3.0.0. First of all I will give you a short background information, what the target will be: I have a SIP-server on place A, 3 gateways on place B and 3 gateways on place C). All (6) gateways are registered with the same username on SIP server (A), but with different Q-values (e.g. GW-B1=1.0, GW-C1=1.0, GW-B2=0.8, GW-C2=0.8, GW-B3=0.6, GW-C3=0.6). The target is, that a call for a gateway MUST be signalled on gateway(s) of place B and C in parallel (forking) until the call is finally established over one of the two involved gateways on place B or C. When (one of) the prime gateway(s) (with the highest Q-value) fail (e.g. they do not send a provisional response within a timeout or send a negative response), the next gateway(s) should be addressed (with the next lower Q-value), a.s.o. The parallelism for signalisation on place B and C is necessary, because of redundancy / safety reasons. Configuration for serial forking is done as described in the README of the TM module. It is a mixture of the main route and a failure_route in combination with the functions t_load_contacts() and t_next_contacts(). The loaded contacts for serial forking differ (from standard scenarios) in that way, that each sequential target consists of a pair of "users" instead of a single user. This configuration works well in case that both gateways on place B and C fail at the same time (BOTH do not send a response or BOTH send a negative response or one does not send a response and the other one sends a negative response). However, in case that only ONE of the two (always in parallel) addressed gateways fails, it does not work as expected. It does nothing and/or waits for an eventually negative response or the call setup timeout. Only then the next gateway-pair will be addressed. => is the behaviour of TM correct/"as designed" to make an AND-relation with both branches and access the failure_route only then, when both branches fail? Does anybody have an idea how this problem could be solved or any alternative solution for this special requirement of parallel signalisation in any case? Thanks in advance! Regards, Klaus -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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[Kamailio-Users] TM module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! ERROR: tm [tm_load.c:47]: tm:load_tm: Module not initialized yet, make sure that all modules that need tm module are loaded after tm in the configuration file Kamailio 3.0 was installed from GIT. Kamailio starts success. But the error appears. I think because of this I have a problem with making a call through kamailio. Calls were failed because kamailio send CANCEL or 487 code immediately after send INVITE to some outgoing gateways (after t_relay() has been made). Is this really need load all modules for tm working properly? I read documentation and fount that almost all modules that need tm loaded do not need in my config. Thank you for any help.

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Re: [Kamailio-Users] Bad URI
by Jon Farmer 28 Jan '10

28 Jan '10

2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: > El Miércoles, 27 de Enero de 2010, escribió: >> 2010/1/27 Iñaki Baz Castillo <ibc(a)aliax.net>: >> > El Miércoles, 27 de Enero de 2010, Jon Farmer escribió: >> >> Hi >> >> >> >> Does anyone know why I am getting >> >> >> >> ERROR: parse_uri: bad host in uri (error at char >> >> >> >> from this ( i hidden the domain and IP) >> > >> > How does the domain look? To show it replace the letters but keep the >> > symbols (as underscore and so). >> >> Hi, >> >> Sorry I am not entirely sure what you mean could you give me an example >> please. >> > > Hi, please keep the thread in the maillist. > > I just mean that, to keep your domain hidden, just change the letters but no > the symbols. Example: > > - Real domain: mycompany-cool-voip.com > - Showed domain: lalalaala-lolo-lili.com > > -- > Iñaki Baz Castillo <ibc(a)aliax.net> > Sorry about the wrong reply address, blame gmail :-) Here you go INVITE sip:01684299007@mydomain.net:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.20.240:5060;branch=z9hG4bK-d8754z-ed473d0ab724045f-1---d8754z-;rport. Max-Forwards: 70. Contact: <sip:500@78.32.132.193:5060>. To: <sip:01684299007@mydomain.net:5060>. From: "Michele "<sip:500@mydomain.net:5060>;tag=fd5b583a. Call-ID: NzU1MTZiMjZjZGM0ZTY5ZmFhMWRjNDlkNGQxZTZmODM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. User-Agent: 3CXPhoneSystem 7.0.4708.0. Authorization: Digest username="500",realm="mydomain.net",nonce="4b606dcc81900a95d9962d3c737cc570e624d409",uri="sip:01684299007@mydomain.net:5060",response="1ac466e73c744ed5936179152b5b27d4",algorithm=MD5. Content-Length: 278. . v=0. o=3cxPS 444025798656 5368709121 IN IP4 78.32.132.193. s=3cxPS Audio call. c=IN IP4 78.32.132.193. t=0 0. m=audio 9000 RTP/AVP 0 8 3 101. c=IN IP4 78.32.132.193. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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Re: [Kamailio-Users] ACC on Failure route
by Daniel-Constantin Mierla 28 Jan '10

28 Jan '10

On 1/25/10 2:10 PM, alex pappas wrote: > > What exactly is iirc? common abbreviation in net-slang, helping to save the planet, think green, by sending less bits on net :-): http://www.internetslang.com/IIRC.asp Cheers, Daniel -- Daniel-Constantin Mierla * http://www.asipto.com/

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[Kamailio-Users] problem with ACC module
by Denis Putyato 28 Jan '10

28 Jan '10

Hello everybody! I have a such problem with ACC module. Calls was accounted but only INVITE and BYE requests has appeared in acc table. ACK is absent. Kamailio 3.0 was installed from GIT. Thank you for any help # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "detect_direction", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "failed_transaction_flag", 29) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_url", "xxxxxxxxxxxxxxxxxxxxxxxx") modparam("acc", "db_extra", "src_user=$avp(i:2);src_domain=$si;dst_ouser=$avp(i:1);dst_user=$rU;dst_doma in=$rd") route { sip_trace(); setflag(30); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(1); setflag(29); } if (is_method("ACK")) { xlog("L_NOTICE", "ACK detected"); # I see this line in syslog setflag(1); setflag(29); } route("relay"); exit; } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { setflag(1); route("relay"); exit; } else { exit; } } } }

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[Kamailio-Users] Dynamic setting of dispatcher route set
by Alex Balashov 27 Jan '10

27 Jan '10

Unfortunately, ds_select_dst()/ds_select_domain() do not seem to take PVs for their first argument (the route set ID from which to choose). Is there any workaround for this, e.g. when the route set is indeterminate until runtime? It would be very inconvenient to have to work around this by implementing own rollover logic when the dispatcher module is so good... perhaps the possibility of a quick and easy patch to the git trunk? Thanks in advance! -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/

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[Kamailio-Users] Bad URI
by Jon Farmer 27 Jan '10

27 Jan '10

Hi Does anyone know why I am getting ERROR: parse_uri: bad host in uri (error at char from this ( i hidden the domain and IP) INVITE sip:xxxxxxxx@mydomain.net:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bKea312c9f5673C67A. From: "200" <sip:200@mydoamin.net>;tag=D6C112A0-D880344D. To: <sip:xxxxxxxx@mydomain.net;user=phone>. CSeq: 2 INVITE. Call-ID: f1f44bbc-58cbed36-2ea0cdab(a)xx.xxx.xxx.xxx. Contact: <sip:200@xx.xxx.xxx.xxx:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.6.3.0067. Supported: 100rel,replace. Allow-Events: talk,hold,conference. Authorization: Digest username="200", realm="mydomain.net", nonce="4b605fcd30e54ca4d6b52ffc9fc6128bc355cf71", uri="sip:xxxxxxxx@mydomain.net:5060;user=phone", response="7c82f9511843dc55aeaed27f468efe39", algorithm=MD5. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 254. . v=0. o=- 1264546283 1264546283 IN IP4 xx.xxx.xxx.xxx. s=Polycom IP Phone. c=IN IP4 xx.xxx.xxx.xxx. t=0 0. a=sendrecv. m=audio 10010 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

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