sr-users June 2012

sr-users@lists.kamailio.org

78 participants
108 discussions

[Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
by Samuel Muller 26 Sep '12

26 Sep '12

Hello all, I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose is to have several interconnections with PSTN. I configured it like this : Audiocodes registers as a gateway to the Kamailio, using a dedicated port (5062). Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the proxy. I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm. But the audiocodes returns some errors about SIP headers sent by Kamailio : ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected Here you have the example of an INVITE from a SIP phone to the PSTN : ** audiocodes debug ** 4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from 77.246.81.132:5060 ---- INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0 Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes> Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0 Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96 From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0 To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>> Call-ID: 944d8aec-27503ee6(a)192.168.0.113 CSeq: 102 INVITE Max-Forwards: 49 Contact: "Sam" <sip:0123451010@77.246.81.162:15170> Expires: 240 User-Agent: Linksys/SPA941-5.1.8 Content-Length: 281 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER Supported: 100rel, replaces Content-Type: application/sdp P-Asserted-Identity: <0123451010> Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes v=0 o=- 26933860 26933860 IN IP4 192.168.0.113 s=- c=IN IP4 77.246.81.133 t=0 0 m=audio 35038 RTP/AVP 18 0 8 101 a=rtpmap:18 G729a/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=nortpproxy:yes ( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time: 12:30:26] ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol '0' in scheme. ALPHA expected ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26] ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26] ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26] The outgoing INVITE from Kamailio is exactly the same received by the AudioCodes. When I searched over Google, I just found 2 answers about Asterisk / Audiocodes unsolved problem, but no more informations. I supposed that the problem is as indicated : " s=- " where source is empty in place of "NULL" / "0" or something like this ... Someone can confirm or already met the problem ? Many thanks all :) .Sam.

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[Kamailio-Users] Having problems using RTPProxy to bridge internal/external networks
by Joe Hart 15 Sep '12

15 Sep '12

Hi all, For a project on which I'm currently working, I am having some problems figuring out how to correctly configure Kamailio to communicate with RTP Proxy in order to send media into and out of a network with private IP address ranges. I have a proxy set up to send the SIP traffic, and all of this is working fine. However, I'm having some trouble getting the RTP Proxy set up. Currently, when the call is connected, the offer/answer is made and RTP Proxy seems to be taking over, but I'm having trouble getting my audio to flow in both directions. Examination of the traffic coming into and out of this machine seems to indicate that the IP addresses aren't being mangled correctly. Specifically, it appears the internal IP address isn't being changed to reflect the IP address of the machine on which RTP Proxy is running, so that when the caller tries to send audio back, the IP it's given to reply to is 10.10.x.x, which obviously won't work. I have tried experimenting with specifically setting IP addresses in the rtpproxy_offer() and _answer() methods to no avail, as well as setting various flags in those methods. However, I must admit that I'm not entirely sure what's happening under the hood with these methods, or what rtpproxy is doing with that information when it gets it. Rather than continue to hack at this by trial and error, I'm hoping someone here can point me in the right direction. Any advice, example code or pep talks would be greatly appreciated. Thanks in advance, -- Joe Hart Voice Systems Integrator Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0673

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Suggestions for time utils
by Andreas Granig 16 Aug '12

16 Aug '12

Hi, I'd like to implement a couple of helper functions for time handling, e.g. checking day of week, day of month etc. from within the kamailio config file. What would you prefer, a new module ("timeutils" maybe?) or adding it to cfgutils (there are already time-based functions there like sleep and usleep)? The idea is to implement time based call-forwards, and an approach could be to provision various time-related values in usr_preferences table, and then check it in the config. For example, when doing call-forwards from Monday to Friday only, I could put this into usr_preferences: attribute: cf_weekday value: [1, 2, 3, 4, 5] (each entry is a separate row in usr_preferences) And in kamailio config, I'd call this: avp_db_load(...); if(is_weekday("$avp(s:cf_weekday)")) { do CF } So is_weekday would iterate over the entries in the avp list and return true if the weekday at the time of routing matches an entry in the list. A module config param could control whether to use gmtime or localtime for matching. Does this make sense? Suggestions for other approaches? I'd rather prefer to do it directly in config instead of using some external interpreter like lua, python etc. Andreas

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Possible bad lcr module, after upgrade to 3.3.0 from 3.2
by Alexey Mechanoshin 13 Aug '12

13 Aug '12

Hello. After upgrade kamailio and db, lcr module bad routing if entered *any value*to from_uri column in lcr_rule table: mysql> select * from lcr_rule where id=5; +----+--------+--------+-------------------------------+-------------+---------+---------+ | id | lcr_id | prefix | from_uri | request_uri | stopper | enabled | +----+--------+--------+-------------------------------+-------------+---------+---------+ | 5 | 1 | yyyy | sip:x[xx]\d{2}@xxx.xxx.xx.xx | NULL | 0 | 1 | +----+--------+--------+-------------------------------+-------------+---------+---------+ and load_gw(...) skip this rule, in log: INFO: lcr [lcr_mod.c:1493]: skipping disabled <gw/rule> = <1/5> If remove value from 'from_ui' - rule enabled and all work. Why is turned off rule ? Linux Debian squeeze, upgraded from "deb http://deb.kamailio.org/kamailiosqueeze main" -- "В связи с отсутствием интереса общественности, конец света отменяется" (с) ? С Уважением, Механошин Алексей

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Reply 503 was replaced to 500
by Uri Shacked 18 Jul '12

18 Jul '12

Hi, Kamailio server is behind our company's softswitch and acts as a sip application server. I notice that there are calls that the softswitch replied with 503 "service unavailable" and kamailio sent to the originator leg 500 "service unavaileable". When kamailio recieved 504 or 502 it sends them back as is. shouldn't it be the same with 503? It also does not have a "to tag" in the CDR. And the "to tag" in the 503 that was recieved is not equal to the 500 reply "to tag" kamailio sent back. any ideas? BR, Uri

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Kamailio Unexpectedly Terminating
by Akan 12 Jul '12

12 Jul '12

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 9(3369) INFO: <core> [main.c:817]: INFO: signal 15 received 10(3370) INFO: <core> [main.c:817]: INFO: signal 15 received 11(3371) INFO: <core> [main.c:817]: INFO: signal 15 received 12(3372) INFO: <core> [main.c:817]: INFO: signal 15 received 13(3373) INFO: <core> [main.c:817]: INFO: signal 15 received 14(3374) INFO: <core> [main.c:817]: INFO: signal 15 received 0(3360) DEBUG: presence_xml [presence_xml.c:347]: start 0(3360) ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Permission denied (13) 0(3360) DEBUG: <core> [db_pool.c:102]: removing connection from the pool 0(3360) DEBUG: db_postgres [km_pg_con.c:122]: PQfinish(100842470) 0(3360) DEBUG: db_postgres [km_pg_con.c:126]: pkg_free(1004c1f30) 0(3360) DEBUG: tm [t_funcs.c:122]: DEBUG: tm_shutdown : start 0(3360) DEBUG: tm [t_funcs.c:125]: DEBUG: tm_shutdown : emptying hash table 0(3360) DEBUG: tm [t_funcs.c:127]: DEBUG: tm_shutdown : removing semaphores 0(3360) DEBUG: tm [t_funcs.c:129]: DEBUG: tm_shutdown : destroying tmcb lists 0(3360) DEBUG: tm [t_funcs.c:132]: DEBUG: tm_shutdown : done Thanks Nathaniel

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Binding arbitrary TLS socket
by Andreas Granig 02 Jul '12

02 Jul '12

Hi, Got some problems when I tried this, not sure how this is intended to work: listen=tls:192.168.51.133:4343 When starting kamailio 3.3, I get the following error: ERROR: <core> [tcp_main.c:2915]: ERROR: tcp_init: bind(10, 0x7f5eea45db24, 16) on 192.168.51.133:4343 : Address already in use ERROR: tls [tls_init.c:314]: Error while initializing TCP part of TLS socket 192.168.51.133:4343 What works though is if I explicitely bind a TCP socket one port below the TLS socket, like this: listen=tcp:192.168.51.133:4342 listen=tls:192.168.51.133:4343 Is this how it's intended to work? Is there a specific reason for this behavior? Andreas

2 2

cfgutils lock / unlock + $sht() variables.
by Vitaliy Aleksandrov 02 Jul '12

02 Jul '12

Hi all, I use $sht() pseudo-variabled provided by htable module for caching various information about sip customers. I thought that htable has some locking mechanism inside it and today while i was reading core-cookbook i payed attention to one example with the lock/unlock functions and $sht(): lock("calls-to::10.10.10.10"); $sht(a=>calls-to::10.10.10.10) = $sht(a=>calls-to::10.10.10.10) + 1; unlock("calls-to::10.10.10.10"); According to that example an explicit locking mechanism should be used when i want to write something to a hash table. On the other hand there are no locks in htable documentation. So the question is when do the explicit locks should be used and for what purpose ?

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Re: [SR-Users] how to talk across domains, my domain to say, iptel.org
by Alex Balashov 30 Jun '12

30 Jun '12

Note that from a security point of view, this is not necessarily a desirable trait, without appropriate programmatic restrictions and/or ACLs. -- Alex -- 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/Auro Tripathy <Auro.Tripathy(a)wdc.com> wrote:Thanks Klaus, non-local routing appears to work out-of-the-box. Thank you again for confirming what I observed. -Auro From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Sent: Friday, June 29, 2012 3:52 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Auro Tripathy Subject: Re: [SR-Users] how to talk across domains, my domain to say, iptel.org The default configuration supports that already. Usually inbound routing does not differ if the call comes from a local subscriber or from an external domain. Outbound routing is performed if the domain in the request URI (ruri) is non-local. regards Klaus Am 29.06.2012 19:46, schrieb Auro Tripathy: Hi All: I’m a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I’m able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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how to talk across domains, my domain to say, iptel.org
by Auro Tripathy 30 Jun '12

30 Jun '12

Hi All: I'm a newbie, excuse the simple (no pun intended) question. I installed Kamailio and I'm able to set it up so that all subscribers in the *same* domain can talk/discover-presence, etc. What do I do so that subscribers in other domains, for example the iptel.org domain, can talk to subscribers in my domain? -Auro

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sr-users June 2012