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42 participants
37519 discussions

How to do accounting for Kamailio-generated BYEs?
by Yufei Tao 27 Feb '12

27 Feb '12

Hi I've got a mediaproxy that detects no media, and sends dlg_end_dlg to Kamailio via mi_datagram socket. Then Kamailio sends BYE to both calling parties. It all works fine. But my question is, how to do accounting for these Kamailio-generated BYEs? They don't appear in the acc table. I'm using Kamailio 3.1.5. Thank you very much! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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CallControl and MediaProxy
by Reda Aouad 27 Feb '12

27 Feb '12

Hi, When I use the function call_control( ) of the call_control module, it automatically engages mediaproxy if it finds the mediaproxy module loaded. If the mediaproxy module is not loaded, call_control doesn't even try to engage it. I need mediaproxy for NAT traversal in some cases, but don't want it to be engaged on every call. How can I disable this behavior? Thanks Reda

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Social networking event, London, March 6, 2012
by Daniel-Constantin Mierla 27 Feb '12

27 Feb '12

Hello, we are already several folks planing to go out after the first day of UC Expo and continue discussions about latest news of Kamailio/SER project and VoIP/Telecomunications. If you want to join, drop me a message to send you the details of the location. Looking forward to meeting many of you next week, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] How to do request URI rewrites using database tables
by Javier Gallart 27 Feb '12

27 Feb '12

Hello Ben dialplan module should help you: http://kamailio.org/docs/modules/3.2.x/modules/dialplan.html Regards Javi > ------------------------------ > > Message: 5 > Date: Fri, 17 Feb 2012 12:49:04 +1300 > From: Ben WIlliams <benwilliams(a)joobworld.com> > Subject: Re: [SR-Users] How to do request URI rewrites using database > tables? > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CANfYGQkeK3i=vBbdox+ekGeeKmjWhhgarUJcAo3oV=rrO=Wwig(a)mail.gmail.com > > > Content-Type: text/plain; charset=ISO-8859-1 > > I've managed to get dbaliases and lcr to do most of this. But the lcr > prefix does not allow regular expressions. Is there any other module > that allows you to store in the database a regular expression rewrite > rule? > > On Thu, Feb 16, 2012 at 1:21 PM, Ben WIlliams <benwilliams(a)joobworld.com> > wrote: > > Hi, can someone please recommend the most appropriate modules to > > rewrite R-URIs based on a database lookup table? I've read the > > documentation for lcr and carrierroute but not sure if they can do > > this. > > > > In most cases it will be a simple R-URI rewrite but I also need to > > rewrite based on From user. > > > > ie > > > > R-URI match ? ? ? ? ? ? From match ? ? ? ? ? ? ? ? ?new R-URI > > =========== ? ? ? ? ? ? ========== ? ? ? ? ? ? ? ? ?========= > > *97(a)example.com ? ? ? ? 6[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.1 > > *97(a)example.com ? ? ? ? 7[0-9](a)example.com ? ? ? ? ?*97(a)10.0.0.2 > > 6[0-9](a)example.com ? ? ?match any ? ? ? ? ? ? ? ? ? rewrite domain to > 10.0.0.1 > > abc(a)example.com ? ? ? ? match any ? ? ? ? ? ? ? ? ? def(a)example.com > > > > Thanks > > Ben > > > >

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Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service (Have-my-domain! - host your own SIP domain at iptel.org). I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank, CTO SuperCel.tel

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Have-my-domain! - host your own SIP domain at iptel.org
by SuperCel.mx.tc 26 Feb '12

26 Feb '12

Sirs: We cannot get our domain to configure to the iptel SIP Service. I have attached a copy of how we've tried setting the DNS, but we cannot find anywhere that gives us info on the "SIP" and/or "protocol UDP". Can you assist us so that we can use our own domain (http://supercel.tel) so that we can send SIP calls across this domain? Sincerely, Frank Riley, CTO SuperCel.tel

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Kamailio segmentation fault
by Fabian Borot 24 Feb '12

24 Feb '12

HelloOur kamailio proxy (version 1.5.2 ) crashed today due to a segmentation fault:Feb 22 16:52:10 AW-PROXY-01 kernel: kamailio[3667]: segfault at 000000000000000b rip 00002ab2b47bf93f rsp 00007ffff62dd270 error 6Apparently due to lack of private memory:[root@AW-PROXY-01 log]# cat proxy.log | grep ERROR | grep dbFeb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_rows: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_convert_result: converting rows failed Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_store_result: failed to convert result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:core:db_do_query: error while storing result Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:db_unixodbc:db_unixodbc_fetch_result: no private memory left Feb 22 16:22:10 AW-PROXY-01 /usr/local/kamailio-install-dir/sbin/kamailio[3667]: ERROR:htable:ht_db_load_table: Error while fetching result I was hoping to see a core file somewhere but I haven't found one yet. I did not have the line "disable_core_dump=no" set on the config file because supposedly is set to no by default: >From the online guide for my version: //////////////////// Disable_core_dumpCan be 'yes' or 'no'. By default core dump limits are set to unlimited or a high enough value. Set this config variable to 'yes' to disable core dump-ing (will set core limits to 0).Default value is 'no'. ///////////////////How can I make sure that a core file is created after a situation like this? Where will it be created? How can I check the status of the private memory? I use the "kamctl fifo get_statistics all" but I don't know which one relates to "private memory": Example:shmem:total_size = 2147483648 shmem:used_size = 16860928 shmem:real_used_size = 17033968 shmem:max_used_size = 19174528 shmem:free_size = 2130449680 shmem:fragments = 1850I know that with so little information is hard to diagnose, but if anybody has an idea as to what else I can do I greatly appreciate it. Thank you in advancefborot

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avp_delete problem [kamailio 1.5]
by Sven Knoblich 24 Feb '12

24 Feb '12

Hello all, i am currently confused by using the avpops function avp_delete. When i run avp_delete without the flag-value \g only the last value will be unset (like an undo). Is this the wanted behaviour? EXAMPLE: $avp(test) = "test" if( $avp(test)){xlog("L_NOTICE","result1:$avp(test)\n");} $avp(test) = "nooo"; if( $avp(test)){xlog("L_NOTICE","result2:$avp(test)\n");} avp_delete("$avp(test)"); if( $avp(test)){xlog("L_NOTICE","result3:$avp(test)\n");} RESULT: result1:test result2:nooo result3:test Could anybody helps me to understand that? thanks in advance, Sven

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Adding parameter to headers
by Uri Shacked 24 Feb '12

24 Feb '12

Hi, I need to add a parameter to the contact header. suppose to look like: Contact: <sip:anonymous@domain:port;param1=XXX;param2=YYY> How do I add the parameters? Should i concatenate the $ct with the parameters? Or is there a better way? BR, Uri

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fix_nated_sdp issue
by Ric Marques 24 Feb '12

24 Feb '12

Greetings, I'm not sure if I found a bug, or if I just have something completely misconfigured... I'm a total newb with Kamailio, working on a proof of concept design. Here's my configuration: provider -> nat firewall -> kamailio/rtpproxy -> asterisk For outbound calls from a phone registered to asterisk via kamailio, I'm trying to use fix_nated_sdp("2", "10.50.50.8") to rewrite the media ip address to resolve my audio issues, where 10.50.50.8 is the address outside my firewall. What I'm running into is the 'c=' line doesn't get re-written properly... it inserts the specified address in front of the existing address, and I end up with the following line in my INVITE: c=IN IP4 10.50.50.810.0.10.10 I have the fix_nated_sdp command under route[sipout], because I only want to use it on calls being sent outside the nat firewall. Here's the sip invite without the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19165551212@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as5498b77e;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK4b3a.960f6466.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK145db73e;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as5498b77e To: <sip:19165551212@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 06b8bb1b7dd7801d7b3b9c917fcb9b12@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:06:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 309 P-hint: outbound v=0 o=root 604360056 604360056 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.0.10.10 t=0 0 m=audio 9702 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Here's the sip invite with the 'fix_nated_sdp' command: -------------------------------------------------------------------------------------------------------------- INVITE sip:19167828326@xxx.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.0.10.10;lr=on;ftag=as49e00c81;nat=yes> Via: SIP/2.0/UDP 10.50.50.8.;branch=z9hG4bK1eab.800c4724.0 Via: SIP/2.0/UDP 10.0.10.11:5060;branch=z9hG4bK20d28324;rport=5060 Max-Forwards: 69 From: "1009" <sip:1009@10.0.10.11>;tag=as49e00c81 To: <sip:19167828326@xxx.xxx.xxx.xxx> Contact: <sip:1009@10.0.10.11:5060> Call-ID: 4def5539675b6f644b99bb300e8ec8d6@10.0.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r356107 Date: Wed, 22 Feb 2012 03:18:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 347 P-hint: outbound v=0 o=root 1009117068 1009117068 IN IP4 10.0.10.10 s=Asterisk PBX SVN-branch-1.8-r356107 c=IN IP4 10.50.50.8.10.0.10.10 t=0 0 m=audio 13540 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=oldmediaip:10.0.10.11 a=nortpproxy:yes -------------------------------------------------------------------------------------------------------------- Is this a bug, or is it likely I have something else screwed up? Thank you in advance for your assistance - this list is an incredible resource! -Ric

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