Re: [SR-Users] modulo operator
Hello, On 2/28/11 9:15 PM, Klaus Darilion wrote: Hi! Using kamailio 3.1.1, I failed to use '%' as described in the core cookbook. Using 'mod' instead seems to work. % was used in SER for some attributes AFAIK -- looking at cfg.lex -- so I changed it to 'mod' only in 3.x. I should check again if the conflict really exists and/or can be avoided. For now using 'mod' is the option to go. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.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] Segfault in Kamailio 1-3-4 to 1.5.5
On 3/1/11 8:41 AM, Andrew O. Zhukov wrote: I someone interested in . It's the old coredumps from 1.3.4. It's really much stabile then 1.5.5 I did degrade today night version 1.3.x is openser only which became later kamailio, practically is no other option for this version. Have you considered upgrading to latest stable (3.1.x) instead of downgrade? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.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] Segfault in Kamailio 1-3-4 to 1.5.5
On 2/28/11 8:06 AM, Andrew O. Zhukov wrote: As I understood you do not provide any support for a legacy versions. In the first place, the problem is you are using very old versions and it is very unlikely someone has a testbed for them. I and many others still have such versions running, but never happened to crash, it has to be something specific, like a not very common module or particular sip request that triggers this one. I tried to help you in the spare time, which didn't happen to be that much lately. Your way of answering the questions was also consuming a lot of such cycles. Normally, yes, we officially support the latest two stable version, those being now 3.0 and 3.1. And it is really advisable to use the latest stable. But as you could see, we don't mind doing it for older version when we can, but that is not always possible we current constraints of time and load. Even if you are willing to get paid support, it is not always possible to get it from a day to the next one, people travel or have other project booked some time ago. Cheers, Daniel On 02/25/2011 09:00 AM, Andrew O. Zhukov wrote: In continue of letters: Kamailio 1.5.5 No TLS Segmentation Fault After upgrade from openser 1.3.4 to kamailio 1.5.5 the same crash set Can someone from developers provide me commercial support to fix this bug in malloc module. If so, contact me directly. -- Daniel-Constantin Mierla http://www.asipto.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] Segfault in Kamailio 1-3-4 to 1.5.5
On 03/01/2011 10:49 AM, Daniel-Constantin Mierla wrote: On 3/1/11 8:41 AM, Andrew O. Zhukov wrote: I someone interested in . It's the old coredumps from 1.3.4. It's really much stabile then 1.5.5 I did degrade today night version 1.3.x is openser only which became later kamailio, practically is no other option for this version. Have you considered upgrading to latest stable (3.1.x) instead of downgrade? Daniel, I sent you my config. How can I do it on a hi usage production server for a one night. The lot of fixes for a different buggy customers SIP and NAT devices which is impossible to retest again. I'll try opensips and possible will be back. Cheers, Daniel ___ 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] Consulting needed
See http://www.kamailio.org/w/business-directory/ and http://www.kamailio.org/w/business/ for consultants. regards Klaus Am 28.02.2011 22:57, schrieb Pete Ashdown: I have been trying to accomplish a couple tasks with Kamailio over the past month with no luck. What I need is a bit of one-on-one training with someone who knows the lay of the land. If you do this kind of consulting and can use Skype with possibly a shared-screen terminal, please drop me an email with your rate. ___ 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 Installation
Hello, I don't get why you have errors regarding the xml files. Have you set the FLAVOUR=kamailio? Maybe you can follow the next tutorial and adapt it for redhat: http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x-from-git Practically it is about the installation of dependencies. The compileinstall 'make' commands are the same. Cheers, Daniel On 2/28/11 7:08 AM, Suresh Bhandari wrote: Hello Community, I am new to Kamailio, and this list as well. I am trying to install Kamailio 3.1.2, but I am getting too many errors. I have fixed some but still not getting the way. I am using Red Hat Enterprise Linux (RHEL) 5, and /usr/local directory. When I ran the following command: make group_include=standard standard-dep mysql include_modules=carrierroute peering install it prompted not found error for the file docbookx.dtd, I found it (modules/auth/auth.xml, and modules_s/acc_syslog/acc_syslog.xml) and fixed it as it was errorenous URL location. For reference earlier it was http://www.oasis-open.org/docbookid/id/g/4.5/docbookx.dtd, which I changed to http://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd. Now when I run the previous command again, I am getting the follwing errors: /nsgmls:URLhttp://www.oasis-open.org/docbook/xml/4.5/docbookx.dtd:116:17:E: X20AC is not a function name/ If I ignore this, and continue, I am not able to find the sip-router service in /etc/init.d. The entire errors is attached here. Please help me solve the issue. TIA Suresh ___ 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 ___ 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] modulo operator
Am 01.03.2011 09:46, schrieb Daniel-Constantin Mierla: Hello, On 2/28/11 9:15 PM, Klaus Darilion wrote: Hi! Using kamailio 3.1.1, I failed to use '%' as described in the core cookbook. Using 'mod' instead seems to work. % was used in SER for some attributes AFAIK -- looking at cfg.lex -- so I changed it to 'mod' only in 3.x. I should check again if the conflict really exists and/or can be avoided. For now using 'mod' is the option to go. I added some text to the core coookbooks. regards 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
Re: [SR-Users] Segfault in Kamailio 1-3-4 to 1.5.5
On 3/1/11 10:02 AM, Andrew O. Zhukov wrote: On 03/01/2011 10:49 AM, Daniel-Constantin Mierla wrote: On 3/1/11 8:41 AM, Andrew O. Zhukov wrote: I someone interested in . It's the old coredumps from 1.3.4. It's really much stabile then 1.5.5 I did degrade today night version 1.3.x is openser only which became later kamailio, practically is no other option for this version. Have you considered upgrading to latest stable (3.1.x) instead of downgrade? Daniel, I sent you my config. How can I do it on a hi usage production server for a one night. The lot of fixes for a different buggy customers SIP and NAT devices which is impossible to retest again. Sending the config is not enough, since I can not use it in my server, I do not have your kind of traffic. The config is good when is some misrouting or syntax error, but for this specific case the investiagation of core and adding some patches to print more information when the crash is happening is the way to solve. I sent you some patches, that were not good enough because I had no 1.5 around and I was offline. More than that, I can count 3-4 more developers that tried to help you on the public mailing list, even you play with very old versions. As said, everyone tries to do it in available time and its own conditions. I would need access to the server to investigate the core dump myself -- you offered that but being traveling was not for me at that time. My interest is to discover if it something that affects 3.x, although we changed the internal architecture a lot, might be some cases existing in 1.x still applying in 3.x What I don't understand is the complain regarding testing. When you did the upgrade to 1.5 from 1.3, you had to do changes everywhere, there were major versions. Same would be for a migration from 1.5 to 3.1. You can even have them both installed, using shared database so you can start/restart with older or newer versions. I did it many times and it goes smooth, just few tables have changed the structure, for that case you can use different databases. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.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] modulo operator
On 3/1/11 10:26 AM, Klaus Darilion wrote: Am 01.03.2011 09:46, schrieb Daniel-Constantin Mierla: Hello, On 2/28/11 9:15 PM, Klaus Darilion wrote: Hi! Using kamailio 3.1.1, I failed to use '%' as described in the core cookbook. Using 'mod' instead seems to work. % was used in SER for some attributes AFAIK -- looking at cfg.lex -- so I changed it to 'mod' only in 3.x. I should check again if the conflict really exists and/or can be avoided. For now using 'mod' is the option to go. I added some text to the core coookbooks. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.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] Segfault in Kamailio 1-3-4 to 1.5.5
On 03/01/2011 11:02 AM, Andrew O. Zhukov wrote: On 03/01/2011 10:49 AM, Daniel-Constantin Mierla wrote: On 3/1/11 8:41 AM, Andrew O. Zhukov wrote: I someone interested in . It's the old coredumps from 1.3.4. It's really much stabile then 1.5.5 I did degrade today night version 1.3.x is openser only which became later kamailio, practically is no other option for this version. Have you considered upgrading to latest stable (3.1.x) instead of downgrade? Daniel, I sent you my config. How can I do it on a hi usage production server for a one night. The lot of fixes for a different buggy customers SIP and NAT devices which is impossible to retest again. I'll try opensips and possible will be back. Hello, I had been checking the coredumps you provided for a while. I don't think they are very useful because for me this looks like an Heisenbug. The coredumps only show the result of the memory corruption and not the cause. Daniel has asked you for some input when compiling with memory debug on (to see if canary values where overwritten by what operations). I have not seen this output yet (don't know if you send it privately to him, or I have missed it on the list). Try to compile again with debug memory support, and set logging to a apropriate level. An idea is to set a special server with a special version of Kamailio. Minimize the number of children (use only one worker) and use just part of the traffic so you have a easier debugging. The bug affects private memory so the number of children should not be an impact. Also you might want to increase the PKG_MEM_SIZE from the default value of 4 MB to something bigger (try 10-15 MB or more). See if this has an impact on the bug (it might be caused by fragmentation in this case a bigger pool might help). If all else fail, I strongly suggest dropping pkg_malloc all together, and using libc's Malloc() instead. This is done at compile time by removing the -DPKG_MALLOC from Makefile.defs and recompiling.. This should fix your bug Marius Cheers, Daniel ___ 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] rtpproxy and connection information field
Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12...@.xxx;transport=udp SIP/2.0 Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp To: 2205sip:12...@.xxx From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:x Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s= c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates c= IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the c= field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez ___ 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 and connection information field
I have to use r flag. Sorry for my quick posting... De: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] En nombre de Dominguez Jover, Ricardo Enviado el: martes, 01 de marzo de 2011 13:16 Para: sr-users@lists.sip-router.org Asunto: [SR-Users] rtpproxy and connection information field Hi all, I'm using rtpproxy 1.2.1 and kamailio. RTPproxy is working fine in almost every case. However I have a problem with some call. When softphone A using sip2sip.info account calls softphone B using my Kamailio server account, the Kamilio receives SIP packets from IP1 (81.23.228.129). The invite packet has IP2 (81.23.228.150) in the conecction information field (c=IN IP4 81.23.228.150): INVITE sip:12...@.xxx;transport=udp SIP/2.0 Record-Route: sip:81.23.228.129;lr;ftag=d9e99adf;did=e81.3538c317 Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bKd14b.12802794.0 Via: SIP/2.0/UDP 192.168.xx.xx:7964;received=88.xx.xx.xx;branch=z9hG4bK-d8754z-8ba1b4b47e 37bb89-1---d8754z-;rport=7964 Max-Forwards: 69 Contact: sip:54...@88.xxx.xxx.xxx:7964;transport=udp To: 2205sip:12...@.xxx From: R sip2sipsip:54...@sip2sip.info;tag=d9e99adf Call-ID: OGE5MTFiMzljZjE0NTYxN2M0N2VkZjgzNDRhMzE2ZTA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces User-Agent:x Content-Length: 279 v=0 o=- 12943454020854250 1 IN IP4 192.168.xxx.xx s= c=IN IP4 81.23.228.150 t=0 0 m=audio 52854 RTP/AVP 107 0 8 18 101 a=rtpmap:107 BV32/16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Then Kamilio translates c= IP address to my RTPproxy IPaddress and sends the INVITE to softphone B. After the OK softphone B sends RTP packets to the RTPproxy as specified in the c= field, however the RTPproxy sends RTP packets to IP1 instead of send packets to IP2. How can I tell the RTPproxy to send packets to IP2 (the one in the c= field?) Thanks, Ricardo Dominguez ___ 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] about tmx:inuse_transactions stat
Juha Heinanen writes: regarding tmx:inuse_transactions stat, it does not seem to exist among tm.stats: ... or does it have the same value as created - freed? a took a look at the code and tmx inuse_transactions seems to be equal to tm current transactions. -- 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] about tmx:inuse_transactions stat
Hi Juha, On 3/1/11 9:34 AM, Juha Heinanen wrote: Juha Heinanen writes: regarding tmx:inuse_transactions stat, it does not seem to exist among tm.stats: ... or does it have the same value as created - freed? a took a look at the code and tmx inuse_transactions seems to be equal to tm current transactions. you are right, I saw your email but I forgot to answer it. SER core, sl tm exported more stats in regard to transactions and replies, so I kept that version when we integrated and exported them via K stats API. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.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] NAT Traversal
May be I miss some important details? No suggestions? Thank you. Hello, all. Using nathelper + rtpproxy for subj. Kamailio has public and private network interfaces. Asterisk is only private. RTP Proxy is working in bridge mode and relaying traffic from UAC to Asterisks. Everything is working fine, except one configuration. When the client is behind router ( a specific one, I do not have an access there to check ), and this UAC is making a call to other public extension, which is behind router, then RTP Proxy is relaying traffic to the caller, using another UDP port, then the packets arrive. For instance: UAC 1 - UAC 2 PUBLIC_IP:10 KAMAILIO_IP: KAMAILIO_IP:5678 PUBLIC_IP:12 While for the UAC 2 it looks like: PUBLIC_IP:20 KAMAILIO_IP:6767 KAMAILIO_IP:4564 PUBLIC_IP:20 The source and destination UDP ports are the same. As result, I can hear UAC 1 and he cannot hear me. In case of we have UAC 3, which is behind other router, call is working fine with same configuration. It's routers fault you can say, but in the same configuration ( I mean network, not kamailio ) it worked, but when RTPProxy was not in bridge mode and Kamailio and Asterisks were in public network. Reinvites are not allowed in both cases. The question is, why the source and destination UDP ports are different? Using STUN in first case, cause without it, private IP written in contacts and as result, traffic relayed from Kamailio is incorrect, cause heading to private network which is unreachable. Any ideas where to dig? ___ 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
