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37555 discussions

Homer Project
by Alexandr Dubovikov 05 Aug '11

05 Aug '11

Hello sr-users, I would like to present a new project called "Homer". What is Homer ? Homer is a capture solution for SIP logging. How does it work? Homer has 4 components: 1. Capture Agent 2. Capture Node 4. Capture DB 3. Web GUI. The capture agent takes your local SIP traffic and sends it to the defined capture node in HEP protocol. If you are user of FreeSWITCH or Kamailio, use build-in capture agent. In other case, you can use an external generic daemon, but only for SIP/UDP. The capture node is based on kamailio, can receive SIP traffic encapsulated in IP proto 4 (IPIP) or in HEP procotol and store it into MySQL DB. To gain access to the capture DB, download a simple webinterface (PHP based). After it you are able to do search, generate PCAP file, callflow, look at the details of SIP messages etc. Homer works in our company and easy collect ~5 M packets per hour. The patches for kamailio and some other information can be found here: http://homer.googlecode.com Beta testers are welcome :-) I'll help everybody to setup/configure Homer :-) With best regards, Alexandr

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$du gets set automatically before branch_route
by Alex Hermann 05 Aug '11

05 Aug '11

Hello, while trying to upgrade from 1.5 to 3.1 i noticed that between t_relay() and branch_route, $du gets automatically set to $ru if it has not been set before. This is a change from 1.5 i could not find documentation for, so it might have been unintened. Can someone enlighten me on the purpose of this change and if there is a way to determine if $du is (was) unset other than comparing $ru to $du, wich seems a bit inefficient? Is it even guaranteed that $du gets set to $ru in every possible scenario? -- Alex Hermann

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udp vs. tcp receiver processes
by Juha Heinanen 05 Aug '11

05 Aug '11

i have 8 children specified in default file and in cfg file i have: listen=192.98.101.10:5060; listen=udp:127.0.0.1:5070; listen=udp:127.0.0.1:5070; listen=tcp:127.0.0.1:5070; listen=udp:127.0.0.1:5080; listen=tcp:127.0.0.1:5080; listen=tcp:127.0.0.1:6060; that results in 8 tcp receiver processes but 3 x 8 udp receiver processes: # sip-proxy_ctl ps 30720 attendant 30721 udp receiver child=0 sock=192.98.101.10:5060 30722 udp receiver child=1 sock=192.98.101.10:5060 30723 udp receiver child=2 sock=192.98.101.10:5060 30724 udp receiver child=3 sock=192.98.101.10:5060 30725 udp receiver child=4 sock=192.98.101.10:5060 30726 udp receiver child=5 sock=192.98.101.10:5060 30727 udp receiver child=6 sock=192.98.101.10:5060 30728 udp receiver child=7 sock=192.98.101.10:5060 30729 udp receiver child=0 sock=127.0.0.1:5070 30730 udp receiver child=1 sock=127.0.0.1:5070 30731 udp receiver child=2 sock=127.0.0.1:5070 30732 udp receiver child=3 sock=127.0.0.1:5070 30733 udp receiver child=4 sock=127.0.0.1:5070 30734 udp receiver child=5 sock=127.0.0.1:5070 30735 udp receiver child=6 sock=127.0.0.1:5070 30736 udp receiver child=7 sock=127.0.0.1:5070 30738 udp receiver child=0 sock=127.0.0.1:5080 30740 udp receiver child=1 sock=127.0.0.1:5080 30752 udp receiver child=2 sock=127.0.0.1:5080 30754 udp receiver child=3 sock=127.0.0.1:5080 30755 udp receiver child=4 sock=127.0.0.1:5080 30756 udp receiver child=5 sock=127.0.0.1:5080 30758 udp receiver child=6 sock=127.0.0.1:5080 30760 udp receiver child=7 sock=127.0.0.1:5080 30761 slow timer 30763 timer 30765 ctl handler 30803 TIMER NH 30805 TIMER RT 30806 tcp receiver child=0 30808 tcp receiver child=1 30810 tcp receiver child=2 30812 tcp receiver child=3 30813 tcp receiver child=4 30844 tcp receiver child=5 30846 tcp receiver child=6 30847 tcp receiver child=7 30849 tcp main process why cannot one udp receiver process handle all listening addresses like tcp processes seem to be able to do? the above seems like waste of resources. -- juha

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Kamailio on a server with different network cards
by Rocío Ruz 05 Aug '11

05 Aug '11

Hello, My server has two different network cards with different IPs. I want kamailio to use both; the network card to be used depends on the destination. My problem is that I have to add an IP in the configuration file kamailio.cfg: listen=109.239.50.67 Will Kamailio listen only to packets sent to this IP? Can I add the two IPs corresponding to my two network cards here? Thank you RR

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480 Temporarily Unavailable. in kamailio 3.1.4
by hala alramli 04 Aug '11

04 Aug '11

i installed kamailio and freeswitch according to http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc i installed also rtpproxy . i managed to create 101,102,103 and all of them registered , i am not able to make calls between them, 480 temporarily unavailable always the error . i cant find in the configuration file what cause this error. this is the track. i will be thankful for any help . Hala interface: eth0 (192.168.1.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) match: 101 U 2011/08/04 13:16:57.351872 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.352037 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="192.168.1.233", nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX". Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.370381 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-6eabf9b3. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=b27e1a1d33761e85846fc98f5f3a7e58.7d11. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 101 ACK. Max-Forwards: 70. Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. . U 2011/08/04 13:16:57.380065 192.168.1.175:5060 -> 192.168.1.233:5060 INVITE sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. Expires: 240. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 444. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. Content-Type: application/sdp. . v=0. o=- 986176 986176 IN IP4 192.168.1.175. s=-. c=IN IP4 192.168.1.175. t=0 0. m=audio 16450 RTP/AVP 8 0 2 4 18 96 97 98 100 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=fmtp:100 192-193. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 2011/08/04 13:16:57.380329 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. Server: kamailio (3.1.4 (x86_64/linux)). Content-Length: 0. . U 2011/08/04 13:16:57.488511 192.168.1.233:5060 -> 192.168.1.175:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5d37bc9 2011-08-03 09-05-01 -0500. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "kb-102" <sip:kb-102@192.168.1.233>;party=calling;privacy=off;screen=no. . U 2011/08/04 13:16:57.502085 192.168.1.175:5060 -> 192.168.1.233:5060 ACK sip:102@192.168.1.233 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.175:5060;branch=z9hG4bK-eab8c613. From: <sip:101@192.168.1.233>;tag=43c22ff96586a71ao0. To: <sip:102@192.168.1.233>;tag=NDrNpDZ1g0jZe. Call-ID: 9acd1ca5-9162c25e(a)192.168.1.175. CSeq: 102 ACK. Max-Forwards: 70. Proxy-Authorization: Digest username="101",realm="192.168.1.233",nonce="TjqA1U46f6liNM8a6LX5HBJ2wXEBtOIX",uri="sip:102@192.168.1.233",algorithm=MD5,response="a019828b7a7334539ffa6b668dfcfada". Contact: <sip:101@192.168.1.175:5060>. User-Agent: Linksys/PAP2T-5.1.6(LS). Content-Length: 0. .

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How to reset t_on_failure/branch?
by Iñaki Baz Castillo 04 Aug '11

04 Aug '11

Hi, I remember that in Kamailio 1.5.X t_on_branch and t_on_failure still remain after rouning failure_route[XX], and I "think" there was a way to reset them. In 3.X it seems that t_on_branch and t_on_failure must be set again in failure_route to take effect in next t_relay() call, am I right? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

AVPs missing in onreply_route
by Sebastian Damm 04 Aug '11

04 Aug '11

Hi, I'm still fighting problems in our Kamailio 1.5 to 3.1 migration. Right now I'm stuck with a strange behavior of AVPs in my dialplan. I set a lot of AVPs for accounting in the route directive, and I just inserted an avp_print() statement right before t_relay() is called, all values are set. Then I call avp_print() as the first statement of onreply_route(), and it doesn't print out anything. I call it again at the end of onreply_route, and the only AVP printed out is the one I set directly there. In 1.5 there was an option in the tm module to control AVP behavior in replies, but that doesn't exist anymore, I even found a mailing list post from Daniel where he explained that all AVPs I set in route are also available in onreply_route. Am I doing something wrong? Is there still a secret option that I accidentally turned on? What else could make kamailio forget all AVPs after t_relay? I tried both version 3.1.3 and 3.1.4. Any help appreciated. Best regards, Sebastian

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ERROR : when create database in kamailio 3.1.0
by hala alramli 04 Aug '11

04 Aug '11

hello i a trying to instal kamailio and free switch according to the tutorial in http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc but when i try to create the database and the extra tables this error apear could you help me? thanks Hala root@noga1:~# nano /usr/local/etc/kamailio/kamctlrc root@noga1:~# /usr/local/sbin/kamdbctl create MySQL password for root: database engine 'mysql' loaded INFO: test server charset INFO: creating database kamailio ... Creating core table: standard Creating core table: acc Creating core table: lcr Creating core table: domain Creating core table: group Creating core table: permissions Creating core table: registrar Creating core table: usrloc Creating core table: msilo Creating core table: alias_db Creating core table: uri_db Creating core table: speeddial Creating core table: avpops Creating core table: auth_db Creating core table: pdt Creating core table: dialog Creating core table: dispatcher Creating core table: dialplan INFO: Core Kamailio tables succesfully created. Install presence related tables? (y/n): y INFO: creating presence tables into kamailio ... INFO: Presence tables succesfully created. Install tables for imc cpl siptrace domainpolicy carrierroute userblacklist htable purple uac pipelimit? (y/n): y INFO: creating extra tables into kamailio ... Creating extra table: imc Creating extra table: cpl Creating extra table: siptrace Creating extra table: domainpolicy Creating extra table: carrierroute Creating extra table: userblacklist Creating extra table: htable Creating extra table: purple Creating extra table: uac Creating extra table: pipelimit /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql: line 225: /usr/local/share/kamailio//mysql/pipelimit-create.sql: No such file or directory ERROR: Creating extra tables failed!

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Re: [SR-Users] Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, for questions about kamailio (openser), the best place to address them is still the mailing list for users. There are many people there that might be able to answer quickly. Messages sent personally to me for technical questions about kamailio (openser) are discarded, unless they were a result of a discussion on the mailing list where I asked for sensitive data. Cheers, Daniel On 8/3/11 1:47 AM, ifeanyi okoye wrote: > hi Daniel..i would like to discuss some of the issues am having with Openser/Kamailio > -----Original Message----- > Date: Tuesday, August 02, 2011 5:15:39 pm > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> > Cc: ifeanyi okoye<ifeanyi_okoye(a)yahoo.com> > From: "Daniel-Constantin Mierla"<miconda(a)gmail.com> > Subject: Re: [SR-Users] Chicago, ClueCon next week > > Hello Ify, > > On 8/2/11 11:15 PM, ifeanyi okoye wrote: >> Hello Daniel, >> >> My name is Ify and I'm a research voip engineer.I would be attending >> the conference. Please can I reach you via telephone. >> > the phone might not be the most convenient at that time, so the best is > to ask the organizers to point you to me, unless you can spot me based > on the pictures you can find on the web. I will have the presentation on > the 2nd day, but I will be around on 1st day as well. > > Looking forward to meeting you, > Daniel > >> Regards, >> Ify >> >> --- On *Tue, 8/2/11, Daniel-Constantin Mierla /<miconda(a)gmail.com>/* >> wrote: >> >> >> From: Daniel-Constantin Mierla<miconda(a)gmail.com> >> Subject: [SR-Users] Chicago, ClueCon next week >> To: "kamailio users"<sr-users(a)lists.sip-router.org>, >> business(a)lists.kamailio.org >> Date: Tuesday, August 2, 2011, 1:59 AM >> >> Hello, >> >> I'll be in Chicago few days next week for ClueCon conference. If >> you are in the area, but no -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Chicago, ClueCon next week
by Daniel-Constantin Mierla 03 Aug '11

03 Aug '11

Hello, I'll be in Chicago few days next week for ClueCon conference. If you are in the area, but not attending the ClueCon conference, drop me a message in case you want to meet. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kat http://linkedin.com/in/miconda -- http://twitter.com/miconda

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