- sr-users - kamailio.org

Kamailio codecs
by Burak ÖZTÜRK 14 Sep '12

14 Sep '12

Does kamailio supports g729 codec? Or which codecs does kamailio supports? Where or how to learn the supported or unsupported codecs for kamailio? Thanks in advance...

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

That's a rather dated study. But it's better than a nonexistent reference point, true. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Mino Haluz <mino.haluz(a)gmail.com> wrote:According to this (http://transnexus.com/index.php/performance-test-results-for-openser-and-rt…) "For a server hosting both OpenSER and RTPproxy, each 1 GHz of CPU processing capacity can manage a maximum of 325 simultaneous calls." I have 2.4GHz for rtpproxy, but CPU/Mem/network is ok, so the bottleneck should be somewhere else probably.. On Thu, Sep 13, 2012 at 5:56 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > I'm not sure what a single instance of rtpproxy can handle, but most people > squeezing thousand of concurrent calls per box are probably doing it on > multicore boxes by binding multiple instances of rtpproxy with different > core affinities, and round-robining among them. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > 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/ > > Mino Haluz <mino.haluz(a)gmail.com> wrote: > The results: > > - rtpproxy calls count 280 > - sipp calls count 2000 > - iptraf on proxy 4.8MB/s > - G711a codec > > So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), > rtpproxy calls count is really the right value. CPU usage is ok on > every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy > cannot serve more than 270-280 calls ? > > On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: >> Ok, so I put there unforce_rtp_proxy even though I'm using >> rtpproxy_manage. The tip with nc now really shows the calls count. >> >> But the dialog count is still higher and higher, so I have bug >> somewhere in the configuration. I'll check it. >> >> On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov >> <abalashov(a)evaristesys.com> wrote: >>> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE >>> or >>> CANCEL. It'll just figure out what to do on its own. >>> >>> None of this has to do with dialog state, though. Just rtpproxy control. >>> >>> >>> >>> >>> -- Alex >>> >>> -- >>> Sent from my Samsung mobile, and thus lacking in the refinement one might >>> expect from a proper keyboard. >>> >>> 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/ >>> >>> Mino Haluz <mino.haluz(a)gmail.com> wrote: >>> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >>> >>> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> >>> wrote: >>>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>>> >>>>> Peter: Thanks for the tip! Really interesting. But I do not >>>>> understand, why also this list contains the calls that were ended by >>>>> sipp... Should I search for some mistake in my kamaillio config ? >>>> >>>> Perhaps you don't close them with unforce_rtp_proxy: >>>> >>>> if(method=="BYE" || method=="CANCEL"){ >>>> unforce_rtp_proxy(); >>>> } >>>> >>>> -- >>>> With best regards, Peter Lemenkov. >>>> >>>> _______________________________________________ >>>> 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 >>> >>> _______________________________________________ >>> 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 >>> >>> _______________________________________________ >>> 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 >>> _______________________________________________ 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

I'm not sure what a single instance of rtpproxy can handle, but most people squeezing thousand of concurrent calls per box are probably doing it on multicore boxes by binding multiple instances of rtpproxy with different core affinities, and round-robining among them. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Mino Haluz <mino.haluz(a)gmail.com> wrote:The results: - rtpproxy calls count 280 - sipp calls count 2000 - iptraf on proxy 4.8MB/s - G711a codec So if my calculations are right (16kB/s per stream * 280 = 4.5MB/s), rtpproxy calls count is really the right value. CPU usage is ok on every machine (rtpproxy 20-30% CPU). Does anybody know why rtpproxy cannot serve more than 270-280 calls ? On Thu, Sep 13, 2012 at 5:07 PM, Mino Haluz <mino.haluz(a)gmail.com> wrote: > Ok, so I put there unforce_rtp_proxy even though I'm using > rtpproxy_manage. The tip with nc now really shows the calls count. > > But the dialog count is still higher and higher, so I have bug > somewhere in the configuration. I'll check it. > > On Thu, Sep 13, 2012 at 4:53 PM, Alex Balashov > <abalashov(a)evaristesys.com> wrote: >> Correct, but you still need to call rtpproxy_manage() on receipt of a BYE or >> CANCEL. It'll just figure out what to do on its own. >> >> None of this has to do with dialog state, though. Just rtpproxy control. >> >> >> >> >> -- Alex >> >> -- >> Sent from my Samsung mobile, and thus lacking in the refinement one might >> expect from a proper keyboard. >> >> 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/ >> >> Mino Haluz <mino.haluz(a)gmail.com> wrote: >> I'm using rtpproxy_manage, so I assume unforce_rtp is not needed. >> >> On Thu, Sep 13, 2012 at 4:10 PM, Peter Lemenkov <lemenkov(a)gmail.com> wrote: >>> 2012/9/13 Mino Haluz <mino.haluz(a)gmail.com>: >>> >>>> Peter: Thanks for the tip! Really interesting. But I do not >>>> understand, why also this list contains the calls that were ended by >>>> sipp... Should I search for some mistake in my kamaillio config ? >>> >>> Perhaps you don't close them with unforce_rtp_proxy: >>> >>> if(method=="BYE" || method=="CANCEL"){ >>> unforce_rtp_proxy(); >>> } >>> >>> -- >>> With best regards, Peter Lemenkov. >>> >>> _______________________________________________ >>> 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 >> >> _______________________________________________ >> 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 >> >> _______________________________________________ >> 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Really? Interesting, I had no idea. I thought the rtpproxy control protocol was binary and did not lend itself easily to interaction in this manner. Thanks for the tip. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > You can't get it from rtpproxy. You'd really have to use something like the > dialog or htable modules to keep call state and get that from Kamailio. On the contrary it's possible (using raw UDP reads/writes): work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 sessions created: 0 active sessions: 0 active streams: 0 work ~: Where * h1u203u03 is randomly chosen token, * 127.0.0.1 is the rtpproxy's control IP, * 22222 is the rtpproxy's control port, * "-u" means that we're using UDP * -w 1 is the timeout in seconds to wait before closing nc. I can't imagine that someone will use nc in performance testing but I think it looks like a good start. -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

Thanks for this info. Netcat is a really underappreciated tool. I can't count how many times I've unnecessarily written utilities to send stuff on UDP sockets, forgetting that nc can do it. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Peter Lemenkov <lemenkov(a)gmail.com> wrote:2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. Yep, this command protocol allows us to do tricks like this which is quite helpful for debugging. Unfortunately this particular command ( 'I' and its counterpart 'Ib' ) seems to be broken - it doesn't prepend received random cookie at the beginning of the answer. Just FYI - I collected some examples of control commands here (just grep for ser_proto:decode): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_proto_test.erl and here (which could be even more interesting since it contains an example of a real session - create session, lookup, destroy, get stats, etc): * https://raw.github.com/lemenkov/erlrtpproxy/master/test/ser_test.erl > Thanks for the tip. Anytime :) -- With best regards, Peter Lemenkov.

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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

If the dialog module is working correctly, and the SIP flows are standards-compliant, the dialog module should automatically track all subsequent state changes and remove calls from tracking after a BYE is processed. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Mino Haluz <mino.haluz(a)gmail.com> wrote:Ok, I'm tagging dialogs with dlg_manage(), but even if the call ends, it still keeps info about this dialog in list "kamctl fifo dlg_list". Should I somehow close the dialog when the BYE transaction is ended ? Peter: Thanks for the tip! Really interesting. But I do not understand, why also this list contains the calls that were ended by sipp... Should I search for some mistake in my kamaillio config ? On Thu, Sep 13, 2012 at 3:57 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Really? Interesting, I had no idea. I thought the rtpproxy control protocol > was binary and did not lend itself easily to interaction in this manner. > Thanks for the tip. > > > > > -- Alex > > -- > Sent from my Samsung mobile, and thus lacking in the refinement one might > expect from a proper keyboard. > > 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/ > > Peter Lemenkov <lemenkov(a)gmail.com> wrote: > 2012/9/13 Alex Balashov <abalashov(a)evaristesys.com>: >> You can't get it from rtpproxy. You'd really have to use something like >> the >> dialog or htable modules to keep call state and get that from Kamailio. > > On the contrary it's possible (using raw UDP reads/writes): > > work ~: echo "h1u203u03 I\n" | nc -w 1 -u 127.0.0.1 22222 > sessions created: 0 > active sessions: 0 > active streams: 0 > work ~: > > Where > > * h1u203u03 is randomly chosen token, > * 127.0.0.1 is the rtpproxy's control IP, > * 22222 is the rtpproxy's control port, > * "-u" means that we're using UDP > * -w 1 is the timeout in seconds to wait before closing nc. > > I can't imagine that someone will use nc in performance testing but I > think it looks like a good start. > > -- > With best regards, Peter Lemenkov. > > _______________________________________________ > 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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Re: [SR-Users] RTPProxy with kamailio : How to get calls count?
by Alex Balashov 13 Sep '12

13 Sep '12

You can't get it from rtpproxy. You'd really have to use something like the dialog or htable modules to keep call state and get that from Kamailio. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Mino Haluz <mino.haluz(a)gmail.com> wrote:Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino _______________________________________________ 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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RTPProxy with kamailio : How to get calls count?
by Mino Haluz 13 Sep '12

13 Sep '12

Hi, I'm doing the performance test with kamailio + RTPProxy, but I would like to get the real calls count that the rtpproxy is serving. I don't want to use value that I get from sipp. So is there any management tool for rtpproxy, or should I get it somewhere in kamailio config ? Thanks, Mino

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script stops processing when relaying e2e ack?
by Andrew Mortensen 12 Sep '12

12 Sep '12

Hi all. I'm working on a custom module and would like to be able to send UPDATEs to the call legs after the script forwards the e2e ACK from the caller. I had hoped to export a function to call in my ACK handling route after the script forwards the ACK. However, the script stops processing after calling t_relay in my ACK handling route. I tracked this back to the fact that t_relay_to invokes forward_request when handling e2e ACKs, and returns the return code of forward_request to the script. Since forward_request returns 0 on success, the t_relay call in the script returns 0, which stops script processing. Is this intentional? My module currently hooks the TMCB_E2EACK_IN, and I can UPDATE the caller here, but the callee seems to be ignoring the UPDATE if it's received prior to the ACK. Thanks for any advice you can offer. andrew

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Kamailio 3.2, NAT, rtpproxy bridging between interfaces
by Russell McConnachie 12 Sep '12

12 Sep '12

Hello, I'm having some issues getting media bridging working in kamailio and rtpproxy. I've attempted to attach as much relevant information as possible. Thanks in advance for taking a look; I'm completely stuck right now. Actual results:No audio is being passed through the rtpproxy from caller "A" to caller "B" even though both media server and UAC are sending media to the rtpproxy. Expected results:Audio that is sent to rtpproxy is bridged/proxied across from caller "A" to caller "B" and vice-versa. ------------------------------------------------------------------------------- Signaling trace: ------------------------------------------------------------------------------- 08:34:57.543224 IP (tos 0x0, ttl 49, id 3237, offset 0, flags [none], proto: UDP (17), length: 1456) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 1428 INVITE sip:9991@r18.test-bwks.local:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Route: <sip:x.x.x.102:5060;lr> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@192.168.2.250:51620;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 619 v=0 o=MxSIP 0 0 IN IP4 192.168.2.250 s=SIP Call c=IN IP4 192.168.2.250 t=0 0 m=audio 51720 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:on - - - - a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=ptime:30 a=sendrecv 08:34:57.545766 IP (tos 0x10, ttl 64, id 58136, offset 0, flags [none], proto: UDP (17), length: 439) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 411 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280;received=x.x.x.199 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Server: kamailio (3.2.4 (x86_64/linux)) Content-Length: 0 08:34:57.547619 IP (tos 0x10, ttl 64, id 47936, offset 0, flags [+], proto: UDP (17), length: 1500) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 1472 INVITE sip:9991@10.0.0.29 SIP/2.0 Record-Route: <sip:10.0.0.38;r2=on;lr=on;nat=yes> Record-Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes> Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0 Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060> Call-ID: b2af8b83e20b0740 CSeq: 29663 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "99999999" <sip:99999999@x.x.x.199:31280;transport=udp;srcadr=192.168.2.250:5060>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D2DF3BB>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 57i/2.6.0.1007 Content-Type: application/sdp Content-Length: 629 v=0 o=MxSIP 0 0 IN IP4 10.0.0.38 s=SIP Call c=IN IP4 10.0.0.38 t=0 0 m=audio 29682 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:106 BV16/8000 a=rtpmap:107 BV32/16000 a=rtpmap:113 L16/16000 a=rtpmap:110 PCMU/16000 a=rtpmap:111 PCMA/16000 a=rtpmap:112 L16/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:115 G726-32/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event 08:34:57.551126 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 409) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 381 SIP/2.0 100 Trying Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060> Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Content-Length:0 08:34:57.561084 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 723) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 695 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.562404 IP (tos 0x10, ttl 64, id 58137, offset 0, flags [none], proto: UDP (17), length: 672) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 644 SIP/2.0 180 Ringing Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Content-Length:0 08:34:57.609771 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 953) 10.0.0.29.55620 > 10.0.0.38.sip: SIP, length: 925 SIP/2.0 200 OK Via:SIP/2.0/UDP 10.0.0.38;branch=z9hG4bK4ed5.ec1e41a.0,SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:133 v=0 o=BroadWorks 196 1 IN IP4 10.0.0.24 s=- c=IN IP4 10.0.0.24 t=0 0 m=audio 10498 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 08:34:57.612047 IP (tos 0x10, ttl 64, id 58138, offset 0, flags [none], proto: UDP (17), length: 930) x.x.x.102.sip > x.x.x.199.31280: SIP, length: 902 SIP/2.0 200 OK Via:SIP/2.0/UDP 192.168.2.250:51620;received=x.x.x.199;branch=z9hG4bK1158b9f850c008ff7.6835b1eb6e3bffee5;rport=31280 From:"99999999"<sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To:"9991"<sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID:b2af8b83e20b0740 CSeq:29663 INVITE Supported: Contact:<sip:10.0.0.29:5060> Record-Route:<sip:10.0.0.38;r2=on;lr=on;nat=yes>,<sip:x.x.x.102;r2=on;lr=on;nat=yes> P-Asserted-Identity:"Auto Attendant"<sip:9991@10.0.0.29;user=phone> Privacy:none Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Content-Type:application/sdp Content-Length:161 v=0 o=BroadWorks 196 1 IN IP4 x.x.x.102 s=- c=IN IP4 x.x.x.102 t=0 0 m=audio 29774 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sdpmangled:yes 08:34:57.844645 IP (tos 0x0, ttl 49, id 3239, offset 0, flags [none], proto: UDP (17), length: 506) x.x.x.199.31280 > x.x.x.102.sip: SIP, length: 478 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.250:51620;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Route: <sip:x.x.x.102;r2=on;lr=on;nat=yes>, <sip:10.0.0.38;r2=on;lr=on;nat=yes> Max-Forwards: 70 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 08:34:57.845182 IP (tos 0x10, ttl 64, id 47937, offset 0, flags [none], proto: UDP (17), length: 509) 10.0.0.38.sip > 10.0.0.29.sip: SIP, length: 481 ACK sip:10.0.0.29:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.38;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 192.168.2.250:51620;rport=31280;received=x.x.x.199;branch=z9hG4bK15c3c5e900b3f6d36.26b037b92d0bba7c9 Max-Forwards: 69 From: "99999999" <sip:99999999@r18.test-bwks.local:5060>;tag=1cf0b02fab To: "9991" <sip:9991@r18.test-bwks.local:5060>;tag=1693866893-1347366897367 Call-ID: b2af8b83e20b0740 CSeq: 29663 ACK User-Agent: Aastra 57i/2.6.0.1007 Content-Length: 0 ------------------------------------------------------------------------------- The call session is up at this point. CPE Router, Source NAT IP x.x.x.199, Internal Network: 192.168.2.0/24 192.168.2.250 is an Aastra 6757i IP Phone, behind the x.x.x.102 is the external interface on our SIP proxy allowing traffic on: Inbound: port 5060/udp Inbound: range 16384 - 32768/udp 10.0.0.38 is the internal interface facing our BroadWorks lab cluster 10.0.0.29 is the internal interface of our BroadWorks Application Server 10.0.0.24 is the internal interface of our BroadWorks Media Server RTP Proxy has been spawned with the following command line arguments: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid -u rtpproxy \ -u root \ -s udp:localhost:7722 \ -l x.x.x.102/10.0.0.38 \ -d DBUG LOG_LOCAL5 \ -m 16384 -M 32768 We've changed the /etc/kamailio/kamailio.cfg and added the #!define's at the top of the configuration file: #!define WITH_NAT #!define WITH_DEBUG In the route[NATMANAGE] { } block we've added a few additional lines to handle the setup of the rtpproxy_manage() function which handles the rtpproxy_offer(), rtpproxy_answer() and unforce_rtp_proxy() convientenly in one function. The logic we've got in place currently is: if ((src_ip =~ "10\.0\.0\.*") && (dst_ip =~ "10\.0\.0\.*")) { xlog("L_INFO", "Media session for ($ci) with source address ($si): Internal > External: SDP contact/origin will be x.x.x.102"); rtpproxy_manage("FAIEOC", "x.x.x.102"); } else if (!(src_ip =~ "10\.0\.0\.*") && (dst_ip == x.x.x.102)) { xlog("L_INFO", "Media session for ($ci) with source address ($si): External > Internal: SDP contact/origin will be 10.0.0.38"); rtpproxy_manage("FAEIOC", "10.0.0.38"); } else { xlog("L_INFO", "!!! Uncaught condition for media session for ($ci) with source address ($si) !!!"); } This rtpproxy log is not from the call above, but the same source #, destination # were used: Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session 84bf2a7cb9d4ab24, tag dcfa6a860e;1 requested, type strong Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: new session on a port 28282 created, tag dcfa6a860e;1 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling caller's address with 192.168.2.250:51720 Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: lookup on ports 28282/22500, session timer restarted Sep 11 08:50:27 sip-proxy1 rtpproxy[3101]: INFO:handle_command: pre-filling callee's address with 10.0.0.24:10502 ... Call is currently active and then hung-up ... Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:process_rtp: session timeout Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0 relayed, 0 dropped Sep 11 08:51:28 sip-proxy1 rtpproxy[3101]: INFO:remove_session: session on ports 28282/22500 is cleaned up Example of the RTP: 08:53:13.555834 IP (tos 0x0, ttl 48, id 510, offset 0, flags [none], proto: UDP (17), length: 200) x.x.x.199.32052 > x.x.x.102.16552: UDP, length 172 x.x.x.199:32052 > x.x.x.102:16552 Why is this packet not being forwarded to the media proxy (rtpproxy) to 10.0.0.24 (specified in the 200 OK from the BroadWorks Application Server) 08:53:13.562206 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) 10.0.0.24.10504 > 10.0.0.38.17968: UDP, length 172 10.0.0.24:10504 > 10.0.0.38:17968 Why is this packet not being forwarded from rtpproxy to the UAC at x.x.x.199? ------------------------------------------------------------------------------- System details: ------------------------------------------------------------------------------- OS: CentOS 5.8 x86-64, Hypervisor is qemu-kvm Memory: 2GB Disk: 40G VCPU: 1 /etc/sysctl.conf: # Controls IP packet forwarding net.ipv4.ip_forward = 1 ------------------------------------------------------------------------------- Current software versions: ------------------------------------------------------------------------------- [root@sip-proxy1 kamailio]# rpm -qa | grep -E 'kamailio|rtpproxy' | xargs rpm -qi Name : rtpproxy Relocations: (not relocatable) Version : 1.2.1 Vendor: Fedora Project Release : 2.el5 Build Date: Sun 19 Sep 2010 08:12:23 AM EDT Install Date: Mon 10 Sep 2012 09:23:28 PM EDT Build Host: x86-10.phx2.fedoraproject.org Group : Applications/Internet Source RPM: rtpproxy-1.2.1-2.el5.src.rpm Size : 105682 License: BSD Signature : DSA/SHA1, Mon 20 Sep 2010 12:40:46 PM EDT, Key ID 119cc036217521f6 Packager : Fedora Project URL : http://www.rtpproxy.org Summary : A symmetric RTP proxy Description : This is symmetric RTP proxy designed to be used in conjunction with the SIP Express Router (SER) or any other SIP proxy capable of rewriting SDP bodies in SIP messages that it processes. Name : kamailio Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:32 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 10200784 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : Kamailio, very fast, reliable and flexible SIP Server Description : Kamailio is a very fast, reliable and flexible SIP (RFC3261) proxy server. Written entirely in C, kamailio can handle thousands calls per second even on low-budget hardware. A C Shell like scripting language provides full control over the server's behaviour. It's modular architecture allows only required functionality to be loaded. Among available features: IPv4, IPv6, digest authentication, accounting, CPL scripts, instant messaging, MySQL and UNIXODBC support, SIMPLE presence agent, radius authentication, record routing, SMS gateway, ENUM, UDP, TCP, TLS and SCTP, transaction and dialog module, OSP module, statistics support, registrar and user location, SNMP, SIMPLE Presence and Perl programming interface. Name : kamailio-cpl Relocations: (not relocatable) Version : 3.2.4 Vendor: kamailio.org Release : 1.1 Build Date: Fri 03 Aug 2012 05:14:14 AM EDT Install Date: Tue 11 Sep 2012 08:22:33 AM EDT Build Host: build08 Group : Productivity/Telephony/SIP/Servers Source RPM: kamailio-3.2.4-1.1.src.rpm Size : 296947 License: GPL Signature : DSA/SHA1, Fri 03 Aug 2012 05:14:38 AM EDT, Key ID c9a75909941fdbdd Packager : Daniel-Constantin Mierla <miconda(a)gmail.com> URL : http://kamailio.org/ Summary : CPL module (CPL interpreter engine) for Kamailio Description : The kamailio-cpl package provides a CPL interpreter engine for Kamailio

2 1

Max-Forwards: and loops. Testing purpose.
by Anton Kvashenkin 12 Sep '12

12 Sep '12

Hi, List. I'm using sipsak to fire some predefined sip message with Max-Forwards: header equal 0 to test mf_process_maxfwd_header function. I'm doing this exclusively for learning purpose. As far as we know, this exported function from maxfwd module services only for adding Max-Forwards header, decrement or detect Max-Forwards: 0 header to prevent loops. So, my question guys, how can I create this loop in my lab, with two proxies for example? What message should I craft?

2 1

using kamailio with clients in a nat environment
by David Thomson 12 Sep '12

12 Sep '12

hi, I was wondering if in clients in a nat environment need to be in the dmz or need to have ports forwarded for kamailio to be used to connect the clients? I have setup kamailio with the nat settings but am still only getting one way audio. chances are it has not been setup correctly but just checking. If i try to connect the two clients from the same network i can get zrtp and 2 way communication working correctly. if one of the clients is on a different network the call initiates and answers but zrtp and 2way audio does not work. kamailio 3.3 running on ec2 ttyl,Dave

2 4

Re: [SR-Users] Jurisdictional (inter vs intra calling) LCR on Kamailio
by Alex Balashov 11 Sep '12

11 Sep '12

William, This is a business-layer function that is far outside the scope of what built-in Kamailio modules can do. In fact, even the 'lcr' module is something of a misnomer: it does not do LCR in a truly financialised sense, to say nothing of jurisdictionally specific regulatory awareness. It's just a basic weighted routing table. The appropriate place to find such functionality, unless you want to write it yourself, is in a variety of ITSP-oriented commercial platforms built on top of Kamailio. They exist precisely to fill these gaps and bring the somewhat general capabilities of Kamailio into an applied dimension that is useful in specific business and product contexts. Many members of this list represent software or consulting companies that offer such solutions. However, as it is a noncommercial list, I'll refrain from plugging them directly. You may want to ask the question on Kamailio-Business. You can also email me privately and I would be happy to give you some unbiased pointers about the commercial ecosystem. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/William Fulton <wfulton(a)thirdhatch.com> wrote:I can’t find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don’t know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

Jurisdictional (inter vs intra calling) LCR on Kamailio
by William Fulton 11 Sep '12

11 Sep '12

I can't find any information anywhere on how to get jurisdictional lcr configured. I have intra and interstate rates for each carrier rate deck and don't know how to let the system know when one is appropriate versus the other. Can someone point me to a tutorial or some documentation on this? Thanks so much! William

1 0

rls notify delay
by Juha Heinanen 11 Sep '12

11 Sep '12

one rls thing that stilll wonders me is the about 2-5 sec delay that occurs when backend notifies have been received and notify to subscriber ua is sent: Sep 11 17:27:41 siika /usr/sbin/sip-proxy[4495]: INFO: Routing PUBLISH <sip:foo@test.fi> to presence server Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Handling PUBLISH <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: Routing locally generated NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> to <sip:jh@test.fi> from <sip:foo@test.fi> Sep 11 17:27:41 siika /usr/sbin/pres-serv[19270]: INFO: presence [notify.c:1575]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:foo@test.fi for event presence Sep 11 17:27:41 siika /usr/sbin/pres-serv[19271]: INFO: Handling in-dialog NOTIFY <sip:rls@127.0.0.1:5082;transport=tcp> from <sip:foo@test.fi> to <sip:jh@test.fi> Sep 11 17:27:46 siika /usr/sbin/pres-serv[19265]: INFO: Routing locally generated NOTIFY <sip:cyvrphbz@192.98.103.1:60122;alias=192.98.103.1~47419~2;transport=tcp> to <sip:jh@test.fi> from <sip:jh-buddies@test.fi> Sep 11 17:27:46 siika /usr/sbin/sip-proxy[4482]: INFO: Routing in-dialog NOTIFY <sip:cyvrphbz@192.98.103.1:60122;transport=tcp> from <sip:jh-buddies@test.fi> to <sip:192.98.103.1:47419;transport=tcp> this does not depend on the pua patch nor which db_mode i have in rls module. -- juha

2 2

rls does not generally work
by Juha Heinanen 11 Sep '12

11 Sep '12

it turned out that presence rls module does not work unless pua module is configured to use database mode and database is postgre. the issue is serious, since besides not working, it makes presence/rls server not to accept any new requests. since there is no easy cure, should i go and write a warning in rls readme file so that others don't stumble on this? -- juha

2 6

siptrace and homer
by Krishna Kurapati 11 Sep '12

11 Sep '12

Hi, We have been using homer for some time. The logging is stopping once the disk is full, which make sense. However, it would nice if either kamailio or homer has an option to auto-truncate the siptrace database tables once they grow large. Is there such a feature? Regards. Krishn Kura

2 1

Продукция из Украины
by andrei＠iptel.org 11 Sep '12

11 Sep '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: ðÒÉÏÂÒÅÔÅÎÉÅ, ÔÁÍÏÖÅÎÎÏÅ ÏÆÏÒÍÌÅÎÉÅ É ÄÏÓÔÁ×ËÁ ÌÀÂÏÊ ÉÎÔÅÒÅÓÕÀÝÅÊ ÷ÁÓ ÐÒÏÄÕËÃÉÉ ÉÚ õËÒÁÉÎÙ. éÍÅÅÍ ÍÎÏÇÏÌÅÔÎÉÊ ÏÐÙÔ ÜËÓÐÏÒÔÎÙÈ ÏÐÅÒÁÃÉÊ. ôÅÒÒÉÔÏÒÉÁÌØÎÏ ÎÁÈÏÄÉÍÓÑ × ÓÁÍÏÍ ÃÅÎÔÒÅ õËÒÁÉÎÙ. éÍÅÅÍ ×ÓÅ ÎÅÏÂÈÏÄÉÍÙÅ ÜËÓÐÏÒÔÎÙÅ ÒÁÚÒÅÛÅÎÉÑ É ÌÉÃÅÎÚÉÉ. ó 1998 ÇÏÄÁ Ñ×ÌÑÅÍÓÑ ÄÅÊÓÔ×ÕÀÝÉÍÉ ÞÌÅÎÁÍÉ ôÏÒÇÏ×Ï ðÒÏÍÙÛÌÅÎÎÏÊ ðÁÌÁÔÙ õËÒÁÉÎÙ. [...] Content analysis details: (16.9 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.0 FSL_HELO_NON_FQDN_1 FSL_HELO_NON_FQDN_1 3.8 HELO_LOCALHOST HELO_LOCALHOST 2.8 FUZZY_XPILL BODY: Attempt to obfuscate words in spam 3.0 BAYES_95 BODY: Bayes spam probability is 95 to 99% [score: 0.9893] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [123.27.115.159 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?123.27.115.159>] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [123.27.115.159 listed in bb.barracudacentral.org] 0.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [123.27.115.159 listed in dnsbl.sorbs.net] 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Re: [SR-Users] MTREE reload question
by Daniel-Constantin Mierla 11 Sep '12

11 Sep '12

Hello, the used size is pretty much the same after the reload, so doesn't look as leak. If you do 2-3 reloads is shmem:used_size staying around same value? It is clear boost in the fragments, I would say that after start the number of fragments is quite low, because it should contain the good records in memory as well. Btw, the version 3.3.x has options for defragmentation, see the core cookbook, you can enable it. Cheers, Daniel On 9/9/12 10:18 AM, Uri Shacked wrote: > Hi, > here is the statistics after kamailio starts: > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > here it is after reload of the num table: > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > here is the number of raws from the DB: > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > and attached is the memory log. > thanks, > Uri > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > is this taken only after startup? Get one at startup and another > one after reload, so it can be compared. > > Cheers, > Daniel > > > On 9/6/12 9:22 AM, Uri Shacked wrote: >> Hi, >> here: >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> >> On 9/5/12 3:06 PM, Uri Shacked wrote: >> >> Hi, >> I use MTREE to load 5 million rows from the database. it >> takes about 30 sec to start kamailio and it is running great. >> Whern I check the shmem usage I see the data take about >> 0.8G out of 4G i set on shmem. >> When i reload the data while kamailio is running, the >> memory usage rises to 1.3G and stays there (the second >> and next reloads stays on 1.3G as well). >> Why doesn't it return to 0.8G after the reload is completed? >> >> >> can you send the statistics related to shmem? >> >> kamctl fifo get_statistics shmem: >> >> There are different values there, some usage come from >> overhead of memory chunks management. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda >> <http://twitter.com/#%21/miconda> - >> http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >> http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> >> > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 8

Memory allocation
by David 11 Sep '12

11 Sep '12

Hello, I originally asked this on IRC, but realized that this is probably better suited for the mailing list : If I set memory usage to 100mb ( -m 100 ) and set 100 children ( children=100 in kamailio.cfg ). Will Kamailio use 100 mb or 10, 000 mb ( 10 gb ) of memory ? Is the memory limit per child or global for kamailio ? Is there an MI command I can send to check what kamailio's health ? ( load, free memory, etc... ) ? Thanks, David

2 3

Rtpproxy re-INVITE handling
by Spencer Thomason 11 Sep '12

11 Sep '12

Hello, I'm using Kamailio 3.2.4 for NAT traversal using rtpproxy_manage() in a largely stock script. Everything works great until the far end (on a public ip) sends a t.38 re-INVITE. The 200OK from the NATed UAC then doesn't trigger rtpproxy and the private IP in the sdp causes the fax to fail. Any help handling these re-INVITEs would be greatly appreciated. I'm happy to post traces if that helps describe the situation. The network topology looks like this: NATed UAC -> Kamailio on a public IP -> UAS on a public IP Thanks in advance, Spencer Connected by DROID on Verizon Wireless

2 12

pua 'Could not convert temporary dialog into a dialog' error
by Juha Heinanen 10 Sep '12

10 Sep '12

when testing presence/rls, i'm getting a pua error when presence server handles subscribe: Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Handling SUBSCRIBE <sip:test@test.fi> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: Routing locally generated NOTIFY to <sip:rls@127.0.0.1:5082;transport=tcp> Sep 8 20:36:12 siika /usr/sbin/pres-serv[22932]: INFO: presence [notify.c:1581]: NOTIFY sip:jh@test.fi via sip:rls@127.0.0.1:5082;transport=tcp on behalf of sip:test@test.fi for event presence Sep 8 20:36:12 siika /usr/sbin/pres-serv[22931]: INFO: <core> [tcp_main.c:2787]: quick connect for 0x7f7114cfa490 Sep 8 20:36:12 siika /usr/sbin/pres-serv[22933]: ERROR: pua [send_subscribe.c:689]: Could not convert temporary dialog into a dialog what does pua error message 'Could not convert temporary dialog into a dialog' mean? when does it happen? -- juha

2 9

Example required for XMLRPC module.
by SamyGo 10 Sep '12

10 Sep '12

Hello, I recently posted email relating to using MI_XMLRPC for kamailio monitoring. As Daniel suggested and as mentioned in module docs Mi-xmlrpc is getting obsolete and don't compile on newer OSs. So I need to use XMLRPC module for pulling statistics from kamailio like get_statistics all: Please can anyone point me to some help other than module doc. I need a simple example script to pull data from kamailio and print it. Any help or pointers will be greatly appreciated. Thanks, Sammy G.

1 0

event_route[dialog:failed] and Warning "script writer didn't realese transaction"
by Uri Shacked 10 Sep '12

10 Sep '12

Hi, I started using testing the event_route[dialog:failed]. On my scenario, I receive an INVITE, use t_newtran(), check some stuff and send a failure reply like 3XX or 4XX. I use send_reply() and can see that the TM sends the reply. Now, if I use event_route[dialog:failed], I keep on getting the warning "script writer didn't release transaction". If I delete the event_route[dialog:failed] from the script, the warning does not appear. My solution was to work stateless, which is OK for this scenario. But, still, What does this event_route has that shows this warning? Why doesn't the transaction being released? (I tried to use t_reply as well and still the same). Thanks, Uri

2 1

Re: [SR-Users] MTREE reload question
by Uri Shacked 09 Sep '12

09 Sep '12

> > Hi, > > here is the statistics after kamailio starts: > > shmem:fragments = 28 > shmem:free_size = 3800871312 > shmem:max_used_size = 494132368 > shmem:real_used_size = 494095984 > shmem:total_size = 4294967296 > shmem:used_size = 342642072 > > here it is after reload of the num table: > > shmem:fragments = 9161531 > shmem:free_size = 3654274496 > shmem:max_used_size = 959885552 > shmem:real_used_size = 640692800 > shmem:total_size = 4294967296 > shmem:used_size = 342654552 > here is the cfg part for mtree: > > #------- mtree params ------------- > modparam("mtree", "db_url", CFGDB) > modparam("mtree", "mtree", "name=odr;dbtable=service_odr_view;type=0;") > modparam("mtree", "mtree", "name=oper;dbtable=service_oper_type;type=0;") > modparam("mtree", "mtree", > "name=permis;dbtable=service_permisions_to_oper;type=0;") > modparam("mtree", "mtree", > "name=num;dbtable=service_numbers_to_areas_view;type=0;") > modparam("mtree", "char_list", "0123456789") > modparam("mtree", "pv_value", "$avp(mtval)") > modparam("mtree", "pv_values", "$avp(mtvals)") > > here is the number of raws from the DB: > > SELECT count(*) FROM `service`.`service_numbers_to_areas_view` = 4195528 > > i have the memory log in a file if needed > > thanks, > Uri > > > On Fri, Sep 7, 2012 at 10:26 AM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> is this taken only after startup? Get one at startup and another one >> after reload, so it can be compared. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 9:22 AM, Uri Shacked wrote: >> >> Hi, >> >> here: >> >> shmem:fragments = 143898 >> shmem:freesize = 3446570952 >> shmem:max_used_size = 861854768 >> shmem:real_used_size = 848396344 >> shmem:total_size = 4294967296 >> shmem:used_size = 319676976 >> >> >> >> On Thu, Sep 6, 2012 at 9:49 AM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 9/5/12 3:06 PM, Uri Shacked wrote: >>> >>>> Hi, >>>> I use MTREE to load 5 million rows from the database. it takes about 30 >>>> sec to start kamailio and it is running great. >>>> Whern I check the shmem usage I see the data take about 0.8G out of 4G >>>> i set on shmem. >>>> When i reload the data while kamailio is running, the memory usage >>>> rises to 1.3G and stays there (the second and next reloads stays on 1.3G as >>>> well). >>>> Why doesn't it return to 0.8G after the reload is completed? >>>> >>> >>> can you send the statistics related to shmem? >>> >>> kamctl fifo get_statistics shmem: >>> >>> There are different values there, some usage come from overhead of >>> memory chunks management. >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.com >>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - >>> http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >>> http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> > >

1 0

Possible bug in authentication
by martian＠centrum.sk 07 Sep '12

07 Sep '12

Hello to everybody. I am currently working with Kamailio 3.3.1 on RedHat. The "loose_route" function was not working correctly and I observed some very strange behaviour (not as one described in the documentation of the function). I have found that there needs to be a port included in the "alias" variable for the loose_route function to work correctly. However, upon adding the port to alias, the INVITE messages were no longer authenticated (Kamailio just accepted them and didn't send proxy-auth header in 407 message). My alias: alias="domain.ch:5060" Examining default routing logic, I found the problem here: if (is_method("REGISTER") || from_uri==myself) { # authenticate requests ... } The "from_uri==myself" was no longer evaluated as true, because there was a port at the end of the alias. The FROM Header of the INVITE messages looks like: From: "acc1" ;tag=12345 ..so .. no port number there. Btw, I have fixed this with replacing the "myself" list with my own defined variable MY_DOMAIN. #!define MY_DOMAIN ".*(a)domain.ch" So now the condition looks like this: if (is_method("REGISTER") || from_uri=~MY_DOMAIN) { ... } I am not sure if this is a bug that needs to be fixed or not. I am just pointing my finger at it and I hope it will contribute to the development. Also, a valid description of this behavior (when using port in alias) would be appreciated. Cheers, Martin

3 8

Wrong index on expires column in pua module
by Juha Heinanen 07 Sep '12

07 Sep '12

looks like unique constraint on expires_idx has not been removed yet, because i still see Duplicate entry for key 'expires_idx' errors. i'll go and make the changes. -- juha

1 0

FW: Kamailio as sipRedirect
by Fatima Chahrour~Vanrise Support 07 Sep '12

07 Sep '12

Dears, I have an installed 3CX where I want to generate a call from it to the Kamailo to get a list of suppliers to terminate the call to. Thus, the Kamailio should act as a sip redirect to the 3CX (for routing decision). Have you tried such configuration before? And how do I configure the 3CX on the Kamailio? Regards, F Charour

3 12

kamctlrc SQL definitions and db/schema/entities.xml
by Vieri 07 Sep '12

07 Sep '12

Hi, What is the purpose of the "SQL definitions" settings found in kamctlrc? It is also stated that one should edit db/schema/entities.xml but I can't seem to find this file. kamctlrc contains: # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2020-05-28 21:32:15" # DEFAULT_ALIASES_EXPIRES=$FOREVER # DEFAULT_Q="1.0" # DEFAULT_CALLID="Default-Call-ID" # DEFAULT_CSEQ="13" # DEFAULT_LOCATION_EXPIRES=$FOREVER Thanks, Vieri

2 5

siptrace module with Kamailio 3.3.0 and Siremis 3.2
by Chris Telford 07 Sep '12

07 Sep '12

Hi, Were attempting to get siptrace working on Kamailio 3.3.0 with Siremis 3.2... We've enabled the module, and configured the basic parameters, however were having problems with where exactly in Kamailio's config to call siptrace(). We currently have it at the very end of FAILURE_ROUTE, ONREPLY_ROUTE, BRANCH_ROUTE as recommended at the end, and it is logging traces sucessfully. The problem were having, is it is only producing traces with inbound sip messages only. It is also duplicating almost every message (so were clearly call siptrace() too many times per request). Is there a guide on this? or any sample config files to help us speed this process up a little? Regards, Chris Telford

1 0

Re: [SR-Users] sr-users Digest, Vol 88, Issue 19
by Vijay Thakur 07 Sep '12

07 Sep '12

On Friday 07 September 2012 02:34 PM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Monitoring Kamailio using mi_xmlrpc good or bad ? > (Daniel-Constantin Mierla) > 2. Re: Video Call Quality (Tayeb Meftah) > 3. Re: FW: Kamailio as sipRedirect (Fatima Chahrour~Vanrise Support) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 07 Sep 2012 10:36:50 +0200 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > Subject: Re: [SR-Users] Monitoring Kamailio using mi_xmlrpc good or > bad ? > To: SamyGo <govoiper(a)gmail.com> > Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <5049B222.6080405(a)gmail.com> > Content-Type: text/plain; charset="iso-8859-1"; Format="flowed" > > Hello, > > On 9/7/12 10:21 AM, SamyGo wrote: >> Thanks alot for your time and valuable feedback on the topic. I just >> wanted to be sure that when if comes to collecting stats like 150+ values > did you defined custom stats? IIRC, we don't have so many in the code. > >> from Kamailio per 30 seconds will it impact on the actual performance >> of Kamailio as sip server. I will definitely look for XMLRPC only as >> well if thats scalable. > Stats are just counters, long values, access to them should be very > fast. mi_xmlrpc is a dedicated process, not handling sip at all. > > Cheers, > Daniel >> Thanks >> Sammy >> >> >> >> On Fri, Sep 7, 2012 at 12:35 PM, Daniel-Constantin Mierla >> <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: >> >> Hello, >> >> if you fetch the statistics, should be ok. >> >> xmlrpc module should be more scalable that mi_xmlrpc, but uses a >> different command format and output. >> >> Cheers, >> Daniel >> >> >> On 9/6/12 3:24 PM, SamyGo wrote: >>> Hello, >>> >>> I've just finished monitoring all the core modules of kamailio >>> via XML-RPC. I use a script placed on kamailio server which sends >>> command to Kamailio over MI_XMLRPC and fetch details of each >>> requested module. The results fetched are then sent back to the >>> monitoring tool for graphing using SNMP OIDs. >>> >>> *My question* here is that if I query kamailio like this for like >>> 7~10 module over xmlrpc per 30/45 seconds how much overhead will >>> it cause to the Kamailio server !? Is this a good approach or bad ? >>> >>> >>> Thanks, >>> Sammy >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu >> >> Thanks, Tayeb for your reply. What can be checked at client end. I am using Kamailio 3.2. With thanks in advance. Vijay Thakur

1 0

Video Call Quality
by Vijay Thakur 07 Sep '12

07 Sep '12

Dear All Experts, Recently, i have configure my first kamailio server. After some technical hurdles, it is working very fine. There is a issue related to video call. When i make a video call from jitsi to jitsi, iphone to iphone or android, the image is blur, choppy. After googling for this issue, i have found no satisfactory answer. Kindly suggest me the reasons for this issue and what action can be taken. Thanks in advance , Vijay Tha

2 1

Monitoring Kamailio using mi_xmlrpc good or bad ?
by SamyGo 07 Sep '12

07 Sep '12

Hello, I've just finished monitoring all the core modules of kamailio via XML-RPC. I use a script placed on kamailio server which sends command to Kamailio over MI_XMLRPC and fetch details of each requested module. The results fetched are then sent back to the monitoring tool for graphing using SNMP OIDs. *My question* here is that if I query kamailio like this for like 7~10 module over xmlrpc per 30/45 seconds how much overhead will it cause to the Kamailio server !? Is this a good approach or bad ? Thanks, Sammy

2 3

[ ... ] operator
by Vitaliy Aleksandrov 07 Sep '12

07 Sep '12

"Migrating kamailio v1.5.x to v.3.0.0" says that "test" operator no longer exists. But it is mentioned in Core Cookbook 3.2 (and 3.3) and can confuse someone. Can somebody remove it from the documentation ?

2 3

MTREE reload question
by Uri Shacked 07 Sep '12

07 Sep '12

Hi, I use MTREE to load 5 million rows from the database. it takes about 30 sec to start kamailio and it is running great. Whern I check the shmem usage I see the data take about 0.8G out of 4G i set on shmem. When i reload the data while kamailio is running, the memory usage rises to 1.3G and stays there (the second and next reloads stays on 1.3G as well). Why doesn't it return to 0.8G after the reload is completed? BR, Uri

2 3

User accounting
by Ali Jawad 06 Sep '12

06 Sep '12

Hi Is it possible to store user logon/logout dates + user agent used. I know that the info is in the location table but it is not persistent, and I am using the accounting table to store calls. But I am not sure there is a way to store logon/logout/user agent. Any input please ? Regards

2 8

Siremis v3.3.0 Released
by Elena-Ramona Modroiu 06 Sep '12

06 Sep '12

Hi, Siremis v3.3.0 was released - the web management interface compatible with Kamailio v3.3.x. This version is an update of v3.2.1 to be able to work with the database schema of Kamailio v3.3.x. The news of the release, including links to installation and screenshots, is available at: - http://siremis.asipto.com/2012/09/06/siremis-v3-3-0-released/ Regards, Ramona

1 0

Meeting in London
by Carsten Bock 06 Sep '12

06 Sep '12

Hi, anyone from the Kamailio-Community in for a beer tomorrow evening in London? I have several meetings during the day (last one at 5PM) and i'll be travelling back to Germany on Saturday, so i'm free for some chat about Kamailio, SIP, IMS or IPv6 during the evening... :-) If yes, just drop me a mail... Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 34927219 Fax +49 40 34927220 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

Информация для бухгалтера
by jan＠iptel.org 06 Sep '12

06 Sep '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: òÁÂÏÔÁÅÔÅ ÂÕÈÇÁÌÔÅÒÏÍ? ðÏÍÎÉÔÅ, ËÁË ÓÌÏÖÎÏ ×ÁÍ ÒÁÎØÛÅ ÂÙÌÏ ÎÁÊÔÉ ÏÂÕÞÅÎÉÅ ÐÏ ÎÕÖÎÏÊ ÔÅÍÅ ôÅÐÅÒØ, ÎÁ ÓÁÊÔÅ ÎÁÛÅÇÏ ÕÞÅÂÎÏÇÏ ÃÅÎÔÒÁ ÎÁ ÜÔÏ ÕÊÄÅÔ ×ÓÅÇÏ ÎÅÓËÏÌØËÏ ÍÉÎÕÔ! é ÎÅ ×ÁÖÎÏ, × ÞÅÍ ×ÁÍ ÎÕÖÎÏ ÒÁÚÏÂÒÁÔØÓÑ - × îäó, ÒÁÓÞÅÔÅ ÏÔÐÕÓËÏ×, ÏÐÌÁÔÅ ÔÒÕÄÁ, ÎÁÌÏÇÏÏÂÌÏÖÅÎÉÉ, ÓÔÒÁÈÏ×ÙÈ ×ÚÎÏÓÁÈ, ÐÏÄÇÏÔÏ×ÉÔØÓÑ Ë ×ÏÚÍÏÖÎÙÍ ÐÒÏ×ÅÒËÁÍ ÉÌÉ ÒÁÚÏÂÒÁÔØÓÑ Ó ÄÏËÕÍÅÎÔÏÏÂÏÒÏÔÏÍ. úÁÈÏÄÉÔÅ ÎÁ ÎÁÛ ÓÁÊÔ - ÕÞÉÔØÓÑ ÓÔÁÌÏ ÇÏÒÁÚÄÏ ÐÒÏÝÅ. é ÍÙ ×ÓÅÇÄÁ ÐÏÍÏÖÅÍ, ÒÁÓÓËÁÖÅÍ, ÏÂßÑÓÎÉÍ É ÐÏÄÂÅÒ£Í ÔÏ, ÞÔÏ ×ÁÍ ÐÏÄÏÊÄ£Ô! [...] Content analysis details: (9.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 0.9981] -0.0 SPF_PASS SPF: sender matches SPF record 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [2.146.222.10 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [2.146.222.10 listed in bb.barracudacentral.org] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Problem with dumping offline messages
by Stoyan Mihaylov 06 Sep '12

06 Sep '12

Following examples, I tried to add processing of offline messages. Structure of my system is: SIP client <->Kamailio<->Asterisk<->Kamailio<->SIP client Where Kamailio in above is same server, but SIP clients are different. Now I have no problems with audio, video calls and messages, if both clients are online. All messages received from client are sent to Asterisk, where I can or not process them and then I forward back to Kamailio. When messages came from Asterisk server, then they are sent directly to end user. Sending messages to end user - I call this function, only if message come from allow_source_address(): route[MSILO_MESSAGE]{ if (lookup("location")){ xlog(" Relying"); t_relay(); }else{ xlog("Storing MSILO_MESSAGE $ru-$tu-$ou"); m_store("$ru"); t_relay(); t_reply("202", "Accepted"); }; exit(); } When client register: route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location",0x04)) sl_reply_error(); # MSILO - dumping user's offline messages if (m_dump("$fu")){ xlog("MSILO: offline messages for $fu dumped - if they were \n"); }else{ xlog("MSILO: no offline messages dumped\n"); }; exit; } } I am sending message to offline contact, and I see record in silo table. Record looks perfect. Then offline contact become online, I see in debug MSILO: offline messages for {offlined contact} dumped - if they were But client do not receive nothing. In case all contacts are online, then message go to contact, and is not stored in silo. Setup of msilio is: loadmodule "msilo.so" modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MY DOMAIN") modparam("msilo","contact_hdr","Contact: registrar@MYDOMAIN:5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","***$rU***$rb***") By the way - I receive in Asterisk offline message. Obviously, I am doing something wrong, but I cant understand what. Any help is welcome. Version of Kamailio is: #~> kamailio -V version: kamailio 3.2.3 (x86_64/linux) 59f87e flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 59f87e compiled on 14:43:47 Jul 19 2012 with gcc 4.5.2

2 1

frequency of dispatcher probing
by Juha Heinanen 05 Sep '12

05 Sep '12

i have defined: modparam("dispatcher", "flags", 2) modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshhold", 3) modparam("dispatcher", "ds_probing_mode", 2) and still i see that dispatcher is pinging inactive node at least once per second: U 2012/09/05 13:47:04.143977 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:04.644185 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:05.143991 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.143975 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... U 2012/09/05 13:47:06.144056 192.98.103.10:5060 -> 192.98.103.3:5070 OPTIONS sip:192.98.103.3:5070 SIP/2.0. ... what is that i'm missing in order to make ping interval 10 seconds? -- juha

2 2

Re: [SR-Users] memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

> The 20% initialy loaded is because I load many data with MTREE, > HTABLE,DIALPLAN etc....(5 million rows with MTREE..) > Regarding the PKG, how do i set it? I read a few months ago that the > shared and private memory on version 3.2.2 are the same. > Thanks for the help :-) > > On 09/05/2012 11:44 AM, Uri Shacked wrote: > >* Hi, *>* Thanks, *>* How do i find memory leaks? *>* when my > application starts the it shows the following values: *>* shmem:fragments > = 143898 *>* shmem:freesize = 3446570952 *>* shmem:max_used_size = > 861854768 *>* shmem:real_used_size = 848396344 *>* shmem:total_size = > 4294967296 *>* shmem:used_size = 319676976 *>* I divided the > "real_used_size" with the "total_size" and notice that *>* it is 20%. So, > this is my initial value of the application. *>* After 2 months with > 40,000 calls per day, the relations between these *>* values is 39%. > (doing a reload of the data that occupies 20% failed *>* due to "no > memory") *>* So, 19% occupation of memory is normal for this amount of > calls? or is *>* it a memory leak? if it is a memeory leak, how do i see > where? *Hello, > > Too little information to go on here. A reload might fail mostly because > data from the DB (e.g. with carrierroute case) is first loaded into > private memory (the sql result). Depending of the module there are other > tricks you could do (decrease the number of fetch rows the query returns > etc.). The out of memory condition affects shared or private mem. Try to > use at least 15 MB for PKG mem. > > If the 39% is with traffic, then it's normal, if you don't have any > traffic I still would not worry. The thing that worries me a little is > the value, you are using from the start around 0.8 GB of used shared > memory... this is a lot. From the percentages you provided it looks ok.... > > Marius > >* Uri *>* On 09/05/2012 10:19 AM, Uri Shacked wrote: *>* >/Hi, />/When I > start kamailio it loads some data into memory. />/Now *>* calls are > starting to gather up. How can I check how mach memory *>* />/does my > avp's take? />/BR, />/Uri /Hello Uri, *>* *>* avp's are associated with > the transactions, so they free'd when the *>* transaction stops(and calls > are composed of transactions). This is true *>* for dialog information as > well (dlg_var's etc.). Continuous growth in *>* memory at a constant call > rate (incoming sip requests) might mean a *>* leak. If you need > statistics about the total memory usage you can use *>* `kamctl fifo > get_statistics all` and check for the shmem real_used_size *>* and > max_used_size fields. *>* *>* Cheers, *>* Marius *>* *>* -- *>* Zbihlei > Marius *>* *>* Head of *>* Linux Development Services Romania *>* * > > -- > Zbihlei Marius > > Head of > Linux Development Services Romania > > 1&1 Internet Development srl Tel KA: 754-9152 > Str Mircea Eliade 18 Tel RO: +40-31-223-9152 > Sect 1, Bucuresti mailto: marius.zbihlei at 1and1.ro<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users> > 71295, Romania > >

1 0

memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, Thanks, How do i find memory leaks? when my application starts the it shows the following values: shmem:fragments = 143898 shmem:freesize = 3446570952 shmem:max_used_size = 861854768 shmem:real_used_size = 848396344 shmem:total_size = 4294967296 shmem:used_size = 319676976 I divided the "real_used_size" with the "total_size" and notice that it is 20%. So, this is my initial value of the application. After 2 months with 40,000 calls per day, the relations between these values is 39%. (doing a reload of the data that occupies 20% failed due to "no memory") So, 19% occupation of memory is normal for this amount of calls? or is it a memory leak? if it is a memeory leak, how do i see where? Uri On 09/05/2012 10:19 AM, Uri Shacked wrote: >* Hi, *>* When I start kamailio it loads some data into memory. *>* Now calls are starting to gather up. How can I check how mach memory *>* does my avp's take? *>* BR, *>* Uri *Hello Uri, avp's are associated with the transactions, so they free'd when the transaction stops(and calls are composed of transactions). This is true for dialog information as well (dlg_var's etc.). Continuous growth in memory at a constant call rate (incoming sip requests) might mean a leak. If you need statistics about the total memory usage you can use `kamctl fifo get_statistics all` and check for the shmem real_used_size and max_used_size fields. Cheers, Marius -- Zbihlei Marius Head of Linux Development Services Romania

2 1

memory usage
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, When I start kamailio it loads some data into memory. Now calls are starting to gather up. How can I check how mach memory does my avp's take? BR, Uri

2 1

event_route[dialog:failure]
by Uri Shacked 05 Sep '12

05 Sep '12

Hi, i seems to me that the "event_route[dialog:failure]" does not recognize dialog failures. i use dlg_manage when i recieve the INVITE. i get 301, 302, 402 or send_reply with some +300 responses. on the manage_failure i print the responses, on the event_route[dialog:start] and [dialog:end] i get good results as well. but, the dialog:failure] does not rise.... any ideas? BR, Uri

2 2

LUA Authentication.
by David | StyleFlare 05 Sep '12

05 Sep '12

I think I saw once an example from miconda using LUA for Auth? Does anyone else remember seeing that? I wanted to do a custom Auth using LUA. Thanks in advance for any pointers. David.

4 3

Re: [SR-Users] t_check_status not found in readmes
by Alex Balashov 05 Sep '12

05 Sep '12

It is true. It disappeared from the docs after the 3.0 sip-router convergence. It is present in 'tm' docs in <= 1.5.x, though. The current docs do have t_grep_status(), which seems in many respects identical. I wonder if one is an alias for another now. -- 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/, http://www.alexbalashov.com Juha Heinanen <jh(a)tutpro.com> wrote: >looks like t_check_status is not documented in any readme file. > >-- juha > >_______________________________________________ >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

2 1

t_check_status not found in readmes
by Juha Heinanen 05 Sep '12

05 Sep '12

looks like t_check_status is not documented in any readme file. -- juha

1 0

Cassandra module - query question
by Boudewyn Ligthart 03 Sep '12

03 Sep '12

Hi, I would to query the Cassandra database for other information with either CQL or other commands. Could this be done through means of either avpops or sqlops, like an MySQL database. This could be done through an external script but I would like to utilize the connection that is already established. Thanks, Bou

3 4

Why append_branch should be called in a failure route ?
by Vitaliy Aleksandrov 03 Sep '12

03 Sep '12

Below is the example from http://kamailio.org/docs/modules/3.2.x/modules/tm.htm : failure_route[0]{ if (t_check_status("5[0-9][0-9]")){ # I do not like the 5xx responses, # so I give another chance to "foobar.com", # and I drop all the replies to make sure that # they are not forwarded to the caller. t_drop_replies(); rewritehostport("foobar.com"); append_branch(); t_relay(); } } Why do append_branch() is present before t_relay(). I'm asking because I didn't used it in failure routes and call rerouting on various conditions worked great.

1 0

Catch error "status rewrite by UAS" in on_reply
by Alex Hermann 03 Sep '12

03 Sep '12

Hello all, i sometimes get a very late reply from a broken client which triggers the following error 9because the call was already continued in another branch): ERROR: tm [t_reply.c:1193]: ERROR: t_should_relay_response: status rewrite by UAS: stored: 408, received: 200 Not a problem perse, but i would like to catch this error in the script's on_reply route because I need to prevent some actions on the reply if it will not be forwarded to the UAC. Is there a way to do this? -- Greetings, Alex Hermann

1 0

Celebrating 11 Years
by Daniel-Constantin Mierla 03 Sep '12

03 Sep '12

Hello, eleven years ago, the first piece of code of the project was committed. Last year we organized a public event, for the 10 years celebration, presentations and more details about it at: * http://sip-router.org/10-years-ser/ For the 11th year, I made a news with the summary of the past year - definitively it was lot of work, adding many new features: * http://www.kamailio.org/w/2012/09/kamailio-at-11-years/ Looking forward to the next year celebration, surely it will list another great set of new features! Many thanks to everyone helping the project, from advocating to development! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

2 1

kamailio and clustered database
by Vieri 01 Sep '12

01 Sep '12

Hi, I'm new to Kamailio and would like to set it up with a "circular replication database" on several servers (master-to-master). Before running 'kamdbctl create' I edited kamctlrc and set DBENGINE to MYSQL (I can use either MySQL or PostgreSQL or both - I just need to use a cluster of servers). What would you suggest? Should I set DBENGINE=MYSQL and configure a cluster of databases as described here: http://dev.mysql.com/doc/refman/5.1/en/mysql-cluster-replication-multi-mast… Or does kamailio provide another way to do the same job? What does the db_cluster module do and where/how should I use it? Thanks, Vieri

1 0

weird dp_translate result
by Juha Heinanen 01 Sep '12

01 Sep '12

i have this piece of config code: modparam("dialplan", "attrs_pvar", "$var(plan)") ... xlog("L_INFO", "Translating <$var(calling_number)> based on dpid 4\n"); if (dp_translate("4", "$var(calling_number)")) { xlog("L_INFO", "Found attrs <$var(plan)>\n"); ... dpid 4 has three entries: 109 4 10 1 ^\+1 0 na 107 4 10 1 ^\+358 0 fi 108 4 10 1 ^\+46 0 se thus in each entry attrs field has some two letter value. when the above piece of code is executed, i get to syslog Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Translating <+35892345670> based on dpid 4 Aug 31 19:06:51 wheezy2 /usr/sbin/sip-proxy[7162]: INFO: Found attrs <0> if i restart the proxy, attrs value may change to some other bogus value. any ideas? -- juha

2 10

OpenIMSCore icscf problem
by Siavash Safi 01 Sep '12

01 Sep '12

Hi, I fetched the latest source code from OpenIMSCore SVN repository and generated the debian packages on squeeze. Everything is working fine so far except the icscf module which fails to start. I think that there is something wrong with the configuration. Here is the output messages: root@spawn:~# icscf.sh Listening on udp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 tcp: xxx.xxx.xxx.xxx [xxx.xxx.xxx.xxx]:5060 Aliases: tcp: spawn:5060 udp: spawn:5060 *: ims.mydomain.com:* *: icscf.ims.mydomain.com:* ------ Semaphore Arrays -------- key semid owner perms nsems root@spawn:~# 0(4597) ERROR: mk_proxy: could not resolve hostname: "" 0(4597) ERROR: fixup_hostport2proxy: bad host name in URI <> ERROR: error -478 while trying to fix configuration 0(4597) INFO:destroy_diameter_peer(): Bye Bye from C Diameter Peer test As you can see mk_proxy is called with a empty hostname! I have pasted my configuration and more verbose debug output: configuration: http://pastebin.com/nDQM1KLM debug output: http://pastebin.com/d6jh0CHF Any help would be appreciated. -- Regards, Siavash Safi Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

1 0

Kamailio - Nonce validity between Register and Invite
by patrice.bodeven＠orange.com 31 Aug '12

31 Aug '12

Hello, I am working on Kamailio 3.2.2. There is no traffic, only functional test done. Based on the SIP Client used until now (Xlite), the INVITE is systematically authenticated by 407 as there is no Proxy-Authorization in the initial INVITE. Expected/normal behavior. But when using an internal SIP client, the initial INVITE is generated with a Proxy-Authorization containing the nonce value used by the successful registration. In Authorization of Register (the answer is 200OK) Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:ims.v0.pftest.net", response="4ed51b8e62cf769b19f8c4771e068ad3", cnonce="06440000000529330fe1", qop=auth, nc=00000011 In Proxy-Authorization of Invite Digest username="+33296488922(a)ims.v0.pftest.net", realm="sip:ims.v0.pftest.net", nonce="UDP43lAzpH7SjicT6+9/KDDloW4OTfTXoGWlZurBVut0JV604jox/QY+tVeA", uri="sip:0157361149@ims.v0.pftest.net", response="b247052d6d5b37e0b6677c816390cb77", cnonce="06440000000529330fe1", qop=auth, nc=00000012 As the nonce expire is high (see nonce_expire), I don't understand why Kamailio requests the Authentication of the INVITE by 407. 407 with Proxy-Authenticate: Digest realm="ims.v0.pftest.net", nonce="UDP6pVAzpkWDF+jSlP3zzVYcuNPhY8NQBj62WIA=", qop="auth", stale=true Does it mean that the REGISTER and INVITE are using different Nonce value ? ==> Information on my configuration file. modparam("auth_db", "db_url", DBURLAUTH) modparam("auth", "auth_checks_register", 6) # callid and From TAG modparam("auth", "qop", "auth") # enable qop=auth modparam("auth", "nonce_count", 1) # enable nonce_count support modparam("auth", "nonce_expire", 21600) # Set nonce_expire to 6 hours For register, I do if (!www_authorize("$ru", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { www_challenge("$ru", "17"); exit; } } For invite, I do if (!proxy_authorize("$fd", "subscriber")) { if($? == -2 || $? == -3) { sl_send_reply("403","Forbidden auth ID"); exit; } else { proxy_challenge("$fd", "17"); xlog("L_INFO","INVITE challenged by 407 to :$di:$dp \n"); exit; } } Followed by consume_credentials(); Thanks in advance for your support. Patrice B _________________________________________________________________________________________________________________________ Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration, France Telecom - Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci. This message and its attachments may contain confidential or privileged information that may be protected by law; they should not be distributed, used or copied without authorisation. If you have received this email in error, please notify the sender and delete this message and its attachments. As emails may be altered, France Telecom - Orange is not liable for messages that have been modified, changed or falsified. Thank you.

3 9

invite/ack avps/transactions
by Juha Heinanen 31 Aug '12

31 Aug '12

i tried to access an avp that was set during invite processing while processing ack to 200 ok, but i guess the avp is already gone by that time, because positive ack is its own transaction? what i tried to do is to save somewhere invite's $du and then t_relay ack to the same place without calling loose_route. should i go for htable as storage or is there a better choice? -- juha

2 1

LCR (next_gw) question
by Nord7 31 Aug '12

31 Aug '12

Hello everyone. Configures the module LCR. I can't get him to take the next on the list of priorities gateway if the first does not work. Function "next_gw" capable of doing that? Did anyone realties that? If possible show example. Kamailio 3.3 Regards, Kirill

2 1

spam messages breaking through filters
by Daniel-Constantin Mierla 31 Aug '12

31 Aug '12

Hello, there were some spam messages going through our mailing list filters. I was updating the filters on mailman, but there are new and new ideas coming from this guys. Sorry for the pollution and I wanted to say that these messages are marked as spam by spamassasin, you can filter them locally as well looking for the header X-Spam-Level: *** You can adjust the number of '*'. Some email services do that automatically, like gmail.com for example. Of course, if anyone has better ideas of protection, please write here. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 1

db_cluster together with the registrar module = signal 11
by Øyvind Kolbu 30 Aug '12

30 Aug '12

Hi, I upgraded our system to Kamailio 3.3 to take advantage of the new db_cluster module. Currently we have two registration/location servers each with their own sql-server. Authenticated REGISTER messages are forwarded to the other server, which uses the 0x02 flag to save() in order to store it without generating an extra reply. This works fine. My goal was to use db_cluster to update both databases at once with save() and use the failover mode when reading with lookup(). Have anyone tested a similar setup? Anyhow, here is how I tried to do it: modparam("db_cluster","connection","voip1_data2=>postgres://user:pass@hostname1/voipt1_data2") modparam("db_cluster","connection","voip2_data2=>postgres://user:pass@hostname2/voipt2_data2") modparam("db_cluster", "cluster", "data2=>voip1_data2=8s7p;voip2_data2=9s7p") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "cluster://data2") But I get: : : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 33 : ALERT: <core> [main.c:785]: child process 28275 exited by a signal 11 : ALERT: <core> [main.c:788]: core was generated : INFO: <core> [main.c:800]: INFO: terminating due to SIGCHLD Same with db_mode = 1 or 3, and either modparam("db_cluster", "cluster", "data2=>voip1_data2=8s9p") or modparam("db_cluster", "cluster", "data2=>voip2_data2=8s9p"). No difference between newly booted phones or phones reregistering. Backtrace from gdb: (gdb) bt #0 0x002fd7da in db_cluster_insert (_h=0xb7d2dc88, _k=0xbf9ac954, _v=0xbf9ac7ec, _n=18) at dbcl_api.c:334 #1 0x00fbb1cc in db_insert_ucontact (_c=0xb5d828d8) at ucontact.c:565 #2 0x00fcfc5d in insert_ucontact (_r=0xfd3a00, _contact=0xb7d3ce90, _ci=0xc5cc80, _c=0xbf9acb84) at urecord.c:478 #3 0x00c55bb0 in add_contacts (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:493 #4 save (_m=0xb7d05e40, _d=0xb5d63060, _cflags=0, _uri=0x0) at save.c:864 #5 0x00c4dd43 in w_save2 (_m=0xb7d05e40, _d=0xb5d63060 "00ֵ", _cflags=0x0) at reg_mod.c:407 #6 0x080586ad in do_action (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1157 #7 0x0805fe5c in run_actions (h=0xbf9acf48, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1644 #8 0x080604e1 in run_actions_safe (h=0xbf9adcb8, a=0xb7c405c0, msg=0xb7d05e40) at action.c:1708 #9 0x080f96b7 in rval_get_int (h=0xbf9adcb8, msg=0xbf9ac954, i=0xbf9ad268, rv=0xb7c406cc, cache=0x0) at rvalue.c:920 #10 0x080fcf38 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad268, rve=0xb7c406c8) at rvalue.c:1914 #11 0x080fd009 in rval_expr_eval_int (h=0xbf9adcb8, msg=0xb7d05e40, res=0xbf9ad544, rve=0xb7c40a70) at rvalue.c:1922 #12 0x080585fe in do_action (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1121 #13 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41550, msg=0xb7d05e40) at action.c:1644 #14 0x08058657 in do_action (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1136 #15 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7c41650, msg=0xb7d05e40) at action.c:1644 #16 0x0805853d in do_action (h=0xbf9adcb8, a=0xb7bd64d8, msg=0xb7d05e40) at action.c:761 #17 0x0805fe5c in run_actions (h=0xbf9adcb8, a=0xb7bc7128, msg=0xb7d05e40) at action.c:1644 #18 0x08060464 in run_top_route (a=0xb7bc7128, msg=0xb7d05e40, c=0x0) at action.c:1729 #19 0x080e11e4 in receive_msg (buf=0x82c4f20 "REGISTER sip:hometest.voip.uio.no SIP/2.0\r\nVia: SIP/2.0/UDP 193.157.149.56;branch=z9hG4bKb8e88f2eE34DCDDD\r\nFrom: \"kolbutest\" <sip:2549619@hometest.voip.uio.no>;tag=34CE9BA6-70FCC135\r\nTo: <sip:2549619@"..., len=809, rcv_info=0xbf9ade98) at receive.c:209 #20 0x08177f7b in udp_rcv_loop () at udp_server.c:544 #21 0x080b0426 in main_loop () at main.c:1633 #22 0x080b3d02 in main (argc=17, argv=0xbf9ae174) at main.c:2546 Core dumps and debug logs (debug=3) are available. -- Øyvind Kolbu University of Oslo

3 21

Job ad - see details! Sent through Search engine
by serusers＠iptel.org 30 Aug '12

30 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (6.7 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [189.4.159.143 listed in bb.barracudacentral.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [189.4.159.143 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [189.4.159.143 listed in bl.score.senderscore.com] 1.3 RCVD_IN_BL_SPAMCOP_NET RBL: Received via a relay in bl.spamcop.net [Blocked - see <http://www.spamcop.net/bl.shtml?189.4.159.143>] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Kamailio Crash using Cassandra 1.1.2 during SIP unregister
by Boudewyn Ligthart 29 Aug '12

29 Aug '12

Hi, Kamailio crash using Cassandra 1.1.2 during SIP unregister (registering works fine) The version of Kamailio is the latest GIT version: version: kamailio 3.4.0-dev3 (x86_64/linux) 10327c flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 10327c compiled on 14:53:11 Aug 22 2012 with gcc 4.6.3 BT of Core : Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `kamailio -ddddd'. Program terminated with signal 11, Segmentation fault. #0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527 527 if(unlikely(f->u.nxt_free!=NULL)) { (gdb) bt #0 fm_free (qm=0x7fa55daf7010, p=0xd22317971716b6a) at mem/f_malloc.c:527 #1 0x00007fa55d35ac53 in db_free_columns (_r=0x7fa55db38c30) at db_res.c:86 #2 0x00007fa55d83f87d in db_cassa_query (_h=<optimized out>, _k=0x7fa55adc6aac, _op=<optimized out>, _v=<optimized out>, _c=0x7fff72ef0e30, _n=<optimized out>, _nc=16, _o=0x7fa55afca990, _r=0x7fff72ef0f10) at dbcassa_base.cpp:659 #3 0x00007fa55adb58eb in db_load_urecord (_c=0x7fa55db218d0, _d=0x7fa5576eee38, _aor=0x7fff72ef11b0) at udomain.c:592 #4 0x00007fa55adb93cd in get_urecord (_d=<optimized out>, _aor=0x7fff72ef11b0, _r=0x7fff72ef1200) at udomain.c:1017 #5 0x00007fa55ab91698 in lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38, _uri=<optimized out>) at lookup.c:156 #6 0x00007fa55ab957f2 in w_lookup (_m=0x7fa55db40c48, _d=0x7fa5576eee38 "\350\355nW\245\177", _uri=<optimized out>) at reg_mod.c:437 #7 0x000000000041c624 in do_action (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1153 #8 0x000000000041b493 in run_actions (h=0x7fff72ef17f0, a=0x7fa55db23098, msg=0x7fa55db40c48) at action.c:1646 #9 0x0000000000423a24 in run_actions_safe (h=<optimized out>, a=<optimized out>, msg=<optimized out>) at action.c:1710 #10 0x00000000004d79a0 in rval_get_int (h=0x7fff72ef2790, msg=<optimized out>, i=0x7fff72ef1c30, rv=<optimized out>, cache=<optimized out>) at rvalue.c:920 #11 0x00000000004dc82c in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef1c30, rve=0x7fa55db23218) at rvalue.c:1914 #12 0x00000000004dc8f9 in rval_expr_eval_int (h=0x7fff72ef2790, msg=0x7fa55db40c48, res=0x7fff72ef20d0, rve=0x7fa55db238e8) at rvalue.c:1922 #13 0x000000000041cb66 in do_action (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1123 #14 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db26d38, msg=0x7fa55db40c48) at action.c:1646 #15 0x000000000041c7e1 in do_action (h=0x7fff72ef2790, a=0x7fa55db15b38, msg=0x7fa55db40c48) at action.c:763 #16 0x000000000041b493 in run_actions (h=0x7fff72ef2790, a=0x7fa55db118c8, msg=0x7fa55db40c48) at action.c:1646 #17 0x0000000000423ab0 in run_top_route (a=0x7fa55db118c8, msg=0x7fa55db40c48, c=<optimized out>) at action.c:1731 #18 0x00000000004b39ee in receive_msg (buf=<optimized out>, len=<optimized out>, rcv_info=<optimized out>) at receive.c:209 #19 0x000000000054de1a in udp_rcv_loop () at udp_server.c:544 #20 0x0000000000478879 in main_loop () at main.c:1635 #21 0x000000000041b002 in main (argc=<optimized out>, argv=<optimized out>) at main.c:2548 Thanks, Bou

2 2

[OT] sip:provider CE v2.6 is out!
by Andreas Granig 29 Aug '12

29 Aug '12

Hi all, I'm excited to announce the release of the Sipwise sip:provider Community Edition v2.6, a free and open source turn-key platform, which uses Kamailio, Sems and Asterisk in its core to provide a full-blown and feature-rich VoIP soft-switch. http://www.sipwise.com/news/announcements/spce-v2_6-release/ New core features are *full presence/xcap/im support* together with the awesome Jitsi client, allowing you to deploy your own Skype-like service for encryted voice/video communication, buddy lists, instant messaging, screen sharing and remote desktop control. We also support *media features for error announcements*, like busy, blocked, unknown number etc. Check our VM images to get started within a couple of minutes and learn what can be built with the power of Kamailio as a stateless load-balancer and a stateful proxy/registrar, and Sems as an SBC and application server. http://www.sipwise.com/products/spce/quickinstall/ Thanks a lot to all developers, contributors and members of this mailing list to make this possible and for helping us out! Andreas

1 0

Work offer inside
by serusers＠iptel.org 29 Aug '12

29 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: I would like to take this time to welcome you to our hiring process and give you a brief synopsis of the position's benefits and requirements. If you are taking a career break, are on a maternity leave, recently retired or simply looking for some part-time job, this position is for you. [...] Content analysis details: (7.4 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4587] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [186.32.220.19 listed in bb.barracudacentral.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [186.32.220.19 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

[ot] virtualization systems
by Daniel-Constantin Mierla 29 Aug '12

29 Aug '12

Hello, just asking to see your experience deploying sip platforms on virtual systems. So far I was running Kamailio in virtual machines and no problems, but I insisted that media servers to be on physical machines. Lately is more pressure from the market to go everything virtual. So the question is more about having everything on virtual systems, proxy and media server, where the media server can deal with transcoding, conference rooms and IVRs. Any strong comments pro or against? What is your preferred virtualization system for such deployments? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

10 15

[OT] OverSIP SIP Proxy released
by Iñaki Baz Castillo 28 Aug '12

28 Aug '12

Hi all, I announce the first stable release of OverSIP SIP Proxy: http://www.oversip.net OverSIP features: ----------------------------------------------------------------------------- - SIP over UDP, TCP, TLS and WebSocket transports. - Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA). - Fully programmable using Ruby programming language. - The perfect Outbound (RFC 5626) Edge Proxy. - Fully asynchronous event-based design. OverSIP is not a replacement for well-known proxies as Kamailio (sip-router). OverSIP does not implement basic proxy features such as Digest authentication, registration or parallel forking. OverSIP can behave as an Outbound Edge Proxy between clients and SIP registrars, or can be placed anywhere for building custom SIP / WebSocket applications. Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 1

Re: [SR-Users] [ot] virtualization systems
by JR Richardson 28 Aug '12

28 Aug '12

>> just asking to see your experience deploying sip platforms on virtual >> systems. So far I was running Kamailio in virtual machines and no problems, >> but I insisted that media servers to be on physical machines. Lately is more >> pressure from the market to go everything virtual. >> >> So the question is more about having everything on virtual systems, proxy >> and media server, where the media server can deal with transcoding, >> conference rooms and IVRs. >> >> Any strong comments pro or against? >> >> What is your preferred virtualization system for such deployments? >> I host ~120 asterisk virtual serves on OpneVZ for more than 3 years, runs great. These are business customers so they can be very demanding, allot of custom apps and such. I host up to 150 phones in virtual environment, if more is needed, I switch over to a physical server. This is just a personal preference, not limited virtually, but customers with that many or more phones usually want a dedicated server and can certainly afford the up charge. Ensure your resources for each VM are enough for the voice apps you want to run, be cautious about disk I/O like high volume recording, database dips per call and such. I just converted my physical sip-routers to virtual about a week now on OpenVZ, so far so good. Transcoding required higher proc and mem loads so partition resources accordingly. I do run Asterisk 10.0 conf bridge server virtual, SIP only, no need for timing interface like DAHDI. This is running well, no problem that I'm aware of, just works. I use VirtualBox and VMware for demo systems and lab work but OpenVZ for production due to higher density you can achive per physical server. Whatever virtualization you want to use or are comfortable with, ensure you know it backwards and forwards and what to do if it breaks and how to increase resource when needed and by all means do your lab work ahead of time. You will not be able to simulate every situation in the lab, but you can certainly stress test your environment with high call volume, disk I/O recordings, databases operations, live migration, upgrades, downgrades, backups, resource monitoring, alarm triggers, test, test, test and then test some more. Virtualizing voice systems has been around a long time, most virt platforms can be implemented to perform well for voice, just got to do your homework. Good luck!. JR -- JR Richardson Engineering for the Masses

1 0

AVP string or integer ID
by Vitaliy Aleksandrov 28 Aug '12

28 Aug '12

Hi all. I'm working with kamailio for about 1 year and still don't understand difference between integer and string IDs of AVPs. Is it just a question of usability ?

2 1

Trunk Utilization graphing using MI COMMANDS
by phillman25 28 Aug '12

28 Aug '12

Dear List I am trying to run the below command from a REMOTE server where i use Cacti to graph all my data. I am trying to graph on a per trunk basis. kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}' This command yields an output on the local server. Could someone perhaps point me in the right direction? Thanks!

2 6

DROUTING nub question
by Luis Fernando Urrea 28 Aug '12

28 Aug '12

I haven't been able to find the answer in the documentation. It seems DROUTING does not take regex in the dr_rules table to match routable numbers, are there any kind of wilcards that could be used. How does one specify different rules for 8 digit numbers and 10 digit numbers. TIA -- O: 4000-1020 C: 6040-3624

3 7

Re: [SR-Users] ADSL client-side problem
by Daniel-Constantin Mierla 27 Aug '12

27 Aug '12

Hello, please do not send private messages from mailing lists discussions, keep the mailing list cc-ed so other can participate or learn from archives. On 8/27/12 5:22 PM, Simon Hintermann wrote: > Hello, > > I will try with a smaller "Subscription Expiry (s):" in my SNOM. > > Just to be clear with the method of test: > > - firewall redirects SNOM to my first ISP > - telephone works perfectly, registered in Kamailio... > - firewall redirects SNOM to my second ISP > - telephone is unregistered in Kamailio, impossible to "re-register" > the identity1, says "network error" > - only way to recover is to change the SNOM's IP address what do you mean by 'firewall redirects Snom to Nth ISP'? Is the IP of snom static or taken via dhcp from adsl router? Cheers, Daniel > > Thanks for your support, greetings > > Simon > > Le 27.08.2012 09:04, Daniel-Constantin Mierla a écrit : >> Hello, >> >> what is the expire time of your registrations? 24 hours seems a lot >> to re-register. >> >> From kamailio point of view, the contact is expired based on >> un-registration or timeout. You can look with 'kamctl ul show' to see >> if the phone is still registered or not. >> >> Anyhow, in such cases is good to set registration time quite small, >> say 5 or 10 minutes, but if there is a problem in the client side, it >> is not much to do in server side. >> >> What is the adsl routed, does it do SIP ALG? I have snom on ADSL >> using tcp, does not seem to make big problems. >> >> Cheers, >> Daniel >> >> On 8/16/12 2:10 PM, Simon Hintermann wrote: >>> >>> Hello, >>> >>> I am experiencing some disconnections with my VOIP phones. >>> >>> Let me explain: >>> >>> My server is an Asterisk + Kamailio in a datacenter with a fixed IP, >>> and a bridge firewall in front of it, so no NAT here. >>> >>> My client (several offices) have usually no problem when they have a >>> fixed IP, but encounter fatal disconnection when they have a dynamic >>> IP. I had the case with Siemens, SNOM and Aastra phones, without any >>> difference. They are of course natted behind an ADSL router. I also >>> have clients doing load-balance between two ADSL lines, which is >>> also problematic. >>> >>> The only way for me to recover is to change the private IP of the >>> VOIP phones. Seems more like a TCP problem or a routing problem, but >>> I am no network guru. >>> >>> I have two ADSL lines to test it, and when I switch from one line to >>> another, every phone becomes "Not registered". Even falling back tot >>> he first line does not help recovering. I did not try to wait for >>> longer than 24 hours to recover. >>> >>> So my question is, more generally: is it possible to allow the >>> clients (VOIP phones) to re-connect with a new IP address without >>> having closed correctly the first connection (SIP session, I >>> presume)? It seems not by default, but I played with the timeouts >>> server-side and phone-side, without luck. >>> >>> I saw many posts with PBX having problems when behind a dynamic IP, >>> but nothing about the clients having dynamic addresses or >>> load-balancing. Normally, having a dynamic IP does not mean that >>> your IP will change « on-the-fly », but each time you shut down and >>> reboot later your ADSL router, but we came to the conclusion that >>> every client NOT having a fixed IP was experiencing fatal >>> disconnects. So I managed to test and validate this behavior with >>> our two ADSL lines. >>> >>> Hope someone can help me. >>> >>> Greetings >>> >>> Simon >>> >>> >>> >>> _______________________________________________ >>> 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 >> >> -- >> Daniel-Constantin Mierla -http://www.asipto.com >> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 0

SMS module & IMS
by Raman Kumar 27 Aug '12

27 Aug '12

Hi, Can any one help me .. how to configure sms module? what changes required in scscf.cfg or any other cfg also?? Thanks and regards, Raman Kumar =====-----=====-----===== Notice: The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail or telephone and immediately and permanently delete the message and any attachments. Thank you

2 1

Re: [SR-Users] centos (fedora) service startup file
by Daniel-Constantin Mierla 27 Aug '12

27 Aug '12

Do not send private messages from the mailing list discussions. Maybe others can jump in the thread and provide good information, if people from discussion cannot sort out things. Also, mailing list is archived online, so people can search later on topics they are interested in. Cheers, Daniel On 8/27/12 9:43 AM, andre wrote: > 27.08.2012 10:09, Daniel-Constantin Mierla пишет: >> Hello, >> >> On 8/14/12 11:31 AM, andre wrote: >>> 14.08.2012 12:21, Daniel-Constantin Mierla пишет: >>>> Hello, >>>> >>>> thanks, it was committed in git master branch, backports to follow >>>> before minor releases. >>>> >>>> Do you build your own rpms or you install from the opensuse build >>>> system? >>> From the opensuse build system. >> >> ok, I will update that system as well. is 'success' available in all >> distros using rpms (like opensuse)? > Not shure about Suse. I even have no someone to ask > > Better add to RH distros only. >> >> Cheers, >> Daniel >> >>>> >>>> Cheers, >>>> Daniel >>>> >>>> On 8/2/12 6:24 PM, andre wrote: >>>>> Somwhere losted since openser 1.0 :) >>>>> >>>>> >>>>> Please fix in: >>>>> >>>>> start() { >>>>> echo -n $"Starting $prog: " >>>>> ..... >>>>> [ $RETVAL = 0 ] && touch something >>>>> to >>>>> [ $RETVAL = 0 ] && touch someting && success >>>>> >>>>> I mean call function >>>>> success >>>>> in the end of the expression. >>>>> >>>>> Thanks in advance >>>>> >>>>> _______________________________________________ >>>>> 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 >>>> >>> >> > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat

1 0

Advisement and promotion company seeking out local affiliate
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (7.0 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [213.177.0.199 listed in zen.spamhaus.org] -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [213.177.0.199 listed in bb.barracudacentral.org] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4633] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

centos (fedora) service startup file
by andre 27 Aug '12

27 Aug '12

Somwhere losted since openser 1.0 :) Please fix in: start() { echo -n $"Starting $prog: " ..... [ $RETVAL = 0 ] && touch something to [ $RETVAL = 0 ] && touch someting && success I mean call function success in the end of the expression. Thanks in advance

2 3

ADSL client-side problem
by Simon Hintermann 27 Aug '12

27 Aug '12

Hello, I am experiencing some disconnections with my VOIP phones. Let me explain: My server is an Asterisk + Kamailio in a datacenter with a fixed IP, and a bridge firewall in front of it, so no NAT here. My client (several offices) have usually no problem when they have a fixed IP, but encounter fatal disconnection when they have a dynamic IP. I had the case with Siemens, SNOM and Aastra phones, without any difference. They are of course natted behind an ADSL router. I also have clients doing load-balance between two ADSL lines, which is also problematic. The only way for me to recover is to change the private IP of the VOIP phones. Seems more like a TCP problem or a routing problem, but I am no network guru. I have two ADSL lines to test it, and when I switch from one line to another, every phone becomes "Not registered". Even falling back tot he first line does not help recovering. I did not try to wait for longer than 24 hours to recover. So my question is, more generally: is it possible to allow the clients (VOIP phones) to re-connect with a new IP address without having closed correctly the first connection (SIP session, I presume)? It seems not by default, but I played with the timeouts server-side and phone-side, without luck. I saw many posts with PBX having problems when behind a dynamic IP, but nothing about the clients having dynamic addresses or load-balancing. Normally, having a dynamic IP does not mean that your IP will change << on-the-fly >>, but each time you shut down and reboot later your ADSL router, but we came to the conclusion that every client NOT having a fixed IP was experiencing fatal disconnects. So I managed to test and validate this behavior with our two ADSL lines. Hope someone can help me. Greetings Simon

2 1

Advisement and promotion company seeking out local affiliate
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (9.6 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [139.195.43.246 listed in zen.spamhaus.org] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.5000] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Re: [SR-Users] Vacancy #120
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (15.1 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [186.63.33.68 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [186.63.33.68 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] 0.8 BAYES_50 BODY: Bayes spam probability is 40 to 60% [score: 0.4970] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

Re: [SR-Users] Vacancy #371
by serusers＠iptel.org 27 Aug '12

27 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: As a company we deal with advising other business and end users on reusable materials, green advancements and nontypical forms of energy and also produce and promote projects that are web media focused. [...] Content analysis details: (10.3 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.6 HELO_DYNAMIC_IPADDR2 Relay HELO'd using suspicious hostname (IP addr 2) 0.0 TVD_RCVD_IP TVD_RCVD_IP -0.0 SPF_PASS SPF: sender matches SPF record 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [184.77.159.149 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.6 URIBL_PH_SURBL Contains an URL listed in the PH SURBL blocklist [URIs: euroinjobs.com] -0.0 BAYES_40 BODY: Bayes spam probability is 20 to 40% [score: 0.2745] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [184.77.159.149 listed in zen.spamhaus.org] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

SEAS Module
by Fabian Borot 24 Aug '12

24 Aug '12

Hello, I would like to use the module and create our own version of the Application Server using the SEAS protocol but there are some pieces of information missing from the web site (links to images that contains information about headers etc ). The basic information on how it works is there, the AS registers to the OpenSER, receives information, sends information but the details of how to register, the format of the messages etc are not there. I checked for information in previous versions only to find the same situation.(3.1 thru 3.3) Is the information about the SEAS module specific to the WeSIP Application Server? Thank you fborot

2 4

Wrong index on expires column in pua module
by Andreas Granig 24 Aug '12

24 Aug '12

Hi guys, In the pua table, there is a unique key on the "expires" column, which seems wrong to me, because you'll sooner or later end up with different pua entries having the same expiry time, no? At least we ended up with messages like this in the logs, I think it's caused by pua_reginfo tests: ERROR: db_mysql [km_dbase.c:122]: driver error on query: Duplicate entry '1345474412' for key 'expires_idx' ERROR: <core> [db_query.c:210]: error while submitting query ERROR: pua [pua.c:1171]: while inserting in db table pua Any comments on that? Andreas

2 1

db_cassandra
by Boudewyn Ligthart 24 Aug '12

24 Aug '12

Hi, I am trying to get the db_cassandra working in kamailio 3.1.1 with thrift 0.6.1 API and however I get the following compile error. ol.h:33, from dbcassa_base.h:38, from dbcassa_base.cpp:47: ../../lib/srdb1/db.h:388:20: error: expected unqualified-id before âdeleteâ dbcassa_base.cpp: In function âColumnVecPtr cassa_translate_query(const db1_con_t*, str* const*, const db_val_t*, str* const*, int, int, int*)â: dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 7 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 5 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 6 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 4 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] dbcassa_base.cpp:538:8: warning: format â%dâ expects argument of type âintâ, but argument 4 has type âstd::vector<org::apache::cassandra::KeySlice>::size_type {aka long unsigned int}â [-Wformat] ../../lib/srdb1/db_res.h: At global scope: ../../lib/srdb1/db_res.h:105:19: warning: inline function âdb1_res_t* db_new_result()â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:121:12: warning: inline function âint db_allocate_columns(db1_res_t*, unsigned int)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:98:12: warning: inline function âint db_free_columns(db1_res_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:129:12: warning: inline function âint db_allocate_rows(db1_res_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_row.h:74:12: warning: inline function âint db_allocate_row(const db1_res_t*, db_row_t*)â used but never defined [enabled by default] ../../lib/srdb1/db_res.h:111:12: warning: inline function âint db_free_result(db1_res_t*)â used but never defined [enabled by default] make[1]: *** [dbcassa_base.o] Error 1 make: *** [install-modules] Error 1 Anybody has got any ideas? Thanks in advance, Bou

2 6

Продается коттедж в 42км от Москвы, у горнолыжного курорта
by jiri＠iptel.org 23 Aug '12

23 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: ðÒÏÄÁÅÔÓÑ ÚÄÅÓØ www.ËÕÐÉ-ÄÏÍ-ÚÄÅÓØ.ÒÆ [...] Content analysis details: (10.2 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 3.5 BAYES_99 BODY: Bayes spam probability is 99 to 100% [score: 1.0000] 0.8 RCVD_IN_SORBS_WEB RBL: SORBS: sender is an abusable web server [116.203.165.242 listed in dnsbl.sorbs.net] 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [116.203.165.242 listed in bb.barracudacentral.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [116.203.165.242 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS 0.0 UNPARSEABLE_RELAY Informational: message has unparseable relay lines

1 0

Kamailio with asterisk for outbound calls
by Vijay Thakur 23 Aug '12

23 Aug '12

Hi All Kamailio Experts, I have configured Kamailio (kamailio 3.1.5) as media server. All things are working fine. Now i want to use Asterisk (Asterisk 1.6) for Outbound Calls. For this purpose i have followed the web page : http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. In this page, some points are not clear for me , as given below: (1) In case you use *sipregs* you have to create a record for each extension where to set the 'name' to value of 'name' from *sipusers*. The rest is populated by Asterisk from registrations. (2) Be sure you configure Asterisk *to not authenticate* SIP requests coming from Kamailio. I am not sure that my local users chat is working through kamailio or asterisk, who is used for authorization. Any specific Web page to correct the issue will highly appreciated according to my scenario. Kindly guide me. Thanks in advance. -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 3

interfacing with kamailio
by Carlos Cruz 22 Aug '12

22 Aug '12

I'm just starting to evaluate Kamailio to be used as a front end to a couple of Asterisk Servers. Currently I have an Adobe Flex application that interfaces with Asterisk via AMI using a socket connection. I searched on how to interface with Kamailio but wasn't able to find anything that helped me. I would appreciate it if someone could point me to where I can find information on how this can be done using Kamailio. For example I would like my Flex Application to connect to Kamailio to know the status of the SIP channels. Thanks for any help!!! Carlos

2 1

new feature: option to remove contacts based on sip nat keepalives
by Juha Heinanen 22 Aug '12

22 Aug '12

daniel, you wrote: This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. why would a sip ua that runs in a mobile network device be configured to use udp instead of tcp? poor battery life seems to be a big problem in so called smart phones. wouldn't this kind of active pinging make it even worse? -- juha

2 1

Websocket INVITE 407 error
by David Patiño 22 Aug '12

22 Aug '12

Hi there, I'm new on kamailio users list. Now I'm developing a client in javascript (using websockets) to SIP calls. I'm using last devel version of kamailio with websocket support. I've already get a correct REGISTER transaction but now I'm not able to get a correct INVITE transaction. Actually I'm not sure if this a kamailio error or a problem with my source code however I'm using the same function to get digest info from received nonce both REGISTER and INVITE transactions. The INVITE transaction always enters in an endless loop with 407 requests. Please, is there anybody who could help me? Those are the frames of my code (obtained from chrome browser console): wsUri: ws://10.1.20.40:80/ SOCKET CREATED SOCKET OPENED SENT: REGISTER sip:10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg From: sip:101@10.1.20.40;tag=Me0arqDG6K8o7l32oJgR To: sip:101@10.1.20.40 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 1 REGISTER Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws> ;reg-id=98 ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>" RESPONSE: SIP/2.0 401 Unauthorized Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1C36k8S3b0OobHUAOLLOae0SbsbEQg;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=Me0arqDG6K8o7l32oJgR To: sip:101@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.000c Call-ID: 0EQSBkoG9RGgfg3E CSeq: 1 REGISTER WWW-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: REGISTER sip:10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037 From: sip:101@10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612 To: sip:101@10.1.20.40 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 2 REGISTER Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws> ;reg-id=98 ;+sip.instance="<urn:uuid:beb-68a9-6396e5f4>" Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:10.1.20.40", response= "9b209eb35ee88192e2e0c468f8cf91da" RESPONSE: SIP/2.0 200 OK Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKl521S8acS8nVEcejv2GOJ8r46Jb037;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=Pb8d1gV8K5qJo1gOv612 To: sip:101@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b2e4 Call-ID: 0EQSBkoG9RGgfg3E CSeq: 2 REGISTER Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws>;expires=120;received="sip:10.1.2.229:50906;transport=WS";pub-gruu="sip:101@10.1.20.40;gr=urn:uuid:beb-68a9-6396e5f4";temp-gruu="sip:uloc-5033344d-5211-1-1de173@10.1.20.40;gr";+sip.instance="<urn:uuid:beb-68a9-6396e5f4>";reg-id=98 Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKOIosJ8TMPgvfHme6NIm4GSn24Fu7f7;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: ACK sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKAmmLhK71tr2QoAiV9f5QmdKF9dGkke From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.f545 Call-ID: 9KoKnFdh285k2jg4 CSeq: 1 ACK Max-Forwards: 70 Supported: path, outbound, gruu SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Proxy-Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:102@10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798" Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK1GRkmlTVo7Cb4fg8OMe9g39g096hmn;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 SENT: ACK sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bKG9v5g7i1LNsoVGo18ULlgqdA1EcFGb From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.0007 Call-ID: 9KoKnFdh285k2jg4 CSeq: 2 ACK Max-Forwards: 70 Supported: path, outbound, gruu SENT: INVITE sip:102@10.1.20.40 SIP/2.0 Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40 Call-ID: 9KoKnFdh285k2jg4 CSeq: 3 INVITE Max-Forwards: 70 Supported: path, outbound, gruu Contact: <sip:101@KMkjMf37lVeM.invalid;transport=ws ;gr=urn:uuid:beb-68a9-6396e5f4;ob> Proxy-Authorization: Digest username="101", realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re", opaque="", uri="sip:102@10.1.20.40", response= "f4bbe128abf3acb3a4e11882749d8798" Content-Type: application/sdp RESPONSE: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WS KMkjMf37lVeM.invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229 From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7 Call-ID: 9KoKnFdh285k2jg4 CSeq: 3 INVITE Proxy-Authenticate: Digest realm="10.1.20.40", nonce="UDND2VAzQq0Tz8wvIFWNh2eYRxdtu4Re" Server: kamailio (3.4.0-dev3 (i386/linux)) Content-Length: 0 .......and so on in an endless loop with 407 requests Thanks a lot Kind Regards -- --DAVID--

4 9

SIP routing configuration howto
by Kaplan, Michael A 22 Aug '12

22 Aug '12

Folks, I'm working on a 5 node network in the following topology: Host1 - Edge Proxy 1 --- Core Proxy - Edge Proxy 2 - Host2 The proxy machines contain SIP Server installations, the hosts contain user agent software. I am able to make a call end to end between the hosts, but when I look at the traffic in wireshark, the core proxy is actually just forwarding the messages at the IP-layer. I would like actual SIP routing to occur, where a host would send the INVITE to the edge, and the edge has to forward it to the core etc, with these messages actually being processed at the SIP layer for all proxy nodes. I've been trying to use the dynamic routing module, thought I'm not sure this is really right for what I need, simple static routing would likely work as well. Is it necessary for me to rewrite the route{} functions of a proxy script? I'm not sure if I'm approaching this problem the right way. Eventually I would like to have multiple core servers and be able to failover between them, but trying to keep it simple for now. Thanks, Mike -------------------------------- Michael A. Kaplan Senior Research Scientist Applied Communication Sciences 732-699-3805 (office) 973-699-6775 (cell) mkaplan(a)appcomsci.com

2 1

Advertising and instructing company seeking region rep
by serusers＠iptel.org 22 Aug '12

22 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (11.3 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [189.58.67.119 listed in bb.barracudacentral.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL [189.58.67.119 listed in zen.spamhaus.org] 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [189.58.67.119 listed in bl.score.senderscore.com] 0.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [189.58.67.119 listed in dnsbl.sorbs.net] 0.7 SPF_SOFTFAIL SPF: sender does not match SPF record (softfail) 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0001] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: euroinjobs.com] 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS

1 0

New Job position
by serusers＠iptel.org 21 Aug '12

21 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (10.6 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [91.54.224.170 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL -0.0 SPF_PASS SPF: sender matches SPF record 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [91.54.224.170 listed in bb.barracudacentral.org] 1.3 RCVD_IN_RP_RNBL RBL: Relay in RNBL, https://senderscore.org/blacklistlookup/ [91.54.224.170 listed in bl.score.senderscore.com] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 1.6 URIBL_SBL Contains an URL listed in the SBL blocklist [URIs: euroinjobs.com] 1.0 RDNS_DYNAMIC Delivered to internal network by host with dynamic-looking rDNS

1 0

Send/Receive IMs
by Marty Beyers 21 Aug '12

21 Aug '12

Hey guys. I am very new to Kamailio. Basically, at this time, I want to generate a SIP message (from an SMS) within Kannel, and send this message to my Kamailio server, which will forward the SIP message to another SIP server. Therefore my Kamailio Sip server would act as a single user agent connecting to only one other SIP server. So what I am unsure of is: 1. Is there a Kamailio interface (such as an HTTP server) which Kannel can connect to in order for Kannel to send multiple messages to my Kamailio Server, which in turn would be forwarded as SIP instant messages to another SIP server? 2. Which Kamailio api can I use to formulate and send the SIP instant messages? 3. Which Kamailio callback function could be called when I receive acknowledgement responses to the SIP IMs I have sent? 4. Which Kamailio callback function could be called when I receive new SIP instant messages sent to me? It would be great if someone could help point me in the right direction. Many thanks. Fred

2 1

vacancy #428
by serusers＠iptel.org 21 Aug '12

21 Aug '12

Spam detection software, running on the system "mail.iptel.org", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: Collectively, we focus on different sources of power, new green tech, and reusable materials. We are advertising for someone from your region to be employed by us. We require: [...] Content analysis details: (7.5 points, 5.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- -0.0 SPF_PASS SPF: sender matches SPF record 0.0 T_FRT_PROFILE2 BODY: ReplaceTags: Profile (2) 0.4 RCVD_IN_XBL RBL: Received via a relay in Spamhaus XBL [109.100.26.103 listed in zen.spamhaus.org] 3.3 RCVD_IN_PBL RBL: Received via a relay in Spamhaus PBL 1.4 RCVD_IN_BRBL_LASTEXT RBL: RCVD_IN_BRBL_LASTEXT [109.100.26.103 listed in bb.barracudacentral.org] 1.7 URIBL_DBL_SPAM Contains an URL listed in the DBL blocklist [URIs: euroinjobs.com] 1.7 URIBL_BLACK Contains an URL listed in the URIBL blacklist [URIs: euroinjobs.com] -1.9 BAYES_00 BODY: Bayes spam probability is 0 to 1% [score: 0.0000] 0.8 RDNS_NONE Delivered to internal network by a host with no rDNS

1 0

SUBSCRIBE/NOTIFY record route
by Spencer Thomason 21 Aug '12

21 Aug '12

Hello all, I'm using Kamailio (on a public IP) as a border proxy that handles NAT traversal (detection and keep alives) for several Freeswitch servers which have public IPs as well. I'm having a problem record-routing SUBSCRIBEs and NOTIFYs for nated UACs. If the UAC is behind NAT, I am adding a Record-Route to the initial SUBSCRIBE and NOTIFY request. This works great until the UAC (in this case a Cisco SPA509G) refreshes the subcription and sends the SUBSCRIBE directly to the Freeswitch server. Since the request doesn't pass through the proxy, Freeswitch then sends NOTIFYs directly to the UAC which only recieves the first NOTIFY as the NAT pinhole is still open from the SUBSCRIBE. The UAC did receive a NOTIFY validating the subscription, and continues to refresh the subscription directly to the freeswitch server. The completely breaks presence. What is the correct way to ensure all SUBSCRIBE/NOTIFYs pass through the proxy? I have found that I can force this behavior by modifying the Contact header in the 202 Accepted to point back to the proxy but I'm not sure if this is what I "should" be doing. Thanks for any assistance and please forgive my ignorance when is comes to SIP presence! Spencer

2 1

Re: [SR-Users] Siremis login issue
by dave mccann 20 Aug '12

20 Aug '12

Hi All I managed to figure out that the table had not been created in the database. So i deleted the install.lock file in the Siremis directory to run the install again. This time I clicked on all the boxes that were on step 2: Create Siremis Database - this create database 'siremis' Import Default Data - this imports initial data into database 'siremis' Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules Replace DB Config - this updates Siremis configuration file with the details to access databases Then after it had finished I was able to login using the default admin/admin. Thanks for the reply Max greatly appreciated for the time taken to do so - Ivan - I didnt get chance to test what you had put however I will take a note of it for future reference.......hopefully I may be able to return the favor in the future if you guys need a hand. Cheers guys Dave > From: sr-users-request(a)lists.sip-router.org > Subject: sr-users Digest, Vol 87, Issue 64 > To: sr-users(a)lists.sip-router.org > Date: Mon, 20 Aug 2012 16:17:22 +0200 > > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Siremis login issue (dave mccann) > 2. Re: Siremis login issue (Max M?hlbronner) > 3. Re: [sr-dev] new feature: option to remove, contacts based on > sip nat keepalives (Yufei Tao) > 4. Re: Siremis login issue (Ivan Milivojevic) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 20 Aug 2012 12:51:57 +0100 > From: dave mccann <davemac2626(a)live.co.uk> > Subject: [SR-Users] Siremis login issue > To: <sr-users(a)lists.sip-router.org> > Message-ID: <DUB117-W1040D3073CA62A3AE627CF787B90(a)phx.gbl> > Content-Type: text/plain; charset="iso-8859-1" > > > Hi All > > I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error: > > "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, > T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, > T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE > T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base > table or view not found: 1146 Table 'siremis.user' doesn't exist" > > I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro'). I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared. I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files. > > Can anyone please help as this is that last bit that I need to overcome :) > > Many thanks and best regards. > > Dave > >

1 0

UAC module
by Ron Hopson 20 Aug '12

20 Aug '12

Hello, I cannot create a configuration consistent with uac_reg_lookup. Does someone have a configuration with a simple example of uac_reg_lookup that works with db_text file uacreg? I have kamailio 3.3 running on linux. Based on suggestions in http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html and http://by-miconda.blogspot.com/2010/10/best-of-new-in-kamailio-310-10.html I created this record in db_text file uacreg: 2:12345678:daniel:mysipserver.com:daniel123:sipprovider.com:sipprovider.com:daniel123:xxxxx:"sip:sipprovider.com" I assume that function uac_reg_lookup retrieves registration information from this record after it is loaded into a memory table. One version of the configuration file is attached. Here are three examples of failures to create a configuration. $rU = 12345678; if(uac_reg_lookup("$rU", "$ru")){ #line 73 0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 1 0(13856) DEBUG: <core> [rvalue.c:2861]: RV is int: 12345678 0(13856) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13856) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13856) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:73 --------- $rU = "12345678"; if(uac_reg_lookup("$rU", "$ru")){ #line 73 0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13973) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678" 0(13973) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13973) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13973) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:73 -------- $rU = "12345678"; $ru = "sip:mysipserver.com"; if(uac_reg_lookup("$rU", "$ru")){ #line 74 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "12345678" 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 2 0(13976) DEBUG: <core> [rvalue.c:2865]: RV is str: "sip:mysipserver.com" 0(13976) DEBUG: <core> [rvalue.c:2857]: RV fixing type 4 0(13976) DEBUG: <core> [route.c:929]: fixing uac_reg_lookup() 0(13976) ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:74 --------------------- regards, Ron Hopson

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Re: [SR-Users] [sr-dev] new feature: option to remove, contacts based on sip nat keepalives
by Yufei Tao 20 Aug '12

20 Aug '12

That's a very useful feature for updating presence information on behalf of some clients that do not support presence, or simply crashed. I will be watching this space for TCP/TLS support :) Is there any other existing way to achieve this for clients over TCP/TLS? Cheers, Yufei On 20/08/12 09:13, sr-users-request(a)lists.sip-router.org wrote: > ------------------------------ Message: 5 Date: Mon, 20 Aug 2012 > 10:13:12 +0200 From: "Olle E. Johansson" <oej(a)edvina.net> Subject: Re: > [SR-Users] [sr-dev] new feature: option to remove contacts based on > sip nat keepalives To: miconda(a)gmail.com, Development mailing list of > the sip-router project <sr-dev(a)lists.sip-router.org> Cc: "SIP Router - > Kamailio $OpenSER$ and SIP Express Router $SER$ - Users Mailing > List" <sr-users(a)lists.sip-router.org> Message-ID: > <1CD23BC5-1002-4A6E-B42B-09C73530DB11(a)edvina.net> Content-Type: > text/plain; charset="us-ascii" 20 aug 2012 kl. 10:06 skrev > Daniel-Constantin Mierla: >> > Hello, >> > >> > over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests). >> > >> > The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs). >> > >> > Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3). >> > >> > In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory). >> > >> > http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244 >> > >> > Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp. >> > >> > This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. >> > >> > There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down. >> > > On my personal wish list is to have an event route when a registered device becomes unreachable - like a closed TCP connection for a device behind NAT or failure to response to NAT pings. > > /O -- 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

1 0

Siremis login issue
by dave mccann 20 Aug '12

20 Aug '12

Hi All I have installed Kamailio with Siremis, Kamailio is working fine and starts and I have been through the initial setup for Siremis however when i go to login to the web management page it gives an error: "Error in query:: SELECT T0.id, T0.username, T0.password, T0.enctype, T0.email, T0.status, T0.lastlogin, T0.lastlogout, T0.create_by, T0.create_time, T0.update_by, T0.update_time FROM `user` T0 WHERE T0.username='openser' and status='1' LIMIT 1. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'siremis.user' doesn't exist" I have tried the default "admin/admin" login as mentioned in the install guide and have also tried the default passwords for Kamailio "openser - (with default password 'openserrw') and openserro - (with default password 'openserro'). I have also tried root/*password* and also siremis/siremis which is contained in the /var/www/siremis-3.2.1/siremis/config.xml however none were successful and the same error appeared. I read that it was a permissions error however have set directory /var/www/siremis-3.2.1 to chmod 777 including all sub dirs and files. Can anyone please help as this is that last bit that I need to overcome :) Many thanks and best regards. Dave

3 2

new feature: option to remove contacts based on sip nat keepalives
by Daniel-Constantin Mierla 20 Aug '12

20 Aug '12

Hello, over the weekend I committed the code that allows to remove contacts from location table if the device does not respond to several attempts of SIP nat keepalives (usually OPTIONS requests). The feature works only for SIP over UDP, being the only transport for which nathelper module sends keepalives (or sip ping requests as written in some docs). Also, the lack of keepalive reply is tracked only for location records stored in memory (so it does not work with db only mode for usrloc). This is because the keepalives are sent stateless, not using tm module, to avoid overloading of timers and shared memory when dealing with large subscriber base -- therefore the implementation is very lightweight in using resources, targeting of course scalability. Detection of replies for keepalives is done based on from tag which carries the internal unique id of the location contact (which is available from v3.3). In order to enable this feature you have to set the keepalive_timeout parameter of nathelper module -- that's the only addition you have to add to a config that sends SIP nat keepalives. Internally, the modules store the timestamp of the last SIP keepalive reply (nathelper detects the right reply and calls usrloc API for storing the timestamp in contact structure in memory). http://kamailio.org/docs/modules/devel/modules_k/nathelper.html#id2535244 Before sending a new keepalive to a contact, it is checking if the last keepalive reply is older than keepalive_timeout value and if yes, will lower the expires to 10 seconds (if the expires is higher). There is no special handling of keepalive reply codes, all of them will update the last keepalive timestamp. This feature should be useful in mobile networks, to avoid stacking lot of invalid contacts which could result in lot of branches causing many retransmissions. There is plant to go for tcp/tls as well and detect when the connection is down, but looks like no spare time in the near future. However, right now using set_forward_no_connect() will avoid trying to create new connections for natted uses whose registration connection is down. Testing and feedback will be welcome. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

2 1

does AVP's stay availeable after the transaction was answered?
by Uri Shacked 20 Aug '12

20 Aug '12

Hi, I am trying to use an AVP the I set on the INVITE when I recieve the BYE. for some strange reason, the AVP is NULL when I test it. Any ideas? Uri

3 2

avp usage limit
by Uri Shacked 20 Aug '12

20 Aug '12

Hi, I wonder, is there a limit for the amount of AVP's i can use? Let's say i store 6-9 characters (string) each time i set an AVP. Thanks, Uri

4 3

does AVP's stay availeable after the transaction was answered?
by Uri Shacked 20 Aug '12

20 Aug '12

OK, I understan the AVP's are per transaction. How can I save info from the invite and use it on the bye? $sht(...)? $dlg(flag), $dlg(set) ? Thanks, Uri > juha writes >it is not strange reason, since avps are transaction specific and bye >does not belong to the same transaction as invite. > juha >>Uri Shacked writes: >* >I am trying to use an AVP the I set on the INVITE when I recieve the BYE.* *> *>* for some strange reason, the AVP is NULL when I test it. *

2 1

Request for advice, running SIP-Router in OpenVZ Container
by JR Richardson 19 Aug '12

19 Aug '12

Hi All, I'm considering running Kamailio as a virtual machine, with such low utilization, it doesn't seem to make sense to keep running it on a physical host server. I've been virtualizing Asterisk PBX's for years and run a host of other virtualized servers with OpenVZ, VMware and MS V-Server without issues. I'd like any opinions or thoughts about pitfalls or caveats I should be aware of specific to virtualized SIP-Router/Kamailio. Thanks in advanced. JR

4 3

append_branch with pseudo variables
by Carlos Ruiz Díaz 17 Aug '12

17 Aug '12

Hello guys, Why I can only append_branch() a SIP URI as a literal constant string? Why not from a pseudovariable? It's the same for most of the core functions. My intention is to do something like this: append_branch("$cnx3a(route1)"); t_relay(); where cnx3a is a variable containing the SIP URIs. The only "fix" I have found for this is by doing it as follows: $ru = $cnx3a(route1); append_branch(); t_relay(); Is there any other (and more elegant) way? Regards. Carlos.

2 9

Kamailio packet sourcing
by Luis Fernando Urrea 17 Aug '12

17 Aug '12

Hi all, I am sort of a new comer to kamailio and I have a config in which Kamailio listens on different IPs that correspond to SSH VPN tunnels. The config handles a MULTIDOMAIN scenario with FreeSWITCH as multi tenant media server. The situation I am facing is that when a REGISTERED user calls in through one of the SSH tunnels to the corresponding listening IP, Kamailio does not seem to source the INVITE that goes to FreeSWITCH from that IP since on FS I use "destination_field=network_addr" to try to apply different dialplan rules per domain and I see the call on FS matching a non corresponding dial plan entry sourced with an IP of a different SSH tunnel. Do I have any other option to send the call to FS using some other field to match to my convenience? I hope the description is clear enough, otherwise I could elaborate if required. Your input is highly appreciated. Regards, Luis -- O: 4000-1020 C: 6040-3624

2 1

Kamailio NAT traversal
by Spencer Thomason 17 Aug '12

17 Aug '12

Hello, I'd like to use Kamailio in from of FreeSWITCH to handle NAT traversal. Is there a way I can allow freeswitch to handle all auth but store the registration in usrloc as I need to send an options ping to the endpoints? My thought is something like this: natted endpoint -> kamailio -> freeswitch If the response is 200OK, save the registration. Thanks, Spencer

4 5

Accessing received after lookup()
by DanB 17 Aug '12

17 Aug '12

Hey All, I was wondering if anyone could point me to the right direction on the following issue: I use registrar with path to store contacts after main proxy. There I have properly stored received db field (original address of the NAT router). When I query the registrar, I get the user contact details (natted ones) in 302 redirect. However at this point I cannot access received avp (or I did not find a way yet, except calling reg_fetch_contacts which I would prefer not to since it doubles the work lookup does). Is there any catch which could save me in this scenario? After loopkup() is there any pseudovariable which I can inspect to find out user's contact data? Any other way to extract received contact data out of contact info? Thanks in advance for any kind of tip! DanB

2 5

sms module & IMS
by Oleksandr Shchirskyi 16 Aug '12

16 Aug '12

Hi, I'm using OpenIMSCore based on the SER. At the moment I try to add SMS-GW to it. I've found, that SER has sms module, and I tried to use it. But, unfortunately, I couldn't make it work. It's not very clear for me how SIP MESSAGEs can be routed to the sms module. This module works not as application server (or I'm wrong?) At the moment I have such configuration in the scscf.cfg: ============================== modparam("sms","modems","Nokia [d=/dev/ttyACM0;b=9600]") modparam("sms","networks","D1[m=10]") modparam("sms","links","Nokia[D1]") modparam("sms","domain","open-ims.test") modparam("sms","max_sms_parts",3) modparam("sms","use_contact",1) # main routing logic route{ if(method=="MESSAGE"){ sms_send_msg_to_net("D1"); } ... } ============================== As a hardware GSM gateway my cell phone is using. As I see, nothing happens when s-cscf processes SIP MESSAGEs. I don't see that these messages are forwarded to the sms module, or smth like that. Final question - is it possible to use this module as IP-SMS-GW for the OpenIMSCore? And how? I would be very appreciated for any help. Thank you in advance. -- Oleksandr Shchirskyi | Software Testing Engineer GlobalLogic Inc. | Innovation by Design ARGENTINA | CHILE | CHINA | INDIA | ISRAEL | UKRAINE | UK | USA Office: +044.492.96.95, (ext. 4296) www.globallogic.com

2 1

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

3 17

Re: [SR-Users] rtpproxy module issue
by phillman25 16 Aug '12

16 Aug '12

Dear SamyGo Thanks for your response. Yes you're right a folder called rtpproxy.so was left in the directory /usr/local/lib/kamailio/modules_k that had not been updated to the date kamailio was compiled and once i deleted it everything was then ok. I checked in directory /usr/local/lib/kamailio/modules and there i found a new rtpproxy.so folder that was updated on the installation date. Seems like rtpproxy.so in the new version is installed in /usr/local/lib/kamailio/modules and not in /usr/local/lib/kamailio/modules_k. Is that correct? Well everything is working ok now! Thanks for your assistance! Regards Phillip > Message: 4 > Date: Thu, 16 Aug 2012 13:29:35 +0500 > From: SamyGo <govoiper(a)gmail.com> > Subject: Re: [SR-Users] rtpproxy module issue > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: > <CAJUJwtjB7fmm1pa60wJMLPPte= > Qu_AeZ1z2aqysQaZch_Pb+dQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Though not an expert, I got few things in mind for your case. In the > upgrade process the modules folder remain the same ? right ! if that is so > is there any chances that the new installation skipped installation of > newer rtpproxy module and hence the old version rtpproxy module could've > been left in place. That could be a reason for not identifying the older > module. > This whole speculation could be wrong if the newer version supports the > older version modules. > > BR > Sammy > > > On Thu, Aug 16, 2012 at 10:56 AM, phillman25 <phillman25(a)gmail.com> wrote: > > > Dear List > > > > I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1 > > However, when trying to start kamailio its giving me the below error: > > > > > > ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open > module > > </usr/local/lib/kamailio/modules_k/rtpproxy.so>: > > /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: > qm_realloc > > : <core> [cfg.y:3591]: parse error in config file > > /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to > > load module > > > > > > Does anyone know why rtpproxy module cannot be loaded? > > > > > > This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: > > > > > > > > shell-init: error retrieving current directory: getcwd: cannot access > > parent directories: No such file or directory > > linux-gate.so.1 => (0x00276000) > > libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) > > libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 > (0x00768000) > > libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) > > /lib/ld-linux.so.2 (0x00114000) > > > > > > > > Regards Phillip > > > > _______________________________________________ > > 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 > > > > >

1 0

Is Route Header in SIP Invite Mandatory?
by Apoorv Khare 16 Aug '12

16 Aug '12

Hi All, Does Kamailio has a mandatory requirement for SIP Invite to have Route header? Although RFC-3261 doesn't have. My requirement is to take out route header from the client if I do that I don't see PRACK being send out from Kamilio to Media GW which causes call drop eventually in 15sec. Any thoughts.......... BR, Apoorv

2 1

rtpproxy module issue
by phillman25 16 Aug '12

16 Aug '12

Dear List I have just upgraded from kamailio V3.2.2 to kamailio V 3.3.1 However, when trying to start kamailio its giving me the below error: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to load module Does anyone know why rtpproxy module cannot be loaded? This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: shell-init: error retrieving current directory: getcwd: cannot access parent directories: No such file or directory linux-gate.so.1 => (0x00276000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) /lib/ld-linux.so.2 (0x00114000) Regards Phillip

2 1

Debian packages still installing 3.3.0
by DanB 16 Aug '12

16 Aug '12

Hey Guys, Not sure if important, but what I noticed is that the latest debian stable as well as latest 33 packages, they both install still kamailio 3.3.0. In order to get 3.3.1 I needed to use nightly build packages. Is this what is desired? Thanks, DanB

2 1

Kamailio and ACME
by Fatima Chahrour 15 Aug '12

15 Aug '12

Sent from iPhone

4 7

Dispatcher: ds_is_from_list does not longer populate grp_avp
by DanB 15 Aug '12

15 Aug '12

Hey Guys, Something I noticed after upgrading Kamailio from 3.3.0 to 3.3.1. Having the following code: """ if ds_is_from_list() { xlog("Request coming from SCloud"); xlog("Checking grp_internal, having: $avp(s:dsgrp)"); } """ will produce the following output to syslog (dsgrp null instead of group id): """ Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: Request coming from SCloud Aug 15 11:24:28 PrxyDev /usr/sbin/kamailio[2818]: ERROR: <script>: Checking grp_internal, having: <null> """ I have in dispatcher set: modparam("dispatcher", "grp_avp", "$avp(dsgrp)") and I am loading the rules from file. I should mention that this block was working fine before upgrade. Am I doing something wrong? Thanks in advance! DanB

2 8

A much more modest initial goal: get Kamailio dev version up and running
by Juan Castro 14 Aug '12

14 Aug '12

1 1

Re: [SR-Users] rtpproxy module issue
by phillman25 14 Aug '12

14 Aug '12

Hello This is the output of ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so: shell-init: error retrieving current directory: getcwd: cannot access parent directories: No such file or directory linux-gate.so.1 => (0x00276000) libkmi.so.1 => /usr/local/lib/kamailio/libkmi.so.1 (0x00b64000) libkcore.so.1 => /usr/local/lib/kamailio/libkcore.so.1 (0x00768000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0x00277000) /lib/ld-linux.so.2 (0x00114000) Regards Phillip Message: 3 Date: Tue, 14 Aug 2012 08:49:13 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] rtpproxy module issue To: sr-users(a)lists.sip-router.org Message-ID: <502A4949.2000707(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Please post output of: ldd /usr/local/lib/kamailio/modules_k/rtpproxy.so On 08/14/2012 08:47 AM, phillman25 wrote: > Dear List > > I have just installed kamailio V 3.3.1 > However, when trying to start kamailio its giving me the below error: > > > ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open > module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: > /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc > : <core> [cfg.y:3591]: parse error in config file > /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to > load module > > > Does anyone know why rtpproxy module cannot be loaded? > > > Thanks > Phillip > > > _______________________________________________ > 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 > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

rtpproxy module issue
by phillman25 14 Aug '12

14 Aug '12

Dear List I have just installed kamailio V 3.3.1 However, when trying to start kamailio its giving me the below error: ERROR: <core> [sr_module.c:572]: ERROR: load_module: could not open module </usr/local/lib/kamailio/modules_k/rtpproxy.so>: /usr/local/lib/kamailio/modules_k/rtpproxy.so: undefined symbol: qm_realloc : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 245, column 12-24: failed to load module Does anyone know why rtpproxy module cannot be loaded? Thanks Phillip

2 1

Remove m=video from SDP header
by phillman25 14 Aug '12

14 Aug '12

Dear List I am trying to remove specific lines from the following original SDP body: Content-Type: application/sdp Content-Length: 406 v=0 o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx b=CT:384 t=0 0 m=audio 11904 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11602 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 a=sendrecv I need to remove the following lines from the above SDP body as my International carrier does not support Video capabilities: m=video 15042 RTP/AVP 34 99 a=rtpmap:34 H263/90000 a=rtpmap:99 H264/90000 I used the following code in Kamailio config: if(has_body("application/sdp") && search_body("m=video")){ subst_body('#m=video ([0-9]+) RTP/AVP (.*)$# #'); subst_body('#a=rtpmap:34 (.*)$# #'); subst_body('#a=rtpmap:99 (.*)$# #'); subst_body('#a=sendrecv(.*)$# #'); } Content-Type: application/sdp Content-Length: 325 P-hint: outbound v=0 o=root 3048 3048 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx b=CT:384 t=0 0 m=audio 11904 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv As you can see the m=video body has been removed, however, calls are still failing. Is there something i have missed? I am using Kamailio v3.2.2 Thanking you in advance! Phillip

3 3

How best to experiment with SIP over Websocket?
by Juan Castro 13 Aug '12

13 Aug '12

As per the instructions here... http://www.kamailio.org/w/2012/07/websockets ...I downloaded the latest source from Git... http://www.kamailio.org/w/download/ ...but I don't see any mention of Websocket anywhere in the source. I suspect I didn't actually download the development tree, but the 3.3 branch. Last time I messed with any VoIP code was in 2008 (OpenSER and Asterisk) and now I'm starting all over again. I apologize for the dumb newbie question, but how do I test the Websocket code? -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100

2 4

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

3 11

Kamailio Presence with XCAP not working accordingly
by SamyGo 13 Aug '12

13 Aug '12

Hi, I've followed the tutorial on kab.asipto.com for presence using built-in xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple I'm using Kamailio version 3.3.1 and did minor changes in modparams and rtpproxy function calls and the kamailio accepted the configurations file posted on the page and started. But the problem is that I don't get the presence status of the contacts still. Please suggest what to look for and how to troubleshoot this. I get this error on Jitsi - image attached. ERROR: "http://ip.of.server/xcap-root/xcap-caps/global/index resource can not be read" Regards, Sammy

2 9

Best tool for crafting sip messages
by Anton Kvashenkin 13 Aug '12

13 Aug '12

Hello, guys. What is the best tool for crafting sip messages for testing purpose? For example, simple REGISTER or INVITE message. I'm new to kamailio (not sip) so it would be great for me to use debbuger module for step-by-step digging into cfg.

6 7

$avp reset to <null> with no reason?
by Uri Shacked 13 Aug '12

13 Aug '12

Hi,**** ** ** I encountered a problem. While doing some avp's settings I notice that one avp that I use is changing to <null> without me doing anything.**** ** ** Here is the part of the cfg file:**** ………….**** ………….**** Route[SET]{**** …………..**** $avp(Sfeaturetype)=$avp(next_feature_type);**** …………**** Route(DIDSRV);**** }**** ** ** route[DIDSRV] {**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV type = $avp(Sfeaturetype), index = $avp(SfeatureIndx) $C(xx)\n");**** if($var(srvcount)>3){**** xlog("L_NOTICE","$C(rg)Too many services loops $C(xx)\n");**** update_stat("nts_subs_srv_loop", "+1");**** $avp(TRMCS)="10";**** t_reply(403,"too many loops");**** exit;**** }**** $var(srvcount)=($var(srvcount)+1);**** xlog("L_CRIT","$C(rg) SCRIPT: in DIDSRV before switch to $avp(SfeatureType) or $avp(next_feature_type) $C(xx)\n");**** switch ($avp(SfeatureType)){**** case "0" :**** ……….**** ………**** ** ** On the route "SET" I put the value from the $avp(next_feature_type) in $avp(Sfeaturetype).**** After that I call the route "DIDSRV" and use switch($avp(Sfeaturetype)). Before the switch I xlog the value of the avp twice.**** As you can see in the log below, on the first xlog print, the value is 0 (as I expect). On the second print it resets to <null>**** WHY?**** ** ** Log:**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe6f9aff000, 72) returns address 0x7fe6fcc4d330 frag. 0x7fe6fcc4d300 (size=72) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT: before DIDSRV type = 0, index = 8888888889 #033[0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV type = 0*, index = 8888888889 #033<8888888889%C2%A0%20%23033> [0;39;49m**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 848) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 848) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: CRITICAL: <script>: #033[0;31;42m SCRIPT*: in DIDSRV before switch to <null>* or 0 #033[0;39;49m **** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:369]: qm_malloc(0x7fe7ff765010, 928) called from <core>: rvalue.c: rval_new_empty(236)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:413]: qm_malloc(0x7fe7ff765010, 928) returns address 0x7fe7ff8f25b0 frag. 0x7fe7ff8f2580 (size=1232) on 1 -th hit**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:437]: qm_free(0x7fe7ff765010, 0x7fe7ff8f25b0), called from <core>: rvalue.c: rval_destroy(141)**** Aug 12 13:38:50 net-ivr-KamIN-Test kamailio[29309]: <core> [mem/q_malloc.c:461]: qm_free: freeing frag. 0x7fe7ff8f2580 alloc'ed from <core>: rvalue.c: rval_new_empty(236)

1 1

Kamailio LCR Module test scenario
by Fatima Chahrour~Vanrise Support 12 Aug '12

12 Aug '12

Dears, Am trying to apply Kamailio LCR feature lab test, after am finally able to run Kamailio with no errors, am not able to reach my target clarified in this following scenario: Call 961312345 - route the call to 192.x.x.15 in first place and if failed route the call to 192.x.x.10. I applied needed configuration in lcr tables (attached image) and the routing script is: route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); }; if (is_method("INVITE")) { if (!load_gws(1)) { sl_send_reply("503", "Unable to load gateways"); exit; } } if (uri==myself) { if (method=="REGISTER") { save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using ourUSRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } BUT when attempting a call am getting the message 'Call failed: Not found .' Any help to achieve the successful call using above scenario is highly appreciated. Thanks in advance. F.Chahrour

2 2

postgres in kamailio 3.3.0-9.1
by Andrew O. Zhukov 09 Aug '12

09 Aug '12

The same trouble with: completely updated Centos 5 the last Kamailio RPM from http://download.opensuse.org/repositories/home:/kamailio:/telephony/RedHat_… and postgresql-libs.x86_64.8.1.23-5.el5_8 several month ago I try postgresql84-libs.x86_64 with the same result. It's not possible to use Kamailio RPM with postgres backend. Need to assemble it manually. Bruno Bresciani wrote: > Hi, > > I configure the Kamailio 3.1.2 with postgres but I cann't start. In the > log file is generated the following error: > > // > > ERROR: load_module: could not open module > </home2/local/kamailio/lib/kamailio/modules/db_ > postgres.so>: /home2/local/kamailio/lib/kamailio/modules/db_postgres.so: > undefined symbol: PQdescribePrepared Have you built Kamailio yourself? Looks like db_postgres.so can not find the postgresql libraries, or it was built with a different library version. Maybe ldd can give you some details: ldd /home2/local/kamailio/lib/kamailio/modules/db_postgres.so regards Klaus

4 8

Re: [SR-Users] Problem Starting kamailio on Debian Squeeze [solved]
by JR Richardson 08 Aug '12

08 Aug '12

On Wed, Aug 8, 2012 at 12:13 PM, JR Richardson <jmr.richardson(a)gmail.com> wrote: > Hi All, > > I'm running redundant kamailio 3.0.4 servers in production, have been > for a long time with great success. They were installed on debian > Lenny. One of my servers crashed. I can't seem to do a debian lennyy > install because that version is archived now. I'm trying to install > on debian squeeze but still using kamailio 3.0.4 for compatibility > reasons. The install went pretty normal except during boot up, > kamailio starts before mysql and networking so it starts then exists. > > Once the server is fully booted, kamailio starts with the init script > without error. > > I have adjusted the init script to check for networking and mysql to > start first and performed an 'update-rc.d kamailio defaults' but still > no luck. > > Any help will be appreciated. > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses Yep, this is an http://wiki.debian.org/LSBInitScripts/DependencyBasedBoot issue. Thanks Alex. The old init scrip that ships with Kamailio 3.0.4 that I am using is not 100% compatible so I pulled a hack out of the hat and inserted 'sleep 20' before of 'check_fork ()' in the kamailio init script. Kamailio now waits long enough for all services to start before launching, simple but effective. Thanks. JR -- JR Richardson Engineering for the Masses

3 4

Porblem Starting kamailio on Debian Squeeze
by JR Richardson 08 Aug '12

08 Aug '12

Hi All, I'm running redundant kamailio 3.0.4 servers in production, have been for a long time with great success. They were installed on debian Lenny. One of my servers crashed. I can't seem to do a debian lennyy install because that version is archived now. I'm trying to install on debian squeeze but still using kamailio 3.0.4 for compatibility reasons. The install went pretty normal except during boot up, kamailio starts before mysql and networking so it starts then exists. Once the server is fully booted, kamailio starts with the init script without error. I have adjusted the init script to check for networking and mysql to start first and performed an 'update-rc.d kamailio defaults' but still no luck. Any help will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Kernel Module Issue
by Vijay Thakur 08 Aug '12

08 Aug '12

Thanks for the hint. What is the file name in /etc/modprob.d/ where i am supposed to enter this module name to disable it. Thanks Vijay Tha Message: 1 Date: Mon, 06 Aug 2012 09:01:03 -0400 From: Richard Fuchs<rfuchs(a)sipwise.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<501FC00F.6000604(a)sipwise.com> Content-Type: text/plain; charset="iso-8859-1" On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 1

AVPOPS / TM behavior
by Brandon Armstead 08 Aug '12

08 Aug '12

Hello, I am curious if there is any documentation on how AVP's processing works in the following scenario below. UAC 1 -> KAMAILIO -> KAMAILIO -> DEST It seems that AVP's I set between UAC 1 -> KAMAILIO are lost once I relay back to the same KAMAILIO proxy (self)? Is there any documentation on why or when this would occur? Is there a better way to handle such a scenario? i.e. more dynamic internal routing, vs relaying to self. Thanks as always in advance! Sincerely, Brandon Armstead

2 5

local_timer: add_timeout: 0 expire timer added
by Barthel Marco (CI/AFU1) 08 Aug '12

08 Aug '12

Hi, Occasionally I can see the following log output from Kamailio in our systems: WARNING: <core> [local_timer.c:97]: WARNING: local_timer: add_timeout: 0 expire timer added Can someone explain what it means? Many thanks. Mit freundlichen Grüßen / Best regards Marco Barthel Robert Bosch GmbH (CI/AFU1) www.bosch.com<http://www.bosch.com> Tel. +49 711 811-3602341 marco.barthel(a)de.bosch.com<mailto:marco.barthel@de.bosch.com> Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000; Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Volkmar Denner, Siegfried Dais; Stefan Asenkerschbaumer, Bernd Bohr, Rudolf Colm, Dirk Hoheisel, Christoph Kübel, Uwe Raschke, Wolf-Henning Scheider, Werner Struth, Peter Tyroller

1 0

Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 08 Aug '12

08 Aug '12

As your first solution with UAC module: "As a plan B we instead opted for sending via uac module the invite to an oversip instance (it's open-source since a week or two), which triggers an HTTP request towards APNS, and on kamailio check with inv-timers every few seconds whether the client came online in the meanwhile, then complete the call. The drawback is that you can wait only for so many loop interations due to the max branch limitation in kamailio (you'd need to recompile it to set it higher)." I read about the UAC module but feeling very hard to implement it as your suggestion, http://kamailio.org/docs/modules/3.3.x/modules_k/uac.html Please give me and others more details about this? Which functions those we need ? Are they in: UAC module, Exec module, tm module ?? Thanks

1 0

set_advertised_port using pseudovariable
by Neven Boric 07 Aug '12

07 Aug '12

Hi, I'm trying to use Kamailio as an outbound proxy behind a NAT (all clients and kamailio itself are behind the same NAT): UACs -> Kamailio -> NAT router -> PBX (hosted, public server) I figured I could detect the external source port used by the router by periodically sending an OPTIONS request to the public server, then capture the rport value in the reply, and use that port as an input to set_advertised_port. Because I'm doing this periodically, it should have the added value of keeping the NAT mapping alive. Otherwise, I should be able to detect it, as the rport value will change. I managed to periodically send the OPTIONS request, read the rport value on the reply message and store it on a pseudovariable using: $var(rport) = $sel(v.rport); But when I try to use that value to set the advertised port, I get an error, as if set_advertised_port only accepted literal values: if( $var(rport) != 0) { set_advertised_port($var(rport)); } Not starting Kamailio: invalid configuration file! -e 0(4692) : <core> [cfg.y:3591]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 24-34: syntax error 0(4692) : <core> [cfg.y:3594]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 504, column 35: bad argument, string expected ERROR: bad config file (2 errors) So, am I missing something or does set_advertised_port actually only accept literal values? Thanks Neven Boric

2 2

Re: [SR-Users] sr-users Digest, Vol 87, Issue 18
by Vijay Thakur 07 Aug '12

07 Aug '12

Thanks all of your for reply. Here is the output of command: root@li496-23:~# lsmod | grep -i sip root@li496-23:~# This command is giving blank output. This is linnode server. So the zcat /proc/config.gz command reveals the kernel loaded modules. You can see complete output at pastebin: http://pastebin.com/a0iqpc4D. Thanks in advance Vijay Th. On Monday 06 August 2012 08:26 PM, sr-users-request(a)lists.sip-router.org wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Kernel Droping SIP packet (Richard Fuchs) > 2. Re: Kernel Droping SIP packet (Alex Balashov) > 3. Re: Kernel Droping SIP packet (Konstantin M.) > 4. Re: Kernel Droping SIP packet (Alex Balashov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 06 Aug 2012 09:01:03 -0400 > From: Richard Fuchs <rfuchs(a)sipwise.com> > Subject: Re: [SR-Users] Kernel Droping SIP packet > To: sr-users(a)lists.sip-router.org > Message-ID: <501FC00F.6000604(a)sipwise.com> > Content-Type: text/plain; charset="iso-8859-1" > > On 08/06/12 06:08, Vijay Thakur wrote: >> Hi all, >> >> I have configure Kamailio 3.1.5 Server. All things are working fine. >> When i make a call from Soft phone (X-Lite) to iphone, all is working >> fine. But in other case call from iphone to Softphone is not working, >> even not ringing. During checking the logs i am getting the error: >> >> Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= >> MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 >> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF >> PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 >> RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) > This is coming from nf_conntrack_sip, which is a netfilter connection > tracking kernel module for SIP. I've never used it, but judging from > what Google brings up, it seems to be very buggy. You should be able to > just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work > and/or if you want to keep it from auto-loading, you can blacklist it in > /etc/modprobe.d/ and then reboot. > > HTH > >

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LCR - inconsistent return results for gateway selection.
by Graham Wooden (personal) 06 Aug '12

06 Aug '12

Hello, I am noticing that some of the time (not always consistent) that load_gw() / next_gw() will not always return the gateway with the lowest priority number. For example, when I look up a route for 205595, I have three grp_ids (gateways) that services the destination; with them being separated with 1, 2, and 3 priorities: id prefix from_uri grp_id priority 7468 205595 NULL 1 1 7469 205595 NULL 2 2 7470 205595 NULL 7 3 About 1/2 of the time, it will chose the priority 1 gateway, as it should. But randomly, it will choose the other gateways on the first try. In this particular case, all three gateways are going to the same destination IP (just using different hostnames in the "gw" mySQL table) - so I know the gateways are up. Also, I am not noticing any failures and going to the next gateway, like it choosing #1, it failing and then going to #2. What kind of debugging can I do on this gateway selection process? I am running 1.5.5 notls, and have approximately 425K routes. Kamailio is started with "-m 512" for the additional memory. Thanks, -graham

2 3

Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Can you paste this output too? lsmod | grep -i sip -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Thanks for prompt reply. Here is the out put if command iptables -L -n: Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To: sr-users(a)lists.sip-router.org Message-ID: <17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

2 1

Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

Ah, you short-circuited where I was going with this. Couldn't remember the name of the module. Yep, Vijay. What Richard said. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Richard Fuchs <rfuchs(a)sipwise.com> wrote:On 08/06/12 06:08, Vijay Thakur wrote: > Hi all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. > When i make a call from Soft phone (X-Lite) to iphone, all is working > fine. But in other case call from iphone to Softphone is not working, > even not ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF > PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 > RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is coming from nf_conntrack_sip, which is a netfilter connection tracking kernel module for SIP. I've never used it, but judging from what Google brings up, it seems to be very buggy. You should be able to just unload it by issuing "rmmod nf_conntrack_sip". If that doesn't work and/or if you want to keep it from auto-loading, you can blacklist it in /etc/modprobe.d/ and then reboot. HTH _______________________________________________ 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

1 0

Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH

2 1

Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/phillman25 <phillman25(a)gmail.com> wrote:Hello Alex Will try with SEMS first, found something called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/ if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation? Thanks again! On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/ phillman25 <phillman25(a)gmail.com> wrote: Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

2 1

Re: [SR-Users] Kernel Droping SIP packet
by Vijay Thakur 06 Aug '12

06 Aug '12

Thanks for prompt reply. Here is the out put if command iptables -L -n: *Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination Thanks for prompt reply. This is a Ubuntu 10.04 Kamailio 3.1 Server. Vijay TH -============================================================================== * Date: Mon, 06 Aug 2012 06:35:04 -0400 From: Alex Balashov<abalashov(a)evaristesys.com> Subject: Re: [SR-Users] Kernel Droping SIP packet To:sr-users@lists.sip-router.org Message-ID:<17t1fd0fbly84yy2vuy8qlcp.1344249304618(a)email.android.com> Content-Type: text/plain; charset="utf-8" You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself.? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Vijay Thakur<vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug? 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables :http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH * *

1 0

Re: [SR-Users] B2BUA issues
by Alex Balashov 06 Aug '12

06 Aug '12

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/phillman25 <phillman25(a)gmail.com> wrote:Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

2 1

Re: [SR-Users] Kernel Droping SIP packet
by Alex Balashov 06 Aug '12

06 Aug '12

You might consider pasting the actual output of: iptables -L -n This lists the actual rules straight from netfilter at runtime. I wouldn't worry too much about what some distro-specific config file or script says. Real truth comes from iptables itself. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Vijay Thakur <vijay.thakur(a)loopmethods.com> wrote:Hi all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) I have not implemented any firewall. You can check the out put of my iptables : http://pastebin.com/i3zUfVeb The SIP server is hosted at linnode. With thanks in advance. Sorry dual posting. Vijay TH _______________________________________________ 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

1 0

Re: [SR-Users] B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Hi Alex Thanks for your prompt reply. The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work. Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios? >From your experience what do you think would be a better solution? Thanks again! Phillip ======================== Message: 2 Date: Mon, 06 Aug 2012 04:26:28 -0400 From: Alex Balashov <abalashov(a)evaristesys.com> Subject: Re: [SR-Users] B2BUA issues To: sr-users(a)lists.sip-router.org Message-ID: <501F7FB4.8040700(a)evaristesys.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed What is the larger objective? Are you using the PGW purely as a B2BUA? If so, that's a colossally overblown waste of resources; just use something like SEMS or Asterisk. On 08/06/2012 04:24 AM, phillman25 wrote: > Dear List > > I am trying to accomplish the following: > > > Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco > PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> > Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) > > When trying the above scenario, the call is silent and drops after a few > seconds. In syslog i observe the following error: > > *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing > name-addr (sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060>>)* > > Looking at the sip trace i see that his might be caused by the ACK > message received from the ASTERISK PABX? : > > ACK sip:22030305@192.168.10.189:5060 > <http://sip:22030305@192.168.10.189:5060> SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport > Route: > <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b. 66623da5>,<sip:22030305@ > yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx; lr=on;ftag=as166b1eea> > Max-Forwards: 70 > From: "22498045" <sip:22498045@192.168.10.189 > <mailto:sip%3A22498045@192.168.10.189>>;tag=as166b1eea > To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 > Contact: <sip:22498045@192.168.10.189:5060 > <http://sip:22498045@192.168.10.189:5060>> > Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 > <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060> > CSeq: 102 ACK > User-Agent: FPBX-2.8.1(1.8.12.0) > Content-Length: 0 > > > After contacting Cisco they informed us that issue is cause by B2BUA > from our current version of Cisco PGW 2200 that doesn't support this > feature. Is there a module, solution that i can implement on Kamailio > that could temporarily resolve this issue? > > Thanking you in advance. > > Phillip

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B2BUA issues
by phillman25 06 Aug '12

06 Aug '12

Dear List I am trying to accomplish the following: Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Cisco PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189) When trying the above scenario, the call is silent and drops after a few seconds. In syslog i observe the following error: *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing name-addr (sip:22030305@192.168.10.189:5060>)* Looking at the sip trace i see that his might be caused by the ACK message received from the ASTERISK PABX? : ACK sip:22030305@192.168.10.189:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport Route: <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@ yyy.yyy.yyy.yyy ;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea> Max-Forwards: 70 From: "22498045" <sip:22498045@192.168.10.189>;tag=as166b1eea To: <sip:22030305@xxx.xxx.xxx.xxx>;tag=as6d578713 Contact: <sip:22498045@192.168.10.189:5060> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060 CSeq: 102 ACK User-Agent: FPBX-2.8.1(1.8.12.0) Content-Length: 0 After contacting Cisco they informed us that issue is cause by B2BUA from our current version of Cisco PGW 2200 that doesn't support this feature. Is there a module, solution that i can implement on Kamailio that could temporarily resolve this issue? Thanking you in advance. Phillip

2 1

Question about Proxy-Authorization header in ACK
by Ivan Milivojevic 06 Aug '12

06 Aug '12

Hi all, I have a one question regarding ACK and Proxy-Authorization header. I am testing Kamailio 3.3 as SIP proxy,default config. I made one call with 2 phones where one has Public IP while other is behind NAT. SIP clients are Panasonic KX-UT133 and other is 1-NET (ex Sweden Mobile&CDMA provider ,over 50M users) Everything is working just fine, both RTP and SIP. On the other hand one of the clients,1-NET one, send Proxy-Authorization in ACK when he receives 200 OK from Kamailio. PUBLIC_IP_USER1 - user 1000 PUBLIC_IP_USER2 - user 1001 KAMAILIO ==> 1-NET SIP/2.0 200 OKr\n Record-Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport=5064;branch=z9hG4bK934894606 Call-ID: 221374358 From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 CSeq: 21 INVITE Allow: INVITE,ACK,CANCEL,BYE,INFO,UPDATE,OPTIONS,NOTIFY,REFER Contact: <sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027> Require: replaces Content-Type: application/sdp Server: Panasonic_KX-UT133NE/01.081 (0080f0cedd83) Content-Length: 182 v=0 o=- 1343736535 1343736535 IN IP4 KAMAILIO_PUBLIC_IP s=- c=IN IP4 KAMAILIO_PUBLIC_IP t=0 0 m=audio 20412 RTP/AVP 18 a=rtpmap:18 G729/8000 a=sendrecv a=ptime:20 a=nortpproxy:yes 1-NET ==> KAMAILIO ACK sip:1001@ <sip:1001@PUBLIC_IP_USER2:1027> PUBLIC_IP_USER2:1027 SIP/2.0 Via: SIP/2.0/UDP PUBLIC_IP_USER1:5064;rport;branch=z9hG4bK642028490 Route: <sip: <sip:KAMAILIO_PUBLIC_IP;lr=on;nat=yes> KAMAILIO_PUBLIC_IP;lr=on;nat=yes> From: blabla <sip:1000@ <sip:1000@KAMAILIO_PUBLIC_IP> KAMAILIO_PUBLIC_IP>;tag=507511069 To: <sip:1001@ <sip:1001@KAMAILIO_PUBLIC_IP:5060> KAMAILIO_PUBLIC_IP:5060>;tag=2148378512 Call-ID: 221374358 CSeq: 21 ACK Contact: <sip:1000@ <sip:1000@PUBLIC_IP_USER1:5064> PUBLIC_IP_USER1:5064> [truncated] Proxy-Authorization: Digest username="1000", realm="KAMAILIO_PUBLIC_IP", nonce="UBfL8lAXysb5tJCs80ZnthyPl9IzmRZk", uri="sip:1001@KAMAILIO_PUBLIC_IP:5060", response="7824519cdad9f1c2c79027a2d7522344", algorithm=MD5, cnonce="0a4f113b", q Max-Forwards: 70 User-Agent: Serbia_2.00 Content-Length: 0 I attach txt file with call flow, I can send pcap also. I think that the issue is related with bad client but I need another opinion. Does anyone has an idea about this issue? Best Regards, Ivan

2 1

REGISTER replication to multiple hosts
by Henning Holtschneider 06 Aug '12

06 Aug '12

Hello, I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch("sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this: add_sock_hdr("Local-Sock"); add_rcv_param(); append_branch(); seturi(sip:1.1.1.1:5060"); t_replicate("sip:2.2.2.2:5060"); exit; Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction? Thanks, Henning Holtschneider -- LocaNet oHG - http://www.loca.net Lindemannstrasse 81, D-44137 Dortmund tel +49 231 91596-25, fax +49 231 91596-55 sip 25(a)voip.loca.net Registergericht Amtsgericht Dortmund HRA 14208 Geschäftsführer Sven Haufe, Henning Holtschneider

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Re: [SR-Users] sr-users Digest, Vol 87, Issue 9
by Vijay Thakur 06 Aug '12

06 Aug '12

Hi Konstantin, Thanks for your kind reply. You can check the iptables output at : http://pastebin.com/i3zUfVeb. I hope that this will give you enough clue in right direction. With Regards, Vijay Thakur ====================================================================================================================== Message: 3 Date: Fri, 3 Aug 2012 17:38:23 +0300 From: "Konstantin M."<evilzluk(a)gmail.com> Subject: Re: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Message-ID: <CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Looks like your firewall is passing a state RELATED,ESTABLISHED and don't have a permit rule for a state NEW. Check your iptables or simply do a command: iptables-save > /tmp/iptables.txt and paste this file (/tmp/iptables.txt) to pastebin 2012/8/3 Vijay Thakur<vijay.thakur(a)loopmethods.com> > Hello all, > > I have configure Kamailio 3.1.5 Server. All things are working fine. When > i make a call from Soft phone (X-Lite) to iphone, all is working fine. But > in other case call from iphone to Softphone is not working, even not > ringing. During checking the logs i am getting the error: > > Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= > MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 > DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP > SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK > URGP=0 OPT (0101080A000E20610932B25A) > > This is server is hosted on Linnode. > > Kindly guide me to solve the problem > > -- > Best Regards, > > Vijay Thakur > (Assistant Manager - Networks) > Mobile : +91 8744018065 > Mail :vijay.thakur@loopmethods.com

1 0

Gorka left a message for you
by Badoo 05 Aug '12

05 Aug '12

Gorka left a message for you Only you can see the sender and content of your message, and you can delete it anytime. You can instantly reply using our message exchange system: http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322… Some other people in the area who are on Badoo Ali (Jeddah, Saudi Arabia) Fnoo meni jnoo (Jeddah, Saudi Arabia) مجووووودي (Jeddah, Saudi Arabia) http://eu1.badoo.com/0106730996/in/UJF4Z.6PYhQ/?lang_id=3&m=63&mid=501e9322… If the link in this message does not work, try copying and pasting it into your browser. This email is part of our delivery procedure for the message sent by Gorka. If you have received this email by mistake, please ignore it. The message will be deleted soon. Have fun! The Badoo Team You have received this email from Badoo Trading Limited (postal address below). http://eu1.badoo.com/impersonation.phtml?lang_id=3&email=serusers%40iptel.o… Badoo Trading Limited is a limited company registered in England and Wales under CRN 7540255 with its registered office at 12 Red Lion Square, London, WC1R 4QD.

1 0

Getting started with Kamailio
by Markus 05 Aug '12

05 Aug '12

Hello! I'm new to this list and Kamailio. First, thanks to all the programmers that developed this software. I'd like to ask several questions on how to get started with Kamailio and the basic understanding of it. I'm trying to figure out Kamailio only since yesterday. Right now I have a single Asterisk server that handles signaling and media. For accounting and user administration I'm using a2billing with MySQL on it. For redundancy and scalability I would like to create the following setup: France: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. Spain: BGP anycast 195.5.5.0/24 2 nodes with Kamailio (IP 195.5.5.5) active + standby 2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster. (a2billing MySQL DB will be always synchronous over all nodes in Spain and France) This will give me: - Network redundancy (data center burns down in France, I stay online in Spain). This will be handled by BGP anycast. - Signaling redundancy (Kamailio active + standby). This will be handled by Linux-HA or something like that. - Media redundancy (2x Asterisk per country). This will be handled by Kamailio dispatcher. - Application/database redundancy (2x a2billing per country + synchronous MySQL DB everyhwere). This will be handled by DRBD and MySQL. = High redundancy, I can sleep and it can scale. What I would like to achieve besides the above: - Give only a single IP address to all customers and termination providers (the same IP address), 195.5.5.5. I'm a SIP noob, so I have to ask: - How do I do the Kamailio part? ;-) ... I have seen this how-to for Kamailio/Asterisk realtime: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb But it feels like "overkill". If I would not have any users that actually REGISTER (e.g. in a pure wholesale termination environment), I would not need Kamailio/Asterisk realtime integration. Correct? - Is there no way around changing the Asterisk side (activating realtime, new MySQL DB) when I have users that do need to REGISTER? If I would not be using a2billing I could probably handle all the registrations in Kamailio only? - Since Kamailio and Asterisk will not be on the same box, what is the recommended way for Kamailio securely communicating with the MySQL database on the Asterisk server? Does Kamailio support SSL with MySQL? - If I use RTPproxy on the Kamailio server, every customer and termination provider would connect to 1 single IP address, because both media and signaling comes from that IP. Correct? - If I don't use RTPproxy, and have canreinvite=yes on my Asterisk servers, customers would get the media, when placing PSTN calls, directly from my termination providers (I would like to avoid that). Correct? - If I don't use RTPproxy, and have canreinvite=no on my Asterisk servers, customers will get the media directly from my Asterisk servers, but termination providers that filter based on IP addresses they would have to allow all Asterisk IP addresses in their filters (same for customers, actually). Correct? Right now I didn't have to worry about such things because media and signaling were handled by a single Asterisk box with canreinvite=no everywhere. - Last question for now: why does it seem like important developers of SIP software such as Kamailio and yate are originating from Romania and are female? Just a coincidence? :-) Regards Markus

2 2

RTP over TCP
by Techie Sup 05 Aug '12

05 Aug '12

Hi All, Is there any possibility of sending RTP over TCP, because we are building a softphone based on PJSIP, but seems that some ISP are blocking RTP packets over UDP. Any solutions for this. Regards Gloria

2 3

kernel: nf_ct_sip: dropping packetIN= OUT=eth0 Error
by Vijay Thakur 03 Aug '12

03 Aug '12

Hello all, I have configure Kamailio 3.1.5 Server. All things are working fine. When i make a call from Soft phone (X-Lite) to iphone, all is working fine. But in other case call from iphone to Softphone is not working, even not ringing. During checking the logs i am getting the error: Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT= MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77 DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A) This is server is hosted on Linnode. Kindly guide me to solve the problem -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

2 1

302 redirect with 2 numbers registered
by Mino Haluz 03 Aug '12

03 Aug '12

Hi, one number is registered on 2 phones. Phone1 has Always redirect set to another number. When incoming call is initiated, Phone2 is ringing and Phone1 sends 302 to the proxy. However the proxy does not send 302 to the caller (for ex. GW), but it waits for timeout of the Phone2. Then the proxy sends 302 to the caller. Can I do in kamailio, that it will ring on the Phone1 and also on the number where it is redirected? I know kamailio is a proxy and cannot initiate a call, but is there any solution? Thanks. Mino

2 1

Question on ACK behavior
by Varsha Venkatraramani 03 Aug '12

03 Aug '12

** ** * * ** ** Hi, ** ** I have a question regarding the below ACK response Kamilaio receives from a carrier. Can someone please help me understand why the “*Route: sip:callmanager@192.168.160.43:5060> *sent from Kamailio is missing a “<” as shown in the captures below? Our Internal proxy is treating that as a malformed header and dropping the packet. ** ** ** ** *ACK from CARRIER* * * U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060**** *ACK sip:callmanager@192.168.160.43:5060 SIP/2.0.* Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 70.**** Route: <sip:192.168.160.47:5060;lr=on>.**** *Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060>.* Content-Length: 0.**** .**** *ACK FORWARDED TO SIP PROXY* ** ** U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060**** *ACK sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060 SIP/2.0.* Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.**** Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.**** From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.**** To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.**** Call-ID: 295226_50030734(a)4.55.18.227.**** CSeq: 18079 ACK.**** Max-Forwards: 69.**** Content-Length: 0.**** *Route: sip:callmanager@192.168.160.43:5060>.* I have attached the config file for you reference. Kamailio version is 3.2.3 **** ** ** Thank You**** Varsha****

2 2

centos (fedora) service startup file
by andre 02 Aug '12

02 Aug '12

Somwhere losted since openser 1.0 file kamailio.init Please fix in function start: start() { echo -n $"Starting $prog: " ..... [ $RETVAL = 0 ] && touch something to [ $RETVAL = 0 ] && touch someting && success I mean call function success in the end of the expression. Thanks in advance.

1 0

Kamailio v3.3.1 Released
by Daniel-Constantin Mierla 02 Aug '12

02 Aug '12

Hello, Kamailio SIP Server v3.3.1 stable release is out. This is a maintenance release of the latest stable branch, 3.3, that includes fixes since release of v3.3.0. There is no change to database schema or configuration language structure that you have to do on installations of v3.3.0. Deployments running previous v3.x.x versions are strongly recommended to be upgraded to v3.3.1. For more details about version 3.3.1 (including links and hints to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2012/08/kamailio-v3-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Outbound and Registrar. No binding update
by José Luis Millán 02 Aug '12

02 Aug '12

Hi, I'm using Kamilio 3.3.0 as registrar server. I`m using an outbound proxy so 'use_path' parameter or 'registrar' module is enabled. According to RFC 5626, a re-registration from a specific combination of AoR, instance_id and reg_id must update the binding. "" If the registrar receives a re-registration for a specific combination of AOR, and instance-id and reg-id values, the registrar MUST update any information that uniquely identifies the network flow over which the request arrived if that information has changed, and SHOULD update the time the binding was last updated. "" In my installation this is not fullfilled as shown: "" AOR:: jmillan Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=a570655c14 Q= Expires:: 181 Callid:: 1hgq3khalq2rzfr Cseq:: 108 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:a570655c14@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-328e-1 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 Contact:: sip:jmillan@MY_IP;transport=ws;ov-ob=1dd97b4d51 Q= Expires:: 193 Callid:: 1hgq3khalq2rzfr Cseq:: 110 User-agent:: JsSIP 0.1.0 Path:: <sip:OUTBOUND_IP:9090;transport=tcp;lr;ovid=de0c0b9f>,<sip:1dd97b4d51@OUTBOUND_IP :10080;transport=ws;lr;ovid=de0c0b9f;ob> State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tcp:KAM_IP:5060 Methods:: 783 Ruid:: uloc-50119d99-3290-2 Instance:: <urn:uuid:38dce009-ae1f-4fd1-91dc-99ed9affddc1> Reg-Id:: 1 "" The scenario is such that when the client looses the connection with the Outbound Server, it reconnects and re-registers to Kamailio in order to replace registration and be able to receive in-dialog messages. Any comment is wellcome. Regards,

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Re: [SR-Users] Regarding - Add your own module
by Alex Balashov 02 Aug '12

02 Aug '12

Does TimesTen have a UnixODBC connector? If so, you can use existing Kamailio modules. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Ananth Kollipara <infoananthk(a)gmail.com> wrote:Hi Alex, Thanks for the response. Yes, i meant developing of a custom module and integrating with ser/kamailio. I am looking for the following. 1) On receiving 302 response, proxy shall use the contact header and re-send the msg to the correct destination. 2) Integration of kamailio with Times Ten database (query Times ten DB) Please advice. Regards, Ananth On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request(a)lists.sip-router.org> wrote: Send sr-users mailing list submissions to sr-users(a)lists.sip-router.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.sip-router.org You can reach the person managing the list at sr-users-owner(a)lists.sip-router.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Add Own Plug-in (Ananth Kollipara) 2. Re: Add Own Plug-in (Alex Balashov) ---------------------------------------------------------------------- Message: 1 Date: Tue, 31 Jul 2012 14:33:52 +0530 From: Ananth Kollipara <infoananthk(a)gmail.com> Subject: [SR-Users] Add Own Plug-in To: sr-users(a)lists.sip-router.org Message-ID: <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS=ZQ(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, Can you please provide the steps to add a plug-in? Regards, Ananth

2 9

planning release of v3.3.1
by Daniel-Constantin Mierla 02 Aug '12

02 Aug '12

Hello, I am planning to package v3.3.1 out of the latest branch 3.3 this Thursday, August 2. If you are aware of issues that are not listed on the tracker (http://sip-router.org/tracker/), add them there to see what can be sorted out before. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

Regarding - Add your own module
by Ananth Kollipara 01 Aug '12

01 Aug '12

Hi Alex, Thanks for the response. Yes, i meant developing of a custom module and integrating with ser/kamailio. I am looking for the following. 1) On receiving 302 response, proxy shall use the contact header and re-send the msg to the correct destination. 2) Integration of kamailio with Times Ten database (query Times ten DB) Please advice. Regards, Ananth On Tue, Jul 31, 2012 at 3:30 PM, <sr-users-request(a)lists.sip-router.org>wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Add Own Plug-in (Ananth Kollipara) > 2. Re: Add Own Plug-in (Alex Balashov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 31 Jul 2012 14:33:52 +0530 > From: Ananth Kollipara <infoananthk(a)gmail.com> > Subject: [SR-Users] Add Own Plug-in > To: sr-users(a)lists.sip-router.org > Message-ID: > <CACJ9FatODfYWEG+y6_U_=-xxsxt65Ca=5FV2NFBQrYw+grS= > ZQ(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > Can you please provide the steps to add a plug-in? > > Regards, > Ananth >

1 0

Re: [SR-Users] Segfault running 3.3.0 with registrar/p_usrloc/path
by DanB 01 Aug '12

01 Aug '12

Hey Guys, Just a quick follow up to see if anything happened with p_usrloc fixup. Any chance to expect it running on new 3.3.1 release? Ta, DanB

3 2

cannot allocate memory & increase shared memory
by Alex Solt 01 Aug '12

01 Aug '12

Hi, We are getting "cannot allocate memory" error in the log file. I was wondering which one is the right way to increase the shared memory: 1) adding the following to /etc/openser/openserctl STARTOPTIONS="-m 128" 2) modify the following in /etc/default/openser MEMORY=1024 Thanks,AS

2 1

New Wiki Site has problems?
by Andrew Pogrebennyk 31 Jul '12

31 Jul '12

At least some links at http://www.kamailio.org/wiki/ don't work at the moment: Install Kamailio v3.3.x From GIT Upgrade Kamailio v3.1.x to v3.2.0 Upgrade Kamailio v3.2.x to v3.3.0 you are redirected to some instruction from DokuWiki Installer when trying to visit them. Could somebody please check it? Thanks. Andrew

2 1

routing based on username
by Richard Zheng 31 Jul '12

31 Jul '12

Hi, We are trying to follow http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdbto setup Kamailio as front end and multiple PBXs as back end. We need to forward sip messages to the respective servers based on the contact. For example, users 101, 102 go to PBX at 1.1.1.1 and users 201, 202 go to PBX at 1.1.1.2. Where is a good place to save this info and how to route them in the config? Also we have some users with fixed IP addresses which don't use authentication. How do we route the calls based on source IP? Since we have plenty of these users, we'd like to save them in database too. Any advice is appreciated! Richard

3 2

Add Own Plug-in
by Ananth Kollipara 31 Jul '12

31 Jul '12

Hi, Can you please provide the steps to add a plug-in? Regards, Ananth

2 1

ACK not relayed
by David Notivol 31 Jul '12

31 Jul '12

In a UAC-Kamailio-UAS scenario, we've found a case where the ACK coming from uac is not relayed by our proxy to the uas. This is the log for the ACK message: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:624]: SIP Request: Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:626]: method: <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:628]: uri: <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:630]: version: <SIP/2.0> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 235, <rport> = <n/a>; state=6 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:1286]: Found param type 232, <branch> = <z9hG4bKKc4gKXyemS9HD>; state=16 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_via.c:2561]: end of header reached, state=5 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:511]: parse_headers: Via found, flags=2 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers: this is the first via Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:149]: After parse_msg... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [receive.c:190]: preparing to run routing scripts... Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sl [sl_funcs.c:396]: DEBUG : sl_filter_ACK: to late to be a local ACK! Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: maxfwd [mf_funcs.c:85]: value = 65 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=6ne1x6d6-CC-23 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:188]: DEBUG: get_hdr_field: <To> [52]; uri=[sip:59138553189@79.170.68.157] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:190]: DEBUG: to body [ <sip:59138553189@79.170.68.157> <sip:59138553189@79.170.68.157>] Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:168]: get_hdr_field: cseq <CSeq>: <31362871> <ACK> Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:202]: DEBUG: get_hdr_body : content_length=0 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/msg_parser.c:104]: found end of header Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:178]: DEBUG: add_param: tag=0cS8trtaF196F Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: <core> [parser/parse_to.c:802]: end of header reached, state=29 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: sanity [mod_sanity.c:255]: sanity checks result: 1 Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: siputils [checks.c:106]: totag found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: ESTAMOS EN WITHIN Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:108]: No Route headers found Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: DEBUG: rr [loose.c:829]: There is no Route HF Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]: ERROR: <script>: NO LOOSE ROUTE However, in a pcap trace I clearly see the Route header: This is the Record-Route in the 200 message Record-Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> And this is the Route header in the ACk: Route: <sip:79.170.68.157;lr;ftag=330cdf48-co1022-INS033> I haven't been able to figure out what's going on. We have plenty of traffic working in that proxy. What am I missing? Thanks in advance David.

2 8

BYE Race Condition
by Brandon Armstead 31 Jul '12

31 Jul '12

Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

2 2

Error 403 not relaying
by Tayeb Meftah 30 Jul '12

30 Jul '12

hello, i'm using Kamailio with a simple registrare configuration i used the apt repository to install the developmant package of kamailio domain sip.tmvoip.net have 2 records, A record and AAA record (dual stacked) i configured it in my kamailio using siremis and in the kamctlrc config below: Kamailio.cfg: http://paste.debian.net/180972/ Kamctlrc: http://paste.debian.net/180973/ siptrace: http://paste.debian.net/180974/ note that 172.28.0.0/21 and 41.221.20.12/30 is routed, no nat. my current issue is if i register or i invite, i get error 403 not relaying trying to investigate this issu but i have no clue thank you everyone, -- Voice of the blind ™ broadcast freedom Phones: Algeria: +213-660-347746 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

3 3

Re: [SR-Users] Can't initialize shared memory
by Uri Shacked 30 Jul '12

30 Jul '12

> > this is the log when debug is 4... > > Jul 24 10:19:37 RNDSRV kamailio: : <core> [cfg.y:3532]: parse error in > config file /usr/local/etc/kamailio/global_params.cfg, line 121, column 54: > Can't initialize shared memory > Jul 24 10:19:37 RNDSRV kamailio: ERROR: mi_fifo [mi_fifo.c:271]: FIFO stat > failed: Bad address > > > not much. > line 121 on the file is the modparam for mi_fifo indication the > /tmp/kamailio_fifo file. > > # ----- mi_fifo params ----- > modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") > any ideas? > > >

2 1

Reply to one topic correctly
by Duong Manh Truong 30 Jul '12

30 Jul '12

Hi list, I am a newbie of the maillist, I'm wondering that whenever i reply to a specific topic with the subject of "RE: [SR-Users]....", my email come to maillist in the way dislike what i wish, my email appears like a NEW topic instead of SUB topic (reply message form) I try several ways but still not get the correct method of reply, please help me to fix this, thanks.

2 1

Re: [SR-Users] Kamailio with APNS
by Duong Manh Truong 29 Jul '12

29 Jul '12

Thank Klaus, I've try with t_suspend() and t_continue() as your guide, *#Failure route to react with negative messages failure_route[1] { if (t_check_status("5[0-9][2-9]")) { if(t_suspend()) { xlog("proceesing suspended in trasaction [$T(id_index):$T(id_label)]\n"); xlog("before sleep \n"); } if(async_sleep("20")) xlog("After sleep 20s \n"); if(t_continue('1','1','MYROUTE')) xlog("Continuation successfully \n"); exit; };* And i found that t_suspend() and async_sleep() runnung ok but t_continue() seems not running. I read about TMX module and TMX module Pseudo-Variables but still can not figure out it :( Secondly, in case of t_continue() runs successfully , please tell me how to let Kamailio resend the INVITE message to client B (before the t_continue(), i think so ) - while client B is waking up ? Which functions and modules that i need to do this? Thanks. ----------------------------------------------------------------- Hi! You have to use a failure route (http://kamailio.org/docs/modules/3.3.x/modules/tm.html#t_on_failure). In the failure route check if the reply code is 404 and if yes, trigger the APNS logic and then put the call e.g. on a stack and resume e.g. every second and ofrward to the "SIP server" again to see if the client is registered meanwhile. For the asynchronous processing take a look at these functions: t_suspend(),t_resume()[1]. For example you can call t_resume in a timer route every second. For the timer route, to know which transactions are currently suspended and are waiting for further processing, you could put the transaction identifiers in a message-queue [2] and read the queue from the timer route. regards Klaus [1] http://kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 [2] http://kamailio.org/docs/modules/3.3.x/modules/mqueue.html On 24.07.2012 06:02, Duong Manh Truong wrote: >* Hi list,*>**>* My team are developing a SIP client for mobile running iOS ,*>**>* then having to deal with the "back ground mode" problem while this soft*>* is installed on mobiles*>**>* One of the solutions is using APNS. (Apple Push Notification Service)*>**>* I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server*>**>* The model is like that:*>**>* User A <---------> Kamailio <---------> SIP server <--------> User B*>**>* IF UserA call UserB, UserB is not available now (not registered yet),*>* normally, SIP server will reply Kamailio with "404 Not found"*>**>* In that case, i want to set up Kamailio do the communication with APNS*>* and waiting for UserB "Wake up" then RESEND the Invite message ......*>* (instead of forwarding this 404 message to UserA)*>**>* I am getting stuck at how Kamailio can control "404 Not found" message*>* which it receives from SIP server .*>**>* I do searching around several days but still can not get any clearer*>* points.*>**>* Please help me with some instruction/ configuration of Kamailio for this*>* case !*>**>* Thank you very much.*>**>**>* _______________________________________________*>* SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list*>* sr-users at lists.sip-router.org <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>*>* http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users*>**

1 0

How to change _only_ next hop and dpn't touch content of message?
by intel＠intrans.baku.az 29 Jul '12

29 Jul '12

I want retransmit some me ssages to other sip server as is. f.e server sip1.com receives invite message INVITE user(a)somewhere.org i want to send it to server sip2.com as is. Unfortunately, if i use rewritehostport() function, or modify $rp $rd pseudovariables fe $rp='sip2.com', kamailio changes invite uri, and send to sip2.com INVITE user(a)sip2.com The To: filed left untouched How can I resend message without modifications? I've tried also modify $dp and $dd pseudovariables, but then message wasn't sent anywhere,and in log I can see this message: Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [t_fwd.c:1523]: ERROR: t_forward_nonack: no branches for forwarding Jul 27 15:44:54 dsdb01 /usr/sbin/kamailio[17786]: ERROR: tm [tm.c:1375]: ERROR: w_t_relay_to: t_relay_to failed What else can I try? --

2 1

BYE Race Condition
by Brandon Armstead 28 Jul '12

28 Jul '12

*** Please forgive if this is a duplicate *** Hello, I am running into an issue where there is a race condition happening. I am looking for opinions / ideas on how to handle the following below scenario. Scenario. UAC places an outbound call -> upstream carrier. The call is disconnected on both ends at the exact same time, UAC -> sends BYE upstream Upstream Carrier -> sends BYE downstream Upstream 200 OK's the BYE UAC sends 481 back to Upstream Carrier for their generated BYE. The upstream carrier is complaining about receiving the relayed 481 responses -- so my first thought was simply to drop() these from relaying upstream. I am curious how other people are handling this? Would you suggest simply dropping the relay from being sent back upstream on the 481? Would you simply always 200 OK a downstream BYE from trusted carriers regardless of UAC response, and create separate transaction to send BYE downstream? Thank you as always. Look forward to your thoughts / suggestions / ideas. Sincerely, Brandon Armstead

1 0

lcr.dump_gws problem
by Gary Chen 27 Jul '12

27 Jul '12

Kamailio 3.2.0 When testing Kamailio 3.2.0, I notice that lcr.dump_gws command does not display ip_addr correctly. Here is my output: lcr_id: 3 gw_id: 15 gw_index: 2 gw_name: gateway_1 scheme: sip ip_addr: 1273060816.0.0.0 hostname: port: 5060 params: strip: 0 prefix: tag: flags: 0 defunct_until: 0 LCR is still working correctly. It just does not display IP in the right form. Does any body also has the same problem? Gary

3 4

Re: [SR-Users] Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

1 0

Add CallerID to SIP header
by Dominic Raferd 27 Jul '12

27 Jul '12

I'm not yet using Kamailio. I am looking for something to do a job which sounds really simple but may not be. Our PBX connects to a SIP service (voicetrading.com) but does not supply our CallerID, this means that voicetrading.com provide some very weird callerids for the person at the other end (international numbers etc). Which is not good. There is no way to get our PBX to add CallerID itself, so I thought maybe I could set up a local sip proxy? or sip redirect? server (Kamailio?) to do nothing other than add in the callerid details to the sip header and then forward the sip message on to voicetrading i.e. be otherwise 'invisible'? If Kamailio is not the right thing to do this with, is there something else that could do it? Thanks for any suggestions. Dominic

2 1

Cant make from_uri, request_uri work for LCR module
by Antanas Masevicius 27 Jul '12

27 Jul '12

Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT

3 5

Re: [SR-Users] Cant make from_uri, request_uri work for LCR module
by Alex Balashov 26 Jul '12

26 Jul '12

When calling load_gws(), try putting $rU and $fu in double quotes, e.g. "$rU" rather than $rU bare. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/Antanas Masevicius <antanas.masevicius(a)ntt.lt> wrote:Hello, having trouble running LCR in kamailio 3.3 (head). Here is my DB excerpt which i use to reproduce the problem: INSERT INTO `lcr_rule` VALUES (2617960,90,'118','.*',1,1,'.*'); INSERT INTO `lcr_rule_target` VALUES (7771454,90,2617960,1424,0,100); INSERT INTO `lcr_gw` VALUES (1424,90,'OP1','10.10.10.10','',5060,NULL,1,0,NULL,'11#',1,NULL,NULL); I stripped all other LCR data from my DB and only this rule with from_uri and request_uri seems to bother me. Tried this version on debian lenny and debian squeeze. On Lenny i am not able to load lcr tables at all, get a segfault. On squeeze, lcr tables loads up, but this rule gets marked as disabled. Here is config part which i use to load a rule: if (load_gws(90, $rU, $fu)) { if (next_gw()) { t_on_failure("FAIL_GW"); } else { route(NO_ROUTE); } On squeeze, if i add following code to lcr_mod.c:1249 : LM_INFO("LOADING lcr rule <%u> \n", rule_id); right after: rule_id = (unsigned int)VAL_INT(ROW_VALUES(row)); rule_id is still printed, but i get segfault like this: Jul 24 10:17:43 sp1 kernel: [23482626.105865] kamailio[15136]: segfault at a4 ip 00007fde15cb8802 sp 00007fff39d68320 error 4 in lcr.so[7fde15ca8000+23000] Not sure what i do terribly wrong. Setting from_uri and request_uri to anythig but NULL makes the code to segfault with that print line added. Without print line, kamailio starts, but never loads that rule. Any help would be appreciated. best regards, Antanas Masevicius NTT _______________________________________________ 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

5 8

siremis errors
by Tony Ferrario 26 Jul '12

26 Jul '12

I just installed a Siremis 3.2.1 on a server with an already working Kamailio 3.2.1. bellow is a typical error message I keep getting. Can you point me to a solution -Thanks [http://192.168.105.196/siremis/themes/default/images/icon_forbidden.gif] System Internal Error The detailed error message are below: [2012-07-26 19:14:11 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.method, T0.from_tag, T0.to_tag, T0.callid, T0.sip_code, T0.sip_reason, T0.time, T0.src_ip, T0.dst_user, T0.dst_domain, T0.dst_ouser, T0.src_user, T0.src_domain, T0.cdr_id FROM `acc` T0 WHERE method='INVITE' AND T0.src_user='admin' LIMIT 10. SQLSTATE[42S22]: Column not found: 1054 Unknown column 'T0.src_ip' in 'field list' Script name and line number of error: /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php:447 function: _run_search ( Array(2) ) @ /var/www/siremis-3.2.1/openbiz/bin/data/BizDataObj_Lite.php 243 function: fetch ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 690 function: fetchDataSet ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 69 function: renderSmarty ( Object(EasyForm), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/grid.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/FormRenderer.php 45 function: render ( Object(EasyForm) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1648 function: renderHTML ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyForm.php 1538 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 83 function: renderSmarty ( Object(EasyView), "/var/www/siremis-3.2.1/siremis/modules/sipuser/template/view.tpl" ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/ViewRenderer.php 40 function: render ( Object(EasyView) ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 274 function: _render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/easy/EasyView.php 237 function: render ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 221 function: renderView ( "sipuser.view.AccListView", "", "", Null, "" ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 107 function: dispatchRequest ( ) @ /var/www/siremis-3.2.1/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/siremis-3.2.1/openbiz/bin/BizController.php" ) @ /var/www/siremis-3.2.1/siremis/bin/controller.php 6 function: include ( "/var/www/siremis-3.2.1/siremis/bin/controller.php" ) @ /var/www/siremis-3.2.1/siremis/bin/_forward.php 102 function: include ( "/var/www/siremis-3.2.1/siremis/bin/_forward.php" ) @ /var/www/siremis-3.2.1/siremis/index.php 3 ________________________________ Please ask system administrator for help... Tony Ferrario Manager SVT & Ops OnMobile Global Limited 3950 Civic Center Drive , Suite 200 San Rafael, Ca 94903 Phone: 415-223-8206 Cell: 707-508-7285 tony.ferrario(a)onmobile.com ________________________________ DISCLAIMER: The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message in error. Further, this e-mail may contain viruses and all reasonable precaution to minimize the risk arising there from is taken by OnMobile. OnMobile is not liable for any damage sustained by you as a result of any virus in this e-mail. All applicable virus checks should be carried out by you before opening this e-mail or any attachment thereto. Thank you - OnMobile Global Limited.

1 0

Overload Control
by Kaplan, Michael A 26 Jul '12

26 Jul '12

Folks, I'm trying to figure out if either Kamailio or OpenSIPs provides the various overload control algorithms recommended in the IETF or elsewhere. I haven't found documentation to this effect. Has anyone looked into this or used such techniques? Thanks, Mike

3 6

rtimer execution period - round two
by Alex Balashov 25 Jul '12

25 Jul '12

Hi Daniel, I wanted to go back to this thread for a minute: http://lists.sip-router.org/pipermail/sr-users/2012-June/073663.html Sorry that I didn't reply to the thread; I no longer have the last message. In your reply, you said: > there is no timeout on the execution of that task. Of course, you > can start many rtimer processes to do same task, up to you, > via configuration parameters. A matter of how rtimer executed > route is built, a rtimer process may not rest (e.g., sleep) and > keep consuming mqueue items as long as the queue is not empty." However, I am seeing some behaviour that is rather different. Roughly, I have a single rtimer task that runs every 3 seconds and consumes an mqueue full of accounting events. The rtimer request route looks like something this: route[ACCT_WRITER] { while(mq_fetch("acct_write")) { # Some stuff to write $mqv(...) values to PostgreSQL. ... sql_query("ca", "SELECT * FROM acct_event_proc(...)"); ... mq_pv_free("account_write"); } } And the rtimer task is parameterised thusly: modparam("rtimer", "timer", "name=acct_write;interval=3;mode=1") modparam("rtimer", "exec", "timer=acct_write;route=ACCT_WRITER") And the mqueue: modparam("mqueue", "mqueue", "name=acct_write") I have a stored procedure - let's call it acct_event_proc() - that is invoked from within the rtimer task. When I looked at the live query activity from Kamailio to Postgres by doing a packet capture on loopback, like this: tcpdump -i any -A -s 0 -tttt -n 'tcp port 5432' | grep acct_event_proc What I saw was some events being written for about ~3 seconds and then the process would stop. Then, after about half a second, it would burst again for 3 seconds, then stop again. And so on and so forth. When I lowered the interval to 1 second, it got faster and more responsive, and would push more events through in one interval. But still, it appeared to be a limited "burst" of SELECT calls punctuated by 1 second of silence. When I checked out branch 3.3 and lowered the interval to 50000u, it got even faster, almost real-time enough to keep up. Still, I did not really know what was happening, so I added an MI command to HEAD to check mqueue size live: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=41f771… (I think it would be quite useful anyhow, as there is no runtime mqueue info available right now.) Using it, I confirmed my impression: the bigger the rtimer interval, the larger the queue grows, and the slower the events seem to be written overall. The shorter the rtimer interval, the faster they are dequeued, and the smaller the mqueue is overall, in the course of normal call processing. Thus, my impression is that the execution time of the rtimer task is bounded by the interval, and that the core kills the forked rtimer task currently running prior to starting another one. Can you shed some light? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

3 12

Cleanup location table on failed NAT ping
by David 25 Jul '12

25 Jul '12

Hello, I am using kamailio 3.2.3. I am using the nathelper module to "ping" users in the location table to keep the NAT bindings alive. In the event of 3 failed NAT pings, I would like to mark the entry as "unavailable" or completely delete it from the location table to avoid having stale data. Is this a feature that exists ? Is not, are their plans for it in the future ? When the REGISTER expires after 3600, if the phone doesn't register, the location entry gets cleaned up. So I have to wait up to one hour before the bad entry is purged. I was hoping to have it purged sooner. Thanks, David version: kamailio 3.2.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3

3 2

Error Loading db_postgres.so on Startup
by Akan 25 Jul '12

25 Jul '12

I am getting this error when trying to run kamailio on Centos 5.5. I have Postgresql 9.1.4 installed and I have tried with both OpenSSL v0.9.8e and v1.0.1c and I still get the same error. Any help would be appreciated. ERROR: <core> [sr_module.c:565]: ERROR: load_module: could not open module </usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so>: /usr/local/kamailio-3.2/lib64/kamailio/modules/db_postgres.so: undefined symbol: TLSv1_method Thanks Nathaniel

2 4

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

3 3

Kamailio with APNS
by Duong Manh Truong 24 Jul '12

24 Jul '12

Hi list, My team are developing a SIP client for mobile running iOS , then having to deal with the "back ground mode" problem while this soft is installed on mobiles One of the solutions is using APNS. (Apple Push Notification Service) I wanna to use Kamailio as SIP Redirect / Proxy for my current SIP server The model is like that: User A <---------> Kamailio <---------> SIP server <--------> User B IF UserA call UserB, UserB is not available now (not registered yet), normally, SIP server will reply Kamailio with "404 Not found" In that case, i want to set up Kamailio do the communication with APNS and waiting for UserB "Wake up" then RESEND the Invite message ...... (instead of forwarding this 404 message to UserA) I am getting stuck at how Kamailio can control "404 Not found" message which it receives from SIP server . I do searching around several days but still can not get any clearer points. Please help me with some instruction/ configuration of Kamailio for this case ! Thank you very much.

2 1

Re: [SR-Users] Kamailio with APNS
by Nguyen Thanh Tien 24 Jul '12

24 Jul '12

I think you can find the 404 message at the onreply_route[]. I tried and found the 200 OK here.

1 0

SIREMIS
by Adam Fyfe 24 Jul '12

24 Jul '12

Hi, I am testing Siremis, and have become stuck. How does one add an outbound SIP trunk for calls ? thanks

2 1

Kamailio setup problem
by Vijay Thakur 23 Jul '12

23 Jul '12

All Experts, I have installed Kamailio Server on Linode Cloud. I have followed the following URL for Setup: http://kb.asipto.com/kamailio:presence:k31-made-simple This is done in ubuntu 10.04. I am only able to make chat. Video chat, conference, Presence, Call, Screen Sharing is not working. But followed the same how to install kamailio in my local system and all features are working fine. Does there anyone who can point out the flaw in setup. Thanks and regards, Vijay Thakur

2 1

Can't initialize shared memory
by Uri Shacked 23 Jul '12

23 Jul '12

Hi, Anyone got this message " Can't initialize shared memory" ? When i run kamailio using "kamailio -m 512" it works OK When i run "/etc/init.d/kamailio start" i get the message above. my init script anf defaults are set with memory = 512. I played a little with 3.3.0 and now i am back to 3.2.4 can it be something to do with it? BR, Uri

2 1

Re: [SR-Users] How to rewritehostport() dynamically?
by Alex Balashov 20 Jul '12

20 Jul '12

Try modifying the request URI with $ru/$rU/$rd/$rp instead. See the pseudovariables page on the dokuwiki for more information. rewritehostport() is a legacy function that does not deal with variables. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/intel@intrans.baku.az wrote:Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? -- _______________________________________________ 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

1 0

How to rewritehostport() dynamically?
by intel＠intrans.baku.az 20 Jul '12

20 Jul '12

Can I use rewritehostport() function with argument selected at time of call processing? I have a lua script, which gets address/port of next hop(from base) But how to call rewritehostport() with it? I can't find, how to call this function from lua script directly (I've tried sr.tm.rewritehostport() and rewritehostport() - no success) If i set pseudovariable (f.e $avp(relay)) and call rewritehostport("$avp(relay)") from kamailio.cfg later, i can see in logs such message: ERROR: tm [ut.h:327]: failed to resolve "$avp(relay)" Seems it use name of pv, not value? What should I do to call this function with correct parameter? --

1 0

Kamailio and IMS
by Apoorv Khare 20 Jul '12

20 Jul '12

Hi Everyone, I am trying to integrate IMS(*carstenbock/ims*) patch in Kamailio. But so far I could not find any proper documentation to load configuration files - pcscf, icscf, scscf, mgcf etc? Please let me know if there is anything available! Only relevant link I found - http://www.kamailio.org/w/2011/01/ims-extensions-available-for-testing/ AND this link does not seems to be update - http://www.ng-voice.com/our-solution/installation/ BR, Apoorv

3 5

uac module From header rewriting problems
by Timo Teras 20 Jul '12

20 Jul '12

Hi, I fixed some uac From rewriting issues in commit e1d1c774c9ac0b4d, however even with the latest versions I'm still seeing corruption in From headers when they are rewritten for the whole dialogue (from_restore_mode=auto, the default). I spent some time analyzing the problem and the problem seems to be connected with the fact that only the bitwise difference of the original and modified URI are stored (XOR). The only purpose for this appears to be the idea to save only the difference in the rr params instead of two full URIs. However, it appears that in many cases (especially call transfer) the From header will be altered during dialogue. This is explicitly allowed in RFC 3261 § 20.20: The From header field indicates the initiator of the request. This may be different from the initiator of the dialog. I've seen this in practise with few commercial SIP stacks. Sometimes the changes are minor, or even trivial - but with the difference encoding having even one character changed (e.g. case), being deleted or added will result in the whole header being corrupted. I'm now wondering if we should just store both URIs in the rr params. While taking little more space, it should fix us sending garbage to the wire (which usually results in dropped call as the remote will reject the garbage messages). Any better ideas? Cheers, Timo

3 7

Kamailio as SBC and far end nat
by Martin Schiøtz 19 Jul '12

19 Jul '12

Hi I'm trying to configure kamailio version 3.3.0 to be a Session Border Controller which only purpose is to handle far-end-nat. On the far end site I have a SIP client with IP '192.168.1.2' which is NAT'ed to IP 'zz.zz.85.230'. The SIP ALG is disabled on the NAT firewall. The SIP client is communicating with the kamailio server and the Kamailio server is forwarding every thing to an Avaya SIP server 'xx.xx.173.15' and vice versa. The kamilio is konfigured to do this stateless. The REGISTER and SUBSCRIBE part works fine. The kamailio is using the 'Via:' field and I also needed to do subst on the 'Contact:' field in order for our SIP server to communicate back to the kamailio. But when I get to the NOTIFY packet from the SIP server to the kamailio there is no 'Via:' field or any other information about the SIP clients IP/port and the Kamailio just sends back 'SIP/2.0 404 Not here.' to the SIP server. I guess that this can not be solved with a statless kamailio? I guess that I need some kind of state table in the kamailio - is that possible without doing REGISTER directly with the kamailio? Any other good ideas are welcome? Cheers, Martin Tell me if need more info the issue? ----- # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [AP] NOTIFY sip:2001012@yy.yy.159.211:5060;transport=TCP SIP/2.0. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Contact: <sip:xx.xx.173.14:15060>. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Event: message-summary. Content-Type: application/simple-message-summary. Content-Length: 69. Subscription-State: active;expires=299. Max-Forwards: 68. User-Agent: AVAYA-SM-6.1.4.0.614005. . Messages-Waiting: no. Message-Account: sip:2001012@bsosip.domain.dk. ## T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4;rport=38059. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. . # T xx.xx.173.15:38059 -> yy.yy.159.211:5060 [A] ...... # T yy.yy.159.211:5060 -> xx.xx.173.15:38059 [AP] SIP/2.0 404 Not here. Call-ID: NjQ5ZDNlYzRjOWFjMzMzN2JlYzAwMTg4ODYwZTJlYWY.. From: "2001012" <sip:2001012@bsosip.domain.dk>;tag=-667195155*1*016asm-callprocessing.sar229435981~1342570131099~1170998434~1. To: "2001012" <sip:2001012@bsosip.domain.dk>;tag=7ca0af7d. CSeq: 3 NOTIFY. Via: SIP/2.0/TCP xx.xx.173.15;rport=38059;branch=z9hG4bK4DEAAD0E478B838008189457-AP;ft=xx.xx.173.15~13c4. Via: SIP/2.0/UDP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838008189457. Via: SIP/2.0/TCP xx.xx.173.14:15070;branch=z9hG4bK4DEAAD0E478B838018189455. Server: kamailio (3.3.0 (i386/linux)). Content-Length: 0. -----

3 2

Compile Kamailio from Source with STUN
by Vijay Thakur 19 Jul '12

19 Jul '12

Dear All Experts, I am following the steps given at http://kb.asipto.com/kamailio:presence:k31-made-simple. I want to compile kamailio with stun support. So guide me that what will be the command to do it. As per the given command : *make PREFIX=/usr/local/kamailio-3.1-xcap FLAVOUR=kamailio \ include_modules="db_mysql presence presence_xml xhttp xcap_server" cfg what text should i use to make stun module. * -- Best Regards, Vijay Thakur (Assistant Manager - Networks) Mobile : +91 8744018065 Mail : vijay.thakur(a)loopmethods.com Loop IT Methods Private Limited 1st Floor, B-10, Sector-7, Noida, (U.P) India Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS) Fax: +91 971 728 330 Web: www.loopmethods.com LOOP Disclaimer ------------------------------------------------------------------------------------------------- This message (including any attachments) contains confidential information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

3 2

Kamailio v3.2.4 Released
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, Kamailio v3.2.4 is out -- this is a minor release packaged from previous stable branch, GIT 3.2. Anyone running 3.2.x should update to it in order to benefit of latest fixes, there is no change required to be done in database structure or configuration file, just re-install over the old deployment. As usual, a backup is recommended to be in the safe side. More details about this release at: * http://www.kamailio.org/w/2012/07/kamailio-v3-2-4-released/ New installations should start with latest stable series 3.3.x, at this time latest Kamailio version being v3.3.0. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 0

Re: [SR-Users] Presence Subscription / Subscription-State: pending
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, please keep the mailing list cc-ed, private emails from mailing list discussions will be ignored. On 7/19/12 8:28 AM, Durmuş Ali Öztürk wrote: > Hi Daniel, > > Thank you for your advise. I have found the problem, I thought it is > enough to subscribe with my own ID but fact is I have to subscribe to > the buddies of my contact list.This was not clear for me at the beginning. ok, great you found it. > > It is better to subscribe only to the presence.wmi? Or should I > susbcribe to presence and presence.wmi? Do you mean presence mwi? mwi is for message waiting indication, like voicemail, so you get notifications when someone leaves you a message on the media server. Cheers, Daniel > > Best Regards, > Ali > > 2012/7/18 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > when you updated the presence rules via xcap, did you executed the > function to update the notifiers as well? I mean the two function at: > > http://kamailio.org/docs/modules/stable/modules_k/presence.html#id2496049 > > Cheers, > Daniel > > > > On 7/18/12 9:29 AM, Durmuş Ali Öztürk wrote: >> Hi All, >> >> I am trying to subscribe to the presence event but unfortunaly I >> am getting following Notify with the "Subscription-State: >> pending" status. >> >> NOTIFY sip:5000@192.168.127.138:55583 >> <http://sip:5000@192.168.127.138:55583> SIP/2.0 >> Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 >> To: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=2786be25cb234561b8dea2e067fd769c-5000 >> From: sip:5000@192.168.127.230 >> <mailto:sip%3A5000@192.168.127.230>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e >> CSeq: 2 NOTIFY >> Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop >> Content-Length: 0 >> User-Agent: kamailio (3.2.3 (i386/linux)) >> Max-Forwards: 70 >> Event: presence >> Contact: <sip:192.168.127.230:5060;transport=udp> >> Subscription-State: pending;expires=3670 >> >> >> For the implemantation I followed the guide from >> http://kb.asipto.com/kamailio:presence:k31-made-simple >> >> The xcap table has already the needed resource-lists >> and pres-rules record. >> >> Is there another guide exists for the presence module which can >> be helpfully for me? >> >> Thank you in advance >> ali >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu > Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw > > -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -http://asipto.com/u/kpw

3 3

planning release of v3.2.4
by Daniel-Constantin Mierla 19 Jul '12

19 Jul '12

Hello, later this week (like Thursday) I am considering packaging v3.2.4 from the latest git branch 3.2. If anyone has patches to push it there, please do it asap in case you want them in this release. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw

1 1

Presence Subscription / Subscription-State: pending
by Durmuş Ali Öztürk 18 Jul '12

18 Jul '12

Hi All, I am trying to subscribe to the presence event but unfortunaly I am getting following Notify with the "Subscription-State: pending" status. NOTIFY sip:5000@192.168.127.138:55583 SIP/2.0 Via: SIP/2.0/UDP 192.168.127.230;branch=z9hG4bK6a8e.19cd7bb7.0 To: sip:5000@192.168.127.230;tag=2786be25cb234561b8dea2e067fd769c-5000 From: sip:5000@192.168.127.230;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6a8e CSeq: 2 NOTIFY Call-ID: e4ec6031-99e1-de11-8845-5000@vvt-laptop Content-Length: 0 User-Agent: kamailio (3.2.3 (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:192.168.127.230:5060;transport=udp> Subscription-State: pending;expires=3670 For the implemantation I followed the guide from http://kb.asipto.com/kamailio:presence:k31-made-simple The xcap table has already the needed resource-lists and pres-rules record. Is there another guide exists for the presence module which can be helpfully for me? Thank you in advance ali

2 1

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

2 6

Topoh address
by Alex Balashov 17 Jul '12

17 Jul '12

A whole ago, it was mentioned on the list that there was some bug or issue in Topoh that precluded the use of the actual host IP of the proxy as the Topoh spoof address. Is that still true in 3.3? Or has that been fixed some time ago? Thanks! -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 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/

2 1

How to use regular expression for $var ?
by Gary Chen 17 Jul '12

17 Jul '12

Kamailio 3.3.0 I have a variable $var(s:dst). It can store either a number or IP. How do I check to determine whether it is a number of IP? I tried the following and it did not work: If ($var(s:dst) =~ "^\d+\.\d+\.\d+\.\d+$"){ It is a IP. }else{ It is a Number } Thanks Gary

4 3

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