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[Serusers] Memory leak issue
by Piyush.Bansal＠relianceada.com 02 Dec '09

02 Dec '09

Hi All, I'm using SER version 2.0.0-rc1 as a SIP server and in that facing lot of memory leaks issue. I'm not able to enable the memory logs in SER. Can anybody help me identifying memory leaks or tell me how to enable memlogs so that I myself will try removing all the leaks... Plz help me ASAP, its urgent... Thanx n regards, --Piyush The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

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[Kamailio-Users] Kamailio -> Asterisk Redirect/Reinvite and Remote-Party-ID
by Florian Meister 02 Dec '09

02 Dec '09

Hi, basically I'm using this structure at the moment: SIP Users <----> Kamailio <-----> Asterisk <-----> PSTN I have to add a diversion-functionality at kamailio-level, so to simply rewrite $ru with something else defined in the database. That's working without problems. For billing issues, I also have to add a Remote-Party-ID header, set to the SIP user, which initiated the redirect in the database. Now to the problem: When a call is coming from PSTN, it's passing the asterisk server, then at kamailio level $ru is rewritten and sent back to asterisk (I'm talking about a redirect to a number in PSTN here) What I've seen from the logs is, that asterisk is seeing that it gets an invite back with the same call-id, and therefore it cancels the original invite and handles the whole call internally via the Local Channel. The Problem is, that in the invite sent from kamailio back to asterisk, I've set a Remote-Party-ID header to tell asterisk to set the Callerid correctly for billing purposes. Now it seems that asterisk is _ignoring_ this header from the second invite. So is this an expected behavior ? If yes, how to do it correctly ? Below you can see the verbose output of asterisk. Since the call is handled at "Local" Channels the function to read sip headers does not work. The only message I get is "thanks to SIP/tpsiptestproxyu01-00d0a0b8". -- Called tpsiptestproxyu01/+435572949012 -- Now forwarding DAHDI/2-1 to 'Local/066480588134@from-internal' (thanks to SIP/tpsiptestproxyu01-00d0a0b8) -- Executing [066480588134@from-internal:1] NoOp("Local/066480588134@from-internal-d69e;2", "435572501134") in new stack [Dec 1 15:14:50] WARNING[20506]: chan_sip.c:15797 func_header_read: This function can only be used on SIP channels. -- Executing [066480588134@from-internal:2] NoOp("Local/066480588134@from-internal-d69e;2", "") in new stack -- Executing [066480588134@from-internal:3] Dial("Local/066480588134@from-internal-d69e;2", "DAHDI/G0/066480588134") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G0/066480588134 -- DAHDI/124-1 is proceeding passing it to Local/066480588134@from-internal-d69e;2 -- Local/066480588134@from-internal-d69e;1 is proceeding passing it to DAHDI/2-1 In the SIP debug you can see that asterisk is cancelling the dialog with kamailio and doing it itself: 13:49:30.589054 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 938 E.......@..L..,L..,N........INVITE sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.591037 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 342 E..r..@.@.[0..,N..,L.....^.aSIP/2.0 100 Trying Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.594345 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 1093 E..a..@.@.XA..,N..,L.....M+LINVITE sip:066480588134@[--ASTERISK--]:5060;transport=udp SIP/2.0 Record-Route: <sip:[--KAMAILIO--];lr;ftag=as27658014> Via: SIP/2.0/UDP [--KAMAILIO--];branch=z9hG4bK06.05390227.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 Max-Forwards: 69 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Tue, 01 Dec 2009 12:49:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 263 Remote-Party-ID: "435572949012" <sip:435572949012@tpseru01.tele.net>;party=caller;privacy=none;screen=yes v=0 o=root 930830518 930830518 IN IP4 [--ASTERISK--] s=Asterisk PBX 1.6.1.5 c=IN IP4 [--ASTERISK--] t=0 0 m=audio 16924 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 13:49:30.594605 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 460 E.......@..)..,L..,N....... CANCEL sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 13:49:30.596307 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 387 E.....@.@.[...,N..,L.......KSIP/2.0 200 canceling Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 CANCEL Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596801 IP [--KAMAILIO--].5060 > [--ASTERISK--].5060: SIP, length: 396 E.....@.@.Z...,N..,L.......kSIP/2.0 487 Request Terminated Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport=5060 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 INVITE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 13:49:30.596842 IP [--ASTERISK--].5060 > [--KAMAILIO--].5060: SIP, length: 539 E..7....@.....,L..,N.....#.oACK sip:+435572949012@[--KAMAILIO--] SIP/2.0 Via: SIP/2.0/UDP [--ASTERISK--]:5060;branch=z9hG4bK5629d66b;rport Max-Forwards: 70 From: "435572501134" <sip:435572501134@[--ASTERISK--]>;tag=as27658014 To: <sip:+435572949012@[--KAMAILIO--]>;tag=45db18648893e7acabf725621374d382-4ddb Contact: <sip:435572501134@[--ASTERISK--]> Call-ID: 4bbee84339a9e2d30850185317983625@[--ASTERISK--] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.5 Remote-Party-ID: "435572501134" <sip:435572501134@[--ASTERISK--]>;privacy=off;screen=yes Content-Length: 0 Thanks, Florian

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[Kamailio-Users] client registers with private IP
by Joao Gomes Pereira 01 Dec '09

01 Dec '09

Hello I'm trying to register in Kamailio, an Asterisk working behind Nat. Here is the trunk: The problem is: Asterisk is registering with its internal IP (192.168.1.25), as you can see here: sip:s@192.168.1.25 Q= Expires:: 81 Callid:: 480b40aa13ddd8707787b21a69656535(a)127.0.0.1 Cseq:: 103 User-agent:: Asterisk PBX How can I force Asterisk to register with its public IP? Is there some configuration I can do in Kamailio? Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespereira(a)startel.pt

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[Kamailio-Users] CDRTool Mediaproxy etc.. AG stuff
by Brian Chamberlain 01 Dec '09

01 Dec '09

Hi List, Apologies if you get this twice, for some reason the last post didn't seem to appear on the list.. I am running Debian lenny. Can I just change the repos in sources.list from lenny to testing and unstable to get the correct python dependencies etc for the ag-project tools? Thanks, Brian

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[Kamailio-Users] presence send system info
by alex pappas 01 Dec '09

01 Dec '09

Dear friends, I'm trying to send some system info through presence in my X-Lite client , following the example http://www.kamailio.org/dokuwiki/doku.php/presence:pua-modules#using_with_x… Does anyone knows what configuration should i have in the X-Lite side? Thanks Alex

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[Kamailio-Users] t_on_failure() t_on_reply()
by BERGANZ François 01 Dec '09

01 Dec '09

Hello, Is it possible to do that : ? t_on_reply("1"); t_on_failure("1"); t_relay(); François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.

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[Kamailio-Users] Fwd: Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 01 Dec '09

01 Dec '09

yes it does: [root@atl-sipgateway1 ~]# ip addr show dev eth0 2: eth0: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast qlen 1000 link/ether 00:1e:c9:b6:4c:fa brd ff:ff:ff:ff:ff:ff inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global eth0 inet 67.220.XX.XX/27 brd 67.220.XX.XX scope global secondary eth0:1 inet6 fe80::21e:c9ff:feb6:4cfa/64 scope link valid_lft forever preferred_lft forever On Mon, Nov 30, 2009 at 12:15 PM, Alex Hermann <alex(a)speakup.nl> wrote: > On Sunday 29 November 2009 20:57:59 Geoffrey Mina wrote: >> I have a Kamailio machine that has multiple IPs assigned to a single >> interface. >> >> eth0 = 1.2.3.4 >> eth0:1 = 1.2.3.5 > > No idea if it has any influence on Kamailio behaviour, but those seem to be > two interfaces as far as applications are concerned. Assigning multiple IP's > to one interface means that for all IP's, the interface should be the same. > > Does 'ip addr show dev eth0' show 2 IP's? > > -- > Alex Hermann > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >

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[Kamailio-Users] 487
by Anders 30 Nov '09

30 Nov '09

Hi, This question may belong in a general SIP forum - but I'll drop it here anyway, - hope I stir no big anger ;-) I'm running Kamailio 1.5, - use it for wholesale, and my customers typically send me calls from their Asterisk platform. I'm experiencing a very share of calls ending up in 487. This should be the client terminating the call - but is it my client (the provider with the Asterisk platform) or is it the final client - my client's client? Could Asterisk send a 487 for some reason? I've researched this (maybe not good enough!), but cannot find the answer: Does a 487 always mean that the client gave up and hung up? If this is the case, could high PDD be my problem - it's taking too long and the client gives up? Thanks!

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[Kamailio-Users] Binding to multiple IP addresses on the same interface.
by Geoffrey Mina 30 Nov '09

30 Nov '09

Hello, I have a Kamailio machine that has multiple IPs assigned to a single interface. eth0 = 1.2.3.4 eth0:1 = 1.2.3.5 In my Kamailio config i have listen=eth0:5060 And it doesn't appear to be binding to my second IP address. What is the proper way to tell the server to bind to all IPs on a single interface? I don't want to specify the IP directly for binding as I use the same config across 4 Kamailio servers. listen=eth0:1:5060 (i think that's not correct) Thanks, Geoff p.s. I would do some testing, but this is a live machine and I don't want to bring it up and down.

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Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails
by Martin Koenig 30 Nov '09

30 Nov '09

Hello Laszlo, Good hint. We are building against openssl-0.9.7f-7.10. For testing purposes I have commented out the INCLUDE of md5.h in nonce.c, now the module compiles with the following warning: gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -ftree-vectorize -mtune=prescott -Wold-style-definition -Wmissing-field-initializers -Wredundant-decls -DMOD_NAME='"auth"' -DNAME='"kamailio"' -DVERSION='"1.5.3-tls"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 4.0.2"' -D__CPU_i386 -D__OS_linux -D__SMP_yes -DCFG_DIR='"/usr/local/etc/kamailio/"' -DPKG_MALLOC -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DUSE_LOCAL_ROUTE -DF_MALLOC -DUSE_TLS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_EPOLL -DHAVE_SIGIO_RT -DHAVE_SELECT -I/usr/local/ssl/include -I/usr/local/include -I/usr/include/openssl -c nonce.c -o nonce.o nonce.c: In function âcalc_nonceâ: nonce.c:116: warning: implicit declaration of function âMD5Initâ nonce.c:126: warning: implicit declaration of function âMD5Updateâ nonce.c:129: warning: implicit declaration of function âU_MD5Finalâ I'll try to see if this is functional now. However, it feels like a "dirty hack". Best regards, Martin ________________________________ Von: voipfreak.net(a)gmail.com [mailto:voipfreak.net@gmail.com] Im Auftrag von Laszlo Gesendet: Montag, 30. November 2009 17:57 An: Martin Koenig Betreff: Re: [Kamailio-Users] Compiling nonce.o (Auth Module) with TLS=1 fails Hi Martin, [...] Which version of openssl you have? You might find this interesting: http://bugs.gentoo.org/146780 (botton of the page) -Laszlo

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