sr-users September 2011

sr-users@lists.kamailio.org

69 participants
109 discussions

by Jon Bonilla

Hi all with the release of kamailio 3.1.5, I've done a small change to the debian repositories: Nightly builds of kamailio_3.0 branch have been dropped and the repository has been erased. There is no development in this branch and it has no sense to keep nightly builds for it. Those (few) people using kamailio_3.0-nightly repositories are advised to move their sources.list to the stable 3.0 repository of upgrade to 3.1 or master branches. kamailio 3.0 will continue to be packaged if any new stable version is released for etch, lenny, squeeze and lucid. For 3.1 branch, nightly builds and stable repos will be provided for lenny, lucid and squeeze When the time for 3.2 (master branch at the moment) comes, I'll add Wheezy support. Not sure about Ubuntu distro support. Should we add support to Ubuntu 11.10 or wait until the lts 12.04 is released? Any opinions welcomed cheers, Jon

12 years, 7 months

rtpproxy VS Turn

by Mikael Williams

Dear All, I would like to ask please if and in case I'm using STUN/TURN for Nat traversal to drp the rtpproxy from my solution as NAT traversal and relay server as I'm using SYUN for NAT traversal and TURN as relay in case of Symmetric NAT Regards

12 years, 7 months

Communication from IPv6 to Ipv4

by Loïc Fofana

My architecture is described below: TDM device----------- Call server[IPv4]---------------[IPv4]Kamailio[IPv6]------------[IPv6]Asterisk---------[IPv6]Linphone * * * * * * * * * * * * Media Server Rtpproxy Media Server and Rtpptoxy carry the media. Call Server carry only signaling. I have succeeded to establish audio communication between an IPv6 domain & an IPv4 domain i.e. : from an TDM device to a Linphone. However, When i try to establish a call from (IPv6 to IPv4), i.e. from Linphone to TDM device, the signaling is OK, but there's no voice. Because, Rtpproxy forward the media to the Call server instead the Media Server. Hence, there can't be voice. I have been stucking on this for a couple of months. I will be very glad if somebody have any idea. best regards!

12 years, 7 months

Re: [SR-Users] Kamailio terminating after upgrading from 3.1.2 to 3.1.4

by Phillman25 Kyriacou

Dear Marius Thank you for your prompt reply. I apologise, As i am a newbie could you please explain how to produce the below output that you require? Thanking you in advance for your help! Regards Phillip > Hello > Hello. You should have a coredump . Can you please send the trace (bt full) as seen with gdb ? Thanks Marius > Im facing a big problem after upgrading from version 3.1.2 to 3.1.4. > The kamailio process keeps on terminating every now and then, this is > what i was able to retrieve from my logs. > > > Sep 14 12:52:31 SipProxy1 kernel: [71448.062089] __ratelimit: 15 > callbacks suppressed > Sep 14 12:52:31 SipProxy1 kernel: [71448.062096] kamailio[13963]: > segfault at 14 ip 0816452a sp bfedc940 error 4 in kamailio[8048000+1d9000] > > > Does anyone have an idea why this is happening? > > Your help is much appreciated! > > Regards > Phillip

12 years, 7 months

(no subject)

by Loïc Fofana

This is my scenario: Call server -------------(IPV4)Kamailio(IPV6)----------Asterisk(IPV6)------Linphone(IPV6) * * * Media server

12 years, 7 months

One way communications in Kamailio 3.1.4

by CaptWho

I'm running 3.1.4 on centos and I'm having some trouble with voice only going one way. Both extensions will ring each other, but after they connect, voice will only travel in one direction. One extension hears fine, but can't talk. I've totally opened up the firewalls (including port 5060) and I'm still having the trouble. I'm not behind NAT. I've tried it using X-Lite, VoIP phones and ATAs in a number of combinations. The problem is across the board. X-Lite sometimes automatically disconnects after 32 seconds. Does anyone have any suggestions on where to look? ANY help appreciated. This has been going on for weeks. -- View this message in context: http://old.nabble.com/One-way-communications-in-Kamailio-3.1.4-tp32460597p3… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

12 years, 7 months

Open source HLR (Home Location Register).

by Henrik Aagaard Sørensen

I know this is the mailing list for Kamailio, but hopefully someone can help me out as it is "kinda" related to VoIP etc. Does anyone know any open source / free HLR implementation?

12 years, 7 months

Siremis site access from Romania

by Dani Popa

Hi, There is any reasons why any linked Siremis site is not accessible from Romania ? i tried to access http://siremis.asipto.com/ and other sites and they are not accesible. Thanks, Dani Popa

12 years, 7 months

Re: [SR-Users] planning v3.1.5

by Vitaliy Aleksandrov

There is a bug in UAC module. uac_req_send doesn't restore avps. I tried to backport your patch to 3.1 but with no success. Could you add this patch to a new stable release ?

12 years, 7 months

DNS support

by CT Radu

Dear sirs and madams, I was wondering what is the current support for DNS fallback in the 3.1.x branch . Lets assume we have a new dialog started and the name in the To field needs to be resolved via a NAPTR. Kamailio sends a NAPTR request, gets 3 SRVs , each with a number of A or AAAA records. The best/most suitable SRV is selected, however no response is returned from any of the addresses in the A records. Now, we should go the second SRV and try the A records in there and so on. My question is how much from this kind of dns fallback algorithm is supported ? (I hope my explanation wasn't too confusing) Thank you, Costin Radu

12 years, 7 months

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sr-users September 2011