sr-users June 2007

sr-users@lists.kamailio.org

191 participants
314 discussions

by Maciek Kowalski

Hello All, I have a problem with transactions rewrited to asterisk. In my configuration asterisk and openser are behind NAT on a Local Network.(Openser- 192.168.1.2, Asterisk 192.168.1.5) When a User who is behind different NAT(different LAN) sends an Invite which is then rewrited to Asterisk with (rewritehostport("192.168.1.5:5060");) everything works great till asterisk answers with 200OK. In this message the Contact header field is set to "192.168.1.5:5060". When user recieves this 200OK it sends ACK to IPaddress 192.168.1.5 (and since he is behind different NAT..the message goes nowhere). Since the Initial Invite in this transaction is "record_routed" by Openser I thought that the ACK should first go through Openser. Please corect me if Im wrong, and if possible, suggest some solution. Best Regards, Maciej Kowalski

17 years, 5 months

[Users] SIP Server running but no messages on stdout.

by akkala.naidu＠aricent.com

17 years, 5 months

[Serusers] Buddy Lists

by KUMAR

Hi all, I am using SER-2.0 rc1, and I want to store the buddy Lists on the server so that the users can access their buddy lists from any place. As far as I can understand, this can be done using XCAP server, but I cannot find any free XCAP server. Is there any other way to do this ? Or is there a free XCAP server ? Please help me out. Thank you kumar

17 years, 5 months

[Serusers] Ser + Windows NAT

by Thomas Britis

Hi all, I'm getting a problem when a user behind a windows gateway (NATed client) connects to my ser box. I think this was already discussed here but I didn't find anything on the logs. The problem is with the '407 Proxy Authentication Required' signal that never reaches the client. Here is the scenario: 1- The user sends the INVITE from a higher port to the SIP server. It looks like this: User:63040 -> SER:5060 2- SER responds with a 407 Proxy Authentication Required on the client's local sip port: SER:5060 -> User:5060 At this point, I think windows drops the package because it don't refer to any entry at the NAT table, but I'm not sure about that. I want to know if there's a solution to this 'problem'. Thank you all. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA

17 years, 5 months

[Users] openser radiator radius accounting packets are missing

by Russell Williams

This is a packet dump from radiator.. Where are the attributes to identify the caller , and the callee. The information I need is timestamp which is here . callingstationid calledstationid originatingip and destinationip Any suggestions on how to get them to show up here . BTW the stop record is as lacking in info as this start record. Attributes: Acct-Status-Type = Start Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = Invite Event-Timestamp = 1181607270 Sip-From-Tag = "EELf2-pTUK4" Sip-To-Tag = "N78f2-UF9FM0" Acct-Session-Id = "0PkXg-tEbc4f2(a)207.158.49.3" NAS-IP-Address = 207.158.49.3 NAS-Port-Id = 5060 Acct-Delay-Time = 0 Mon Jun 11 17:15:53 2007: DEBUG: Handling request with Handler 'NAS-IP-Address=xx.xx.xx.xx Thank you Russell Williams

17 years, 5 months

Re: [Users] Accounting Bye packet Issue

by vallimamod abdullah

Hi Dan, > From: "Dan" <fiedler.dan(a)gmail.com> > To: <users(a)openser.org> > Date: Wed, 13 Jun 2007 11:45:41 +0530 > Subject: [Users] Accounting Bye packet Issue > > I am facing a peculiar issue with accounting. BYE packets are getting rejected > with error message: 478 Unresolvable destination. >From your logs, I see: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp> ^^^^ The bye packet is broken as you don't have the username part on the Route header uri. > This happens only in case I use ATA for calling and with some gateways only. > With the same gateway if I call using xlite, accounting works fine. I am > unable to find out the reason. I haven't worked with these phones but it seems to me that they are doing strict routing like some pre-rfc3261 devices. In "normal" case, the first line would have looked like: BYE sip:user@210.68.65.78 SIP/2.0 without the route header. Hope this will help. -- Regards, vma .

17 years, 5 months

[Users] Accounting Bye packet Issue

by Dan

Hi, I am facing a peculiar issue with accounting. BYE packets are getting rejected with error message: 478 Unresolvable destination. This happens only in case I use ATA for calling and with some gateways only. With the same gateway if I call using xlite, accounting works fine. I am unable to find out the reason. Can someone help on this? I am attaching the log when I am calling from a ATA and BYE is being rejected by openser. Regards Dan

17 years, 5 months

[Users] rtp relay & transcoding

by Ryan Mitchell

Regarding RTP relay (e.g. media proxy), and transcoding -- This topic has come up many times before in various forms, but I still have not found any obvious solution that's highly scalable and cheap (well sure, I can dream). We have lots of customers behind various firewalls using various codecs. Customers call each other (both parties behind firewall) and to/from carriers. Certainly this is a common use-case. What do you do? What are best known practices? Commercial media gateway, or open-source solutions? We use both OpenSER and a custom B2BUA written against the NIST Java stack. We need a way to transcode when needed. Obviosly mediaproxy module is great, but doesn't transcode. Ideally, what I'm looking for: * at the point where our system determines that transcoding or rtp relay is needed, a media gateway is chosen based on network proximity (e.g.: 1/2 our customers are in the U.S., 1/2 in Brazil; should pick a media gateway accordingly to minimize network path). * our system then signals the media gateway (mgcp?) to setup the channels, and modifies the SDPs in the SIP path accordingly. * even more ideally: the media gateway signals our system with inband or rfc2833 dtmf events, plus rtcp reports when available. I think Asterisk could be hacked to do this. Does anyone know of a commercial product that's not too expensive? Thoughts? Advice? thanks, -- Ryan Mitchell <rjm(a)tcl.net> Telecom Logic, LLC http://www.tcl.net

17 years, 5 months

[Users] Question about openser with sip trunks

by jamespev

17 years, 5 months

[Serusers] error when compiling SER 2.0 rc1

by James Beatson

Can someone explain the cause for the following compile error when trying to compile SER 2.0 [root@localhost ser-2.0.0-rc1]# make group_include-"standard" all make: *** Warning: File `Makefile.rules' has modification time in the future (2006-09-29 03:52:24 > 2006-06-11 21:31:35.613769) bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex make: *** Warning: File `Makefile.rules' has modification time in the future (2006-09-29 03:52:24 > 2006-06-11 21:31:39.786688) bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce bison -d -b cfg cfg.y cfg.y: conflicts: 1 shift/reduce flex cfg.lex > > [root@localhost ser-2.0.0-rc1]# James Beatson Camiant 781-264-2297 www.camiant.com <blocked::http://www.camiant.com/>

17 years, 5 months

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sr-users June 2007