sr-users November 2005

sr-users@lists.kamailio.org

245 participants
570 discussions

by Douglas Garstang

Ok, I worked out that you need to use append_branch, like this: append_branch (proxy1) append_branch (proxy1) t_replicate (proxy3) That's what the devevelopers say anyway... However, all the t_replicate's are going to exactly the same proxy. Here's a post by two people that have reported _exactly_ the same problem. Is there a solution? http://www.archivum.info/serusers@iptel.org/2004-12/msg00268.html Thanks, Doug. -----Original Message----- From: Douglas Garstang Sent: Wednesday, November 30, 2005 4:11 PM To: users(a)openser.org Subject: [Users] t_replicate after failure In the scenario below, if the first t_replicate fails because 192.168.10.5 returns "Not Found", the second t_replicate command does not execute. Why? How can I make it execute? Call me crazy, but this doesn't seem at all intuitive to me. In a functional programming world, if the first fails, the next should still run. if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("voip.com", "subscriber")) { www_challenge("voip.com", "0"); exit; }; save("location"); t_replicate("192.168.10.5","5060"); t_replicate("192.168.10.7","5060"); exit; }; Thanks, Doug. _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

18 years, 11 months

[Serusers] hierarchical VoIP system

by Joao Pereira

Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt

18 years, 11 months

[Serusers] Authenticate to Active Directory

by Bob McCormick

Does anyone know if there is a way to authenticate SER to MS Active Directory? I'd like to use an existing user database instead of having to manage users myself, and most of our users are in Active Directory. Has anyone tried this? What are most SER users using for authentication? Is anyone here tying into any kind of existing organization user database?

18 years, 11 months

[Serusers] SER Comment

by Selvasundar

hi andres , i request your help to install SIP SER as a proxy . THanks selva The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s)and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender or administrator(a)tataelxsi.co.in

18 years, 11 months

[Users] (open)SER and ICE

by Christophe Irles

Hi, As I'm a newbie on this subject, I'm wondering how two SIP devices with ICE compatibility will interact with a (open)SER+Mediaproxy implementation since ICE implementation could permit to avoid RTP flow on the (open)SER server. Is there any update to do on the conf file on the (open)SER ? Since they are ICE compatible, the SIP flow (signalization) will be analysed by this two devices in order to create a direct RTP flow ? Or is it because i know on the (open)SER side that this devices are ICE compatible I send a special SIP message ? In fact I want to know if a server update is necessary or if it's "free" to have this great feature. :) Thanks, Christophe

18 years, 11 months

[Users] a router error

by messa＠innsof.com

Hi all I use d-link 704P router. When i want to make a call, signalisation is good, but, when the client is hang up, i have voice to the client which is connect in the D-link LAN and the other client has no voice. My server has a public IP. When i'm find the log of the router, i see these messages: unallowed access from 192.168.0.53:to 66.249.93.104:protocol=6 rule=3(by firewall) 192.168.0.53, is the adress of the client behind the d-link lan, the client which has voice and cannot transmit it! what means theses messages? how can i solve the router problem to make my calls? please help best regards! Serge

18 years, 11 months

[Users] Can anyone share his openser.cfg..in which openser is routing calls to pstn gateway

by Lokesh Kumar

Hello, I am trying to configure pstn gateway with openser.when i start openser with default openser file it works fine, but i one direction. i can dial on my SIP user agent from outside. but when i am using pstn.cfg file as a openser.cfg(after rename). I am unable to even start openser server. it gives me errors and crashed.I puts my pstn gateway ip's in that file where requested. Please please help Regards

18 years, 11 months

[Users] AVPs are lost on relayed INVITE errors

by Federico Giannici

I'm using OpenSER 1.0.0 on OpenBSD 3.7 amd64. I have a strange problem with the accounting: I set a couple of AVPs for every message that arrives at the server. I'm sure they are there because they are written in the syslog logging. Sometimes, when an INVITE is relayed (with transactions) and receives an error (488, 422, etc.), in the SQL logging there is no more presence of the AVPs! Is this a known problem? How can I avoid this? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

18 years, 11 months

[Users] Re: [Devel] nicer compiling

by Bogdan-Andrei Iancu

Hi, thanks to Anand, you have now the liberty to get a more friendly output during compilation. Just do: NICER=1 make all normal "make all" still produces same amount of output - it;s the default since it's useful to see the compilation flags...at least for me... also I got rid of the messages regarding the missing dependency files (.d) during first compilation.... so, have a nicer compiling ;) regards, bogdan Anand Kumria wrote: >Hi, > >Attached is a diff that makes it nicer to compile openser. Should you >want to see the exact compilation sequence Q="" make <...> will show it >to you. > >Thanks, >Anand >

18 years, 11 months

[Serusers] Creating CDRs from SERs accouning records

by Frank Fischer

Hi all i'm looking for a free/opensource application that generates CDRs (not XDRs, must not be rated) from the accounting data/events generated by SER (i think this is usually called normalization). I think this would have to be an application that is able to match INVITEs and BYEs and form CDRs with call start timestamp, call duration and so on. I had a look at the list of accounting software on the iptel.org website, but most of there applications are highly integrated accounting/billing solutions, far more than what i need (and also most of them seem to be commercial products). In the mailinglist archive i only found some references to CDRTool which seems also to be commercial. Obv. there must be a free version of CDRTool but neither i'm sure whether it implements exactly what i need so i could "grab it" for my own purposes nor could i find a place to download the free version). Does anyone know if there is such a kind of application available? Or any other idea how to deal with my requirements? Thanks a lot for your help Frank

18 years, 11 months

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sr-users November 2005