Re: [SR-Users] Mid call announcement(kamailio server)

2013-01-18 Thread rabs
> Hello, > As of now am working on Kamailio server. > My task is to announce in between call,my sip clients are EKIGA. > I should able to make my kamailio server act as controller and media > server,for that purpose my first step is : invite hold to called party. > second step is:: invite(sdp for .

Re: [SR-Users] FW: Unsuccessful call using DRouting - Request Timeout

2012-03-19 Thread rabs
>  Hi, > > Thanks for the reply. > So can I say that the script and data in tables are correct? > > The .266 is doing nothing, neither ringing nor showing any response, and > this is the issue! > So what might be prohibitting the call ringing? However without the drouting > module calling 6000 (.26

Re: [SR-Users] FW: Unsuccessful call using DRouting - Request Timeout

2012-03-19 Thread rabs
> Dears, > > > > I started my lab on VMware (debian lenny) where I installed Kamailio 3.2 and > started drouting configuration. > > > > I'm using 2 PCs each having an x-lite softphone and the Kamailio on the > vmware all connected locally. My primary test is to make a call from x-lite > on PC (192.

Re: [SR-Users] SIP Recorder

2011-01-27 Thread rabs
Danny Dias escribió: Hi Iñaki, 2011/1/27 Iñaki Baz Castillo 2011/1/26 Danny Dias : > i mean > signaling: A>PROXY>B (the normal procedure) > At the same time, this must be done: (I'm not sure how to do this...the > proxy could be out of this or not, not sure :() > A ---INVITE---

Re: [SR-Users] SIP Recorder

2011-01-26 Thread rabs
Danny Dias escribió: Many thanks Jaremya, The main problem is that both terminals, SHALL (required and must not be changed, because of standards of EUROCAE ED-137 Part3) initiate a session with the recorder server (a commercial one, can't use Asterisk for my disgrace) sending INVITE and receiv

Re: [SR-Users] SIP Recorder

2011-01-26 Thread rabs
Danny Dias escribió: Thanks Jeremya, but it's a requeriment from the client to record the calls through an external server and not with rtpproxys, my question is how the media should be handled in order to record the conversations if the server is external? Signaling: Phone_A <---> Proxy <--->

Re: [SR-Users] SIP Recorder

2011-01-26 Thread rabs
Danny Dias escribió: Hello my friends, I have a requeriment, which indicates that i have to record every SIP conversation between peers (also for callings to the PSTN); the recording server will be built for our company following this requeriments (also requested for the client): My doubt is:

[SR-Users] Doubd about Dialplan module

2010-09-06 Thread rabs
Hi all, I'm tryting to implement a E164 normalizer using the dialplan module, by now I have 3 simple rules: dpid priority matchop matchex matchlen subsex replaceex attributes 011 (00|\+)([1-9][0-9]+) 0 (00|\+)([1-9][0-9]+) \2 011 ([5-9][0-9]{8}) 0 ([5-9][0-9]{