[OpenSIPS-Users] Playing caller a file before dialing callee

Hi users. I am trying to play caller a file before dialing callee. I used the makeann from rtpproxy to pre encode a wav file didyou8r1c.wav(mono, 16bit, 8k), which gave didyou8r1c.wav.0 and didyou8r1c.wav.8. Both files are located in /opt/test/ directory. My invite block is given below. if

Re: [OpenSIPS-Users] Sending to TLS destinations with dispatcher module (2.1)

Hi Razvan Happy New Year! Yes your answer is clear, thank you. I will do some experimentation but I think I may run into some issues. I am talking to some Cisco equipment which does not like the ;transport= suffix in the request URI so the socket specification is really a preferred option. In

Re: [OpenSIPS-Users] apt.opensips.org not reachable

First of all a Happy New Year to all!!! Is there a reason why http://apt.opensips.org is down? I know we can compile from github but still it is very convenient to use apt. >From a project as Opensips I would expect information to be a little accurate (see

Re: [OpenSIPS-Users] Sending to TLS destinations with dispatcher module (2.1)

Hi, Pete! I did not test this, but I think you can catch the 477 by checking the returned code of the t_relay() exit code. If it fails, simply call the use_next_gw() function. Best regards, Răzvan Crainea OpenSIPS Solutions www.opensips-solutions.com On 01/04/2016 12:43 PM, Pete Kelly

Re: [OpenSIPS-Users] apt.opensips.org not reachable

Hello all! Unfortunately the maintainers of the code itself[1] are not the maintainers of the repositories[2]. We've been trying to get in touch with the maintainers of the debian repo[2] for a while, but unfortunately without any success. The reason why we have not yet started configuring

Re: [OpenSIPS-Users] apt.opensips.org not reachable

That repository has been down since at least the 8th of December 2015. Back then I said that we were interested in hosting the APT repository for some time but heard nothing from the maintainers. That offer is still standing by the way. --- Best regards Frederik Bjerggard Nielsen Technical

Re: [OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

Take a look to OverSIP. Il 05/01/2016 07.42, suganthi karthick ha scritto: Hi all, I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE, DTLS-SRTP. The webrtc clients can be JsSIP or any JSON based webrtc client. The

[OpenSIPS-Users] MenuConfig Compile Fail on Solaris Sparc

I am trying to install Opensips 2.1 on Solaris Sparc 10 and I am getting this error when I perform "make menuconfig": # make menuconfig Usage: grep -hblcnsviw pattern file . . . /bin/sh: proto_: not found make -C menuconfig make[1]: Entering directory

Re: [OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

Thanks for the reply. Whether OverSIPS has support for ICE,STUN,DTLS-SRTP? Since the existing conference bridge platform is in C implementation, we thought of using openSIPS Thanks. On Tue, Jan 5, 2016 at 12:12 PM, suganthi karthick wrote: > Hi all, > > I need to

[OpenSIPS-Users] Can OpenSIPS can be used as a WebRTC gateway for JsSIP client and WebRTC client?

Hi all, I need to implement a WebRTC gateway for an existing conference bridge. The WebRTC gateway has to support Signaling, ICE, DTLS-SRTP. The webrtc clients can be JsSIP or any JSON based webrtc client. The conference bridge is an existing working one for SIP clients, and I am trying to add

[OpenSIPS-Users] presence xpidf

Hello all and Happy New Year! I have a problem with publishing application/xpidf+xml (Xpidf) presence info with OpenSIPS mi (ver11). It seems like it is not supported. The xpidf xml body is something like this: After browsing around the source I think there