Re: [OpenSIPS-Users] [asterisk-users] Register Free Opensips/Asterisk Integration

2013-03-11 Thread Olle E. Johansson
10 mar 2013 kl. 03:04 skrev Nick Khamis sym...@gmail.com: Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an

[OpenSIPS-Users] two UACs behind the same nat

2013-03-11 Thread Nick Chang
Hello I have two UACs behind the same nat. A call B is OK. But it's not voice. Can everyone give me a suggestion?? Thanks Nick ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Register Free Opensips/Asterisk Integration

2013-03-11 Thread Schneur Rosenberg
I have a similar setup and I use the full URI for incoming calls, so lets say the OpenSIPS server is at sip1.mycarrier.com and I want to send the call to a sip user called 101 then I send the call to 1...@sip1.mycarrier.com On Sun, Mar 10, 2013 at 4:04 AM, Nick Khamis sym...@gmail.com wrote:

Re: [OpenSIPS-Users] ACK Timer

2013-03-11 Thread M.Khaled W Chehab
Dear Nicolas, I am using opensips 1.8.2 tls version ,the problem still exist while even while I am using the timer ,please update me if you have a solution . if (has_totag()) { if ( is_method(INVITE)) { $avp(timeout2) = 3; } else

Re: [OpenSIPS-Users] NAT

2013-03-11 Thread Adrian Georgescu
Only in the client you can check such thing. Adrian On Mar 6, 2013, at 7:20 PM, Leonardo Uzcudun wrote: Hello Saul: I'm still trying to configure ICE support. How could i check if it is working? Thanks, Leo. Da: Saúl Ibarra Corretgé s...@ag-projects.com A: OpenSIPS users mailling

Re: [OpenSIPS-Users] Register Free Opensips/Asterisk Integration

2013-03-11 Thread Nick Khamis
Thank you so much for your responses! Schneur, I know that we were working on the similar architectures at some point, and had the same questions starting up. With your approach, do you still have the answering machine functionality defined by Asterisk (e.g., exten = _1XXX,1,Dial(SIP/${EXTEN},

Re: [OpenSIPS-Users] NAT

2013-03-11 Thread leo
Ouch Da: Adrian Georgescu [via OpenSIPS (Open SIP Server)] ml-node+s1449251n7585234...@n2.nabble.com A: leo uzcud...@yahoo.it Inviato: Lunedì 11 Marzo 2013 16:05 Oggetto: Re: NAT Only in the client you can check such thing. Adrian On Mar 6, 2013, at

Re: [OpenSIPS-Users] Register Free Opensips/Asterisk Integration

2013-03-11 Thread Nick Khamis
I hope it's not asking too much, but is there any way you can share your general configuration? I am referring to how SIP Peers/Friends looks like for a particular user and the OpenSIPS proxy. The problem I am having right now is that the INVITES to extensions are getting bounced around between