Hello list, I am trying to solve a problem and after unsucessfully chasing forums and google for some hours, I turn to you in hope of a solution. I feel it's just a configuration issue but I just can't get my head wrapped around it.
The situation is basically this: I have an Asterisk connected to an Alcatel OmniPCX via SIP. Asterisk only ever does SIP and has no dedicated hardware phone interface. The Alcatel PBX is connected to the public phone net, and is configured to forward all calls to a certain number to Asterisk. Also when Asterisk dials out, that number is correctly transmitted by the PBX. Asterisk's job is to implement a special highly dynamic call routing, controlled by a script. I tested all functionality I need first with simple Softphones from my work PC. Everything I needed worked fine. Now connected to the PBX it works too, but in certain situations, I simply get no audio at all. Call setup and dynamically calling the correct recipient works fine, but if the callee picks up the phone, there simply is only silence on the line. More precisely, I have the following situation: I call a number (with my desktop phone), the number is picked up by the Alcatel PBX and is calling Asterisk via SIP on a specific extension. Asterisk determines the target, initiates a call via SIP out over the Alcatel and the other phone rings (say my mobile). I can pick it up and the call is connected. Now, if I have canreinvite=no, meaning the connection goes like this: Desk Phone -> PSTN -> Alcatel PBX -> SIP -> Asterisk and Asterisk -> SIP -> Alcatel PBX -> PSTN -> mobile then I hear nothing. There is only silence. Talking with the Alcatel PBX people, they can tell me that their SIP equipment is allocating codec and compressor resources so the media path is "open". I can also confirm that there is RTP network traffic passing to and from Asterisk. But there is only silence. If I change it so, that either source or destination of the call is not going through the PBX but to one of my Softphones registered at Asterisk, then it works fine. The Softphone can receive or initiate the call and there is audio between the two. If I set canreinvite=yes and I have set directrtpsetup=yes to say Asterisk I want it to "shortcut" anyway, since I'm only interested in the call setup and not really in the actual audio/media data, then the above scenario does work. Desktop phone to mobile phone both via Alcatel PBX works fine, except that I don't get the call disconnected when one side hangs up (seems simply to keep the line open with silence from then on). However while that scenario works, call origination then fails. If I perform a call origination, then the first phone rings, and if picked up, the second phone rings exactly once (or actually, it feels like a fraction of "one ring") and then the first phone gets a NO ANSWER/BUSY response right ahead. If I remove once again the canreinvite=yes then the second phone rings normally on call origination and can be picked up, but again, I am having no audio, only silence. So in short, with canreinvite=yes, everything through Asterisk, call forward and call origination works but no audio canreinvite=no, call forward works (but no hangup detection), call origination fails with the called member only receiving one single ring. I really can't find any hints to this, but I think it must be a simple configuration issue on my end. I can provide configuration snippets, but I think the issue is something basic that if someone knows what I am doing wrong, can immediately point me to the answer. Otherwise, here's the sip.conf part where the connection is defined to the PBX: [pbx] type=peer secret=something defaultuser=something fromuser=7889 ; extension we are called with host=10.64.x.y ; IP of PBX sip gateway fromdomain=172.29.x.y ; our IP. canreinvite=yes context=pbxIN direct call forward out again (the scenario above) extensions.conf: exten => 7889,1,Dial(SIP/pbx/0<phonenumber>) If more debug / config information is required, I'll be happy to provide that. Thanks in advance Rene -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
