search for: siphead

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

I've got one that I haven't been able to solve. Hopefully someone else has had this issue. I'm using the paging script in free pbx, which appears to: Send a sipheader autoanswer, Create a conferece Add the phone to the conference But if the user hits the page extension, all the phones auto answer, and if they hang-up before the phones join the conference I end up with dozens of phones off the hook which never hang up. Anyone else seen this or have a solutio...

Hello everybody, I want to use the TIMEOUT() function, but in the CLI the "show functions" command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message "function LANGUAGE not registered". How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l.

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110316/bda02b4d/attachment.htm>

Hi, I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server

Well, this kinda sux. We have three Asterisk servers. Phones register to a single, primary server. When a phone on one wants to reach a phone on another, we use DUNDi to discover the destination pbx and IAX to transfer the call to the other Asterisk box. This seems to be a fairly common practice amongst Asterisk users, yes? Well, what about setting variables before call placement? Say you want

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit extension. No