search for: voipcall

...sers-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Hadar Pedhazur > Sent: Tuesday, May 09, 2006 2:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] PSTN Incoming call on real line disrupts > VoIPcall over DSL circuit - EXPLAINED > > Juergen K. Zick wrote: > > HI, > > > > well, that was what I expected in my posting yesterday. For me, your > > wiring looks strange. Here in Germany, we have "spiltters" connected to > > the incoming line which have tw...

...ot;1656222"<sip:1656222@192.168.0.1>;tag=as254bbd1a To: <sip:165622270602000@192.168.0.254>;tag=c0a800fe-14 User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport Quintum: 070e00000003008f6506001e03808081 v=0 o=Quintum 2 3131 IN IP4 192.168.0.254 s=VoipCall c=IN IP4 192.168.0.254 t=0 0 m=audio 10240 RTP/AVP 18 c=IN IP4 192.168.0.254 a=rtpmap:18 h723/8000/1 --- (10 headers 8 lines)--- Found RTP audio format 18 Peer audio RTP is at port 192.168.0.254:10240 Found description format h723 Capabilities: us - 0x100 (h723), peer - audio=0x100 (h723)/video=0x0...

Hi all, -------- beacause I am a newbie in the asterisk ralm and the existing documentation could not satisfy I'd like to ask you some Questions: 1. Does somewhere in the Internet exist additional documentations for asterisk configuration ? 2. Does Asterisk work as a standard SIP Proxy ? 3. I am just installing a Asterisk PBX in our institute and additionally I purchased some ot the Snom

...tag=as34cebb79 Call-ID: 71e3976d3e5dffcf3f0eda5b6ecd4b89@192.168.2.20 CSeq: 104 INVITE Record-Route: <sip:98009800@192.168.1.90;ftag=as34cebb79;lr> Contact: <sip:98009800@192.168.1.90:5061> Content-Type: application/sdp Content-Length: 207 v=0 o=Quintum 13544 2493 IN IP4 192.168.1.90 s=VoipCall c=IN IP4 192.168.1.90 t=0 0 m=audio 10672 RTP/AVP 18 101 c=IN IP4 192.168.1.90 a=rtpmap:18 g729/8000/1 a=rtpmap:101 telephone-event/8000/1 10 headers, 9 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.90:10672 Found description format g729 Found descr...

...ioned before, Quintum gives access to their test gateway for the first call, which works good. It delivers following SDP for comparison: sproto |01/01| 2000/01/01|08:09:27:930 v=0 sproto |01/01| 2000/01/01|08:09:27:930 o=Quintum 392 24 IN IP4 208.226.140.40 sproto |01/01| 2000/01/01|08:09:27:930 s=VoipCall sproto |01/01| 2000/01/01|08:09:27:930 c=IN IP4 208.226.140.40 sproto |01/01| 2000/01/01|08:09:27:930 t=0 0 sproto |01/01| 2000/01/01|08:09:27:930 m=audio 10656 RTP/AVP 18 101 sproto |01/01| 2000/01/01|08:09:27:930 c=IN IP4 208.226.140.40 sproto |01/01| 2000/01/01|08:09:27:930 a=rtpmap:18 g729/8000...