I'm not familiar with Quintum, but what codec do you mean at the "allow=" line in sip.conf with "h723"?
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Freddy Setiawan > Sent: Sunday, June 25, 2006 8:37 PM > To: [email protected] > Subject: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode > > Hello, > > I got Quintum A800 with the P5-2-1 firmware. I configure my > asterisk trunk as followed: > > [SIP_BD1] > type=peer > qualify=yes > host=192.168.0.254 > disallow=all > context=from-pstn > allow=h723 > > and inside the quantum I change the config sip useragent to > 5060. Up to this part if I run sip show peers, I got: > > asterisk1*CLI> sip show peers > Name/username Host Dyn Nat ACL Port Status > SIP_BD1 192.168.0.254 5060 > OK (56 ms) > > Which seems that I can connect to the quantum A800, but when > ever I tried to call I can't get the phone connected. I mean > the destination phone was ring and picked up, but on the pap2 > device I didn't hear any voice, as the destination phone also > doesn't heard any voice. > > Followed are my sip debug for the SIP_BD1: > =~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.06.25 00:10:51 > =~=~=~=~=~=~=~=~=~=~=~= > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 200 OK > Call-ID: [EMAIL PROTECTED] > CSeq: 102 OPTIONS > From: "Unknown"<sip:[EMAIL PROTECTED]>;tag=as30cbdfca > To: <sip:192.168.0.254> > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK7ca47b5b;rport > --- (6 headers 0 lines)--- > Destroying call '[EMAIL PROTECTED]' > asterisk1*CLI> > Destroying call '[EMAIL PROTECTED]' > asterisk1*CLI> > We're at 192.168.0.1 port 12580 > Adding codec 0x100 (h723) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > 13 headers, 11 lines > Reliably Transmitting (no NAT) to 192.168.0.254:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Sat, 24 Jun 2006 16:12:21 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 235 > > v=0 > o=root 3131 3131 IN IP4 192.168.0.1 > s=session > c=IN IP4 192.168.0.1 > t=0 0 > m=audio 12580 RTP/AVP 18 101 > a=rtpmap:18 H723/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > --- > asterisk1*CLI> > Retransmitting #1 (no NAT) to 192.168.0.254:5060: > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]> > Contact: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Sat, 24 Jun 2006 16:12:21 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 235 > > v=0 > o=root 3131 3131 IN IP4 192.168.0.1 > s=session > c=IN IP4 192.168.0.1 > t=0 0 > m=audio 12580 RTP/AVP 18 101 > a=rtpmap:18 H723/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > --- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 100 Trying > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > From: "1656222"<sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]>;tag=c0a800fe-14 > User-Agent: Quintum/1.0.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > Quintum: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 100 Trying > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INVITE > From: "1656222"<sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]>;tag=c0a800fe-14 > User-Agent: Quintum/1.0.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > Quintum: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 183 Session Progress > Call-ID: [EMAIL PROTECTED] > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "1656222"<sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]>;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 (nothing), combined - 0x100 (h723) Non-codec > capabilities: us - 0x1 (telephone-event), peer - 0x0 > (nothing), combined - 0x0 (nothing) asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 180 Ringing > Call-ID: [EMAIL PROTECTED] > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "1656222"<sip:[EMAIL PROTECTED]>;tag=as254bbd1a > To: <sip:[EMAIL PROTECTED]>;tag=c0a800fe-14 > User-Agent: Quintum/1.0.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > v=0 > o=Quintum 3 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 > > > any idea what is the problem? 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