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 cant get the phone connected. I mean the destination phone was ring and picked up, but on the pap2 device I didnt hear any voice, as the destination phone also doesnt 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? _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
