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: 5ca18dee412172f54096c30c4f30485b@192.168.0.1 CSeq: 102 OPTIONS From: "Unknown"<sip:Unknown@192.168.0.1>;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 '5ca18dee412172f54096c30c4f30485b@192.168.0.1' asterisk1*CLI> Destroying call '4e3311ae44e8fff01a7600a85a84cec8@192.168.0.1' 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:165622270602000@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a To: <sip:165622270602000@192.168.0.254> Contact: <sip:1656222@192.168.0.1> Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 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:165622270602000@192.168.0.254 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a To: <sip:165622270602000@192.168.0.254> Contact: <sip:1656222@192.168.0.1> Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 CSeq: 102 INVITE From: "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: 0b023236 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 100 Trying Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 CSeq: 102 INVITE From: "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: 0b023236 --- (8 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 192.168.0.254:5060: SIP/2.0 183 Session Progress Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 Content-Length: 162 Content-Type: application/sdp CSeq: 102 INVITE From: "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 (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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 Content-Length: 162 Content-Type: application/sdp CSeq: 102 INVITE From: "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 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?
I'm not familiar with Quintum, but what codec do you mean at the "allow=" line in sip.conf with "h723"?> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Freddy Setiawan > Sent: Sunday, June 25, 2006 8:37 PM > To: asterisk-users@lists.digium.com > 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: 5ca18dee412172f54096c30c4f30485b@192.168.0.1 > CSeq: 102 OPTIONS > From: "Unknown"<sip:Unknown@192.168.0.1>;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 '5ca18dee412172f54096c30c4f30485b@192.168.0.1' > asterisk1*CLI> > Destroying call '4e3311ae44e8fff01a7600a85a84cec8@192.168.0.1' > 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:165622270602000@192.168.0.254 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a > To: <sip:165622270602000@192.168.0.254> > Contact: <sip:1656222@192.168.0.1> > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > 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:165622270602000@192.168.0.254 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a > To: <sip:165622270602000@192.168.0.254> > Contact: <sip:1656222@192.168.0.1> > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > 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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > CSeq: 102 INVITE > From: "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: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 100 Trying > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > CSeq: 102 INVITE > From: "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: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 183 Session Progress > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "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 (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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "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 > 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 >
Freddy Setiawan wrote:> 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: 5ca18dee412172f54096c30c4f30485b@192.168.0.1 > CSeq: 102 OPTIONS > From: "Unknown"<sip:Unknown@192.168.0.1>;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 '5ca18dee412172f54096c30c4f30485b@192.168.0.1' > asterisk1*CLI> > Destroying call '4e3311ae44e8fff01a7600a85a84cec8@192.168.0.1' > 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:165622270602000@192.168.0.254 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a > To: <sip:165622270602000@192.168.0.254> > Contact: <sip:1656222@192.168.0.1> > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > 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:165622270602000@192.168.0.254 SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport > From: "1656222" <sip:1656222@192.168.0.1>;tag=as254bbd1a > To: <sip:165622270602000@192.168.0.254> > Contact: <sip:1656222@192.168.0.1> > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > 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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > CSeq: 102 INVITE > From: "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: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 100 Trying > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > CSeq: 102 INVITE > From: "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: 0b023236 > --- (8 headers 0 lines)--- > asterisk1*CLI> > <-- SIP read from 192.168.0.254:5060: > SIP/2.0 183 Session Progress > Call-ID: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "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 > (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: 44f37f304731f0ef212d4ffd38846d1c@192.168.0.1 > Content-Length: 162 > Content-Type: application/sdp > CSeq: 102 INVITE > From: "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 > 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? > > _______________________________________________ >Try Ulaw. 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)
In the quintum also check you have a codec profile: Example (below has alaw and G729 configure in the codec profile): CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) config-VoiceCodec-2* show VoiceCodec-2 : name : (Not Set) name CodecVoiceCoding : 8 G.711 A-Law CodecPayloadSize : 1280 bits config-VoiceCodec-2* this profile should be attached to you IP Routing Group (IPRG). Neill..;o) =================================== Try Ulaw. 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)> 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:-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060625/8b1a6aaa/attachment.htm
Thanks. Gonna try today. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neill Wilkinson Sent: Monday, June 26, 2006 3:54 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode In the quintum also check you have a codec profile: Example (below has alaw and G729 configure in the codec profile): CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) config-VoiceCodec-2* show VoiceCodec-2 : name : (Not Set) name CodecVoiceCoding : 8 G.711 A-Law CodecPayloadSize : 1280 bits config-VoiceCodec-2* this profile should be attached to you IP Routing Group (IPRG). Neill..;o) =================================== Try Ulaw. 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)> 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:-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060625/57007223/attachment.htm
I tried to change the codec to ulaw but still cannot do anything. I got this on my Asterisk box: ---------------------------------------------------------------------------- ----- Found RTP audio format 0 Peer audio RTP is at port 192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) 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: 2a5bb110693fbc7259ceaf6c3928a050@192.168.0.1 Content-Length: 160 Content-Type: application/sdp CSeq: 102 INVITE From: "1656222"<sip:1656222@192.168.0.1>;tag=as20454a28 To: <sip:16562227279561@192.168.0.254>;tag=c0a800fe-b User-Agent: Quintum/1.0.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK63e689bf;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 0 c=IN IP4 192.168.0.254 a=rtpmap:0 pcmu/8000/1 --- (9 headers 8 lines)--- Found RTP audio format 0 Peer audio RTP is at port 192.168.0.254:10240 Found description format pcmu Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) And this on the quintum box : ---------------------------------------------------------------------------- ----- CH : 29263617:sip[0]: sip:RcvIncomingCall CH : 29263618:sip[0]: osipcall:RcvSetup, my media type=4 CH : 29263618:chsip : bandwidth info: max=-1 cur=12600. CH : 29263618:chsip: Media present in Setup CH : 29263618:chsip: Setting remote rtp port=192.168.0.1:19126. CH : 29263618:Remote side packet saver version = 2. CH : 29263618:CallInfo[0xd3c82c]: origCalled.digit(16562227279561) . CH : 29263618:sip[34/0]: osipcall:stackSendCallProc CH : 29263618:sent message to sip: msg=7; ua=1 CH : 29263618:Routing requested for: public(1) orig=16562227279561 public(1) normalized=16562227279561 route code= tg=0. CH : 29263618:1 match(es) found: 3 CH : 29263618:CasTG[3]: newTermCall: selected line=256 chan=256. CH : 29263618:Route response(34): result=1 cause=0. CH : 29263618:udp connect: 9 11 CH : 29263618: c0a800fe 10240 c0a80001 19126 CH : 29263618:TBCSM[34]: Setup from peer=0xd3c808 NP=0x0 NT=0x0. CH : 29263618:OrigNum=16562227279561 NormNum=16562227279561 TranNum=0227279561 OrigDest=. CH : 29263618:[2: 1] sent message to cas: Setup CH : 29263630:tsi connect: 001 202 01 CH : 29263630:TsiConnXlate: 0:1, 2:2 CH : 29263657:tsi disconnect: 001 202 01 CH : 29263657:TsiDiscXlate: 0:1, 2:2 CH : 29263657:[2: 1] received message from cas: Call-Proc CH : 29263664:tsi connect: 001 210 10 CH : 29263664:TsiConnXlate: 2:10, 0:1 CH : 29263884:tsi disconnect: 001 210 10 CH : 29263884:TsiDiscXlate: 2:10, 0:1 CH : 29263884:[2: 1] received message from cas: Alert CH : 29263884:sip[34/0]: osipcall:stackSendProg CH : 29263884:sent message to sip: msg=9; ua=1 CH : 29263884:tsi connect: 001 209 01 CH : 29263884:TsiConnXlate: 0:1, 2:9 CH : 29263884:tsi connect: 001 209 10 CH : 29263884:TsiConnXlate: 2:9, 0:1 CH : 29263884:sip[34/0]: osipcall:stackSendAlert CH : 29263884:sent message to sip: msg=10; ua=1 Followed are my quintum dsp settings: ----------------------------------------------------------------- Voice Coding algorithm = 9 --> for G711 U-law=9 Voice Information Field size = 1280 bits Silence Suppression = Enable(1) Minimum Jitter buffer = 60 msec Maximum Jitter buffer = 150 msec Receive Gain (PCM -> IP) = 2 dB Transmit Gain (IP -> PCM) = 0 dB Digit Relay = 0 Fax Relay Type = 0 Fax Maximum Rate = 144 Fax Playout FIFO nominal delay = 600 Fax Coding = 0 Packet Saver = Disabled Idle Time = 0 Answer Supervision Options = 0 _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neill Wilkinson Sent: Monday, June 26, 2006 3:54 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode In the quintum also check you have a codec profile: Example (below has alaw and G729 configure in the codec profile): CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) CodecProfile-default : name : (Not Set) name VoiceCodecAttached[1] : VoiceCodec-1 VoiceCodecAttached[2] : VoiceCodec-2 VoiceCodecAttached[3..8] : (unspecified) config-VoiceCodec-2* show VoiceCodec-2 : name : (Not Set) name CodecVoiceCoding : 8 G.711 A-Law CodecPayloadSize : 1280 bits config-VoiceCodec-2* this profile should be attached to you IP Routing Group (IPRG). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060625/e391883f/attachment.htm