[Asterisk-Users] FW: Asterisk Quintum A800 SIP Mode

Sun Jun 25 11:48:49 MST 2006

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 at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Freddy Setiawan
> Sent: Sunday, June 25, 2006 8:37 PM
> To: asterisk-users at 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 at 192.168.0.1
> CSeq: 102 OPTIONS
> From: "Unknown"<sip:Unknown at 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 at 192.168.0.1'
> asterisk1*CLI>
> Destroying call '4e3311ae44e8fff01a7600a85a84cec8 at 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 at 192.168.0.254 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport
> From: "1656222" <sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 192.168.0.254>
> Contact: <sip:1656222 at 192.168.0.1>
> Call-ID: 44f37f304731f0ef212d4ffd38846d1c at 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 at 192.168.0.254 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK45e00b01;rport
> From: "1656222" <sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 192.168.0.254>
> Contact: <sip:1656222 at 192.168.0.1>
> Call-ID: 44f37f304731f0ef212d4ffd38846d1c at 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 at 192.168.0.1
> CSeq: 102 INVITE
> From: "1656222"<sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 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 at 192.168.0.1
> CSeq: 102 INVITE
> From: "1656222"<sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 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 at 192.168.0.1
> Content-Length: 162
> Content-Type: application/sdp
> CSeq: 102 INVITE
> From: "1656222"<sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 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 at 192.168.0.1
> Content-Length: 162
> Content-Type: application/sdp
> CSeq: 102 INVITE
> From: "1656222"<sip:1656222 at 192.168.0.1>;tag=as254bbd1a
> To: <sip:165622270602000 at 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?
> 
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