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

Sun Jun 25 11:37:22 MST 2006

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?