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Freddy Setiawan

2006-Jun-25 11:37 UTC

[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?

Roland Zagler

2006-Jun-25 11:48 UTC

head link

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

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?
> 
> _______________________________________________
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> 
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>

Steve Totaro

2006-Jun-25 11:54 UTC

head link

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

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)

Neill Wilkinson

2006-Jun-25 12:54 UTC

head link

[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:
 

 

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Freddy Setiawan

2006-Jun-25 17:41 UTC

head link

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

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:
 

 

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Freddy Setiawan

2006-Jun-25 18:00 UTC

head link

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

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).

 

 

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asterisk users - Jun 2006 - FW: Asterisk Quintum A800 SIP Mode

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

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

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

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

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

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

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