Posted: Mon Dec 03, 2007 10:49 pm Post subject: [Asterisk-video] 3G to SIP transfer
Hello alls,
I am trying to transfer a 3G video call to a SIP outgoing call using the Dial
command.
Asterisk seems to fail searching a codec translator (I don't know if it's for
the video or the audio stream).
[Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find a
codec translation path from unknown to unknown
[Dec 3 17:37:57] WARNING[18752]: channel.c:3395 ast_channel_make_compatible:
Unable to set read format on channel Local/dial@default-4934,2 to 524288
[Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to drop
call because I couldn't make Local/dial@default-4934,2 compatible with
SIP/octavius.i6net.org-0830c088
Have someone an idea ?
Thanks,
Tech from i6net
Here the full Asterisk CLI traces :
quartus*CLI> sip debug
SIP Debugging re-enabled
-- Accepting call from '699435965' to '912104507' on channel 0/5, span 1
-- Executing [912104507@default:1] Answer("Zap/5-1", "") in new stack
-- Executing [912104507@default:2] h324m_gw("Zap/5-1", "dial@default") in
new stack
-- Executing [dial@default:1] h324m_gw_answer("Local/dial@default-4934,2",
"") in new stack
-- Executing [dial@default:2] Dial("Local/dial@default-4934,2",
"SIP/600@octavius.i6net.org") in new stack
Video is at 193.22.119.85 port 10004
Audio is at 193.22.119.85 port 10050
Adding codec 0x2000 (amr) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.22.9.77:5060:
INVITE sip:600@octavius.i6net.org SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK052dfe35;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>
Contact: <sip:699435965@193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 102 INVITE
User-Agent: Divina
Max-Forwards: 70
Date: Mon, 03 Dec 2007 16:37:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 397
<------------->
--- (12 headers 18 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Found RTP video format 103
Peer audio RTP is at port 62.22.9.77:10024
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Found description format H263 for ID 34
Found description format h263-1998 for ID 103
Capabilities: us - 0x18000c (ulaw|alaw|h263|h263p), peer - audio=0x18000c
(ulaw|alaw|h263|h263p)/video=0x180000 (h263|h263p), combined - 0x18000c
(ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 62.22.9.77:10024
Peer video RTP is at port 62.22.9.77:10026
list_route: hop: <sip:600@62.22.9.77>
set_destination: Parsing <sip:600@62.22.9.77> for address/port to send to
set_destination: set destination to 62.22.9.77, port 5060
Transmitting (no NAT) to 62.22.9.77:5060:
ACK sip:600@62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK39bd2334;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>;tag=as7133a043
Contact: <sip:699435965@193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 102 ACK
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
---
-- SIP/octavius.i6net.org-0830c088 answered Local/dial@default-4934,2
[Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find a
codec translation path from unknown to unknown
[Dec 3 17:37:57] WARNING[18752]: channel.c:3395 ast_channel_make_compatible:
Unable to set read format on channel Local/dial@default-4934,2 to 524288
[Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to drop
call because I couldn't make Local/dial@default-4934,2 compatible with
SIP/octavius.i6net.org-0830c088
Scheduling destruction of SIP dialog
'2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:600@62.22.9.77> for address/port to send to
set_destination: set destination to 62.22.9.77, port 5060
Reliably Transmitting (no NAT) to 62.22.9.77:5060:
BYE sip:600@62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK23d8c153;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>;tag=as7133a043
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 103 BYE
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
I am trying to transfer a 3G video call to a SIP outgoing call using the Dial
command.
Asterisk seems to fail searching a codec translator (I don't know if it's for
the video or the audio stream).
[Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find a
codec translation path from unknown to unknown
[Dec 3 17:37:57] WARNING[18752]: channel.c:3395 ast_channel_make_compatible:
Unable to set read format on channel Local/dial@default-4934,2 to 524288
[Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to drop
call because I couldn't make Local/dial@default-4934,2 compatible with
SIP/octavius.i6net.org-0830c088
Have someone an idea ?
Thanks,
Tech from i6net
Here the full Asterisk CLI traces :
quartus*CLI> sip debug
SIP Debugging re-enabled
-- Accepting call from '699435965' to '912104507' on channel 0/5, span 1
-- Executing [912104507@default:1] Answer("Zap/5-1", "") in new stack
-- Executing [912104507@default:2] h324m_gw("Zap/5-1", "dial@default") in
new stack
-- Executing [dial@default:1] h324m_gw_answer("Local/dial@default-4934,2",
"") in new stack
-- Executing [dial@default:2] Dial("Local/dial@default-4934,2",
"SIP/600@octavius.i6net.org") in new stack
Video is at 193.22.119.85 port 10004
Audio is at 193.22.119.85 port 10050
Adding codec 0x2000 (amr) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.22.9.77:5060:
INVITE sip:600@octavius.i6net.org SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK052dfe35;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>
Contact: <sip:699435965@193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 102 INVITE
User-Agent: Divina
Max-Forwards: 70
Date: Mon, 03 Dec 2007 16:37:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 397
<------------->
--- (12 headers 18 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Found RTP video format 103
Peer audio RTP is at port 62.22.9.77:10024
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Found description format H263 for ID 34
Found description format h263-1998 for ID 103
Capabilities: us - 0x18000c (ulaw|alaw|h263|h263p), peer - audio=0x18000c
(ulaw|alaw|h263|h263p)/video=0x180000 (h263|h263p), combined - 0x18000c
(ulaw|alaw|h263|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 62.22.9.77:10024
Peer video RTP is at port 62.22.9.77:10026
list_route: hop: <sip:600@62.22.9.77>
set_destination: Parsing <sip:600@62.22.9.77> for address/port to send to
set_destination: set destination to 62.22.9.77, port 5060
Transmitting (no NAT) to 62.22.9.77:5060:
ACK sip:600@62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK39bd2334;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>;tag=as7133a043
Contact: <sip:699435965@193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 102 ACK
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
---
-- SIP/octavius.i6net.org-0830c088 answered Local/dial@default-4934,2
[Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find a
codec translation path from unknown to unknown
[Dec 3 17:37:57] WARNING[18752]: channel.c:3395 ast_channel_make_compatible:
Unable to set read format on channel Local/dial@default-4934,2 to 524288
[Dec 3 17:37:57] WARNING[18752]: app_dial.c:1640 dial_exec_full: Had to drop
call because I couldn't make Local/dial@default-4934,2 compatible with
SIP/octavius.i6net.org-0830c088
Scheduling destruction of SIP dialog
'2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:600@62.22.9.77> for address/port to send to
set_destination: set destination to 62.22.9.77, port 5060
Reliably Transmitting (no NAT) to 62.22.9.77:5060:
BYE sip:600@62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK23d8c153;rport
From: "699435965" <sip:699435965@193.22.119.85>;tag=as1d80f1af
To: <sip:600@octavius.i6net.org>;tag=as7133a043
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf@193.22.119.85
CSeq: 103 BYE
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
Posted: Tue Dec 04, 2007 10:16 am Post subject: [Asterisk-video] 3G to SIP transfer
Hi,
When I call from the SIP phone (ulaw), I can access to the video service (play
and record mp4 file with AMR stream).
So I suppose that the transcoder works with the video service.
I have "patched" the Asterisk with the AMR support from fontventa (I have check
the changes, and it seems to be OK).
Have someone succeed in transfering a 3G call to a Video SIP phone ?
Thanks,
Tech from i6net
Selon Mitul Limbani <mitul@enterux.com>:
Quote:
Hello,
Do you have AMR codec in your SIP Phone ?
I think that might be causing the problem.
Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions,
The Enterprise Linux Company (TM),
www.enterux.com
BR
Sergio
----- Original Message -----
From: tech@i6net.com [mailto:tech@i6net.com]
To: asterisk-video@lists.digium.com
Sent: Tue, 04 Dec 2007 11:06:08 +0100
Subject: Re: [Asterisk-video] 3G to SIP transfer
Hi,
When I call from the SIP phone (ulaw), I can access to the video service (play
and record mp4 file with AMR stream).
So I suppose that the transcoder works with the video service.
I have "patched" the Asterisk with the AMR support from fontventa (I have check
the changes, and it seems to be OK).
Have someone succeed in transfering a 3G call to a Video SIP phone ?
Thanks,
Tech from i6net
Selon Mitul Limbani <mitul@enterux.com>:
Quote:
Hello,
Do you have AMR codec in your SIP Phone ?
I think that might be causing the problem.
Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions,
The Enterprise Linux Company (TM),
www.enterux.com
When I call from the SIP phone (ulaw), I can access to the video service (play
and record mp4 file with AMR stream).
So I suppose that the transcoder works with the video service.
I have "patched" the Asterisk with the AMR support from fontventa (I have check
the changes, and it seems to be OK).
Have someone succeed in transfering a 3G call to a Video SIP phone ?
strange - I did this some time ago and it worked.
Quote:
>> [Dec 3 17:37:57] WARNING[18752]: channel.c:3014 set_format: Unable to find
> a
>> codec translation path from unknown to unknown
^^^^^^^ ^^^^^^
this is strange too.
klaus
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