[Asterisk-video] 3G to SIP transfer
tech at i6net.com
tech at i6net.com
Mon Dec 3 16:37:15 CST 2007
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 at 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 at 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 at default:1] Answer("Zap/5-1", "") in new stack
-- Executing [912104507 at default:2] h324m_gw("Zap/5-1", "dial at default") in
new stack
-- Executing [dial at default:1] h324m_gw_answer("Local/dial at default-4934,2",
"") in new stack
-- Executing [dial at default:2] Dial("Local/dial at default-4934,2",
"SIP/600 at 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 at octavius.i6net.org SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK052dfe35;rport
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>
Contact: <sip:699435965 at 193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 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
v=0
o=root 17239 17239 IN IP4 193.22.119.85
s=session
c=IN IP4 193.22.119.85
b=CT:384
t=0 0
m=audio 10050 RTP/AVP 96 8 0 101
a=rtpmap:96 AMR/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10004 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
---
-- Called 600 at octavius.i6net.org
[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 SIP/octavius.i6net.org-0830c088 to 524288
quartus*CLI>
<--- SIP read from 62.22.9.77:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
193.22.119.85:5060;branch=z9hG4bK052dfe35;received=193.22.119.85;rport=5060
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
CSeq: 102 INVITE
User-Agent: Divina
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:600 at 62.22.9.77>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
quartus*CLI>
<--- SIP read from 62.22.9.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
193.22.119.85:5060;branch=z9hG4bK052dfe35;received=193.22.119.85;rport=5060
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>;tag=as7133a043
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
CSeq: 102 INVITE
User-Agent: Divina
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:600 at 62.22.9.77>
Content-Type: application/sdp
ontent-Length: 366
v=0
o=root 19599 19599 IN IP4 62.22.9.77
s=session
c=IN IP4 62.22.9.77
b=CT:384
t=0 0
m=audio 10024 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 10026 RTP/AVP 34 103
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=sendrecv
<------------->
--- (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 at 62.22.9.77>
set_destination: Parsing <sip:600 at 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 at 62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK39bd2334;rport
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>;tag=as7133a043
Contact: <sip:699435965 at 193.22.119.85>
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
CSeq: 102 ACK
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
---
-- SIP/octavius.i6net.org-0830c088 answered Local/dial at 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 at 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 at default-4934,2 compatible with
SIP/octavius.i6net.org-0830c088
Scheduling destruction of SIP dialog
'2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:600 at 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 at 62.22.9.77 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.85:5060;branch=z9hG4bK23d8c153;rport
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>;tag=as7133a043
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
CSeq: 103 BYE
User-Agent: Divina
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (default, dial, 2) exited non-zero on
'Local/dial at default-4934,2'
== Spawn extension (default, 912104507, 2) exited non-zero on 'Zap/5-1'
-- Hungup 'Zap/5-1'
quartus*CLI>
<--- SIP read from 62.22.9.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
193.22.119.85:5060;branch=z9hG4bK23d8c153;received=193.22.119.85;rport=5060
From: "699435965" <sip:699435965 at 193.22.119.85>;tag=as1d80f1af
To: <sip:600 at octavius.i6net.org>;tag=as7133a043
Call-ID: 2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85
CSeq: 103 BYE
User-Agent: Divina
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:600 at 62.22.9.77>
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2d9b830c0f58f7d72fe1ab4d4d708cdf at 193.22.119.85'
Method: INVITE
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