[Asterisk-video] 3G to SIP transfer
Mitul Limbani
mitul at enterux.com
Mon Dec 3 19:25:17 CST 2007
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
Quoting tech at i6net.com:
> 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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