Hi,
We want to make callfile based video delivery to 3g enabled cell phones. After a week of googling we found some info and resources and tried to success. The call is placed and answered but it stucks in dialplan after executing h324m_gw_answer(). Technical details are below. Is it possible to make such a call from sip trunk to directly 3g enabled cell phone? If it is not is there another way to do it without an isdn line? (like umts gsm gateway perhaps?) System config ------------- Asterisk 1.4.7.1 with 10217-asterisk-unrestricted-digital-llc-11595-1.4.17.patch Libpri 1.4.4 with 13055-libpri-1.4.4-llc-transmit-receive-patch-0.1.txt patch I successfully compiled below modules with instructions in http://asterisk-party.net/index.php/Asterisk_Video_3G_FR libh324m.so codec_amr.so app_h324m.so app_mp4.so app_rtsp.so callfile -------- Set: CHANNEL(transfercapability)=VIDEO Set: CHANNEL(userinformationlayer1)=38 Channel: SIP/0090xxxxxxx...@voicetrading Context: msg MaxRetries: 1 RetryTime: 60 WaitTime: 30 Extension: s Priority: 1 cli show codecs --------------- ... ... 8192 (1 << 13) (0x2000) audio amr (AMR NB) 65536 (1 << 16) (0x10000) image jpeg (JPEG image) 131072 (1 << 17) (0x20000) image png (PNG image) 262144 (1 << 18) (0x40000) video h261 (H.261 Video) 524288 (1 << 19) (0x80000) video h263 (H.263 Video) 1048576 (1 << 20) (0x100000) video h263p (H.263+ Video) 2097152 (1 << 21) (0x200000) video h264 (H.264 Video) sip.conf -------- [general] videosupport=yes disallow=all allow=h263p allow=h263 allow=h264 allow=h261 allow=amr allow=gsm allow=alaw allow=ulaw extensions.conf --------------- [msg] exten => s,1,Answer exten => s,2,NoOp(ul1=${CHANNEL(userinformationlayer1)}) exten => s,3,NoOp(ul2=${CHANNEL(transfercapability)}) exten => s,4,h324m_gw(sendvi...@3g_video) [3g_video] exten => sendvideo,1,h324m_gw_answer() exten => sendvideo,2,mp4play(/tmp/demo.3gp) cli output ---------- -- Attempting call on SIP/00905357676...@voicetrading for s...@msg:1 (Retry 1) -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected > Channel SIP/voicetrading-0823a580 was answered. -- Executing [...@msg:1] Answer("SIP/voicetrading-0823a580", "") in new stack -- Executing [...@msg:2] NoOp("SIP/voicetrading-0823a580", "ul1=38") in new stack -- Executing [...@msg:3] NoOp("SIP/voicetrading-0823a580", "ul2=VIDEO") in new stack -- Executing [...@msg:4] h324m_gw("SIP/voicetrading-0823a580", "sendvi...@3g_video") in new stack -- Executing [sendvi...@3g_video:1] h324m_gw_answer("Local/sendvi...@3g_video-157d,2", "") in new stack -- Remote UNIX connection -- Remote UNIX connection disconnected == Spawn extension (msg, s, 4) exited non-zero on 'SIP/voicetrading-0823a580' == Spawn extension (3g_video, sendvideo, 1) exited non-zero on 'Local/sendvi...@3g_video-157d,2' Really destroying SIP dialog '[email protected]' Method: BYE sip trace --------- # U 2010/01/27 15:54:56.933028 91.205.172.196:5060 -> 194.120.0.198:5060 INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 27 Jan 2010 13:54:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 372 v=0 o=root 2685 2685 IN IP4 91.205.172.196 s=session c=IN IP4 91.205.172.196 b=CT:384 t=0 0 m=audio 10266 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 19806 RTP/AVP 34 103 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv # U 2010/01/27 15:54:56.942796 194.120.0.198:5060 -> 91.205.172.196:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]> Contact: sip:[email protected]:5060 Call-ID: [email protected] CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE WWW-Authenticate: Digest realm="sipdiscount.com",nonce="3509265298",algorithm=MD5 Content-Length: 0 # U 2010/01/27 15:54:56.943036 91.205.172.196:5060 -> 194.120.0.198:5060 ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK1b37c1ca;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U 2010/01/27 15:54:56.943339 91.205.172.196:5060 -> 194.120.0.198:5060 INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="username", realm="sipdiscount.com", algorithm=MD5, uri="sip:[email protected]", nonce="3509265298", response="feb2e7315123e783252748325b71446e", opaque="" Date: Wed, 27 Jan 2010 13:54:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 372 v=0 o=root 2685 2686 IN IP4 91.205.172.196 s=session c=IN IP4 91.205.172.196 b=CT:384 t=0 0 m=audio 10266 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 19806 RTP/AVP 34 103 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv # U 2010/01/27 15:54:56.953906 194.120.0.198:5060 -> 91.205.172.196:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]> Contact: sip:[email protected]:5060 Call-ID: [email protected] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 # U 2010/01/27 15:54:57.033535 194.120.0.198:5060 -> 91.205.172.196:5060 SIP/2.0 183 Session progress Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]>;tag=c51710acc52b10ac4af9c8301bd806 b Contact: sip:[email protected]:5060 Call-ID: [email protected] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 200 v=0 o=username 1264600385 1264600385 IN IP4 194.120.0.43 s=SIP Call c=IN IP4 194.120.0.43 t=0 0 m=audio 25510 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 # U 2010/01/27 15:55:02.855279 194.120.0.198:5060 -> 91.205.172.196:5060 SIP/2.0 200 Ok Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK3cadce12;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]>;tag=c51710acc52b10ac4af9c8301bd806 b Contact: sip:[email protected]:5060 Call-ID: [email protected] CSeq: 103 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 200 v=0 o=username 1264600391 1264600391 IN IP4 194.120.0.43 s=SIP Call c=IN IP4 194.120.0.43 t=0 0 m=audio 25510 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 # U 2010/01/27 15:55:02.855713 91.205.172.196:5060 -> 194.120.0.198:5060 ACK sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK73a7c26f;rport From: "Unknown" <sip:[email protected]>;tag=as7a873e05 To: <sip:[email protected]>;tag=c51710acc52b10ac4af9c8301bd806 b Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 # U 2010/01/27 15:55:19.904614 194.120.0.198:5060 -> 91.205.172.196:5060 BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 194.120.0.198:5060;branch=z9hG4bK3cadce12 From: <sip:[email protected]>;tag=c51710acc52b10ac4af9c8301bd806 b To: "Unknown" <sip:[email protected]>;tag=as7a873e05 Contact: sip:[email protected]:5060 Call-ID: [email protected] CSeq: 1 BYE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Length: 0 # U 2010/01/27 15:55:19.904820 91.205.172.196:5060 -> 194.120.0.198:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 194.120.0.198:5060;branch=z9hG4bK3cadce12;received=194.120.0.198 From: <sip:[email protected]>;tag=c51710acc52b10ac4af9c8301bd806 b To: "Unknown" <sip:[email protected]>;tag=as7a873e05 Call-ID: [email protected] CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 # U 2010/01/27 15:55:20.865373 91.205.172.196:5060 -> 194.120.0.198:5060 OPTIONS sip:sip.voicetrading.com SIP/2.0 Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK15310eaf;rport From: "Unknown" <sip:[email protected]>;tag=as3f83dde4 To: <sip:sip.voicetrading.com> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 27 Jan 2010 13:55:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 # U 2010/01/27 15:55:20.875367 194.120.0.198:5060 -> 91.205.172.196:5060 SIP/2.0 200 Ok Via: SIP/2.0/UDP 91.205.172.196:5060;branch=z9hG4bK15310eaf;rport From: "Unknown" <sip:[email protected]>;tag=as3f83dde4 To: <sip:sip.voicetrading.com> Contact: sip:194.120.0.198:5060 Call-ID: [email protected] CSeq: 102 OPTIONS Supported: foo User-Agent: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Accept: application/sdp
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