Hi, I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk rejects the t38.
Anybody know if is possible to transmit t38 fax with Asterisk 1.4? following settings: --- sip.conf --- [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw context=from-outside t38pt_udptl=yes [operator] qualify=no nat=yes host=189.160.126.201 dtmfmode=rfc2833 context=from-outside type=friend canreinvite=yes t38pt_udptl=yes ;t38pt_rtp=no ;t38pt_tcp=no disallow=all allow=ulaw allow=alaw --- channels/chan_sip.c --- static int global_t38_capability = T38FAX_VERSION_0 | T38FAX_RATE_2400 | T38FAX_RATE_4800 | T38FAX_RATE_7200 | T38FAX_RATE_9600; --- logs --- logs [Nov 13 10:21:11] VERBOSE[25087] logger.c: <--- SIP read from 189.160.126.210:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060 Call-ID: [email protected] From: "Teste"<sip:[email protected]>;tag=as41b028c6 To: <sip:[email protected]>;tag=66359f37 CSeq: 102 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER Contact: <sip:[email protected]:5060;user=phone> Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 31955175 31955175 IN IP4 189.160.126.210 s=Sip Call c=IN IP4 189.160.126.210 t=0 0 m=audio 13474 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Nov 13 10:21:11] VERBOSE[25087] logger.c: --- (10 headers 10 lines) --- [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 0 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 8 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found RTP audio format 101 [Nov 13 10:21:11] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL [Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMU for ID 0 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format PCMA for ID 8 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Found audio description format telephone-event for ID 101 [Nov 13 10:21:11] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 13 10:21:11] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 13 10:21:11] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:11] VERBOSE[13464] logger.c: -- SIP/ctbc-08345a10 is making progress passing it to IAX2/nmg010-to-nmg005-trunk1-2748 <--- SIP read from 189.160.126.210:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6de18e85;rport=5060 Call-ID: [email protected] From: "Teste"<sip:[email protected]>;tag=as41b028c6 To: <sip:[email protected]>;tag=66359f37 CSeq: 102 INVITE Contact: <sip:[email protected]:5060;user=phone> Content-Length: 237 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 31955175 31955176 IN IP4 189.160.126.210 s=Sip Call c=IN IP4 189.160.126.210 t=0 0 m=audio 13474 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> [Nov 13 10:21:15] VERBOSE[25087] logger.c: --- (9 headers 10 lines) --- [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 0 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 8 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found RTP audio format 101 [Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Peer doesn't provide T.38 UDPTL [Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMU for ID 0 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format PCMA for ID 8 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Found audio description format telephone-event for ID 101 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) [Nov 13 10:21:15] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Nov 13 10:21:15] VERBOSE[25087] logger.c: Peer audio RTP is at port 189.160.126.210:13474 [Nov 13 10:21:15] VERBOSE[25087] logger.c: list_route: hop: <sip:[email protected]:5060;user=phone> [Nov 13 10:21:15] DEBUG[25087] chan_sip.c: Strict routing enforced for session [email protected] [Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: Parsing <sip:[email protected]:5060;user=phone> for address/port to send to [Nov 13 10:21:15] VERBOSE[25087] logger.c: set_destination: set destination to 189.160.126.210, port 5060 [Nov 13 10:21:15] VERBOSE[25087] logger.c: Transmitting (NAT) to 189.160.126.210:5060: ACK sip:[email protected]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 189.6.70.47:5060;branch=z9hG4bK6618dc53;rport From: "Teste" <sip:[email protected]>;tag=as41b028c6 To: <sip:[email protected]>;tag=66359f37 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Nov 13 10:21:15] VERBOSE[13464] logger.c: -- SIP/ctbc-08345a10 answered IAX2/nmg010-to-nmg005-trunk1-2748 <--- SIP read from 189.160.126.210:5060 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c Call-ID: [email protected] From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:189.160.126.210:5060> Content-Length: 295 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 31955175 31955177 IN IP4 189.160.126.210 s=Sip Call c=IN IP4 189.160.126.210 t=0 0 m=image 13474 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:176 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy <-------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (10 headers 12 lines) --- [Nov 13 10:21:18] VERBOSE[25087] logger.c: Sending to 189.160.126.210 : 5060 (NAT) [Nov 13 10:21:18] VERBOSE[25087] logger.c: Got T.38 offer in SDP in dialog [email protected] [Nov 13 10:21:18] DEBUG[25087] chan_sip.c: Peer T.38 UDPTL is at port 189.160.126.210:13474 [Nov 13 10:21:18] VERBOSE[25087] logger.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid [email protected] [Nov 13 10:21:18] DEBUG[25087] chan_sip.c: Our T38 capability = (16208), peer T38 capability (3872), joint T38 capability (3872) [Nov 13 10:21:18] VERBOSE[25087] logger.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing) [Nov 13 10:21:18] VERBOSE[25087] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- Transmitting (NAT) to 189.160.126.210:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c;received=189.160.126.210 From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 Call-ID: [email protected] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 <------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- Reliably Transmitting (NAT) to 189.160.126.210:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c;received=189.160.126.210 From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 Call-ID: [email protected] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- SIP read from 189.160.126.210:5060 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bKb9f6b05aa5fb314af51ece37c Call-ID: [email protected] From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (8 headers 0 lines) --- [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- SIP read from 189.160.126.210:5060 ---> BYE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bK4cd5c1fb8ed03c364cf46fbbf Call-ID: [email protected] From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 CSeq: 2 BYE Max-Forwards: 70 Content-Length: 0 <-------------> [Nov 13 10:21:18] VERBOSE[25087] logger.c: --- (8 headers 0 lines) --- [Nov 13 10:21:18] VERBOSE[25087] logger.c: Sending to 189.160.126.210 : 5060 (NAT) [Nov 13 10:21:18] VERBOSE[25087] logger.c: <--- Transmitting (NAT) to 189.160.126.210:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 189.160.126.210:5060;branch=z9hG4bK4cd5c1fb8ed03c364cf46fbbf;received=189.160.126.210 From: <sip:[email protected]>;tag=66359f37 To: "Teste"<sip:[email protected]>;tag=as41b028c6 Call-ID: [email protected] CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 thank in advance. -- Marcus ____________________________________________________________________________________ Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
