Boa Tarde Lista. I'm having problems in tramissão a fax using T.38.
My scenario is: Asterisk 1.6.0.5 2 ATA of Intelbras 2210. ReceiveFAX in the asterisk. Unable to fax when it is a ATA to another user on the Asterisk means, if I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX of Asterisk works perfect, but if I try to pass between two Branches using the same ATA does not work ever. rtp.conf: [general] rtpstart=17000 rtpend=33000 udptl.conf: [general] udptlstart=4000 udptlend=4999 T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 udptlfecentries = 3 udptlfecspan = 3 sip.conf: t38pt_udptl = yes t38pt_rtp=no t38pt_tcp=no I did a tcpdump and when after listening to the signal for a fax transmission, the network traffic for RTP, and only the end of the connection timeout in the show some more data from SIP protocol and the connection drops. Example of the signal is given when a fax: 18:05:28.931701 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439 18:05:29.047186 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 573 18:05:29.163231 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439 18:05:31.336965 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36 18:05:36.339933 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36 18:05:41.338790 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36 But when using ATA ReceiveFAX for the RTP traffic is constant until the passage of the fax. The next sip debug peer at the time that the fax is not: <--- Transmitting (NAT) to XX.XX.XX.67:1024 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.100:5060 ;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67 From: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 To: sip:2...@xx.xx.xx.66;tag=as46031e07 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 51 INVITE User-Agent: Asterisk PBX 1.6.0.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2...@xx.xx.xx.66> Content-Length: 0 <------------> Audio is at XX.XX.XX.66 port 30206 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP engeplus*CLI> <--- Transmitting (NAT) to XX.XX.XX.67:1024 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.2.100:5060 ;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67 From: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 To: sip:2...@xx.xx.xx.66;tag=as46031e07 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 51 INVITE User-Agent: Asterisk PBX 1.6.0.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2...@xx.xx.xx.66> Content-Type: application/sdp Content-Length: 283 v=0 o=root 577489170 577489170 IN IP4 XX.XX.XX.66 s=Asterisk PBX 1.6.0.5 c=IN IP4 XX.XX.XX.66 t=0 0 m=audio 30206 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- SIP/2007618-b7cb7cf8 is ringing -- SIP/2007618-b7cb7cf8 answered SIP/2005618-08bd3e50 Audio is at XX.XX.XX.66 port 30206 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to XX.XX.XX.67:1024 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.100:5060 ;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67 From: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 To: sip:2...@xx.xx.xx.66;tag=as46031e07 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 51 INVITE User-Agent: Asterisk PBX 1.6.0.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:2...@xx.xx.xx.66> Content-Type: application/sdp Content-Length: 283 v=0 o=root 577489170 577489171 IN IP4 XX.XX.XX.66 s=Asterisk PBX 1.6.0.5 c=IN IP4 XX.XX.XX.66 t=0 0 m=audio 30206 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> engeplus*CLI> <--- SIP read from UDP://XX.XX.XX.67:1024 ---> ACK sip:2...@xx.xx.xx.66:5060 SIP/2.0 From: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 To: sip:2...@xx.xx.xx.66;tag=as46031e07 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 51 ACK Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-iNjO1len Contact: 2005618<sip:2005...@192.168.2.100:5060> Max-Forwards: 70 User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing <sip:2005...@192.168.2.100:5060> for address/port to send to set_destination: set destination to 192.168.2.100, port 5060 Reliably Transmitting (NAT) to XX.XX.XX.67:1024: INVITE sip:2005...@192.168.2.100:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc;rport Max-Forwards: 70 From: sip:2...@xx.xx.xx.66;tag=as46031e07 To: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 Contact: <sip:2...@xx.xx.xx.66> Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.0.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 365 v=0 o=root 577489170 577489172 IN IP4 XX.XX.XX.66 s=Asterisk PBX 1.6.0.5 c=IN IP4 XX.XX.XX.66 t=0 0 m=image 4729 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:400 a=T38FaxUdpEC:t38UDPFEC --- engeplus*CLI> <--- SIP read from UDP://XX.XX.XX.67:1024 ---> SIP/2.0 100 Trying From: sip:2...@xx.xx.xx.66;tag=as46031e07 To: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 102 INVITE Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from UDP://XX.XX.XX.67:1024 ---> SIP/2.0 200 OK From: sip:2...@xx.xx.xx.66;tag=as46031e07 To: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 102 INVITE Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc Contact: 2005618<sip:2005...@192.168.2.100:5060> User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008 Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER Supported: timer,replaces Content-Type: application/sdp Content-Length: 243 v=0 o=2005618 207176 2 IN IP4 192.168.2.100 s=- c=IN IP4 192.168.2.100 t=0 0 m=image 17002 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:400 a=T38FaxUdpEc:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF <-------------> --- (12 headers 11 lines) --- Got T.38 offer in SDP in dialog ccd6s0-vff0z...@xx.xx.xx.66 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid ccd6s0-vff0z...@xx.xx.xx.66 Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) set_destination: Parsing <sip:2005...@192.168.2.100:5060> for address/port to send to set_destination: set destination to 192.168.2.100, port 5060 Transmitting (NAT) to XX.XX.XX.67:1024: ACK sip:2005...@192.168.2.100:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK732ec166;rport Max-Forwards: 70 From: sip:2...@xx.xx.xx.66;tag=as46031e07 To: 2005618<sip:2005...@xx.xx.xx.66>;tag=74tf2-7Q4xE0 Contact: <sip:2...@xx.xx.xx.66> Call-ID: ccd6s0-vff0z...@xx.xx.xx.66 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.0.5 Content-Length: 0 --- engeplus*CLI> How has the idea to do? Have tried several configurations possible.
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