Boa Tarde Lista.<br><br> I'm having problems in tramissão a fax using T.38.<br><br> My scenario is:<br> Asterisk 1.6.0.5<br> 2 ATA of Intelbras 2210.<br> ReceiveFAX in the asterisk.<br><br> 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.<br>
<br> rtp.conf:<br>[general]<br>rtpstart=17000<br>rtpend=33000<br><br>udptl.conf:<br>[general]<br>udptlstart=4000<br>udptlend=4999<br>T38FaxUdpEC = t38UDPRedundancy<br>T38FaxMaxDatagram = 400<br>udptlfecentries = 3<br>udptlfecspan = 3<br>
<br>sip.conf:<br>t38pt_udptl = yes<br>t38pt_rtp=no<br>t38pt_tcp=no<br><br>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.<br>
<br>Example of the signal is given when a fax:<br>18:05:28.931701 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439<br>18:05:29.047186 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 573<br>18:05:29.163231 IP XX.XX.XX.67.1024 > XX.XX.XX.66.sip: SIP, length: 439<br>
18:05:31.336965 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36<br>18:05:36.339933 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36<br>18:05:41.338790 IP XX.XX.XX.67 > XX.XX.XX.66: ICMP XX.XX.XX.67 udp port 17003 unreachable, length 36<br>
<br>But when using ATA ReceiveFAX for the RTP traffic is constant until the passage of the fax.<br><br>The next sip debug peer at the time that the fax is not:<br><br><--- Transmitting (NAT) to XX.XX.XX.67:1024 ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67<br>From: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>To: sip:2007@XX.XX.XX.66;tag=as46031e07<br>
Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 51 INVITE<br>User-Agent: Asterisk PBX 1.6.0.5<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Contact: <sip:2007@XX.XX.XX.66><br>
Content-Length: 0<br><br><br><------------><br>Audio is at XX.XX.XX.66 port 30206<br>Adding codec 0x100 (g729) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>engeplus*CLI><br><--- Transmitting (NAT) to XX.XX.XX.67:1024 ---><br>
SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67<br>From: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>To: sip:2007@XX.XX.XX.66;tag=as46031e07<br>
Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 51 INVITE<br>User-Agent: Asterisk PBX 1.6.0.5<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Contact: <sip:2007@XX.XX.XX.66><br>
Content-Type: application/sdp<br>Content-Length: 283<br><br>v=0<br>o=root 577489170 577489170 IN IP4 XX.XX.XX.66<br>s=Asterisk PBX 1.6.0.5<br>c=IN IP4 XX.XX.XX.66<br>t=0 0<br>m=audio 30206 RTP/AVP 18 96<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><------------><br> -- SIP/2007618-b7cb7cf8 is ringing<br> -- SIP/2007618-b7cb7cf8 answered SIP/2005618-08bd3e50<br>
Audio is at XX.XX.XX.66 port 30206<br>Adding codec 0x100 (g729) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br><br><--- Reliably Transmitting (NAT) to XX.XX.XX.67:1024 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-0Uic90FJs;received=XX.XX.XX.67<br>
From: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>To: sip:2007@XX.XX.XX.66;tag=as46031e07<br>Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 51 INVITE<br>User-Agent: Asterisk PBX 1.6.0.5<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces, timer<br>Contact: <sip:2007@XX.XX.XX.66><br>Content-Type: application/sdp<br>Content-Length: 283<br><br>v=0<br>o=root 577489170 577489171 IN IP4 XX.XX.XX.66<br>s=Asterisk PBX 1.6.0.5<br>c=IN IP4 XX.XX.XX.66<br>
t=0 0<br>m=audio 30206 RTP/AVP 18 96<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:96 telephone-event/8000<br>a=fmtp:96 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br><------------><br>
engeplus*CLI><br><--- SIP read from UDP://XX.XX.XX.67:1024 ---><br>ACK sip:2007@XX.XX.XX.66:5060 SIP/2.0<br>From: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>To: sip:2007@XX.XX.XX.66;tag=as46031e07<br>
Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 51 ACK<br>Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bKdN0f2-iNjO1len<br>Contact: 2005618<<a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a>><br>
Max-Forwards: 70<br>User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008<br>Content-Length: 0<br><br><br><-------------><br>--- (10 headers 0 lines) ---<br>set_destination: Parsing <<a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a>> for address/port to send to<br>
set_destination: set destination to 192.168.2.100, port 5060<br>Reliably Transmitting (NAT) to XX.XX.XX.67:1024:<br>INVITE <a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc;rport<br>
Max-Forwards: 70<br>From: sip:2007@XX.XX.XX.66;tag=as46031e07<br>To: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>Contact: <sip:2007@XX.XX.XX.66><br>Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.0.5<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>X-asterisk-Info: SIP re-invite (External RTP bridge)<br>Content-Type: application/sdp<br>
Content-Length: 365<br><br>v=0<br>o=root 577489170 577489172 IN IP4 XX.XX.XX.66<br>s=Asterisk PBX 1.6.0.5<br>c=IN IP4 XX.XX.XX.66<br>t=0 0<br>m=image 4729 udptl t38<br>a=T38FaxVersion:0<br>a=T38MaxBitRate:9600<br>a=T38FaxFillBitRemoval:0<br>
a=T38FaxTranscodingMMR:0<br>a=T38FaxTranscodingJBIG:0<br>a=T38FaxRateManagement:transferredTCF<br>a=T38FaxMaxBuffer:400<br>a=T38FaxMaxDatagram:400<br>a=T38FaxUdpEC:t38UDPFEC<br><br>---<br>engeplus*CLI><br><--- SIP read from UDP://XX.XX.XX.67:1024 ---><br>
SIP/2.0 100 Trying<br>From: sip:2007@XX.XX.XX.66;tag=as46031e07<br>To: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc<br>
Content-Length: 0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br><br><--- SIP read from UDP://XX.XX.XX.67:1024 ---><br>SIP/2.0 200 OK<br>From: sip:2007@XX.XX.XX.66;tag=as46031e07<br>To: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>
Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 102 INVITE<br>Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK3380e2dc<br>Contact: 2005618<<a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a>><br>
User-Agent: INTELBRAS ATA GKM2210T - Nov 19 2008<br>Allow: INVITE,BYE,CANCEL,OPTIONS,PRACK,NOTIFY,UPDATE,REFER<br>Supported: timer,replaces<br>Content-Type: application/sdp<br>Content-Length: 243<br><br>v=0<br>o=2005618 207176 2 IN IP4 192.168.2.100<br>
s=-<br>c=IN IP4 192.168.2.100<br>t=0 0<br>m=image 17002 udptl t38<br>a=T38FaxVersion:0<br>a=T38MaxBitRate:14400<br>a=T38FaxMaxBuffer:400<br>a=T38FaxUdpEc:t38UDPRedundancy<br>a=T38FaxRateManagement:transferredTCF<br><br><-------------><br>
--- (12 headers 11 lines) ---<br>Got T.38 offer in SDP in dialog CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>Capabilities: us - 0x102 (gsm|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)<br>set_destination: Parsing <<a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a>> for address/port to send to<br>
set_destination: set destination to 192.168.2.100, port 5060<br>Transmitting (NAT) to XX.XX.XX.67:1024:<br>ACK <a href="http://sip:2005618@192.168.2.100:5060">sip:2005618@192.168.2.100:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP XX.XX.XX.66:5060;branch=z9hG4bK732ec166;rport<br>
Max-Forwards: 70<br>From: sip:2007@XX.XX.XX.66;tag=as46031e07<br>To: 2005618<sip:2005618@XX.XX.XX.66>;tag=74tf2-7Q4xE0<br>Contact: <sip:2007@XX.XX.XX.66><br>Call-ID: CcD6S0-VFf0Z8f2@XX.XX.XX.66<br>CSeq: 102 ACK<br>
User-Agent: Asterisk PBX 1.6.0.5<br>Content-Length: 0<br><br><br>---<br>engeplus*CLI><br><br><br>How has the idea to do? Have tried several configurations possible.<br>