<div dir="ltr">Fair enough Kevin :-) It's just that your documentation for this product is so limited that without extensive search online and the assistance of others, it would have been impossible for us to make any progress and we haven't reached the ReceiveFax part yet ;)<br>
<br>Anyway, specifically, we installed Asterisk 1.6.2.11. As far as we know/understand, the SendFax application is running.<br><br>This is the full log of the call, until it's rejected for the first time. The remote switch resends the INVITEs a few more times, but it's all the same, so I didn't include it:<br>
<br>sip*CLI> -- Attempting call on Local/12345678@outgoing for s@outboundfax:1 (Retry 1)<br>sip*CLI> -- Executing [12345678@outgoing:1] Dial("Local/12345678@outgoing-2c36;2", "SIP/12345678@main,50,tTr") in new stack<br>
sip*CLI> == Using SIP RTP CoS mark 5<br>sip*CLI> == Using SIP VRTP CoS mark 6<br>sip*CLI> == Using UDPTL CoS mark 5<br>sip*CLI> Audio is at yyy.yyy.yyy.yyy port 10714<br>sip*CLI> Adding codec 0x100 (g729) to SDP<br>
sip*CLI> Adding codec 0x2 (gsm) to SDP<br>sip*CLI> Adding non-codec 0x1 (telephone-event) to SDP<br>sip*CLI> Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060:<br>INVITE sip:12345678@xxx.xxx.xxx.xx8 SIP/2.0<br>
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5b6bc617;rport<br>Max-Forwards: 70<br>From: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>To: <sip:12345678@xxx.xxx.xxx.xx8><br>Contact: <sip:98765432@yyy.yyy.yyy.yyy><br>
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 102 INVITE<br>User-Agent: PBX<br>Date: Tue, 19 Oct 2010 16:41:16 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>
Content-Type: application/sdp<br>Content-Length: 312<br><br>v=0<br>o=root 697508180 697508180 IN IP4 yyy.yyy.yyy.yyy<br>s=Asterisk PBX 1.6.2.11<br>c=IN IP4 yyy.yyy.yyy.yyy<br>t=0 0<br>m=audio 10714 RTP/AVP 18 3 101<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>sip*CLI> -- Called 12345678@main<br>sip*CLI> <br>
<--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060<br>From: "Fax" <<a href="http://sip:98765432@10.0.0.3:5060">sip:98765432@10.0.0.3:5060</a>>;tag=as28606a47<br>
To: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 102 INVITE<br>Content-Length: 0<br><br><br><-------------><br>--- (7 headers 0 lines) ---<br>
sip*CLI> <br><--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---><br>SIP/2.0 183 Session Progress<br>Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060<br>From: "Fax" <<a href="http://sip:98765432@10.0.0.3:5060">sip:98765432@10.0.0.3:5060</a>>;tag=as28606a47<br>
To: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 102 INVITE<br>Contact: <sip:12345678@xxx.xxx.xxx.xx8:5060><br>Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH<br>
Content-Length: 262<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br><br>v=0<br>o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8<br>s=SIP Media Capabilities<br>c=IN IP4 xxx.xxx.xxx.xx7<br>
t=0 0<br>m=audio 6256 RTP/AVP 18 101<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><br><-------------><br>--- (11 headers 12 lines) ---<br>
Found RTP audio format 18<br>Found RTP audio format 101<br>Found audio description format G729 for ID 18<br>Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x102 (gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Peer audio RTP is at port xxx.xxx.xxx.xx7:6256<br>sip*CLI> -- SIP/main-0000002a is making progress passing it to Local/12345678@outgoing-2c36;2<br>
sip*CLI> <br><--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---><br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060<br>From: "Fax" <<a href="http://sip:98765432@10.0.0.3:5060">sip:98765432@10.0.0.3:5060</a>>;tag=as28606a47<br>
To: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 102 INVITE<br>Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed<br>
Contact: <sip:12345678@xxx.xxx.xxx.xx8:5060><br>Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH<br>Require: timer<br>Supported: timer<br>Session-Expires: 1800;refresher=uac<br>Content-Length: 262<br>
Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br><br>v=0<br>o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8<br>s=SIP Media Capabilities<br>c=IN IP4 xxx.xxx.xxx.xx7<br>t=0 0<br>m=audio 6256 RTP/AVP 18 101<br>
a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=sendrecv<br>a=maxptime:20<br><br><-------------><br>--- (15 headers 12 lines) ---<br>list_route: hop: <sip:12345678@xxx.xxx.xxx.xx8:5060><br>
set_destination: Parsing <sip:12345678@xxx.xxx.xxx.xx8:5060> for address/port to send to<br>set_destination: set destination to xxx.xxx.xxx.xx8, port 5060<br>Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060:<br>ACK sip:12345678@xxx.xxx.xxx.xx8:5060 SIP/2.0<br>
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK7c7557f9;rport<br>Max-Forwards: 70<br>From: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>To: <sip:12345678@xxx.xxx.xxx.xx8>;tag=gK028217ef<br>
Contact: <sip:98765432@yyy.yyy.yyy.yyy><br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 102 ACK<br>User-Agent: PBX<br>Content-Length: 0<br><br><br>---<br>sip*CLI> -- SIP/main-0000002a answered Local/12345678@outgoing-2c36;2<br>
sip*CLI> > Channel Local/12345678@outgoing-2c36;1 was answered.<br> -- Executing [s@outboundfax:1] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(filename)=/tmp/1585867851.tif") in new stack<br>
[Oct 19 18:41:17] WARNING[31185]: res_fax.c:2454 acf_faxopt_write: channel 'Local/12345678@outgoing-2c36;1' set FAXOPT(filename) to '/tmp/1585867851.tif' is unhandled!<br> -- Executing [s@outboundfax:2] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(ecm)=yes") in new stack<br>
sip*CLI> -- Executing [s@outboundfax:3] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(headerinfo)=Smartel") in new stack<br> -- Executing [s@outboundfax:4] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(localstationid)=+972-72-278-0008") in new stack<br>
-- Executing [s@outboundfax:5] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(maxrate)=14400") in new stack<br> -- Executing [s@outboundfax:6] Set("Local/12345678@outgoing-2c36;1", "FAXOPT(minrate)=2400") in new stack<br>
-- Executing [s@outboundfax:7] SendFAX("Local/12345678@outgoing-2c36;1", "/tmp/1585867851.tif,d") in new stack<br> -- Channel 'Local/12345678@outgoing-2c36;1' sending FAX:<br> -- /tmp/1585867851.tif<br>
sip*CLI> -- Channel 'Local/12345678@outgoing-2c36;1' FAX session '6' started<br>sip*CLI> -- FAX handle 0: [ 000.000152 ], STAT_EVT_STRT_TX st: IDLE rt: IDLENSTX<br> -- FAX handle 0: [ 000.000214 ], STAT_EVT_TX_HW_RDY st: WT_TX_HW_RDY rt: TRDYNHTY<br>
-- FAX handle 0: [ 000.000240 ], P30EVN_SEND_STARTED<br>sip*CLI> == Spawn extension (outgoing, 12345678, 1) exited non-zero on 'Local/12345678@outgoing-2c36;2'<br>sip*CLI> > Channel 'Local/12345678@outgoing-2c36;1' fax session '6', [ 000.075767 ], channel sent 2 frames (40 ms) of silence.<br>
sip*CLI> > Channel 'Local/12345678@outgoing-2c36;1' fax session '6', [ 000.588736 ], stack sent 28 frames (560 ms) of energy.<br>sip*CLI> > Channel 'Local/12345678@outgoing-2c36;1' fax session '6', [ 003.566428 ], stack sent 149 frames (2980 ms) of silence.<br>
sip*CLI> > Channel 'Local/12345678@outgoing-2c36;1' fax session '6', [ 004.089420 ], stack sent 26 frames (520 ms) of energy.<br>sip*CLI> <br><--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---><br>
INVITE <a href="http://sip:98765432@10.0.0.3:5060">sip:98765432@10.0.0.3:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a<br>From: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>
To: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 8530 INVITE<br>Max-Forwards: 70<br>Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH<br>
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed<br>Contact: <sip:12345678@xxx.xxx.xxx.xx8:5060><br>Supported: timer<br>Session-Expires: 1800;refresher=uas<br>Min-SE: 90<br>
Content-Length: 306<br>Content-Disposition: session; handling=required<br>Content-Type: application/sdp<br><br>v=0<br>o=Sonus_UAC 28160 32051 IN IP4 xxx.xxx.xxx.xx8<br>s=SIP Media Capabilities<br>c=IN IP4 xxx.xxx.xxx.xx7<br>
t=0 0<br>m=image 6256 udptl t38<br>a=T38FaxVersion:0<br>a=T38MaxBitRate:9600<br>a=T38FaxRateManagement:transferredTCF<br>a=T38FaxMaxBuffer:262<br>a=T38FaxMaxDatagram:176<br>a=T38FaxUdpEC:t38UDPRedundancy<br>a=sendrecv<br>
<br><-------------><br>--- (16 headers 13 lines) ---<br>Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT)<br>Got T.38 offer in SDP in dialog 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<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>Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.<br><br><--- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---><br>
SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8<br>From: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>To: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>
Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 8530 INVITE<br>Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Contact: <sip:98765432@yyy.yyy.yyy.yyy><br>
Content-Length: 0<br><br><br><------------><br>sip*CLI> <br><--- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 ---><br>SIP/2.0 488 Not acceptable here<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a;received=xxx.xxx.xxx.xx8<br>
From: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>To: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 8530 INVITE<br>
Server: PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces, timer<br>Content-Length: 0<br><br><br><------------><br>sip*CLI> <br><--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---><br>
ACK <a href="http://sip:98765432@10.0.0.3:5060">sip:98765432@10.0.0.3:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK02B020504a6f7a14f9a<br>From: <sip:12345678@xxx.xxx.xxx.xx8:5060>;tag=gK028217ef<br>
To: "Fax" <sip:98765432@yyy.yyy.yyy.yyy>;tag=as28606a47<br>Call-ID: 2d965b0926e0134e0b211f882cbd2cc3@yyy.yyy.yyy.yyy<br>CSeq: 8530 ACK<br>Max-Forwards: 70<br>Content-Length: 0<br><br><br><br>Thanks,<br><br>
Michael<br><br><br><div class="gmail_quote">On Tue, Oct 19, 2010 at 8:56 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">On 10/19/2010 12:01 PM, VoIP Question wrote:<br>
> Digium claims that their FFA is the best and most compatible solution<br>
> and they give one channel for free, but do not provide support for those<br>
> that do not buy more channels, but why buy more channels if the<br>
> free/test one doesn't work?<br>
><br>
> I know they read (and sometimes respond) to this list, so I don't<br>
> understand why they don't clarify this issue.<br>
<br>
</div>When you are asking for free help on a mailing list, patience is a<br>
virtue :-) You posted your question approximately four hours ago.<br>
<div class="im"><br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
</div><div><div></div><div class="h5">-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br></div>