<div dir="ltr">We don't have an ATA and fax machine.<br><br>The whole point (as I specified in the header and initial message) is the attempt to use "Fax for Asterisk" to send the message.<br><br>As I showed in the logs, the remote carrier sends a proper T.38 reINVITE, but our Asterisk doesn't accept, despite the fact that this provider is defined in sip.conf with both canreinvite and t38pt_udptl enabled, so the only question is (as far as we understand) is why in this scenario, the T.38 is rejected.<br>
<br>Here are the logs (sip debug is open) again, since we get the reINVITE:<br><br><--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --->
<br>INVITE <a class="moz-txt-link-freetext" href="sip:1234567@10.0.0.3:5060">sip:1234567@10.0.0.3:5060</a> SIP/2.0
<br>Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8
<br>From: <a class="moz-txt-link-rfc2396E" href="sip:98765432@xxx.xxx.xxx.xx8:5060"><sip:98765432@xxx.xxx.xxx.xx8:5060></a>;tag=gK0d817deb
<br>To: "Fax" <a class="moz-txt-link-rfc2396E" href="sip:1234567@yyy.yyy.yyy.yyy"><sip:1234567@yyy.yyy.yyy.yyy></a>;tag=as0ddeacb5
<br>Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy">74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy</a>
<br>CSeq: 1785 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: <a class="moz-txt-link-rfc2396E" href="sip:98765432@xxx.xxx.xxx.xx8:5060"><sip:98765432@xxx.xxx.xxx.xx8:5060></a>
<br>Supported: timer
<br>Session-Expires: 1800;refresher=uas
<br>Min-SE: 90
<br>Content-Length: 303
<br>Content-Disposition: session; handling=required
<br>Content-Type: application/sdp
<br>
<br>v=0
<br>o=Sonus_UAC 218 7126 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 6202 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
<a class="moz-txt-link-abbreviated" href="mailto:74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy">74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy</a>
<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=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8
<br>From: <a class="moz-txt-link-rfc2396E" href="sip:98765432@xxx.xxx.xxx.xx8:5060"><sip:98765432@xxx.xxx.xxx.xx8:5060></a>;tag=gK0d817deb
<br>To: "Fax" <a class="moz-txt-link-rfc2396E" href="sip:1234567@yyy.yyy.yyy.yyy"><sip:1234567@yyy.yyy.yyy.yyy></a>;tag=as0ddeacb5
<br>Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy">74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy</a>
<br>CSeq: 1785 INVITE
<br>Server: Smartel-PBX
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
<br>Supported: replaces, timer
<br>Contact: <a class="moz-txt-link-rfc2396E" href="sip:1234567@yyy.yyy.yyy.yyy"><sip:1234567@yyy.yyy.yyy.yyy></a>
<br>Content-Length: 0
<br>
<br>
<br><------------>
<br>
<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=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8
<br>From: <a class="moz-txt-link-rfc2396E" href="sip:98765432@xxx.xxx.xxx.xx8:5060"><sip:98765432@xxx.xxx.xxx.xx8:5060></a>;tag=gK0d817deb
<br>To: "Fax" <a class="moz-txt-link-rfc2396E" href="sip:1234567@yyy.yyy.yyy.yyy"><sip:1234567@yyy.yyy.yyy.yyy></a>;tag=as0ddeacb5
<br>Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy">74ca1e4e3e86a1b873428773477e201f@yyy.yyy.yyy.yyy</a>
<br>CSeq: 1785 INVITE
<br>Server: Smartel-PBX
<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
<br>Supported: replaces, timer
<br>Content-Length: 0
<br><br>Thanks.<br><br>Michael<br><br><br><div class="gmail_quote">On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg <span dir="ltr"><<a href="mailto:dbackeberg@gmail.com">dbackeberg@gmail.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 Tue, Oct 19, 2010 at 11:21 AM, VoIP Question <<a href="mailto:voip.question@gmail.com">voip.question@gmail.com</a>> wrote:<br>
> It's set to yes for this peer.<br>
><br>
> also t38pt_udptl is set to yes.<br>
><br>
> :(<br>
<br>
</div>You don't say anything about what you're trying to send / receive against.<br>
<br>
Here's how you should troubleshoot:<br>
<br>
* start with a 'real fax machine' if you have one, on an analog line<br>
if you have one. If you can't receive / send with that against your<br>
target, blame your target.<br>
* move to audio-pass through fax on asterisk. No T.38. If that works.<br>
* add in T.38<br>
<br>
You will learn things in that process and be able to tell at what<br>
layer your troubles are happening.<br>
<br>
It could be coincidental that things give up during the reinvite. It<br>
could actually be giving up for noise on the line, packet drops, etc.<br>
<br>
At the very least, start recording the call. You'll at least be able<br>
to hear up to the re-invite.<br>
<br>
Definitely record the audio passthrough attempt and listen back to it.<br>
<div><div></div><div class="h5"><br>
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