Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
We're learning all the time and made some significant progress and some very nice calls scenarios, but specifically with this issue, is there anything we can do to solve the interop problem with this end-point? Thanks. Original Message Subject: Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here) From: Kevin P. Fleming kpflem...@digium.com To: asterisk-users@lists.digium.com Date: Thursday, 21 October, 2010 16:11:00 On 10/20/2010 11:35 AM, VoIP Question wrote: On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Flemingkpflem...@digium.com mailto:kpflem...@digium.com wrote: This was fixed in Asterisk 1.6.2.12 and later releases, so if you were running the current version, you wouldn't have experienced this specific problem. This was listed in the ChangeLog for 1.6.2.12, but unfortunately the commit message the developer wrote did not explain why the change was made or what problem it was addressing, so you wouldn't have noticed it. In any case, upgrading to 1.6.2.12 or later will cure this problem. I upgraded to 1.6.2.13 and now we get this error (with a specific destination, to which we occasionally need to send faxes): WARNING[857]: udptl.c:1087 ast_udptl_write: (SIP/XXX): UDPTL asked to send 50 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. How can we fix it, without risking incompatibility with other end-points? What's a channel driver configuration and where is it? It appears that you need to spend some time learning the basics of Asterisk. In this case, the channel driver is chan_sip, since the channel involved is a SIP channel, and the 'channel driver configuration' is the sip.conf file. It is unfortunate that you have chosen to tackle a very complex task (T.38 interoperability is fraught with problems due to widely varying implementations) as your first experience with Asterisk... there's a lot you'll need to learn to be able to diagnose and troubleshoot problems. Asterisk alone is not 'point and click', and adding T.38 to the mix makes things more complicated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Thanks Kevin,
We managed to get the ReceiveFAX going, while making some minor changes
to the code, like, for example, using the ${UNIQUEID} for the file name.
Regards,
Michael
Original Message
Subject: Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not
acceptable here)
From: Kevin P. Fleming kpflem...@digium.com
To: asterisk-users@lists.digium.com
Date: Thursday, 21 October, 2010 16:13:02
On 10/20/2010 09:35 AM, VoIP Question wrote:
Thank you Kevin,
We'll upgrade our server to 1.6.2.12 and try again.
Another question: Is there (expect for the admin guide that we didn't
succeed to understand the example in) an example somewhere for
ReceiveFax full extensions.conf diaplan? We would like to allocate one
of the extensions that our SIP provider gives us to a fax storage server
or later to email.
The ReceiveFAX example in the Fax For Asterisk administrator's guide is
very straightforward and easy to follow... if you don't understand it,
then you'll need to spend some time learning how the Asterisk dialplan
works. I would highly recommend reading the O'Reilly Asterisk book
(which you can read online for free)... while it is based on Asterisk
1.4, the dialplan concepts documented in it have not changed much in
Asterisk 1.6, and gaining that basic understanding will go a long way
towards helping you be able to resolve these issues on your own.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 04:07 PM, VoIP Question wrote: Hello again, If I set a peer to use G.711 only, they try to process a sent fax in G.711, but Asterisk doesn't like it: WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on channel 'SIP/Main-000a' and T.38 negotiation failed; aborting. What can I do to enable it? What you can do is read the documentation. The built-in help for the SendFAX application shows you how to enable audio FAX on channels that support T.38 (where audio FAX mode is normally disabled for reliability reasons). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 11:35 AM, VoIP Question wrote: On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.com mailto:kpflem...@digium.com wrote: This was fixed in Asterisk 1.6.2.12 and later releases, so if you were running the current version, you wouldn't have experienced this specific problem. This was listed in the ChangeLog for 1.6.2.12, but unfortunately the commit message the developer wrote did not explain why the change was made or what problem it was addressing, so you wouldn't have noticed it. In any case, upgrading to 1.6.2.12 or later will cure this problem. I upgraded to 1.6.2.13 and now we get this error (with a specific destination, to which we occasionally need to send faxes): WARNING[857]: udptl.c:1087 ast_udptl_write: (SIP/XXX): UDPTL asked to send 50 bytes of IFP when far end only prepared to accept 30 bytes; data loss will occur.You may need to override the T38FaxMaxDatagram value for this endpoint in the channel driver configuration. How can we fix it, without risking incompatibility with other end-points? What's a channel driver configuration and where is it? It appears that you need to spend some time learning the basics of Asterisk. In this case, the channel driver is chan_sip, since the channel involved is a SIP channel, and the 'channel driver configuration' is the sip.conf file. It is unfortunate that you have chosen to tackle a very complex task (T.38 interoperability is fraught with problems due to widely varying implementations) as your first experience with Asterisk... there's a lot you'll need to learn to be able to diagnose and troubleshoot problems. Asterisk alone is not 'point and click', and adding T.38 to the mix makes things more complicated. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/20/2010 09:35 AM, VoIP Question wrote: Thank you Kevin, We'll upgrade our server to 1.6.2.12 and try again. Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. The ReceiveFAX example in the Fax For Asterisk administrator's guide is very straightforward and easy to follow... if you don't understand it, then you'll need to spend some time learning how the Asterisk dialplan works. I would highly recommend reading the O'Reilly Asterisk book (which you can read online for free)... while it is based on Asterisk 1.4, the dialplan concepts documented in it have not changed much in Asterisk 1.6, and gaining that basic understanding will go a long way towards helping you be able to resolve these issues on your own. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Thank you Kevin, We'll upgrade our server to 1.6.2.12 and try again. Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. Michael On Wed, Oct 20, 2010 at 4:25 PM, Kevin P. Fleming kpflem...@digium.comwrote: You have a 'Local' channel in between SendFAX and the SIP channel to your other endpoint. In Asterisk 1.6.2.11, chan_local was not properly aware of T.38 negotiation, so it ends up acting as a sort of 'firewall' between the endpoints. This was fixed in Asterisk 1.6.2.12 and later releases, so if you were running the current version, you wouldn't have experienced this specific problem. This was listed in the ChangeLog for 1.6.2.12, but unfortunately the commit message the developer wrote did not explain why the change was made or what problem it was addressing, so you wouldn't have noticed it. In any case, upgrading to 1.6.2.12 or later will cure this problem. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Wed, Oct 20, 2010 at 10:35 AM, VoIP Question voip.quest...@gmail.com wrote: Another question: Is there (expect for the admin guide that we didn't succeed to understand the example in) an example somewhere for ReceiveFax full extensions.conf diaplan? We would like to allocate one of the extensions that our SIP provider gives us to a fax storage server or later to email. Not that I've ever seen. I built mine by reading mailing list archives, then the source for app_fax. +1 for open source. At least one reason such a thing does not exist is that everybody has a different idea of what 'full extensions.conf dialplan' means. In my case, I ReceiveFax, record the call, give it a naming convention that works, convert tiff to pdf, tail a log to a flat txt file, copy the pdf to Winders. That's not going to be the same thing as what a lot of other people want to do. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello again, If I set a peer to use G.711 only, they try to process a sent fax in G.711, but Asterisk doesn't like it: WARNING[4903]: res_fax.c:1709 sendfax_t38_init: Audio FAX not allowed on channel 'SIP/Main-000a' and T.38 negotiation failed; aborting. What can I do to enable it? Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:98765...@xxx.xxx.xxx.xx8:5060 Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv - --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy 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) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. --- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:1234...@yyy.yyy.yyy.yyy Content-Length: 0 --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Please help. Thank you. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote: Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite take a look at your canreinvite option. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
It's set to yes for this peer. also t38pt_udptl is set to yes. :( On Tue, Oct 19, 2010 at 5:12 PM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Oct 19, 2010 at 10:36 AM, VoIP Question voip.quest...@gmail.com wrote: Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite take a look at your canreinvite option. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote: It's set to yes for this peer. also t38pt_udptl is set to yes. :( You don't say anything about what you're trying to send / receive against. Here's how you should troubleshoot: * start with a 'real fax machine' if you have one, on an analog line if you have one. If you can't receive / send with that against your target, blame your target. * move to audio-pass through fax on asterisk. No T.38. If that works. * add in T.38 You will learn things in that process and be able to tell at what layer your troubles are happening. It could be coincidental that things give up during the reinvite. It could actually be giving up for noise on the line, packet drops, etc. At the very least, start recording the call. You'll at least be able to hear up to the re-invite. Definitely record the audio passthrough attempt and listen back to it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
We don't have an ATA and fax machine. The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. 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. Here are the logs (sip debug is open) again, since we get the reINVITE: --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- INVITE sip:1234...@10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060 Supported: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Content-Length: 303 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 218 7126 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=image 6202 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy a=sendrecv - --- (16 headers 13 lines) --- Sending to xxx.xxx.xxx.xx8 : 5060 (no NAT) Got T.38 offer in SDP in dialog 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy 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) Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. --- Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy Content-Length: 0 --- Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060 --- SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8;received=xxx.xxx.xxx.xx8 From: sip:98765...@xxx.xxx.xxx.xx8:5060sip:98765...@xxx.xxx.xxx.xx8:5060;tag=gK0d817deb To: Fax sip:1234...@yyy.yyy.yyy.yyy sip:1234...@yyy.yyy.yyy.yyy;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e2...@yyy.yyy.yyy.yyy CSeq: 1785 INVITE Server: Smartel-PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 Thanks. Michael On Tue, Oct 19, 2010 at 5:40 PM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Oct 19, 2010 at 11:21 AM, VoIP Question voip.quest...@gmail.com wrote: It's set to yes for this peer. also t38pt_udptl is set to yes. :( You don't say anything about what you're trying to send / receive against. Here's how you should troubleshoot: * start with a 'real fax machine' if you have one, on an analog line if you have one. If you can't receive / send with that against your target, blame your target. * move to audio-pass through fax on asterisk. No T.38. If that works. * add in T.38 You will learn things in that process and be able to tell at what layer your troubles are happening. It could be coincidental that things give up during the reinvite. It could actually be giving up for noise on the line, packet drops, etc. At the very least, start recording the call. You'll at least be able to hear up to the re-invite. Definitely record the audio passthrough attempt and listen back to it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA - it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 11:48 AM, VoIP Question voip.quest...@gmail.com wrote: The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. Asterisk can handle audio passthrough faxing. I'm talking audio faxing over SIP. You compile against this thing called SpanDSP, and then asterisk squawks audio tones over the line. It's amazing. Until you've tried it, you don't know whether it could work. I'm under the assumption that you'd rather have faxing at all than faxing over T.38. The world is littered with broken T.38 implementations. Just because it's a standard, doesn't mean people follow it. Ever heard of HTML? Which browsers follow it to the letter? You seem to have never successfully exchanged a fax with your target, so I don't know why you think the far end isn't broken. Try turning off t38 and see what happens. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. I spent a few hours on Google and saw many similar posts, but no actual valuable answer. Weird... On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas da...@debsinc.com wrote: From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA – it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
I think the generic throw away gmail address will keep many people from answering... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Oct 19, 2010 at 2:01 PM, VoIP Question voip.quest...@gmail.com wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. I spent a few hours on Google and saw many similar posts, but no actual valuable answer. Weird... On Tue, Oct 19, 2010 at 6:08 PM, Danny Nicholas da...@debsinc.com wrote: From what I have read over the last few months, you should invest in Motrin before trying T.38 faxing with or without FFA – it can (possibly) be done, but it has beaten some folks into the ground trying it. Could be a codec issue. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On Tue, Oct 19, 2010 at 1:01 PM, VoIP Question voip.quest...@gmail.com wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I don't know. I'm not using FFA, and I'm doing more channels simultaneously than I want to disclose. It's called app_fax, and it's been built into the 1.6 series for quite some time now. You seem pretty hell bent against spending any money. Not for FFA, not for commercial digium support, not for an analog copper line, not for an ATA. So try app_fax. It's free. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/19/2010 10:48 AM, VoIP Question wrote: We don't have an ATA and fax machine. The whole point (as I specified in the header and initial message) is the attempt to use Fax for Asterisk to send the message. 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. Here are the logs (sip debug is open) again, since we get the reINVITE: Is the SendFAX application running when this re-INVITE is received? You didn't actually include a complete console log, only the SIP traces, so we can't see what was happening in Asterisk at this time. You also didn't include an Asterisk version number (or any version numbers), so we can't be sure whether the problem you are seeing is a known resolved one or not. Please post the complete console log, with all logger levels enabled, so we can see the entire timeline of the call up to the failure occurring. If the re-INVITE arrives before SendFAX is started, there won't be an application to respond to it, and chan_sip will (rightly) assume that T.38 cannot be used on this channel so it will respond with a 488. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
On 10/19/2010 12:01 PM, VoIP Question wrote: Digium claims that their FFA is the best and most compatible solution and they give one channel for free, but do not provide support for those that do not buy more channels, but why buy more channels if the free/test one doesn't work? I know they read (and sometimes respond) to this list, so I don't understand why they don't clarify this issue. When you are asking for free help on a mailing list, patience is a virtue :-) You posted your question approximately four hours ago. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FFA SendFax rejects T.38 reINVITE (488 Not acceptable here)
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 ;) Anyway, specifically, we installed Asterisk 1.6.2.11. As far as we know/understand, the SendFax application is running. 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: sip*CLI -- Attempting call on Local/12345...@outgoing for s...@outboundfax:1 (Retry 1) sip*CLI -- Executing [12345...@outgoing:1] Dial(Local/12345...@outgoing-2c36;2, SIP/12345...@main,50,tTr) in new stack sip*CLI == Using SIP RTP CoS mark 5 sip*CLI == Using SIP VRTP CoS mark 6 sip*CLI == Using UDPTL CoS mark 5 sip*CLI Audio is at yyy.yyy.yyy.yyy port 10714 sip*CLI Adding codec 0x100 (g729) to SDP sip*CLI Adding codec 0x2 (gsm) to SDP sip*CLI Adding non-codec 0x1 (telephone-event) to SDP sip*CLI Reliably Transmitting (no NAT) to xxx.xxx.xxx.xx8:5060: INVITE sip:12345...@xxx.xxx.xxx.xx8 SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK5b6bc617;rport Max-Forwards: 70 From: Fax sip:98765...@yyy.yyy.yyy.yyy;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8 Contact: sip:98765...@yyy.yyy.yyy.yyy Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE User-Agent: PBX Date: Tue, 19 Oct 2010 16:41:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 312 v=0 o=root 697508180 697508180 IN IP4 yyy.yyy.yyy.yyy s=Asterisk PBX 1.6.2.11 c=IN IP4 yyy.yyy.yyy.yyy t=0 0 m=audio 10714 RTP/AVP 18 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sip*CLI -- Called 12345...@main sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Content-Length: 0 - --- (7 headers 0 lines) --- sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Contact: sip:12345...@xxx.xxx.xxx.xx8:5060 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=audio 6256 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - --- (11 headers 12 lines) --- Found RTP audio format 18 Found RTP audio format 101 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x102 (gsm|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port xxx.xxx.xxx.xx7:6256 sip*CLI -- SIP/main-002a is making progress passing it to Local/12345...@outgoing-2c36;2 sip*CLI --- SIP read from UDP:xxx.xxx.xxx.xx8:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK5b6bc617;rport=5060 From: Fax sip:98765...@10.0.0.3:5060;tag=as28606a47 To: sip:12345...@xxx.xxx.xxx.xx8:5060;tag=gK028217ef Call-ID: 2d965b0926e0134e0b211f882cbd2...@yyy.yyy.yyy.yyy CSeq: 102 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: sip:12345...@xxx.xxx.xxx.xx8:5060 Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,MESSAGE,PUBLISH Require: timer Supported: timer Session-Expires: 1800;refresher=uac Content-Length: 262 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 28160 32050 IN IP4 xxx.xxx.xxx.xx8 s=SIP Media Capabilities c=IN IP4 xxx.xxx.xxx.xx7 t=0 0 m=audio 6256 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 - --- (15 headers 12 lines) --- list_route: hop: sip:12345...@xxx.xxx.xxx.xx8:5060 set_destination: Parsing sip:12345...@xxx.xxx.xxx.xx8:5060 for address/port to send to set_destination: set destination to xxx.xxx.xxx.xx8, port 5060 Transmitting (no NAT) to
