Re: [asterisk-users] Fax setup T.38 Help needed
- Original Message - From: Thorben Jensen i...@thorben.dk To: asterisk-users asterisk-users@lists.digium.com Sent: Wednesday, June 20, 2012 8:25:28 PM Subject: [asterisk-users] Fax setup T.38 Help needed Hi, I'm looking for someone who can help us setup Fax with T. 38 on asterisk 10.x.x - We need to be able to do FoIP (Fax over IP) as we have no pstn lines available. Do you know how to setup a reliable fax system, then we will pay you to help us do this. If you're looking for consultants, you may want to try the asterisk-biz mailing list. -- Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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] Fax with T.38
Hi Bill, I'm in exactly the same boat with T.38 and OpenPBX. I too think the Cisco-T.38 Gateway is the most practical at this moment. Where are you on testing this and can you share the 3660 config? In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems. Jon -Original message- From: Bill Gibbs [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Fax with T.38 Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine
RE: [asterisk-users] Fax with T.38
I am waiting for the powers that be to get a dual port PRI card at this time. I think a dial-peer will only need to look similar to this on the Cisco: dial-peer voice 10 voip destination-pattern WHATEVER session protocol sipv2 session target ipv4:openpbx ip dtmf-relay sip-notify rtp-nte fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco Since that's basically what you need to do voice, all this adds is the T38 line. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jbebeau Sent: Saturday, February 24, 2007 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax with T.38 Hi Bill, I'm in exactly the same boat with T.38 and OpenPBX. I too think the Cisco-T.38 Gateway is the most practical at this moment. Where are you on testing this and can you share the 3660 config? In researching the CIsco/voice, there is a TON of hardware options you need, or so it seems. Jon -Original message- From: Bill Gibbs [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 15:02:18 -0500 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: RE: [asterisk-users] Fax with T.38 Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image
RE: [asterisk-users] Fax with T.38
Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. So my call flow will be PRI - Asterisk 1.2.x Out the 2nd PRI to the 3660 3660 dial-peer, with T38 fax settings talk SIP to Asterisk 1.4.x then t38 pass through to my ATA. I have the 3660 there to take the call via TDM and convert to T38. I only have a single PRI which is why I don't want to have to purchase other lines dedicated to a T38 faxserver, and this will give me the ability to use my DIDs already assigned. That's how I plan to set it up. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ray Jackson Sent: Wednesday, February 21, 2007 10:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fax with T.38 Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30
Re: [asterisk-users] Fax with T.38
22 feb 2007 kl. 21.02 skrev Bill Gibbs: Ray, I have been playing with OpenPBX. My core servers are Asterisk so I was playing around with their T38Gateway application. Long story short - I can get the ATA (behind NAT) to talk T38 to the rxfax app on an OpenPBX server but the gateway feature of that product is still under development so I was sending IAX calls to it and it would try to talk T38 to my ATA (behind NAT or public IP) and eventually the call would fail. Clearly T38 was working though, debug output was full of T38 talk. However the wiki clearly states it's experimental still. I personally have decided to go with a 2nd PRI port to a 3660 I have on hand that will do T38 SIP. I am going to set that up to talk to * 1.4.0 and do T38 pass through. I to will be doing NAT for the ATAs so...hopefully it will work. We shall see. As I've stated a few times, T.38 passthrough is broken in 1.4.0. Either use 1.4 from subversion or wait for 1.4.1. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with T.38
Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.html This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Cheers, Ray PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. Thomas Deillon wrote: Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas *De :* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *De la part de* Rajnish Jain *Envoyé :* lundi, 19. février 2007 16:25 *À :* Asterisk Users Mailing List - Non-Commercial Discussion *Objet :* Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, *Thomas Deillon* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 http://0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me
RE: [asterisk-users] Fax with T.38
Ray wrote: Could anybody give me an authoritative answer on whether Asterisk can support T.38 pass-through when the clients are behind NAT? We have Asterisk servicing clients behind NAT (with nat=route, canreinvite=no) and would love to get T.38 going but have had no luck so far. The following case: http://bugs.digium.com/view.php?id=7844 Authoritative? Nope. But I'll try to help anyways... 1. t38pt_udptl must be set to yes in [general] in sip.conf ...suggests that T.38 *does* now work for clients behind NAT but I have the latest SVN trunk but still cannot get it to work? On the other side I have seen on this list only 2 weeks or so ago: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg172556.ht ml This suggests that T.38 does *NOT* work behind NAT? So, can anybody save me the trouble and tell me how it is. Am I on a hiding to nothing trying to get T.38 going with NAT? Please put me out of my misery! :) Part of an age old issue that doesn't bear repeating, but is also not terribly accurate or relevant. Cheers, Ray Capture a debug log of a failed T.38 session and post it on Mantis. Make sure to set: core set verbose 4 core set debug 4 sip set debug Testing and (what little) feedback the developers have received indicate that it SHOULD work with the latest SVN. PS. Does anybody know whether OpenPBX would support T.38 and NAT configurations? This was my backup plan if I couldn't get it to go in Asterisk. No idea. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Thomas Deillon Envoyé : jeudi, 15. février 2007 11:26 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option “FAX without T.38(Use G.711 fax)” On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it’s why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with T.38
A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, Thomas Deillon [EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Yes, the canreinvite means Re invite, but there is a consequence in Asterisk configuration. For sure, all the signalisation traffic will go through the asterisk … but for the RTP traffic? If canreinvite = No, all RTP traffic will go through the Asterisk (useful for NATed phoned without ALG/STUN/…) If canreinvite = Yes, the phones will try to exchange RTP packets directly. Do you thing there is a way to allow Re Invite (because you’re right) without the RTP consequence? Thanks a lot for your help, Thomas De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Rajnish Jain Envoyé : lundi, 19. février 2007 16:25 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Fax with T.38 A T.38 fax call typically begins as a normal voice media call. The call then dynamically switches over T.38 image media on detection of fax handshake tones. The dynamic modification of session from audio to image is accomplished through SIP RE-INVITE messages. I would imagine canreinvite= flag controls if an end-point is allowed to send/recv RE-INVITE to/from Asterisk. If so, you'll need to set it to yes for T.38 to work. On 2/19/07, Thomas Deillon [EMAIL PROTECTED] wrote: Hi all, I make others tests. Analog Fax 1 - PATTON M-ATA - Asterisk - PATTON M-ATA - Analog Fax2 It works only if I use canreinvite= yes. But all my clients are behind a Nat without ALG or stun stuffs... Do you know if canreinvite= yes it's the only way to make it works?? Thanks a lot for your help, Thomas -Message d'origine- De: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ] De la part de Thomas Deillon Envoyé: jeudi, 15. février 2007 11:26 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem… I want to send fax with FoIP. Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→ Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts … Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 …. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk …. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec … more than this, I remove the g729 licence file … Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Hi all, I make mistakes in my explanation, so I will try to re-explain my problem... I want to send fax with FoIP. Analog Fax PATTON SN4960 Asterisk PATTON M-ATA Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts ... Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec ... more than this, I remove the g729 licence file ... Do you have an idea for me ?? Thanks a lot, Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fax with T.38
Hi Thomas, In the Patton SN4960, try making the codec 1 alaw instead of g729. Cheers Dave _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thomas Deillon Sent: Thursday, 15 February 2007 8:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Fax with T.38 Hi all, I make mistakes in my explanation, so I will try to re-explain my problem. I want to send fax with FoIP. Analog Fax PATTON SN4960 Asterisk PATTON M-ATA Analog Fax 2 In the Patton SN4960 configuration I have : profile voip FOIP codec 1 g729 rx-length 20 tx-length 20 no-silence-suppression codec 2 g711alaw64k rx-length 30 tx-length 30 no-silence-suppression codec 3 g711ulaw64k rx-length 30 tx-length 30 no-silence-suppression dtmf-relay signaling dejitter-max-delay 100 fax transmission 1 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax detection fax-frames modem transmission 1 bypass g711alaw64k modem bypass-method nse On Patton M-ATA : 1. codec alaw 2. codec ulaw 3. codec g729 No silence suppression on these codecs. I not use this option FAX without T.38(Use G.711 fax) On asterisk side I have: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw dtmfmode = rfc2833 rtcachefriends=yes realm=vtxvoip useragent=VTX SIP rtupdate=yes language=en tos=184 notifyringing=yes t38pt_udptl=yes And t38pt_udptl=yes in the 2 PATTONs sip accounts . Then, I made a trace when I call from Analog FAX 2 to Analog fax 1 .. I received T.38 packets from the Patton sn4960 but no T.38 packets go through the Asterisk .. And on the asterisk I have 3 WARNINGS: [Feb 15 10:05:24] WARNING[9167]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-01-0071e6b0(256) to SIP/0xxx0379xx-0070a490(8) [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 15 10:05:33] WARNING[9163]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 What I really not understand it's why asterisk try to translate from ulaw to g729 !!! I disallow all and allow just the alaw codec . more than this, I remove the g729 licence file . Do you have an idea for me ?? Thanks a lot, Thomas _ I am using the free version of SPAMfighter for private users. It has removed 8633 spam emails to date. Paying users do not have this message in their emails. Try SPAMfighter http://www.spamfighter.com for free now! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
