Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
On 10/08/2011 02:50 AM, Kevin P. Fleming wrote: On 10/07/2011 07:46 AM, Administrator TOOTAI wrote: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? You included the 'c' option to ReceiveFAX, telling it to act as the 'caller', even though it isn't the caller. This argument is parsed by ReceiveFAX in spite of it not being supported because the older app_fax version did support it, and we didn't want dialplans that included it to silently ignore the 'c' option. The same is true for the 'a' option; you'll note that neither of them are included in the documentation for the ReceiveFAX and SendFAX applications, and shouldn't be used. Why did you specify the 'c' option? Why was the ability to poll dropped from ReceiveFAX? Steve -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/08/2011 04:04 AM, Kevin P. Fleming wrote: On 10/07/2011 02:20 PM, James Sharp wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 Thanks for that; it helps. First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. t38UDPFEC is perfectly valid for version 0 of T.38. It works badly, so it makes no sense to use it, but it is valid. However, it is unlikely those are causing the call failure here. It's hard to say for sure without seeing the contents of the UDPTL packets, but based on their sizes, they are very likely t38-nosignal packets, and if that's all the FAX stack in Asterisk ever received, it would not trigger a FAX transaction to begin. Another possible problem is the repeated 'seq 0' in all the UDPTL packets, but this could be a problem with the UDPTL stack debugging messages themselves (this was just fixed in the Subversion branches for Asterisk 1.8 and later a couple of days ago). If you would, please retry this with the HEAD of the Asterisk 10 branch instead of 10.0.0-beta1, and also capture the UDPTL packets themselves so we can see what they contained. Steve -- _ -- 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] dahdi show status command not avilable in CLI
*[root@astrisks ~]# cat /etc/asterisk/dahdi-channels.conf* ; Autogenerated by /usr/sbin/dahdi_genconf on Thu Oct 6 18:28:14 2011 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings ; ; Span 1: WCFXO/0 Wildcard X100P Board 1 (MASTER) ;;; line=1 WCFXO/0/0 FXSLS (In use) signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel = 1 callerid= group= context=default *[root@astrisks ~]# cat /etc/asterisk/chan_dahdi_additional.conf* ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; * [root@astrisks ~]# cat /etc/asterisk/chan_dahdi_groups.conf* ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; signalling=fxs_ls context=from-analog On Fri, Oct 7, 2011 at 11:11 PM, Sammy Govind govoi...@gmail.com wrote: Please paste the configurations in the #included files as well. On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote: Hi, This is my /etc/asterisk/chan_dahdi.conf file. [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf ; Copied from DAHDI Module of FreePBX [general] #include chan_dahdi_general.conf [channels] ; include dahdi groups defined by DAHDI module of FreePBX #include chan_dahdi_groups.conf ;added by mic 06-oct-20011 #include /etc/asterisk/dahdi-channels.conf ; include dahdi extensions defined in FreePBX #include chan_dahdi_additional.conf Any issues in this ? Michael.k On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling ewiel...@nyigc.com wrote: It is likely you have an error in your /etc/asterisk/chan_dahdi.conf -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Friday, October 07, 2011 9:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I am getting this error message while executing the module load chan_dahdi.so. astrisks*CLI module load chan_dahdi.so Unable to load module chan_dahdi.so Command 'module load chan_dahdi.so' failed. == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_general.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_groups.conf': == Found == Parsing '/etc/asterisk/dahdi-channels.conf': == Found == Parsing '/etc/asterisk/chan_dahdi_additional.conf': == Found Thanks, Michael.k On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling ewiel...@nyigc.com wrote: What happens when you do the module load chan_dahdi.so command? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, astrisks*CLI module unload chan_dahdi.so Unable to unload resource chan_dahdi.so Command 'module unload chan_dahdi.so ' failed. Producing some other error messages ! On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling ewiel...@nyigc.com wrote: In the Asterisk CLI run the commands module unload chan_dahdi.so and module load chan_dahdi.so. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of michael k Sent: Thursday, October 06, 2011 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI Hi, I was run the commands dahdi_genconf and dahdi_cfg outside the CLI as the part of x100p card installation. Before issuing this command the dahdi show status command was available. There may any issues ?
Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX
On 10/08/2011 05:17 AM, Steve Underwood wrote: On 10/08/2011 02:50 AM, Kevin P. Fleming wrote: On 10/07/2011 07:46 AM, Administrator TOOTAI wrote: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? You included the 'c' option to ReceiveFAX, telling it to act as the 'caller', even though it isn't the caller. This argument is parsed by ReceiveFAX in spite of it not being supported because the older app_fax version did support it, and we didn't want dialplans that included it to silently ignore the 'c' option. The same is true for the 'a' option; you'll note that neither of them are included in the documentation for the ReceiveFAX and SendFAX applications, and shouldn't be used. Why did you specify the 'c' option? Why was the ability to poll dropped from ReceiveFAX? I honestly don't remember at this point. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Digium FFA + Gafachi T38 outgoing issues
On 10/08/2011 05:21 AM, Steve Underwood wrote: On 10/08/2011 04:04 AM, Kevin P. Fleming wrote: On 10/07/2011 02:20 PM, James Sharp wrote: On 10/07/2011 12:27 AM, Nasir Iqbal wrote: Check firewall and NAT settings! Monitoring sip and media flow from asterisk cli can help, use sip set debug on, rtp set debug on and udptl set debug on No NAT involved and I shut off IPTables. Still no luck. Debug shows the SIP invite, RTP frames going in out, the SIP reinvite, and then UDPTL frames coming in until timeout. See the entire transaction at http://pastebin.ca/2087758 Thanks for that; it helps. First, we can see that Gafachi's T.38 implementation still has some breakage in it (although the problems are ones that Asterisk has been taught to deal with). In its 200 OK to the T.38 re-INVITE, it has a=T38FaxRateManagement:transferredTCFlocalTCF; this is not valid (the valid values for this are 'transferredTCF' and 'localTCF'). In addition, even though Asterisk sent a=T38FaxUdpEC:t38UDPRedundancy, Gafachi replied with a=T38FaxUdpEC:t38UDPFEC. For T.38 version 0 (which is in use here), the only valid response was either what Asterisk sent, or no T38FaxUdpEC value at all. t38UDPFEC is perfectly valid for version 0 of T.38. It works badly, so it makes no sense to use it, but it is valid. Yes, it's a valid option in an offer, but it's not a valid option in answer unless that's the value that was in the offer; in all T.38 recommendations until the most recent, the answer must include either the same T38FaxUdpEC value as the offer did, or no value at all. The most recent version allows the answer to include a different value from the offer, because it was always reasonable, just not allowed by the recommendation. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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] Digium FFA + Gafachi T38 outgoing issues
Interesting. I just signed up with Gafachi (haven't even tested the service yet) but I planned to make use of their T38 support since they are listed at voip-info as being one of the ITSP's that _do_ support T38. Have you tried contacting Gafachi with these results about their broken implementation? I would hope/expect them to try to fix this, instead of trying to force Asterisk to violate RFCs. It sounds like that Gafachi's T38 implementation is horribly, horribly broken I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( Thanks for your help -- Daniel 2011/10/7 Administrator TOOTAIad...@tootai.net: Hi, I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from deb http://packages.asterisk.org/deb lucid main) including dahdi from this same repository. No FFA involved. On incoming calls (only SIP, no telephony card), fax detection is working but reception failed with -- Executing [fax@from-TOOTAiAudio:19] ReceiveFAX(SIP/tootaiAUDIO-0564, /tmp/1317991071.1614.tiff,c) in new stack [Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX does not support polling == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 'SIP/tootaiAUDIO-0564' What can be the problem? Thanks for any hint. -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburg l...@solvent-llc.com wrote: Interesting. I just signed up with Gafachi (haven't even tested the service yet) but I planned to make use of their T38 support since they are listed at voip-info as being one of the ITSP's that _do_ support T38. Have you tried contacting Gafachi with these results about their broken implementation? I would hope/expect them to try to fix this, instead of trying to force Asterisk to violate RFCs. It sounds like that Gafachi's T38 implementation is horribly, horribly broken I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. Ryan -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/08/2011 02:38 PM, Ryan Wagoner wrote: I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I wonder how many of these customers are just getting fallback to G711 when the T38 stack falls over. Heck, I thought I was getting T38 until I realized that I had SendFAX running with the audio fallback option. Turned that off, and fax fails 100% of the time. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. I found t38faxing.com. I was going to try them until I saw that their opening credit is $10. More than I want to spend to try for just home faxing. -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
On 9/10/2011 1:29 AM, Administrator TOOTAI wrote: Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( The Fallback option to T.30 is 'f'. ReceiveFAX(filename,f) See https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29 Larry. -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On Sat, Oct 8, 2011 at 3:51 PM, James Sharp ja...@fivecats.org wrote: On 10/08/2011 02:38 PM, Ryan Wagoner wrote: I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I wonder how many of these customers are just getting fallback to G711 when the T38 stack falls over. Heck, I thought I was getting T38 until I realized that I had SendFAX running with the audio fallback option. Turned that off, and fax fails 100% of the time. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. I found t38faxing.com. I was going to try them until I saw that their opening credit is $10. More than I want to spend to try for just home faxing. I tried to sign-up with them a week ago, but received an error message. I went to their contact page and saw the grnvoip.com email. It turns out grnvoip and t38faxing are both owned by ez call service. I signed up for grnvoip.com, but was unable to get the t.38 faxing to work. Additionally ez call service's administration panel is not laid out the best and doesn't let you change the static IPs that are allowed to send calls to them. I have tested T38 faxing and pass through with Asterisk 1.8 and combinations of the Linksys SPA2102 ATA, Zoiper, and Asterisk. The faxes are sent and received successfully. Analyzing the packet traces with Wireshark shows they were sent with T38. I just need to find a provider that has a working T38 implementation. Ryan -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
The f/F option for ReceiveFAX is not in the 1.8.x builds. It was a patch for 1.8.x but it is in the 10 builds Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Larry Moore lmo...@starwon.com.au Sent: Saturday, October 08, 2011 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX On 9/10/2011 1:29 AM, Administrator TOOTAI wrote: Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( The Fallback option to T.30 is 'f'. ReceiveFAX(filename,f) See https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29 Larry. -- _ -- 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] Asterisk 1.8.7 and ReceiveFAX
Le 08/10/2011 23:48, Bryant Zimmerman a écrit : The f/F option for ReceiveFAX is not in the 1.8.x builds. It was a patch for 1.8.x but it is in the 10 builds Well, I tried and it is working in 1.8.7 version, so command 'core show application ReceiveFAX' doesn't reflect the real application options, only shows c option which is not present in the link sended by Larry. Well ... FYI, I got this error -- Channel 'SIP/tootaiAUDIO-00ee' receiving FAX '/tmp/1318111488.262.tiff' [Oct 9 00:04:53] WARNING[9039]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ee' refused to negotiate T.38 [Oct 9 00:05:05] WARNING[9039]: res_fax_spandsp.c:368 spandsp_log: WARNING T.30 ECM carrier not found -- Auto fallthrough, channel 'SIP/tootaiAUDIO-00ee' status is 'UNKNOWN' -- Executing [h@from-TOOTAiAudio:1] NoOp(SIP/tootaiAUDIO-00ee, Hangup Cause: 16) in new stack -- Executing [h@from-TOOTAiAudio:2] NoOp(SIP/tootaiAUDIO-00ee, Dial status : ) in new stack but the fax was received. Thanks Larry for the tip. *From*: Larry Moore lmo...@starwon.com.au *Sent*: Saturday, October 08, 2011 5:32 PM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject*: Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX On 9/10/2011 1:29 AM, Administrator TOOTAI wrote: Le 07/10/2011 16:32, Kristijan Vrban a écrit : remove the c argument Done but now I have [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1508 receivefax_t38_init: channel 'SIP/tootaiAUDIO-00ea' refused to negotiate T.38 [Oct 8 19:20:20] WARNING[8771]: res_fax.c:1529 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/tootaiAUDIO-00ea' and T.38 negotiation failed; aborting. [Oct 8 19:20:20] ERROR[8771]: res_fax.c:1734 receivefax_exec: error initializing channel 'SIP/tootaiAUDIO-00ea' in T.38 mode How can I allow Audio FAX? I saw a discussion on asterisk-devel from january 2010 about new spandsp where Kevin P. Fleming told you to do an core show application ReceiveFAX to find out how to enable this feature. I'm perhaps a little bit stupid but can't find any usable information while using this command :-( The Fallback option to T.30 is 'f'. ReceiveFAX(filename,f) See https://wiki.asterisk.org/wiki/display/AST/Application_ReceiveFAX+%28res_fax%29 -- Daniel -- _ -- 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] Digium FFA + Gafachi T38 outgoing issues
On 10/09/2011 02:38 AM, Ryan Wagoner wrote: On Sat, Oct 8, 2011 at 10:41 AM, Luke Hamburgl...@solvent-llc.com wrote: Interesting. I just signed up with Gafachi (haven't even tested the service yet) but I planned to make use of their T38 support since they are listed at voip-info as being one of the ITSP's that _do_ support T38. Have you tried contacting Gafachi with these results about their broken implementation? I would hope/expect them to try to fix this, instead of trying to force Asterisk to violate RFCs. It sounds like that Gafachi's T38 implementation is horribly, horribly broken I'm not tied to them at all, so if their stuff is broken, I'll go somewhere else. I signed up with Gafachi a few weeks ago to use them for T38 as well. I haven't had any luck getting it to work. I have been mainly trying to use Asterisk in T38 pass through mode and have tested with a Linksys SPA2102 and Zoiper. Gafachi basically told me they have many customers utilizing their T38 implementation and that it works. When asked for a list of compatible devices they said there were too many combinations and it was up to me to find a working solution. I am still looking a PAYG service provider that has a working T38 implementation. It seems like these are impossible to find. Ryan Gafachi was one of the few service providers to support T.38 when we first started providing T.38 support in Asterisk and Callweaver. We did get things working reliably with them, by making our software tolerant of a few weird things Gafachi do. Any practical T.38 has to be made to tolerate a lot of weird things other implementations do. So Gafachi has worked in the past, but its entirely possible they have now broken their service further. Steve -- _ -- 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
