similar to: T.38 vs. G.711

All; I'm currently running Asterisk 1.8.15-cert7 and am using realtime to store my voicemail configuration. The voicemail application works fine, but the problem I have is that the 'Directory' app cannot find any entries because there are no entries in the voicemail.conf file. When I add a context and an extension entry in voicemail.conf, it works the way it should. Is there

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-00000074): Buffer overflow detected (59 + 134 > 175) Has anyone ever seen this before? I have the following configuration. udptl.conf:

All; I'm working on a project (using Asterisk 1.8, but 11 would probably work just as well) where so far I've been able to originate over 1,000 concurrent outbound faxes. I have no problem with that so far. Where I have the problem is that Asterisk is dumping core after the faxes are sent. Now two things happen after the faxes are sent. (1) A fax log similar to a CDR is written to

All; I have a few customers that do a lot of faxing, both inbound and outbound. Some use the Spandsp and some use the Digium FFA modules. What I would like to do is when an outbound fax fails, determine whether the remote end was a fax machine or a plain old phone. I would also like to determine that in the dial plan if I could. Any insight at all with this would be extremely helpful.

All; I have a customer who is looking for a particular DID. (I dialed it and it is not in service). I searched through my preferred upstream provider's list but I came up empty. I wrote them, and this is their reply. "We currently do not have that specific number in stock as this number is owned by another carrier that we do not have a business relationship with." So my

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in

Just a thought regarding testing. Create a suitable TIFF file with more than 30 seconds worth of data and send it from Asterisk using SendFAX() to convince yourself whether Asterisk will work with your ITSP, you may still need to enable session timers. Have you considered setting up an extension on your Asterisk server which will receive the fax e.g. using ReceiveFAX() and see if your

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

On 11/12/2014 4:52 PM, Recursive wrote: > Hello, > > at first, thanks for helping! > > In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: > You may very well find getting T.38

Dears I installed digium fax and followed the instruction at http://downloads.digium.com/pub/telephony/fax/README,And as you can see above that t38 is loaded I am using a call file to send fax1.tif file as fax to the gateway named add The problem that Addpac send always Receive 488 Not acceptable here,and lkindly find my debug attached Please advice. Thanks I Advance shark*CLI>

Hi everyone, we're trying to setup a fax gateway from PRI to SIP. Our test setup consists of two asterisk 10 machines connected over PRI and a third one which is acting as a fax endpoint over SIP. On the machine which is receiving over PRI, we see the following message: [Jul 23 12:54:23] DEBUG[18288] res_fax.c: detected v21 preamble on SIP/xxx.xxx.xxx.xxx-00000001, but DAHDI/i1/-2 does not

On 14.12.2014 22:10, Larry Moore wrote: > Asterisk 1.8 with the T.38 Gateway patch (not the one by Niccol? Belli) sends a T.38 invite to the ITSP when the fax tones are detected from the callee, the T.38 Gateway implementation in Asterisk 10 and Niccol?'s back-port for Asterisk 1.8.11 does not behave this way. > Just to be sure: Is this the situation with FAXOPT(gateway)=no or with

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Hello. I've been trying to setup Free Fax for Asterisk on a Debian machine with Asterisk 1.8. I have managed to register and installed the Digium modules. Sending and receiving through it have resulted in failure. The output of fax show capabilities is: Registered FAX Technology Modules: Type : DIGIUM Description : Digium FAX Driver Capabilities : SEND

Hello, at first, thanks for helping! In the meantime, I have done a lot of research and trial and error, and I could solve that specific problem. Obviously, the dialplan application "Answer" was playing a key role here. My original dialplan snippet (which produced that problem) was: exten => _00., 1, NoOp() same => n, Set(FAXOPT(gateway)=yes) same => n,

Hello, I am a newer on asterisk. when I tried to send fax, I can not get success. After doing a lot of reseach, I decide to ask my question at here. Instead of using old fax machine, I want my system can send fax via t.38. Could anyone give me some idea to correct my configuration? Thanks a lot! Here is the detail: my system: *ubuntu 14.04 + asterisk 11.7* I use apt-get to install the ubuntu

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

Hello, Trying to set up res_fax_spandsp. Based on https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in my extensions.conf: exten => 306,1,NoOp(Fax transmission) same => n,Set(FAXOPT(gateway)=yes) same => n,Dial(DAHDI/3) ----->FXS port to fax machine same => n,Hangup() Call flow Im trying to pull out is as follows: Zoiper -->

Hello, I have Asterisk 1.6.1.12 with FAX For Asterisk Components: Applications: 1.6.1.5_1.1.6 Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32) If I use call file with spool -------------------------------------------- Channel: SIP/IP/DEst No MaxRetries: 0 RetryTime: 10 WaitTime: 50 Application:SendFAX Data:/var/spool/asterisk/test.tif