similar to: sip show channel and t.38

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 -->

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:

Hi, I tried to compile asterisk-1.8.0-beta4 but after ./configure && make I've got following error: [CC] res_fax.c -> res_fax.o [LD] res_fax.o -> res_fax.so [CC] res_fax_spandsp.c -> res_fax_spandsp.o res_fax_spandsp.c:117: error: field ?fax_state? has incomplete type res_fax_spandsp.c:118: error: field ?t38_state? has incomplete type res_fax_spandsp.c: In function

I am working on some fax tools for some of my users. I am reading the https://wiki.asterisk.org docs for faxing. Is see Application_SendFax and Application_SendeFax has one been discondinued? Any feed back on using the res_fax module would be apperciated. Any examples or other. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have asterisk 1.6.2.6 on a Debian Lenny system. When I try to send a fax in T.38 mode I receive this error ERROR[15035]: res_fax.c:795 set_fax_t38_caps: channel 'SIP/eutelia-sirio-out-00000000' is in an unsupported T.38 negotiation state, cannot continue. In my sip.config general section I have added this lines t38pt_udptl=yes t38pt_rtp=no t38pt_tcp=no If I comment this lines,

Hello, I have a question regarding incoming fax to local file (on the Asterisk server). While the fax is received properly (I have the tiff file generated as expected) I get the warning 'FAX CNG detected but no fax extension' on the consol. If the fax is received ok then what 'fax extension' does it expect and what should I do there? My Setup: Sender -> Public PSTN ->

Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There

Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Digium res_fax. Receiving is OK. I use Local channel in Call file and context [fax-out] in dialplan. My setup: Asterix<-SIP (T.38)-> Cisco(MERA MSIP v.1.0.2)<-> LocalTelco<->fax machine Debian GNU/Linux 5.0 ; Linux 2.6.26-2-686

Hello! I?m trying to add fax functionality to my asterisk installation. Right now I?m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right? Per book, I made following setup additions: 1. In sip.conf [general] I added: ;FAX stuff faxdetect=yes t38pt_udptl=yes 2.

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 Digium G.711 Licensed Channels : 1 Max Concurrent : 0 Success : 0 Switched to

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 =>

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Hi, Last night I was working on trying to get Fax for Asterisk installed. I followed all the instructions, but noted that the download selector only provider an X86 32 Bit binary. The machine I'm using is: 2.6.26-1-amd64 #1 SMP Fri Mar 13 17:46:45 UTC 2009 x86_64 GNU/Linux Consequently, on load it presents an error: [Sep 12 10:25:35] WARNING[31306]: loader.c:368 load_dynamic_module:

We are using res_fax_digium with a Sangoma PRI card on asterisk 1.8.13 The docs at http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf indicate v34 is supported, but when I enable it I get the message "res_fax_digium.c:1624 dgm_fax_new: V.34 not supported, will be ignored." Is v34 only supported with SpanDSP? Also, the res_fax.conf.sample does not indicate v34 as a valid

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

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old

Hello, We're trying to receive G.711 (aLaw) faxes on the asterisk and convert them to tif. With T.38, we have several issues, so we are trying to use G.711, since the gateway is located in the same LAN, so there's no bandwidth/packet-lose issue. We also use on the same Asterisk Real-Time process for the extensions.conf My question: Is the following syntax for disabling T.38 support

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

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