similar to: res_fax_digium and T.38 error correction

Hi, I installed Free FAX for Asterisk on my home Asterisk 1.8(Centos5.5) server. Everything seems fine but I just saw this WARNING shows up in the log every time I start the asterisk: /[2011-03-07 10:50:41] WARNING[13429] loader.c: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_tech_unregister/ And in later in the log

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 List, I did follow the procedure to install Free Fax for Asterisk successfully till i came accross this isssue: i can't load the fax module: pbx3*CLI> module load res_fax_digium.so Unable to load module res_fax_digium.so Command 'module load res_fax_digium.so' failed. [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module

Since digium is apparently blind to users of their Free Fax for Asterisk, does anyone have advice on how to report a crashing problem with res_fax_digium and Asterisk 1.8.2 ? I have detailed logs/reports and a backtrace ready, but I have no idea who can help. -- Jeremy Kister http://jeremy.kister.net./

Hello, I have asterisk 1.6.2.9-2 I tried to install fax utility as it is shown on pdf documents on asterisk site. I downloaded Opteron compiled res_fax and res_fax_digium files and copied to /usr/lib/asterisk/modules/ where default modules directory is. I created a free fax license and created license file on asterisk server too. WHen i run asterisk it crashed. I noticed that if res_fax.so

Dears , I am running asterisk version 1.6.0.6 with asterisk Fax Version 1.6.0_1.0.10, I installed on it two channels (one Fax for Asterisk channel and one Free Fax for Asterisk) My license files located at /var/lib/asterisk/licenses/ The two channels was working fine until I restart my PC,I did not change anyof my network cards or mac Address , How can I reload the channel license and

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

hi all, any one know, about echo cancellation with digium card, is it actually needed or it okay if we dont purchase because it increase price which half of new card, regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090827/8d6c680a/attachment.htm

"Security only fixes: 2011-04-21" So in six days, no more bugfix patches will committed into 1.4-branch :( Is a prolongation possible? Because 1.4 is so reliable now. It would be a great loss. And no, 1.8 is not (yet) a replacement. Kristijan

Hello, short question: is there a possibility to use asterisk as an outbound proxy? iam open for any suggestions, use asterisk trunk, dirty patches, ugly workarounds, everything. What is want to build is: SIP Phone -> via TLS/SRTP -> Asterisk as outbound proxy -> via UDP/RTP -> VoIP-Provider So Asterisk should just forward any incoming SIP messages (INVITE, REGISTER) to the

Hi, Maybe I missed it while checking it, but which spandsp version is recommended to play with Asterisk 10 and T.38/T.30 gatewaying ? I can see both spandsp-0.0.6pre17.tgz and spandsp-0.0.6pre18.tgz here (http://www.soft-switch.org/downloads/spandsp/) but I couldn't find a changelog documenting differences between them. So I prefer to double check ask for recommendations. Regards

Hi, I would appreciate some advice on the following: how does one use a single BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and Asterisk 1.8? For Asterisk 1.4 I would first install mISDN & mISDNuser and then build Asterisk 1.4. Now I noticed that Digium's HFC chipset based B410P is supported by the latest DAHDI & libpri but reading the source of the

Hello, what about: "This feature means you can restart Asterisk after a failure (or asterisk restart itself with safe_asterisk), and keep existing calls up with only a few seconds of audio dropped." That would be a feature! there is a "other" pbx that has this feature... Anyone else would like to see that feature? just want to start some brainstorming, Kristijan

Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI>

I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXXXXXX.lic but: Executing [17772822954 at FaxIncoming:4] ReceiveFAX("SIP/ec2faxcall18-00000002",

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 All, I seem to have stumbled on a bit of a problem. When trying to send a fax with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the current svn version, with FFA 1.2 I get a core dump each time. Here is an extract form the console: [May 12 22:47:09] DEBUG[22584]: app_queue.c:1084 handle_statechange: Device 'SIP/vltb-sbc01' changed to state '1' (Not in use)

Hello, i just made the reproducible watching: I send a Fax from asterisk (trunk) with spandsp (latest snapshot) via T.38 -> Audiocodes Mediant 2000 (FW 5.60.43.5) -> PSTN Fax With Internal timing Enabled, the Fax break after the first quarter from the first page is transfered. With Internal timing Disabled, the fax is transferred flawless. Both test with pthread timing module on a QEMU

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Hi, trying to compile but have problems: + /root/xen-unstable.hg-rev/tools/../scripts/git-checkout.sh git:// xenbits.xensource.com/qemu-xen-unstable.git82db8de16530f016809264d3179823999d702849 qemu-xen-traditional-dir Cloning into qemu-xen-traditional-dir-remote.tmp... fatal: read error: Connection reset by peer git down? Best Regards, Kristijan Lecnik