similar to: Asterisk+IVR functions trouble

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

Hi, After a whole day of work, I finally complied oh323 0.6.3a successfully. But when I started asterisk, it cann't load oh323. Following is the error: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [chan_oh323.so]Jul 13 09:43:45

Hi, Here is a proposed release schedule for 3.9.1: Nov 21, 2016: Deadline for nominating change to merge to the 3.9 branch Nov 29, 2016: -rc1 Dec 5, 2015: 3.9.1 Release As a reminder, if you would like to request a patch be merged to the 3.9 branch, file a bug at www.llvm.org/bugs with the subject "Merge rxxxxx into the 3.9 branch." and add

Hi, Just to know if somebody had succesfully compile app_radius from http://appradius.minitelecom.org ? Here below my configuration : -> asterisk runing -> mysql running -> freeradius running -> Compiling cpprad : OK -> Compiling app_radius : not OK, here below my error message : "" make[1]: Quitte le r?pertoire `/home/grd/appradius/inc' make[1]: Entre dans le

I read the wiki and looked at the examples, but I'm still having problems. I have a Digium 4 port card with POTS lines plugged into all four ports. How do I play the congestion tone the the caller when they try and dial out but all the lines are in use? should something like this work? [dial-trunklocal] ; Local calls ignorepat => 9 exten => _9NXXXXXX,1,Dial(${TRUNK1}/${EXTEN:1}) exten

Hello All, I downloaded the ChanSpy patch from Mantis and updated to the latest asterisk source from cvs. Everything seems to have installed fine and everything works as it had before, but I can't get ChanSpy to work. I added a line to extensions, as a test: exten => *53,1,ChanSpy(scan) When I dial this extension from a SIP phone, and then make a call (which I am trying to monitor) from

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what

Hi, i'm trying to install in wine-1.0.1 on Ubuntu 9.04 a management software but I receive run-time 713 error. Any solution or tip? Here the log.txt: fixme:ole:OleLoadPictureEx (0x12c8c44,35146,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fac0), partially implemented. fixme:ole:OleLoadPictureEx (0x12c8c44,774,1,{7bf80980-bf32-101a-8bbb-00aa00300cab},x=0,y=0,f=0,0x32fa90),

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

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 try to set up asterisk for making outgoing calls with oh323 channel driver version 0.7.1 with Asterisk CVS-1-01/09/05-01:41:37. Our provider uses Mera MVTS softswitch and supports only H.323. We don't use gatekeeper for connection but provider requires SOURCE PHONE NUMBER for route out calls and I don't know how I can specify this number. Call with this string exten

dear all, we need help for integration asterisk (sip) with mera we have configure for sip.conf and extentions.conf sip.conf [mvts] context=mvts type=friend host=10.10.0.2 dtmf=rfc2833 in extentions.conf [mvts] ; ; mvts exten => _01162.,1,SetCallerID(mvts) exten => _01162.,2,SetCIDName(to mvts) exten => _01162.,3,Dial(SIP/${EXTEN:3}@mvts) i need if i dial 01162 in mera replace with

Hi, I've changed rsync to support the BSD change file flags. However, this raises some compatibility problems, especially when including it with the -a option. If the remote host does not understand the new option for updating file flags, the user gets an error message about an unknown option. How should I handle this? If you'd like to look at the patch (and preferably integrate it with

According to digium's site, "Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue." (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces,

I'm having trouble to send calls to a Mera MVTS softswitch (with SIPHIT) when the asterisk box has a dynamic IP address. If the Asterisk box has a fixed IP, everything is OK. Any ideas? I'm looking for a working sample of the sip.conf in this case... user.cfg (for MVTS) is also appreciated if any special setting should be done there also.

Estimado Carlos Gil Bellosta ¿Cómo está usted? En estos lados de América del sur comienza la primavera, desde la ventana miro la parra contando las posibles uvas, siempre aparece un ave que se arrima a la ventana o incluso llegan hasta la computadora como si supiesen usarla. Ahora en R. En ese esquema un modelo lineal tendría que ir con mlib que es aportada por sparklyr, en ese caso tendría

Estimados En el sitio de https://www.rstudio.com/ hay un aviso sobre http://spark.rstudio.com/index.html ( sparklyr ). Microsoft publico un artículo donde comparan el R Server que está dentro de SQL server (o por separado, depende un poco), o el Microsoft R, junto con algunas librerías que se pueden compilar y obtener lo mismo en Ubuntu. Supongamos que tengo el dinero como para comprar por

Somebody must have! Please, send me a g723 and/or g729 (for Asterisk) to pisac@hotmail.com (antispam subject: codec) Thanks, thanks, thanks... :-)

Hello everyone can anyonone of you send me the file codec_g729.so this file has to be inserted in /urs/lib/asterisk/modules Thank You

Has anyone out there successfully set up their * box to terminate their VONAGE calls? I (and I am sure lots of others) would love to hear how you did it. I'd like to be able to get rid of the extra hardware I have hanging around here and use the ASTERISK machine to handle the SIP termination instead of needing to have a Linksys modem (w/phone) and an additional X100P card. Thanks.