similar to: Problem playing file with G729A

I' got a problem, using asterisk-rc2 :IVR functions (Background...Playback...etc) doesn't works : Executing Background("OH323/RXXXXX", "vm-extension") in new stack channel.c:1650 ast_set_write_fornat: Unable to find path from GSM to G729A---Asterisk box supplied only with network adapter.---Asterisk box registered in Mera (soft-switch with H323 protocol) and doing

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, 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, sorry to bother you, and that this is probably not the right list :-), but I read that some of you might use syn as Editor for .R files. I've released an unofficial Version of the syn Text Editor with improved support for R (I'm the initial developer of this program, btw.). syn is a Windows 32 Program (Win9x, NT4, 2000), but maybe it runs also inside Wine, I didn't try it.

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,

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

Hi list I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk solutions and appliances. I installed TrixBox on a litle PC @ home and a x100p card which is recognized as a Zaptel card, I made some in/outbound routes and they seem to work but I have a problem with SIP softphones. I created 2 estensions 1000 and 1001 they're both in different cities, when I 1000

Hi to everybody, I have the problem that nearly no information are displayed on the Asterisk CLI (asterisk -r). In former times (before patching Asterisk 1.0.2 with ast_data 1.0.2) it looks e.g. like this: --- snip --- -- Registered '96' (AUTHENTICATED) at 212.202.169.118:4569 -- Accepting AUTHENTICATED call from 212.202.169.118, requested format = 1024, actual format = 1024

Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from

New install, Calls are working phone to phone using gsm, ulaw or alaw codec but when try and echo test or voicemail there is no playback. I've tried turning on and off every codec and still no luck. Asterisk says it's playing the sound file but I just don't hear anything. I can't find any reason for this. I've tried the latest tar and CVS with the same result.

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

I've got an SPA-841 SIP hardphone connecting to my asterisk server across the internet through a couple of NAT routers. Everything works great (I can initiate calls, receive calls, hear audio both ways, etc.) except for one thing. When I hang up the phone, the connection in asterisk doesn't disconnect. The phone is idle and things everything is fine, but Asterisk still show an open