search for: echotests

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 'demo-echotest' (language 'en')...

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 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:

Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when they call iax to sip, it works. When they make an echo test..no voice received on iax clients. When they make call from sip to iax ..no sound received on iax clients. I didnt see any clue on debug.

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, I have changed some prompts in /var/lib/asterisk/sounds and Asterisk still play the old ones, even if they does not exist in the file system anymore. There is any cache used to play prompts files? If yes, there is any way to purge that cache? Tried with 'reload' or to restart Asterisk without any luck. I have not tried and I don't want to restart the computer too. Thanks, Dan

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/104a07f3/attachment.htm -------------- next part -------------- Hello In my IVR menu whenever user select the option number 1 then it should jump to echo context, I think call did jump to "echo" context but I always get the following warning and I hear couple of beeps and then

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, There is any experience using Asterisk with VMWare? I think about installing a virtual linux box over VMWare and then Asterisk over it. Thanks, Dan

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

I got four phones: 601 is a SIP phone (no brand) 615 is Snom 190 621 is a Grand stream 628 is a remote SIP phone (no brand) 601, 615, 628 can call each other without any problems 621 used to be able to call remote 628, but after upgrade to CVS Head Nov. 11 the remote party cannot hear me. 615 never could call remote 628, both party hear nothing. 601 can always call 628 [Oct 16 00:52:13] --

Hi all, DIAX 0.9.9g is available for download (including the updated help file and web page) from the following locations: http://www.laser.com/dante or http://www.geocities.com/tdanro What's new in 0.9.9g (from 0.9.9f): - during a call, accept DTMF tones as monitored events to trigger output commands - call timer on the phone display - Swedish language added - can run a command from the

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

> Olle E Johansson wrote: >> >> 23 apr 2007 kl. 19.55 skrev Russell Bryant: >> >>> John Todd wrote: >>>> To morph this into a -dev thread: if this patch were to become (again) >>>> useful and error-free, is there any objection or usefulness in adding it >>>> to TRUNK? Personally, I think there is, if there is a method by which