similar to: SIP channel not diconnecting on hangup

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

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:

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

Hello all I've discovered that SIP channels sometimes get stuck in *. I've read some posts from Fri 29 Aug 2003 which mentions this issue, but there doesn't seem to be any final answers I don't know if this is related to the 0001604 bug? Below is a list from one of the incidents: I know the (d) means that it is scheduled for destruction but the 10.1.1.45 channel hasn't

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

> 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

Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : "sip show channels" [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels" Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0

Hi all, I am trying to use a "Siemens optiPoint 300" IPPhone (H.323 only) with Asterisk (1.0-RC2). So far I have been using the H.323 channel included in the tarball (Nufone ?). I encountered a strange behaviour when I try to make a call from the IPPhone to my Asterisk box : =====> here is the H.323 configuration for the incoming calls (192.168.1.50 is the IP of the

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

'morning everybody, Here is the setup: 5126800422 called 3035 (3035 is a Cisco 7960). The call is g729. 3035 presses 'Conference' on her phone and calls 8327549222. This call is ulaw. (65.72.107.2 is our Cisco 7206 SIP->PRI gateway.) asterisk*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Format 65.72.107.2 8327549222 1758081f67e

Hi, We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity recorded in our full log. Could you help us to explain what had happened. Thanks. === my friend, 801, from his room did a test by dialing echo test in freepbx, *43: [Aug 20 14:42:46] VERBOSE[14427] logger.c: -- Executing [*43 at from-internal:1] Answer("SIP/801-000003f5", "") in new stack [Aug 20

Hi, one of the sip-extensions we created always returns busy when someone tries to call the phone. The extension itself can place calls. We're using snom360 phones with the latest firmware. On every one of those phones when we register with the sip-extension, we've experienced the same problem. This is the output from sip show channels: Peer User/ANR Call ID Seq

I am using a2billing as calling card platform with asterisk 1.2.17. After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?