similar to: Hung IAX Channels

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. >One possibility is that the volume is set to 0. aumix can be handy here. Does

I'm trying to create an extension that will connect caller to asterisk sound card. I've followed the example at http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+console. With no luck. What I get is: Jul 13 09:56:45 VERBOSE[1315]: -- Executing Dial("SIP/300-3bd6", "CONSOLE/dsp") in new stack Jul 13 09:56:45 WARNING[1315]: No channel type registered for

I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with "astgenkey -n office.pbx.bluegrass.net" using the host name for each box of course. I

Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says "Presenation allowed of network provided number" which leads me to believe Asterisk thinks it should not be displaying it. Can anyone

Anyone have anything on this? (I'm sure someone will complain about me bringing it up again, chill out...) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jonathan k. Creasy Sent: Friday, October 14, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No Audio

Greetings, Is there a way to play the Digitally-imported OGG stream using xmms? I've tried it using ogg123, and it seams to work OK! But, I haven't managed to get it running on xmms. I've downloaded the latest plugin (xmms tells me version 1.2.4) and installed the 'RC1' rpms; but this doesn't work yet. (The player fill-ups it buffers; but doesn't start playing; and

Just a poll because I am curious, what kind of call volume do you see? Calls/Day -Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050909/9b8adf09/attachment.htm

Here is my file to parse and load extensions. No wise cracks about my code.... DB.php is the Pear DB module. (pear.php.net) <?php include('DB.php'); $db_host = ''; $db_name = ''; $db_login = ''; $db_pass = ''; $db_table = 'extensions_table'; define(DBINFO,"mysql://$db_login:$db_pass@$db_host/$db_name"); $db =

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

Asterisk Project Security Advisory - AST-2014-014 Product Asterisk Summary High call load may result in hung channels in ConfBridge. Nature of Advisory Denial of Service Susceptibility Remote

I have recently posted a message regarding hung SIP channels when using the Monitor() command. Well, I was wrong.   The channel hanging wasn't caused by the Monitor command. They also hang when they aren't monitored. The cause seems to be our PSTN gateway provider. When for some reason their PSTN gateway crashes or reboots (OK, this should happen, but anyway...) and RTP/SIP data stops

Hi all, is it possible to go on in the current context after the dest channel hung up? For example: exten => 111,1,Dial,Zap/4 If the originating channel is connected to Zap/4 and the destination channel (Zap/4) hangs up, both channels will be destroyed. Is there any option or whatever for preventing the hangup for the originating channel and go on in the current context ? Or, is there a

Can someone tell me how to clear hung SIP channels in asterisk without restarting? Currently I have 62 channels and only show 10 in use.. this is some of the sip show channels output.. xxx.xxx.xxx.xxx 00xxx24xxx 04240xxxxxx 00103/00001 UNKN (d) ?How can I remove these? from * without rebooting? ? .o-------------------------------------------------------o. Brian Fertig Network

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

Hi there, mostly based upon list postings I compiled a couple of administrative suggestions on the Wiki page below. I'd be glad to have this reviewed and commented: http://www.voip-info.org/tiki-index.php?page=Asterisk+administration Cheers, Philipp Adminstrative suggestions Use a GUI client that's based upon the manger API (like gastman or astman etc) to obtain an overview of

Been using mpg123 for moh for the last two years or so. However, when I have * config errors, often times get a endless stream of console messages and need to kill the two mpg123 processes. Is there an alternative to mpg123 that eliminates that issue? I see references in musiconhold.conf relative to madplay, native file format, asterisk-addons, etc. Not sure why the asterisk-addon approach

I am in the middle of rolling RHEL 4 update 5 ( 2.6.9-55) through my servers. We use a development, QA, and production rollout. Unfortunately I have ocfs members in each of these groups that need to share filesystems. The network compatibility issues between ocfs2 tools 1.2.4 and 1.2.5 are causing the members to not be able to operate with ocfs. Hence this breaks my systems or causes me to

I just got the latest update from the 1.0 CVS tree this morning. I was able to make the zaptel drivers just fine, but in the asterisk directory, "make" just sits there. This is under the 2.4 kernel on a SuSE system which has worked just fine until now. I'm making as root, so it's not likely a permission problem. According to top, grep and cat are running with grep sucking

Anyone know what has happened to the local calling guide? http://members.dandy.net/~czg/search.html -Jonathan