similar to: Defining what an extension should do after the Dial() command returns busy.

We have an issue with our Asterisk install where Asterisk produces many Zombie processes (on the order of several hundred per minute) until either the Asterisk server is restarted (and the zombies die a natural death), or the kernel runs out of PID space (happens within hours) and brings the system to a halt. This problem only happens when the server is under some non-trivial load. We were

We have two Asterisk servers. One is a live server supporting our customers, and the other is a backup server that's being upgraded and pressed into service. Both servers have a Digium TE405P T1 card in them, and in order to test the T1 service on the backup server, I've created a T1 crossover cable (as per http://www.voip-info.org/wiki/view/crossover+T1+cable) that goes from port 4 on the

I'm kind of at a loss to diagnose problems like this, yet we get them a lot. - The ATA (Thomson 784 in this particular case) is logged into the Asterisk server. 'sip show peer' shows their IP address, port, and useragent. - The ATA is connected directly to the internet (no NAT, but the sip configuration has nat=always) and logs in to our server, which is also directly connected to the

Hi! Part of extensions.conf: exten => 985,1,Dial(SIP/0317998985&H323/00702221448 at Avaya,20) exten => 985,2,Goto(985-${DIALSTATUS},1) exten => 985-BUSY,1,VoiceMail(0317998985 at inputinterior.se,b) exten => 985-BUSY,2,PlayBack(vm-goodbye) exten => 985-BUSY,3,HangUp() exten => 985-NOANSWER,1,VoiceMail(0317998985 at inputinterior.se,u) exten =>

This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan

On 2016-02-17 15:32, Richard Mudgett wrote: > On Wed, Feb 17, 2016 at 5:15 PM, Ernie Dunbar <maillist at lightspeed.ca> > wrote: > >> Hi everyone. >> >> We have an Asterisk server running Debian Squeeze, with Asterisk >> v1.8.13.1 (basically, the Debian Stable version for Squeeze, but >> with some minor source code changes specific to our site).

Okay, so here's the configuration I have for MySQL Realtime (Asterisk version 1.6.2.17): In /etc/asterisk/extconfig.conf: sipusers => mysql,mya2billing,cc_sip_buddies In /etc/asterisk/res_mysql.conf: [mya2billing] dbhost = localhost dbname = mya2billing dbuser = a2billinguser dbpass = REDACTED dbport = 3306 And here's the error messages I get: voip2*CLI> realtime mysql status

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old

I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 "Moved Temporarily" back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1

Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no answer, the extension hangs up, but the dial status is busy instead of no answer. How do I get this to work -- do I need to update dahdi? The card is an X400p using its FXS module. Thanks in advance for any ideas on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it?

I've seen several examples that use extensions such as; s-BUSY s-NOANSWER etc... It's more or less evident what they do, but I've searched for some FORMAL documentation everywhere and have found nothing. Do they work for anything else than "s-"? (I think I've seen other examples, but can't find them now) Are they standard in any way? What are the allowed values

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from TUN to the phone client. I would suggest

Hi All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

This is a problem that is completely stumping me, and my understanding of Asterisk dialplans tells me this should never be a problem. Moreover, this scenario works on Asterisk 1.4 but not 1.6. We have a customer with several Aastra 6731 phones. They want incoming calls from the PSTN to work and they also want to be able to call each other "internally" on a special non-DID number (like

Hi, Now and then our SIP phones ring with "asterisk" showing as the caller-ID. Upon picking up the receiver, there is about five seconds of silence and then the channel is closed (hangup). Can anyone offer some insight? Here's relevant snippets from my extensions.conf and Master.csv log: This line shows up in Master.csv: