similar to: PRI zap channels not cleared when nomatchincontext for dialed number on inbound call

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 1:48 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the > sametimewithdifferentcaller ID number? > >

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Tuesday, January 31, 2006 8:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels > atthesametimewithdifferentcaller ID number? > > > >

Don't feal bad about not reading. I yell at my 10 y.o. about it all the time. READ, NO more TV, READ!!! > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Damon Estep > Sent: Tuesday, January 31, 2006 10:23 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion >

What is CFU and CFNR? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michel Koenen > Sent: Tuesday, August 30, 2005 1:46 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] How to use * and # as part of > numberindialcommand > > > From: "Damon

The issue I have had with all other FC3 kernels apart from the 2.6.9 one was that the zaptel build would throw lots of warnings up. This would have the knock on of hanging the system, spinlock I think the problem was, on a modprobe -r. Lee ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Troy Settle > Sent: Wednesday, September 14, 2005 7:03 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] MAX PRI for single server (was:Not > enoughlinesavailable for Asterisk implemetation)

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Damon Estep > Sent: Monday, January 30, 2006 11:20 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] dialing 2 channels at the same > timewithdifferentcaller ID number? > > >Exten

In the real world (or at least in my world) we use undersubscribed internet connections that come with a service level agreement (SLA) that guarantees that the jitter, delay, and packet loss with be within defined parameters in the service agreement. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

Anyone out there have success getting caller id name from a pri, through a lucent tnt, to asterisk? What about from other media gateways? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050809/d3f02c3d/attachment.htm

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options

I have considered opening a bug report on this, but wanted to get some feedback and make sure I am not missing something in the way of a simple work around. What is the scenario in which this impacts your implementation? Ours is the desire to use the same realtime SIP database for many asterisk servers, and route the call based on a "home server" value in the realtime database. The

I found that only the kernel is installed. I'd avoid 2.6.12 for now as I had problem with the zaptel driver and stay with 2.6.9. Regards Lee ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Damon Estep Sent: 24 August 2005 22:33 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] fedora