similar to: Park and PARKINGDYNAMIC

Hello, I am trying to create a dynamic call parking lot using https://wiki.asterisk.org/wiki/display/AST/Application_Park But this manual is not enough to fix my problem : Asterisk keeps trying to park the call in the default parking lot : [Jul 2 11:32:14] -- Executing [3333 at from-770000:5] Set("SIP/testacc77000-00000002", "PARKINGDYNAMIC=parkinglot_test") in new

Does anyone have any examples of using realtime database driven call parking lots. I am on version 1.8.x My goal is to be able to do database driven multi-tenant parking lots with out adding sperate entries into Features.conf for each lot. I also need to be able to use the same parking extension pool for each tenant but sand box them into sperate lots. We have been able to do this for every

Hi there. I have the similar problem ("Digium fax - sending fax call file vs manager originate") sending faxes with Asterisk 1.6.2.6 and Digium res_fax. Receiving is OK. I use Local channel in Call file and context [fax-out] in dialplan. My setup: Asterix<-SIP (T.38)-> Cisco(MERA MSIP v.1.0.2)<-> LocalTelco<->fax machine Debian GNU/Linux 5.0 ; Linux 2.6.26-2-686

How can we set the music on hold class using the Dynamic Parking lots? The variables set the PARKINGLOT, PARKINGDYNAMIC, PARKINGDYNPOS,PARKINGEXT,PARKINGDYNCONTEXT I can't find a PARKINGMOH variable. This is becoming a big issue. We are using the current release 11. version We have to be able to set the MOH dynamically I just can't find the mechanism. Any ideas? Thanks

Howdy, folks. I'm having a problem finding a way to auto-record calls (both incoming and outgoing). I know how to make it so either party can initiate recording, but I want it done as soon as both ends are connected (or prior to that if that's what it takes). It's probably right in front of me and I'm just missing it. Any help would be much appreciated. Thanks, Jay

hello, i have trouble with nat + sip outgoing call.when make an outgoing call to a sip gateway, i have no sound. i have 2 sip gateway, one is asterisk. asterisk is on public ip and private ip other sip gateway is on public ip phone are cisco and grandstream on private ip on the same subnet as asterisk. phone are connected by sip to asterisk (i have try with or without nat=yes) incoming call

Hey Everyone; I inherited an Asterisk box where the dialplan is a real mess. ( I would actually be embarrassed to post some of the stuff!) So, here is what I need to do - and again, I am looking for fishing nets and places to cast them - if I don't figure it out, I will never freakin' learn! I have several users configured (101, 102, 105, 155, 211, etc). They are all in different

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hello, I have the following in my dialplan : exten => callpark,n,Set(PARKINGDYNPOS=200-210) exten => callpark,n,Set(PARKINGDYNCONTEXT=parked_001) exten => callpark,n,Park(20000,,,,s,parkinglot_001) I see on the CLI : [Nov 25 15:08:47] -- Executing [callpark at pbx-routing:10] Set("SIP/SipT01-0000000b", "PARKINGDYNPOS=200-210") in new stack [Nov 25 15:08:47]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

Hi, I need to create a simple number checking for authorizing the calls. if a person dial 91800555121212345 where 12345 is the authorization code. If the authorization code is correct the call will go through if not it will play something saying wrong authorization code or just hangup. This my dialplan to get the authorization code AUTH=12345 exten => _9.,1,Answer() exten =>

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040721/e99c8b95/attachment.htm -------------- next part -------------- Hello friends, I got one page from net "http://www.voip-info.org/wiki-CLASS" In that page I saw lot of *xx codes for asterisk feautres. I don't know how to use these codes. If anyone used

Hi everyone, For some reason MixMonitor doesn't record when it should; It actually shows the MixMonitor line just fine on the CLI. How can MixMonitor be debugged for things like privilege issues or filename issues? **I had this working at one point and then stopped working. Not sure what I changed. System Info: Asterisk 1.4.21.2 Queuemetrics 1.6.3.0 [queuedial] ; this piece of dialplan is

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Apologies if this is a resend, but I've sent this 12 hours ago and still can't see it on the list. Hi, I've just started to setup my phones with Buddy watch. Basically, it all works fine when using the simple example on the wiki: exten => 123,hint, SIP/some_sip_reg exten => 123,1,SIP/some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy

Sorry that I wrote in german : Ich benutze asterisk mit dem zaphfc Treiber. Jetzt hab ich folgendes Problem, habe 2 ISDN-Telefone angeschlossen. zaphfc im nt-mode. Anrufe von ausserhalb per sip (sipgate.de) kommen an. Wenn ich aber intern zwischen den zwei Telefonen (Ascom Eurit 30) sprechen m?chte geht das nur wie folgt : Erst die Nebenstelle w?hlen und dann den H?rer am Telefon abnehmen.

Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error

'lo, all, Is there some (possibly undocumented?) way that I can pattern-match on a specified number of repeating digits? (Something similar to regular expressions' {}) Here's an example: let's say I have a string of things that need to be done for both extensions 233 and 255. I can either... A) Repeat the exact same code for both extensions, like so: exten =>

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the