similar to: disconnecting destination channel

Hi all, I have 2 question. First: I use dovecot (version 2.2.9) with mdbox mail format. When I run dsync tool with "mirror" or "backup" parameters my source and destination directory synchronize correctly but if I delete some messages in user mailbox, deleted messages does not synced to destination. For example : atif at domain.com path is /mail/domain.com/atif/ and its

Hello everybody!!! I really need your help guys, I am using the AGI mode in meetme application, and I want that AGI should wait for an input from the client/user i.e. a digit and then proceed, but I have used that AGI function agi->wait_for_digit(), but no use....my agi just passes, or ignores this function, where AGI should stop here and wait for the input.... .....my extension in my

Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

I am using gentoo Linux and Asterisk CVS-HEAD-09/23/04-19:57. I have tested both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

Hello there! Somebody tried the meetme|b which initiates the conf-background AGI. Actually I want to originate another call from a conference.my AGI originates the call and connects it to the conference, but the calleeee is nowhere My extension exten => 21,1,meetme(21|pb) and my AGI **************************************************************************** #!/usr/bin/perl -w

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

this is from my extensions.conf, the first three patterns are for toll-free numbers, and fourth pattern is for other numbers, where an AGI is called for authentication. now when I dial 011448000664327 if falls into the fourth pattern, where as it should be matched by the first pattern. Any suggestions 1 - exten => _01144800XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 2 - exten =>

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

I am configuring a sip-phone, receing calls, excellent voice quality. but it does not place calls, please, can some one sort out. here is my debug output, and below that is sip-debug, Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to 0 Jul 16 11:34:32 DEBUG[163850]: Stopping retransmission on 'iiasPlzFribMJMcW' of Response 1: Found Jul 16 11:34:32 DEBUG[163850]: Setting NAT on RTP to

Hi all, I have a problem with using remote MySQL database server with two Asterisk (1.4.17) servers. PhoneA registers with Asterisk#1 using realtime into MySQL on remote server and everything is working fine and when I call Phone A from Phone B (also registered with Asterisk#1) call is established. Problem is when I call PhoneA (which is registered with Asterisk#1) from PhoneC (which is

Keywords: dovecot, dict, quota, postgre sql, broken pipe, remote disconnected Having Dovecot 2.2.22 (fe789d2) with Postgre SQL 9.5 (9.5.5-0ubuntu0.16.04) as the backend. I do not understand why quota service is not working, not seeing it as a configuration error at least. My quotas are DICT/SQL based. OS: Ubuntu 16.0.4.1 32-bit (Linux XXX 4.4.0-59-generic #80-Ubuntu SMP Fri Jan 6 17:36:54

Dear Xen users, I was wondering if there is a way to change the xen bridge (vif) of a running domain. For example, if I have booted a domU with xenbr1 as vif. Now is it possible to change the vif to xenbr2 while the domU is running? If yes, then can someone guide me a little so as to where to start looking? Best Regards, Muhammad Atif __________________________________________________ Do You

hello there, any one who used ASTCC in a real enviroment, or has successfully handled above 1k simultanous calls. need some evalution of ASTCC. if any one has such an experience please share it with the rest thank you Atif

Hello all, I just installed chan_h323 by Jeremy (nufone) with asterisk-head. well, I have some questions if someone please briefly answer. 1 - can we configure h323 to work in g729 pass-thru mode, like we do in SIP. becaz I had to installed g729+intel libraries to work with g729. where as we don't need to install g729 to work in pass-thru mode. 2 - now most important question is

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Sun, 5 Feb 2017, ygrishin-lists at mail2.ca wrote: > service dict { > unix_listener dict { > mode = 0660 > user = Debian-exim > group = Debian-exim > } > } > > dovecot-lda-erros.log: > ********************** > Feb 04 14:23:33 lda(testuser at XXX): Error: read(/var/run/dovecot/dict) failed: > Remote

Hi there, I'm running Dovecot 2.0.16 just set up the director with cut&paste from the wiki. It's working & running fine for pop/imap connections (as verified by doveadm director status user), however when I specify 'RCPT TO' for a (tcp) lmtp connection straight away I get: 451 4.3.0 <xxx at yyy.com> Temporary user lookup failure and in the logs:

The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint. Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header

I place a call into Asterisk (from SIP phone) and the To header does not have a tag. Asterisk then sends it's Trying response, still no tag in the To header. The phone then replies with OK, this time the To header includes a tag. Is there any way to retrieve this response To header (including the tag field) from the dial plan? I have tried the PJSIP-HEADER read of the To header, but it