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

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

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

-----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

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

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:

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

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 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

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

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

We have frequent timeout problems after upgrading our imap servers from dovecot 1.1.16 to dovecot 1.2.11. One server acts as proxy only, and the other one is the "real" imap server". The credentials for the proxy service are stored in a remote MYSQL database. There were no trouble with dovecot 1.1.16. But now, with the most recent version, we get frequent login failures. It

Samba Version: 3.4.7 OS: Ubuntu Lucid 10.04 Setup: This samba box is a member of a win2k active directory domain and functions as a file server. Files/directories shared out utilize file system acls. smb.conf portion for share in question: [Accounting] comment = Accounting Share path = /netdrives/accounting browsable = yes read only = no map archive = no map system = yes

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should

Hi, I'm currently setting up an Apache SSL reverse proxy for Exchange 2003 Outlook Web Access. The setup that I have works fine on my Gentoo laptop or on a Trustix server, however, when I try to set it up on an RHEL based distro, with the exact same virtual host settings, I get some weird error with the authentication mechanism. I have tried with both CentOS 4.2, based off the server CD