similar to: Help with voicemail

Trixbox 2.2 Beta with freePBX 2.2.0rc1 I have a setup that looks something like this in ASCII art: Teliax IAX Trunk ------+ | V Embarq PRI ----> Tandem switch ----> Ottawa Office Server------+ +--------------> Lima Office Server -----+|

I have an issue where * tries to route a call over IAX to another server even if the server is down. I have included the relevant entries from my iax.conf, extensions.conf, and some debug output. If someone could tell me what I have configured incorrectly, I would appreciate it. Thanks, Stephen -----------iax.conf on voip2---------- [voip1] type=friend username=voip1 host=x.x.x.x (ip

I'm specing out a new box to act as a tandem switch. It will have a TE410P with 4 x PRI and support IAX connections to four other boxes using predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call routing also using gsm. No extensions actually terminate on the tandem, they're all switched to other boxes (highly distributed). On the PRI card, one goes to Embarq, the

Hi, I just upgraded from 1.2 to 1.4. In 1.2, when I did a "show uptime" I used to see a second line telling me the time since the last reload. Has this been removed in 1.4? The following is the output of my two test boxes: Connected to Asterisk 1.4.18.1 currently running on voip2 (pid = 10605) Verbosity is at least 3 voip2*CLI> show uptime System uptime: 15 hours, 55 seconds

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hi I have an account with mynetphone (australia), which gives me two voip (sip) accounts, which i used to have connected to a spa9000. this is behind a firewall, so on the spa9000 I would listen on another port apart from 5060. so on the firewall 5060 would go to voip1 and 5061 to voip2. I moved to asterisk (+tdm410) and the machine was also the firewall and I had no problem - well atleast it

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

Hey, all... If this is too off-topic, I'd be grateful for directions to a more appropriate mailing list. I'm trying to set up Asterisk and some Sipura boxes. I've got an SPA-3000 which is registering twice with Asterisk - once for its FXS/Line1/VoIP1 and once for its FXO/PSTN/VoIP2. My eventual goal is to have inbound calls on its FXO ring four times on its FXS and then fail over to

Hi, I'm trying to get zapras do work, I had downloaded the pppd-source and the 2 patches. I succefull compiled and install the patched version of pppd, but got this error in message-log Jul 15 11:43:32 voip1 pppd[9296]: In file /etc/ppp/filters: unrecognized option 'active-filter' Jul 15 11:43:57 voip1 pppd[9299]: Plugin zaptel.so loaded. Jul 15 11:43:57 voip1 pppd[9299]: Zaptel

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

December 28, 2017 12:43 PM, "Fabian A. Santiago" <fsantiago at garbage-juice.com> wrote: > December 28, 2017 12:14 PM, "Aki Tuomi" <aki.tuomi at dovecot.fi> wrote: > >>> On December 28, 2017 at 7:07 PM "Fabian A. Santiago" <fsantiago at garbage-juice.com> wrote: >>> >>> Hello, >>> >>> After

Hello, I have setup v2.2.12 to mirror mailboxes between our main mail server (vmail.example.com) and a failover server (vmail1.example.com). The problem I have is the behavior of mailboxes used on the main server (*vmail*) over POP3. I have setup an IMAP account on the failover server (*vmail1*) on such a mailbox: the mailbox is used on vmail using POP3 and the same is monitored on vmail1

Hey guys, I am trying to pinpoint where the problem with my dovecot-FTS installation . I am trying to rule out between the Dovecot and the Solr Module. I think if the output below looks OK then the dovecot is fine, and will focus more on the Solr. Appreciate if someone could check and confirm. Thanks root at mail:/etc/dovecot/conf.d# doveadm -D -v fts rescan -u admin at email.net doveadm(root):

Hello there, I'm trying to use the doveadm expunge function, but it doesn't seems to work.. I have installed a mail test server using iredmail, Dovecot and postfix using mysql for authentications. I sent a email to this server, and change the timestamp using touch. the mail boxes are as follow : /home/mail/vmail1/mydomain.fr/n/i/c/nicolas-2013.09.05.14.30.49/Maildir I want to delete

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List I'm trying to create a conference room using H323 channels. If i start asterisk normally (service asterisk restart) and connect to cli using -vvvvvvvr options, when a user enters the Conference, asterisk says "You are the only ..." and then dies, withou any error message, nothing at all. But, if i start asterisk with cli

Hi All, I'm on a middle of an asterisk installation/configuration for my company and I'm testing the TLS configuration. For this reason, I used the ast_tls_cert script to build the ssl certificates for my server. On sip.conf file: tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 and on

I have a small Postfix/Dovecot/MariaDB on Centos 7 with a dozen virtual domains, I'm running out of space in /var but got space can I simply move the entire /var/vmail/vmail1 tree elsewhere ? like say /vmail/vmail1 is it as simple as shutting Dovecot/Postfix, move tree, edit conf and re start ? or am I missing something ..? or is the a better way of doing this ? # grep 'var/vmail'

installed new Dovecot 2.2 on Centos 7, and, migrated some users from old server # dovecot --version 2.2.33.2 (d6601f4ec) works OK, but, noticed some sql and index.cache errors what am I missing ? only noticed this one so far ------------------ # grep "Too many open files" /var/log/dovecot.log Jan 02 15:22:16 auth-worker(1411): Warning: mysql: Query failed, retrying: Got error 24

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an