similar to: Asterisk with MySQL Cluster

I have asterisk 1.8.0 installed and I am not able to forward a voicemail from one users mailbox to another user. I have the user log into their mailbox press 8 to forward a message enter the extension of the user I wish to forward too I don't prepend a audio message and press # to send the message to the other user from a debug perspective I don't see any errors. The only message I see

I am currently running two different versions of Asterisk 11.0.1 11.2.1 I have noticed the bug occur on both servers. The issue is that when I try to dial a phone number sometimes the call will never go out. I will check the Asterisk server with NGREP and see that the SIP messages are making it to Asterisk but Asterisk isn't responding. I do the following command "netstat -nap |grep

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

I have configured my mysql database by following this link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The only difference is that I am using ODBC instead of MySQL with Realtime. Within extensions.conf I have the following for my queue exten => 9**2**1611,1,Answer exten => 9**2**1611,2,Queue(irock.com,tT,,,300) exten => *50,1,Answer exten =>

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

I'm getting the following message in my full log at startup and my realtime sip peers aren't being found. My realtime extensions have no errors. The table sippeers exists in the database. Is this a known problem? res_config_mysql.c: Table sippeers not found in database. This table should exist if you're using realtime.

Hi Guys, I have a strong feeling the loads on my servers will be shooting up soon... anyone got any idea how many calls i can expect to put through a DL360: Dual Quad Core 2.33ghz 4gb RAM with 1gb allocated for a ramdisk (call recordings) This server is recording calls (mixmonitor), codec is gsm (no conversion). I know there's a lot of other things to consider like AGI scripts and such

I am using Asterisk 1.4.36 with Realtime Voicemail from a MySQL database, and whilst I have it all working, I am unable to find a way to customize the content of the email that gets sent to a user when they receive a voicemail. In the past I just edited it in the voicemail.conf file and made the customizations in there, but now that I am using Realtime voicemail from MySQL, my voicemail.conf file

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

Hi all, I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server using freetds and unixodbc, which works with 1.6.1.20. With the same config in 1.8 I get an error when trying to start asterisk which says: [Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined

Hi, I'm playing around with asterisk 1.6.2.0 and the first try was to replace my now non-functionning 'app-realtime' macro which emulated RealTime with REALTIME_HASH() There is very few documentation on the subject except for this bug report: https://issues.asterisk.org/view.php?id=13651#c94998 However when i try this syntax:

Starting with Asterisk 1.2 I have always used realtime static to load my extensions.conf into Asterisk. It worked perfectly up to version 1.6.0.X but starting from 1.6.1.X and upwards it simply does nothing. I can see that the extensions.conf file is mapped to the database: == Parsing '/etc/asterisk/asterisk.conf': == Found == Parsing '/etc/asterisk/extconfig.conf': ==

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Hi, I have an Asterisk behind an OpenSIPS proxy. The proxy handles registrations and also SIP SUBSCRIBE for MWI. The Asterisk are configured to send NOTIFY to the proxy even when the SUBSCRIBE haven't been received. I can configure a user in sip.conf that works: [az5134939706] type=friend host=xxx.xxx.xxx.xxx (IP of proxy) port=5060 nat=no mailbox=1234 at customer subscribemwi=no

Hello, When I create a sip peer? in users.conf then a hint is automatically created for that peer. But when I create a peer in sip.conf or a realtime peer with the same values then this hint is not created. Every time I add such peers I have to add a hint in extensions.conf. Is it possible to have something like?? exten => _XXX,hint,SIP/${EXTEN}? in extensions.conf so that I don't have

Hi list, I hope someone could help me. I've started using Asterisk 1.6.0.14 to get queue logs in real time with odbc (our databases are all PostgreSQL) but it's not working. However, cdr odbc is working well. When asterisk starts next message appears: WARNING[4217] config.c: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available My

Hi List, I have a problem with Asterisk 1.6.1.6 realtime (MySQL databases hosted on a separate machine). When Asterisk is in verbose mode, it prints messages saying "failed to extend from 512 to 664" (quite a few lines in a block) and then the last message is mostly "failed to extend from 512 to 663". The number of lines varies unpredictably. The full message (in the logs)

Hi All, I'm testing some realtime extension apps with Asterisk 1.4.28 and addons 1.4.10 using res_mysql. Localhost database is 5.0.32 with Debian Etch. The apps are working fine all syntax is proper, using Set with (REALTIME) function, Set with (CUT) function, calling a Macro with s extensions, and using a few pattern matching extensions as well. I can certainly detail all database rows if

Hi Everyone, For some reason a few phones connected to a pfSense box can't make or allow in OpenVPN in port 1194 UDP. So, I established the VPN tunnel on 1194 TCP and it works fine. I would like to know if there is any disadvantages to using TCP over UDP for the tunnel when using Asterisk or is just as reliable and solid as a UDP tunnel? Thanks -------------- next part -------------- An HTML

So I''ve inherited a legacy application and I''m trying to work around the edges as I put an admin tool interface on top of the existing code base. I install Devise for user authentication, since I''ve used it in the past. I change none of the default code. And yet, on successful sign in, I get an error: Render and/or redirect were called multiple times in this action.