similar to: clustering

I have a customer that has a Trendnet TEW-435BRM router which has the bad habit of rewriting all external connections so the Asterisk server only sees the IP address of the router itself. Up to today this has not been a problem since all extensions are on the local network but now they want to have a couple external IP phones (SIP). I opened up the ports on the router and my phone can register.

Hello community, I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone. Now as next step I want to extend this to PSTN , My Required scenario: I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected. Which hardware I need for it. Also please

Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i'm looking to update some SRV DNS Record , but i didn't find the correct syntax to handle priority, weight and port. The goal is to higher the priority of one of my different DCs. Thanks for your help Alain

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this

Hi All, Please refresh my memory. I am trying to install asterisk after 2 years. I hav'nt used it since 2008 (version 1.4.2). Now I am trying to install 1.8.0-rc2 on centos 5.5 but getting the following errors. app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory app_mysql.c: In function ?mysql_ds_destroy?: app_mysql.c:135: warning: implicit declaration of function ?mysql_close?

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

Hi, We installed a samba4 AD controller using Gpo for a small group of users (5 users), everything is OK. (Samba4 beta 7 on Ubuntu 12.04) We installed another Samba4 AD controller as a BDC of the first one with the command "samba-tool domain join" with succes. After checking this new installation, we saw that the sysvol share was not replicated from the PDC and all the Policies are

Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other

Hi List, Can anyone please tell me how to use the phoneprov.conf to provision my client's atas. I read the file but dont know how to actually use it. -- Best Regards Rizwan Qureshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101027/7f9de26c/attachment.htm

Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with "Asterisk <Unknown>" caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Hi Everyone, How can I create a TLS based sip trunk between two asterisk servers. I have been trying to do it but i dont know how to defined the client certificate on the asterisk server. Has anyone tried this? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying to use for kind of a call intercept between two GSM users. Call comes through one SIM and goes out through another Sim with our Asterisk in between to log the call. This works fine but we need the original callerid to pass-through through the outgoing SIM. I have tried every possible configuration on Asterisk that

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8

The relevant part of my setup is something like: SIP phones -> local server -> remote server -> SIP-to-PSTN provider I want _some_ of the SIP phones on the local server to be able to get access to SIP-to-PSTN, but not all of them. The local-to-remote connection is IAX2 over VPN. Do I need to set up two separate IAX2 connections, one "privileged" and the other not, or can I