I have a problem with dialing status. Im using Asterisk 1.6 and a patton 4554 gateway for ISDN calls. When I call fixed telephone (not mobile phone) after few ringing the status change to answer but the phone is still ringing, so if I hangup before some..
I cant believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 – Line 1..
From our experience it is not enough. We had to rewrite CDR generation to suite our billing needs. That was on 1.4.xx, we are not using 1.6+ Regards, Mindaugas Kezys Kolmisoft UAB VoIP Billing Solutions e-mail: firstname.lastname@example.org URL: http://www.kolmisoft…
Everyone, I am looking into AMI (using PHP) to record every instance of HOLD that is generated by putting a caller on HOLD (press hold button on the phone set). There is no HOLD in Asterisk but the event Music on Hold is generated when HOLD is press..
Ive had phones before where, with the phone on-hook, it still implements the local dialplan.E.g., if I dialed 0 (on-hook), after three seconds, it would dial the operator, and have the call on speakerphone.Does Polycom allow this functionality?Clear..
Someone has hacked into our system and is making calls overseas. How can I: 1. Find out the where the calls are originating from? 2. Block all calls that are not authorized? Our system is in the USA. Only calls from inside our LAN are allowed. Th..
What format are the actual calls in?Are they in G.711u/a format or are they in something else (perhaps gsm?) format?Im asking to find out if Asterisk would need to transcode them. On Mon, Nov 22, 2010 at 6:47 AM, Vilius Adamkavicius wrote: > All, >..
We are having some problems with crashes in Asterisk, my asterisk
versions are 184.108.40.206 and 220.127.116.11. I have found this:
~/work/asterisk-branch-1.4$ svn log -..
We tried to upgrade our Asterisk from 18.104.22.168 to 22.214.171.124, after trying to install iksemel (jabber support) and spandsp, but now Asterisk doesnt work anymore and we cant get it to run, althorugh we tried to remove it completely and reinstall 1.6.2…