search for: quikcat

Displaying 7 results from an estimated 7 matches for "quikcat".

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2005 Aug 24
2
RealTime ignoringswitch=>Realtime/context@re altime_ext
Thanks John, You are my savior. This is such a great relief. Apparently realtime will not use either '127.0.0.1' or 'localhost' to connect to the database. I had to use the actual IP address attached to the NIC before it worked. My OS is Debian just a note and Asterisk HEAD from August 20, 2005 Details below for those who might be swimming in the same pool with me.
2005 Jan 04
6
Polycom Buddy Feature
Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP
2006 Mar 16
1
Re: transfers/parked calls + polycom 501
This is a dialpaln issue. I solved the same problem recently. For 4 digit extensions you need to append the dialplan statement in the sip.cfg configuration file as follows <digitmap dialplan.digitmap="[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2 -9]xxxT|1xxxT" dialplan.digitmap.timeOut="3"/> Michael > I am not sure what I did but blind transfers do not
2006 Oct 26
10
ECHO Cancellation in SIP Calls
Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the
2004 Aug 11
0
X-Lite behind NATed ADSL router
The X-lite client is installed behind an ADSL router and it seems the ISP also has a firewall supposedly protecting their ADSL customers blocking some netbios ports and non-critical. I can't make outbound calls but I can receive but before then I had to use the following in the account section in SIP.conf [2222] ... type=friend secret=xxxx host=dynamic ;dtmfmode=inband ;
2005 Aug 23
2
RealTime ignoringswitch=>Realtime/context@realti me_ext
I am using the current HEAD of asterisk and for asterisk-addons. I have been trying to setup realtime mysql voicemail but no sucess. I keep getting this error below. The necessary modules are loaded, res_config_mysql.so ... Any pointers will help. Thanks mail*CLI> realtime mysql status Aug 24 00:56:50 ERROR[963]: res_config_mysql.c:596 mysql_reconnect: MySQL RealTime: Failed to connect
2006 Oct 24
6
Callmanager 3.3(5) and Asterisk with ooh323
I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael