Displaying 7 results from an estimated 7 matches for "maraba".
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taraba
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
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.
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