Thanks but it did not help resolve the situation.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Thursday, January 19, 2012 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial
On 22/01/12 08:25, Eyal wrote:
Thanks but it did not help resolve the situation.
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew
2012-01-20 20:09, Matt Hamilton skrev:
Hi,
Once in a while when a SIP channel connected to meetme conference is
hung up, I start getting the following error multiple times:
WARNING[14031]: app_meetme.c:3668 conf_run: Unable to write frame to
channel Local/100203@h
The status of the
I'm not using meetme directly - I'm using SLA which internally uses meetme and
creates conferences for SLA trunks. There are no sound problems for me, either,
but when the caller hangs up and this error occurs, the trunk statuses are not
updated properly and the phones still show them as in
Hi All;
Is there a telephony card that contains analoge ports and E1s at the same time?
Regards
Bilal
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Hi All,
I am currently building GSM Based trunk, for voice and sms.
I am compiling the newest 1.6 (Asterisk
1.6.2.22http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.2.22.tar.gz)
asterisk with --bluetooth support,
and also asterisk-addons for chan_mobile support,
my os
When I call my internal extension and hang up the phone keep ringing, I get:
SIP/11-0048 connected line has changed. Saving it until answer for IAX2/iaxy
Is there a solution for it?
SIP does not detect that IAX has hang up the line.
--
Joseph
--
http://sangoma.com/products/hardware_products/digital_analog_hybrids/b601.html
2012/1/23 bilal ghayyad bilmar...@yahoo.com
Hi All;
Is there a telephony card that contains analoge ports and E1s at the same
time?
Regards
Bilal
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