Displaying 5 results from an estimated 5 matches for "janprunk".
2008 Jul 11
1
Microsoft CRM 4.0 integration with asterisk
...Or alternatively integration with Microsoft Office Communications
server (however trying to avoid this, if it isn't really necessary for
the integration).
I would be glad to receive any links or manuals on this topic, which
helped you to integrate it.
Kind regards,
Jan Prunk
--
Jan Prunk <janprunk AT SPAMFREE gmail DOT com>
Website: http://www.prunk.si PGP key: 00E80E86
Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86
2008 Apr 01
5
ZAPTEL
Hello !
I am having issues since the upgrade in debian from zaptel 1.4.7.1 to 1.4.8.
I did m-a a-i zaptel to upgrade
what i still get is:
cat /sys/module/zaptel/version
1.4.7.1
It doesn't seem that it wants to load the new driver, any ideas ?
Kind regards,
Jan
--
Jan Prunk <janprunk AT SPAMFREE gmail DOT com> http://www.prunk.si
GnuPG: 1024D/00E80E86 77C5 156E 29A4 EB6C 1C4A 5EBA 414A 29F5 00E8 0E86
2008 Jun 19
5
Grandstream Busy Light Fields
...IP/71 ; Sinisa
exten => 73,hint,SIP/73 ; Tomaz Doma
exten => 78,hint,SIP/78 ; Tomaz Ratece
exten => 80,hint,SIP/80 ; Bostjan doma
exten => 82,hint,SIP/82 ; Tomaz
exten => 92,hint,SIP/92 ; Tomaz
exten => 95,hint,SIP/95 ; Test
This is it.
Kind regards,
Jan Prunk
--
Jan Prunk <janprunk AT SPAMFREE gmail DOT com>
Website: http://www.prunk.si PGP key: 00E80E86
Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86
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2007 Sep 13
2
Fwd: Bad FCS error
---------- Forwarded message ----------
From: Jan Prunk <janprunk at gmail.com>
Date: Sep 13, 2007 3:35 PM
Subject: Bad FCS error
To: asterisk-users at lists.digium.com
Hello !
I am getting the following error in asterisk logs:
Sep 11 17:33:30 NOTICE[2187] chan_zap.c: PRI got event: HDLC Bad FCS (8) on
Primary D-channel of span 1
Sep 11 21:45:08 NOTICE[2187...
2008 Nov 06
0
Asterisk trunking
Hello !
I am experiencing some problems with Asterisk trunking, this is the scenario:
There are 3 servers, a DID server provider (VOIP provider) which
delegates us a bunch of DID numbers to our asterisk server number one
(I will call it AA), from which I route the calls to Asterisk server
number 2 (I will call it BB), which then terminate on phone handsets.
The trouble is, that I probably