Re: [Asterisk-Users] Q.SIG support in CVS

2005-02-19 Thread Kurt Bauer
--On February 18, 2005 11:16:11 -0600 [EMAIL PROTECTED] wrote: On Fri, Feb 18, 2005 at 02:18:37PM +0100, Kurt Bauer wrote: I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap

[Asterisk-Users] Q.SIG support in CVS

2005-02-18 Thread Kurt Bauer
Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has

Re: [Asterisk-Users] supposable timing problem with TE100P

2004-11-05 Thread Kurt Bauer
--On Thursday, November 04, 2004 04:41:53 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Is your timing source set correctly? If you are connecting to the pstn the pstn connection should be the primary timing source. connection is to a Ericsson MD110

[Asterisk-Users] supposable timing problem with TE100P

2004-11-04 Thread Kurt Bauer
Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 As this is only a notice and voice worked quite well, despite the messages, I didn't bother. _But_, I wanted to try spandsp/fax

Re: [Asterisk-Users] supposable timing problem with TE100P

2004-11-04 Thread Kurt Bauer
--On Thursday, November 04, 2004 03:19:56 PM +0100 Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 4 Nov 2004, Kurt Bauer wrote: Hi list, every now and then I get the following message in my * logs: chan_zap.c:7379 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1

[Asterisk-Users] SIP authentication problem

2004-09-06 Thread Kurt Bauer
;branch=z9hG4bK1cba.998c27f1.0 Via: SIP/2.0/UDP 131.130.220.101:5060;branch=z9hG4bK03521c86 From: Kurt Bauer sip:[EMAIL PROTECTED];tag=000cce3a7be800087fd8099f-62cc5396 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Date: Mon, 06 Sep 2004 10:01:57 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact

Re: [Asterisk-Users] Going to voicemail instead of queue if no agent is logged in ?

2004-09-03 Thread Kurt Bauer
Hi, I did this the following way: -) define a global variable - AGENTS_AVAIL=0 -) when agent logs in increment - SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} + 1]); -) when agent logs off decrement - SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} - 1]); -) when queue is called evaluate and goto

Re: [Asterisk-Users] strange problem PBX-Asterisk

2004-08-25 Thread Kurt Bauer
Thanks for the hints, 'overlapdial=yes' did the trick. br, kurt --On Tuesday, August 24, 2004 10:08:08 PM +0200 Peter Svensson [EMAIL PROTECTED] wrote: On Tue, 24 Aug 2004, Christian Victor wrote: maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error':

[Asterisk-Users] strange problem PBX-Asterisk

2004-08-24 Thread Kurt Bauer
Hi, maybe I oversee somth. very obvious, but I'm a little puzzled about the following 'error': When I make a call from the PBX to * I get number not available, but on debug I see, that asterisk is searching just for the first digit in the extension, which of course doesn't exist, eg: I dial

[Asterisk-Users] Zaptel Problem after Upgrade

2004-08-20 Thread Kurt Bauer
Hi, after upgrading to latest CVS (20.08) I have a problem with the connection to PSTN: When I try to make a call from PSTN to * I hear the number not available' sound and the following warnings on * console: Aug 20 12:09:55 WARNING[180235]: chan_zap.c:6762 zt_pri_error: PRI: !! Unknown IE