[asterisk-users] Zap channel shows "answered" as soon as outbound ringing starts

yusuf yusuf at ecntelecoms.com
Mon Nov 6 23:35:13 MST 2006


Hi,

I had this problem; when dialing to a Zap channel, it was "answered" as soon as the ZAP phone 
started to ring.  This is because I had callprogress=yes in zapata.conf.  Whne I disabled it, it 
fixed this problem.  I hope this helps!

shadowym wrote:
> Just to follow up on this,
> 
> After some testing tonight I found the following.  Watching the Asterisk
> CLI, when making a call from an extension to a ZAP channel the channel shows
> as "answered" as soon as the zap line starts ringing.  That would explain
> why Followme was not working.  It thought the PSTN line was answered
> 
> So the problem is that ALL outgoing PSTN calls are seen as "answered" as
> soon as the Sangoma card rings a zap channel.  Not the first Asterisk
> generated ring but the second ring right when you hear it switch over to the
> PSTN line.  Is there a trick to prevent this?  I messed around with "wink"
> and "debounce" settings in zapata.conf but that didn't seem to make a
> difference. 
> 
> -----Original Message-----
> From: shadowym [mailto:shadowym at hotmail.com] 
> Sent: Monday, November 06, 2006 1:06 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Follow Me problems
> 
>  
> Hi all,
> 
> I have a production system up and running for a little over a week now.  So
> far it has exceeded my wildest expections.  No problems whatsoever running
> what I consider a fairly complicated dialplan using many advanced features.
> 6 extensions and averaging about 50 calls a day. 
> 
> Today we appear to have discovered our first bug.  We have an extension
> setup to "followme" by ringing that extension + an external cell #
> (ringall).  If nobody answers after 20 seconds the "destination if no
> answer" is set to go to the extensions voicemail in the "followme" module.
> The problem is it just keeps ringing forever.  If we delete the followme it
> forwards to the voicemail as per the default SIP extension configuration
> with voicemail enabled.
> 
> Anyone run into this?  Is there a workaround?  Any advice would be greatly
> appreciated as always.
> 
> Our configuration is:
> Supermicro Pentium D 2.66 Server with 2x512MB Memory 3ware 8006-2LP Hardware
> RAID 1 Sangoma A200D with 8fxo (latest firmware/drivers as of last week)
> CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 FreePBX 2.1.3
> 
> 
> 
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-- 
thanks,
yusuf

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