Has anybody got the polycom acd function to work? I have the following
setup:
Debian 3.1 - 2.6.8 linux
zlib-1.1.4
libpri-1.2.3
zaptel- 1.2.6
Asterisk - the bweschke/polycom_acd_funtions branch version - I get one
error when doing a make install about needing a newer version of libpri and
zaptel, I
I've spent some time now trying to find information on the changes made
to the Asterisk config files. I want to upgrade an old installation to
the latest version, which of course now uses phone.conf. If anyone
could point me in the direction of a set of upgrade notes so that I can
work out w
Does the Sipura web interface on the info page reveal that the spa2100
is successfully receiving CLID?
My SPA2100 passes CLID from asterisk to the connected phone without problem.
On 23/06/06, Jim Lynch <[EMAIL PROTECTED]> wrote:
I have a Uniden wireless phone connected into Linksys/Supura 2100
On Monday 19 June 2006 15:41, Remco Barendse wrote:
> Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a
> x86_64 box (I guess nobody is using x86_64 platform or is able to fix this
> themselves?)
>
> First of all when bristuff is downloaded and compile is started it appears
>
On Thursday 22 June 2006 20:59, John Klimek wrote:
> Any idea why it wouldn't work in my dial plan?
>
Set EXIT_CONTEXT before the Dial command and then populate that context with
the single digit options u want
extens => s,1,.Set(EXIT_CONTEXT=interrupt)
extens => s,n,Dial* ..,d)
[in
On Wednesday 21 June 2006 14:12, Tzafrir Cohen wrote:
> On Wed, Jun 21, 2006 at 01:57:36PM +0200, Mimmus wrote:
> > Hi,
> > after a few of upgrades, I noticed these messages in full debug log:
> >
> > Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype
> > Jun 21 12:58:11 WARNING[27273]
On Fri, 2006-06-23 at 14:58 -0400, Matt Florell wrote:
> P4 3.2GHz
> HT enabled
> 1MB L3 cache
> 2GB RAM
> Asterisk 1.2
>
> meetme participants were Zap or IAX, some rooms recording
Matt,
A bit off-topic but I noticed you have HT enabled. Since you may be
doing transcoding on the MeetMe box, whi
Probably the ring voltage is to low for the phone. Try turning it up a bit.
On 6/24/06, Jonathan Attwood <[EMAIL PROTECTED]> wrote:
Does the Sipura web interface on the info page reveal that the spa2100
is successfully receiving CLID?
My SPA2100 passes CLID from asterisk to the connected phone
On Thursday 22 June 2006 23:51, Roland Zagler wrote:
> Hi to all,
>
> we are searching for a hardware based DSP solution for use
> with Asterisk based on PCI or MiniPCI to reduce main processor
> load and to use embedded boards with Digium E1/T1 cards like
> TE410P.
>
> does anyone know about any m
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are many
other cases.
I have created a number the user can dial to reset this flag. However,
that is written in the manual!!! Who reads a manual anyway
I wan
Is anybody using XEN in conjunction with Asterisk and/or Openser?
I would like to get some info about such an environment and experience
reports.
bye
Ronald Wiplinger
___
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Asterisk-Users mailin
Pessoal,
Facing wierd bug on * using MD3200 modem. It was working ok, then
after a boot bug started:
The calls that came out thru the Zap channel, stopped work. The *
gets the call from an IAX client and set it as active, even before
the destination rings, and finaly when someone pickup de phone
Those are the only files that come to mind. On 6/23/06, Erick Perez <[EMAIL PROTECTED]> wrote:
Tom, just to make sure im on the right track.What files do you tweak?sip.conf, the ones from avaya and anything else?On 6/22/06, Tom Lynn <[EMAIL PROTECTED]> wrote:
> Nope. Let me know if you do. I've
Hello,
I try to configure CDRTool + Asterisk .
After having fixed a lot of files I get too many
problems
with this software.
First I get this error why ?
Error: Table 'asterisk_cdr' was not locked with LOCK
TABLES
The basic soap client does not connect to the soap
server to get the version .
Hey everyone,
I know this isn't a direct Asterisk issue, but some of you may know this
answer.
I recently upgraded the SIP version to 1.6.6 on all of our phones in the
office. Everything is working fine, except one aspect. The phones in the
office reboot randomly for no apparent reason. I ha
As you said, my expectation is wrong in the current release. How can
I implement load balancing using multiple asterisk of current release
with a centralized database in another way? Is it possible? Any
clue? Thanks!
On 6/23/06, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
- unplug <[EMAI
On 24 Jun 2006, at 02:43, James Harper wrote:
I've not used it with VoIP, but we have a speakerphone which we have
used on occasion in a meeting where someone was unable to attend.
When it is turned on, it sends out a 'ping' and measures the echo
response of the room, and perfectly cancels out
On Jun 24, 2006, at 1:37 AM, George Gardiner wrote:
I've spent some time now trying to find information on the changes
made to the Asterisk config files. I want to upgrade an old
installation to the latest version, which of course now uses
phone.conf. If anyone could point me in the direct
Martin Joseph wrote:
Huh, I never looked at that file before (phone.conf).
Actually they seem to refer to a Linux telephony interface?
Anyone please care to elaborate on what the phone.conf file is really
for? The wiki just has a copy of the default file...
It's the conf file for the
I have a single DID number from sellvoip.net.
This works well and has been mostly reliable (HEH). I am wondering if
caller id info is likely be provided for calls to my DID #?
If so, should they show up in the CLI(they don't)? Or how do I go
about looking at them (the caller ID infos that i
Bartosz,
Try changing username=username to user=username
This works for me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bartosz
Wegrzyn - asterisk
Sent: Thursday, June 22, 2006 1:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ia
Kevin Smith wrote:
Any other thoughts as to what may have caused the phone to reboot?
the power supplies on these phones are very underrated and any power
fluctuation will cause them to reboot. I get it when we are on generator
and the A/C cuts in.
--
Chris Mason
(264) 497-5670 Fax: (264)
Dustin,
any updates on this?
Thanks,
Daniel
On Jun 23, 2006, at 1:07 PM, Dustin Wildes wrote:
shadowym wrote:
That feature is called Bridged (or Shared) line appearance. That
is one of
the things Asterisk cannot do and nobody seems very interested in
making it
do that because it is app
Ronald Wiplinger wrote:
If a user calls and hangs up before the destination party rings, than
the in-use flag remains set! This is one case, but maybe there are
many other cases.
I have created a number the user can dial to reset this flag. However,
that is written in the manual!!! Who reads a
Hello Asterisk Community,
I'm using Voismart's GSM PCI cards to connect Asterisk to GSM cellular
network.
The problem I face is DTMF detection; that is, whenever I call to one of
the channels (SIMs) on GSM card through my Mobile phone, and dial DTMF
digits while in the call, the Asterisk receiv
I had that running in my small home Asterisk setup for a while, however I
had to move back to an older kernel to support a SATA driver (mv_sata). It
looks like as of late 2.6.16 and 2.6.17 they've updated the driver so in a
week or so I am going to try builind it again and create a DomU with
As
Hi Brian -
I'm just trying to get my Polycom 601 to have multiple extensions on it. For
example, on
line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want
extension
23. Ideally I'd actually have each extension appear on 2 lines and therefore
filling up all
6. I sho
Hey Chris,
That is interesting. The ones in the office are all connected using a
PoE switch. One would hope that the transformer and support
filtering/feedback circuitry would be able to filter or compensate for
any power fluctuation the switch encounters. I will have to look into
that, and s
Hi,
I have configured asterisk now with ENUM lookups which are working
really perfect.
Now I want to play a small soundfile before dial the number to inform
the caller which protocl is used (SIP, IAX2 or ISDN).
How can I do this?
With Playback it doesn't seems to work:
[iax2-sipport-out]
; with
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