Lance,
I was in a similar situation, though i was rec'ing the event 6 message,
i noticed no degradation of sound and so ignored it. I've since removed
a *load* of unused modules, and it appears that the message is no longer
coming in. I had read that some people were only getting the message
Also, around the same time, I isolated the IRQ that my zaptel cards were
on. (so neither zaptel card shared its IRQ).
you can see what IRQ's are in use with
lspci -vb
This is more likely to be the cause of the fix.
Adam Dobrin wrote:
Lance,
I was in a similar situation, though
I haven't tested this, but we've been thinking the same.
http://lists.digium.com/pipermail/asterisk-users/2004-November/072387.html
Carlos Alperin wrote:
I need to see if we can make the voicemail do the following:
After they reach the voicemail, and left a message or not, they should
be
Rich Adamson wrote:
Thanks for the thorough reply.
I'm aware that there necessarily are inconsistencies between termination
providers; I was just curious to find out if there's some form of
standard one should follow, which may either result in more consistent
behavior, or at least shift culpab
I've read that the auto fax detection for asterisk is built into the
chan_zap software, however i've been experiencing odd behavior. I have
two digium cards, a TDM and a TE110p, and the only time i am getting the
fax detection is on outgoing calls from the TDM card.
zapata.conf has =both.
Chad Osmond wrote:
CVS Head from 07/07/2005
I'm trying to make an IP-501 auto answer a call.
exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans")
exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations
exten => 301,3,Dial(SIP/5001,15)
exten => 301,4,Hangup
Sip.cfg for Polycom phone
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Eric wrote:
I would like to initiate a call in asterisk (say with cron)
so that this call rings on the destination number _and_
on an asterisk extension.
How would I achieve this?
thanks
I'm using Polycom 501's; with stable1.0.8, g729 and a very decent
machine; we have a PRI interface to a T1.
Many users complain that after a given amount of time, say, 30 or 40
minutes on a call, the outside party complains that their sound keeps
'cutting in and out'. I believe that the inc
here, here!
Kevin P. Fleming wrote:
Lee Howard wrote:
Go ahead and have a proprietary fork, sell it, have it specially
licensed. But please, please, please treat the community fairly.
Otherwise it causes unrest in the community, discourages
contribution, encourages forking, and triggers f
"pri debug span 1"
will show you what the PRI is sending to you, before you do that:
have you tried the previous suggestion of issuing a Wait(1) (i actually
suggest 2) prior to Answer or Dial?
Ryan Williams wrote:
How do I go about doing this? I can not find any documentation on "isdn
trace
to force g729 for non PSTN connections, and ulaw
(im in the US) for calls going out the PRI. Has anyone done this..?
I'm still hoping to be able to stick with g729; anyone else experience
this kind of issue?
-a
Adam Goryachev wrote:
On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrot
anyone else have the above issue? this is today's CVS.
thanks.
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I have asterisk running on dual 244's. Everything works fine, the only
special issue i had was installing the g729a codec (required a very tiny
tweak to the asterisk Maiefile). Unfortunately, the system doesn't get
a huge amount of traffic, so I can't testify to capacity.
Running 1.0.8, btw.
Sarge. RHEL/compatible should be fine too.
Wiley Siler wrote:
Did you build it using the 64 bit CentOS or another Distro?
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Dobrin
Sent: Friday, July 22, 2005 4:47 PM
To: Asterisk
As nice as HDLC sounds in theory; we have the same setup, a T1 with afew
lines split off, and i just don't see a need to add the routing load to
the asterisk machine. We have an Adtran 604 which splits the T into a
PRI and 10/100. Incidentally, HDLC in asterisk seems to be.. a hassle
to get w
d be to force g729 for non PSTN connections, and ulaw
(im in the US) for calls going out the PRI. Has anyone done this..?
I'm still hoping to be able to stick with g729; anyone else experience
this kind of issue?
-a
Adam Goryachev wrote:
On Thu, 2005-07-21 at 15:56 -0400, Adam Dob
Bob Goddard wrote:
On Thursday 28 Jul 2005 13:07, Ronald Wiplinger wrote:
before I accuse somebody to "overbill" I would like you to calculate
with me:
Rate: 0.0189 for calling Taiwan via NuFone
Duration: 930 seconds
Lets vote for the answers:0.7269 or 0.2929 ???
Assuming it
You aren't dealing with analog phones, and you aren't transmitting DTMF
signals.. the functional difference between analog and digital systems
kindof precludes what you are looking to do.. meanwhile, once the entire
number has been dialed, the outgoing call should be started almost
instantaneou
And as much as you dislike these kinds of questions; its unfortunate
that the community doesn't have any good answers to them available--they
should be. It would be great if we could get some independent
verification of digium's claims/figures.
voip-info:
http://www.voip-info.org/tiki-index.p
www.emplifyhr.com/pcom
Douglas Garstang wrote:
We purchased our phones through Alliance Systems, a Polycom certified reseller. Getting firmware was difficult, and they where very unresponsive, probably because we didn't pay them additional money for a support contract. Such is life.
-O
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