I have asterisk
1.2.7.1 running on Fedora core 5. Everything looked like it compiled
OK.
When a call is
bumped to voicemail, the message prompts sound fine to the user. However,
when the voice message is retrieved, it sounds "compressed" or speeded
up.
I have checked this
against t
channel drops.
As an aside, I have noted that calls that are hung up before completion, from either the Definity or the BCM, clear with a cause of -1. This can hang a channel on the Definity side as well.
Mark.
From: McQuiggan, Mark xt46480
Sent: Monday, March
The best way to set gains would be to use ztmonitor (located in
/usr/src/zaptel). Make a call and note your channel number. Run
/usr/src/zaptel/ztmonitor -v from a telnet session. check
to see if your levels are too high or too low and adjust your zapata.conf
accordingly. I ended up setting m
I have the
following configuration:
Bell CO.
<---ISDN---> Definity <---ZAP PRI---> Asterisk <--ZAP
PRI---> Nortel BCM.
With SIP
users on the Asterisk box.
I have found
a problem with call tear-downs. When a caller calls in from Outside, or
from a Definity station to a BCM station,
Is it possible that
putting the incorrect switchtype in zapata.conf can cause Asterisk to
crash?
I have a Definity
Generic 3 connected to a TE405 port 1, and a Nortel BCM connected to port
4. In zapata.conf, I configured the Definity connection (group 1) to
switchtype=national as per voi
= 1
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
nsf = none
overlapdial = no
signalling = pri_net
channel => 1-12,25-36
rxgain=-5
group = 2
context = trunk
usecallerid=yes
callerid = asreceived
switchtype = national
overlapdial = no
signalling = pri_net
relaxdtmf = yes
While testing a
problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon
this problem:
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1210770512 (LWP 11346)]
0x002e3fe1 in pri_release_timeout (data="" at
q931.c:2589
2589 q931.c: No such fil
I am now receiving
spontaneous restarts of asterisk on my system, with no apparent rhyme or
reason. I am using version 1.2.4, with zaptel-1.2.3 (downgraded from
1.2.4. I downgraded after the system started restarting spontaneously,
this week. I upgraded last Friday). I see no indication
Chuck:
Try placing your "group=1" line above the "context=incoming-rest" line. You
seem to have to define your group first. I think that your second "group=1"
is redundant as well.
Regards,
Mark McQuiggan
Message: 1
Date: Thu, 10 Nov 2005 11:02:19 -0700
From: Chuck Bunn <[EMAIL PROTECTED]>
S
On voip-info.org there is a claim that asterisk and a BCM can interconnect
via H.323. There is little on the page beyond setting the H.323 connection
on the BCM to "other". Hardware restrictions at the moment make the H.323
solution preferable to ISDN or SIP. I am using oh323.
Every time that
Has anyone been able
to successfully integrate the Avaya AUDIX voicemail system with Asterisk?
I have been
diligently investigating converting our small (Ontario, Canada) office to
Asterisk, and ditching our Avaya PBX. However, our head office (New
Jersey, USA) maintains our AUDIX syst
Title: "Called" ID question - Trying again
The question below was asked before, and bounced back to me...trying again
In the Avaya PBX, when a person calls a local extension within the PBX network, the caller is provided with the called person's name on his/her call display (I suppose th
Title: "Called" ID on local extensions
In the Avaya PBX, when a person calls a local extension within the PBX network, the caller is provided with the called person's name on his/her call display (I suppose that this could be called "Called ID"). For example, if I called Joe Blow from my exte
>Peter Svensson wrote:>>>On Tue, 22 Mar 2005, McQuiggan, Mark xt46480 wrote:>>>> >>>>>I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to
an>>>Asterisk v 1.0.3 PBX. The PBX is also connected
via a ISDN-PRI crossover>>>
I am using a SIP
softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX.
The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity
Generic 3 PBX via a TE405P card. All outside of the office calls go
through the Definity. Here's the issue:
Calls to i
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