Hello,
I recently acquired
a Digium TE210P for a faxing solution I'm working on. I would like to use
the card to send faxes from my infrastructure using Asterisk and HylaFAX.
I'm curious if the zaptel driver will recognize in/out faxes to/from the
TE210P. Is this going to work or should I
Hello,
I'm just starting
out with asterisk and I'm playing around with the system. Currently I have
a Digium TE210P connected to a PRI on the Asterisk server. I have a SIP
soft phone on my laptop for testing that is working fine. When I try to
place a call from my soft phone I get this fr
Some more info to my problem:
ipt-dev01*CLI> zap show status
Description
Alarms IRQ
bpviol CRC4
T2XXP (PCI) Card 0 Span
1
OK
0
0 0
ipt-dev01*CLI> pri show span 1
Primary D-channel: 24
Status: Provision
Thank you for the reply Alexandar.
After I restarted my dev machine I recieved these messages
from asterisk:
chan_zap.c:2290 pri_find_dchan: No D-channels
available! Using Primary channel 24 as D-channel
anyway!
chan_zap.c:8202 pri_dchannel: PRI got event: No more alarm (5) on Primary
D
Hello,
I'm recently having
the problem where Asterisk just stops working. The console gets
disconnected and the process appears to die. I am using Asterisk version
1.2.9.1. Anyone have any ideas on where I should be looking for the cause
of my problem? Also, I notice there is a /var/log/
Hello,
I am having a
problem with transferring calls that come in from the outside. Users have
been calling in to the PRI that's on the Cisco GW, then they are passed into
Asterisk via SIP and to the end phone (Polycom 501/601) using SIP. When
that user tries to transfer that call to anoth
Asterisk 1.2.9.1 is the version I'm
on.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
BrummerSent: Tuesday, July 11, 2006 8:30 AMTo:
asterisk-users@lists.digium.comSubject: [asterisk-users] Issues with
making Transfers
Hello,
I am having a
problem with transferring
-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk stops abruptly
On Tue, 2006-07-11 at 08:20 -0700, Dan Brummer wrote:
> Hello,
> I'm recently having the problem where Asterisk just stops working.
> The console gets disconnected and the process appears to die. I am
&g
ogger reload"
On 7/11/06, Dan Brummer <[EMAIL PROTECTED]> wrote:
> Thank you for the quick response. I assume this change will require
> an Asterisk reload?
>
> Thanks!
>
> -Dan
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTE
with making Transfers
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dan Brummer wrote:
> Asterisk 1.2.9.1 is the version I'm on.
>
> --
> --
> *From:* [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] *On Be
ECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ron
Wellsted
Sent: Tuesday, July 11, 2006 12:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issues with making Transfers
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Dan Brummer wrote:
> Asterisk 1.2.9.
Hello,
Well I had an issue
this morning where the Asterisk process unexpectedly stopped. Below is an
output from the full log:
Jul 18 09:18:51
DEBUG[4892] chan_sip.c: (Provisional) Stopping retransmission (but retaining
packet) on '[EMAIL PROTECTED]'
Request 103: Not Found
Jul 18 09:
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.7.1 Crashing
Dan Brummer wrote:
> Hello,
> Well I had an issue this morning where the Asterisk process
> unexpectedly stopped. Below is an output from the full log:
>
> Jul 18 09:18:51 VERBOSE[
Hello,
Well I was having
transfer issues in 1.2.9.1 so I downgraded to 1.2.7.1. For testing I
installed 1.2.10 on a test server and setup two Polycom SIP phones. Tried
the transfer on this configuration and had the same issues. Here is a log
from the console:
This is how the flow
goes:
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Davies
Sent: Thursday, July 20, 2006 1:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Warm transfer issues in 1.2.10
On 7/19/06, Dan Brummer <[EMAIL PROTECTED]> wrote:
> Hello,
> Well I wa
transfer issues in 1.2.10
On 7/20/06, Dan Brummer <[EMAIL PROTECTED]> wrote:
> Setting canreinvite=no on all the sip peers and gateway made the warm
> transfer work. I'm still noticing SIP entries, is this an
> issue?
>
> == Spawn extension (ANC, 1691, 2) exite
You can try bonding the interfaces into one:
http://www.cyberciti.biz/nixcraft/vivek/blogger/2006/04/linux-bond-or-te
am-multiple-network.php
We have this in our Asterisk setup but recently got recommended to
remove it to combat intermittent issues.
-Dan
-Original Message-
From: [EMAIL
Hello,
I have been given
the task of setting up an extension in asterisk that would dial an outside
number based on the week. For example:
Week 1 - Dial
extension #1
Week 2 - Dial
extension #2
Week 3 - Dial
extension #1
Week 4 - Dial
extension #2
etc...
I found the function
GotoIfTim
Brent, thank you for the reply. I think I asked my question
incorrectly. I'm looking to setup a single extension that changes the
number it goes to each week. If this extension was called on week 1, it
would go to number 1, if called on week 2 then dial number 2, week 3
dial number 1 and so fort
Alexander, thank you for the reply. Do you have more
information on AstDB? Would this require me to switch to something like
RealTime in my extensions.conf or would I be able to use this in my current
dial-plan?
-Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexan
Using the asterisk database and a crontab will get what I want done.
Thank you all for your replies.
-Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Sent: Monday, August 07, 2006 11:21 AM
To: asterisk-users@lists.digium.com
Subject: [asteris
Thanks Alexander, I got it working.
-Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
LopezSent: Monday, August 07, 2006 11:56 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[asterisk-users] By week extension dialing
NO AstDB is al
Hey
guys,
I'm getting the
following message when I start asterisk:
Aug 23 13:42:40 WARNING[29258] loader.c:
/usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol:
__pure_virtual
Aug 23 13:42:40 WARNING[29258] loader.c: Loading
module res_config_mysql.so failed!
I don't know ho
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