I currently use asterisk version 1.0.10 with AMP 1.0.010,
our setup is APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk
server. When I use Asterisk version 10.0.10 everything works
perfectly, however when I use 1.2.4 I lose the ability to receive calls from the
PSTN. All I get is the
I have included two files, one from asterisk 1.0.10 and one from 1.2.4.
Thanks Olle
Chris Modesitt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc
I have included two files, one from asterisk 1.0.10 and one from 1.2.4.
Thanks Olle
Chris Modesitt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson
Sent: Friday, February 24, 2006 7:53 AM
To: Asterisk Developers Mailing List
Cc
: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose
: [asterisk-dev] Possible Bug in SIP Stack.
Chris Modesitt wrote:
I currently use asterisk version 1.0.10 with AMP 1.0.010, our setup is
APX 8000 - Interaction SIP Proxy 3.0.013 - asterisk server. When I
use Asterisk version 10.0.10 everything works perfectly, however when I
use 1.2.4 I lose
I am testing Asterisk Beta 2 in our lab and I have found a
possible bug, the box is setup with a T410P. Call path looks like this:
T1 PRI Asterisk
Server(1.2.0beta2) SIP
Interaction Proxy Asterisk
Server (1.0.9) SIP Phone.
This works perfectly.
SIP Phone Asterisk
Server
I am attempting to unify how numbers come to me from a
specific T1, this T1 acts as an ingress for about 4000 DIDS. About 98% of
those DIDS come in as a 10-digit DNIS, what I would like to do is have asterisk
log when a number comes in 7 or 11 digit so I can contact my upstream provider
Okay here is a quick breakdown of my situation; I have an
asterisk server with AMP installed. Amp stores all of the CDRS in a mysql
database and comes with a nifty web based reporting tool that has worked well
for me. The problem I am running into is that my mysql database is
nearing 2
Asterisk 1.09 works well with it however CVS head has major
issues, wondering if anybody else has seen any RTP like issues and what they
did to work around it.
Thanks
Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --
On Thursday 08 September 2005 10:26, Simone Cittadini wrote:
Is it true ?
My boss is just asking me if it is possible to stuck 4* TE411P in a
single server, for a total of 480 lines, someone can assure me it is
possible/impossible (manageable/unmanageable) from real-life experience ?
Don't
Man I almost passed from laughing when I read this, that is the best
description of bad echo I have ever heard:)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Sunday, July 24, 2005 12:28 PM
To: Asterisk Users Mailing List -
Is there a way to match the last 7 digits of an extension?
So that 1008014445454
8014445454
4445454
Would all match?
I have looked at extension matching and I cant figure
out how to do thisJ
Thanks in advance!
Chris
___
-Commercial Discussion
Subject: RE: [Asterisk-Users]
Extension Matching.
Try _.4445454
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Modesitt
Sent: Wednesday, June 29, 2005
4:17 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users
29, 2005
2:30 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
Extension Matching.
Try _.4445454
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Modesitt
Sent: Wednesday, June 29, 2005
4:17 PM
: RE: [Asterisk-Users]
Extension Matching.
What are you trying to do???
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Modesitt
Sent: Wednesday, June 29, 2005
4:54 PM
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: RE
I know this is slight OT however I have decided that I need to but in some
echo cancellers on my PRI's. I was wondering if anybody else was using a
hardware echo canceller capable of 24 T1's, how well it works and an
approximate price range:)
Thanks
Chris
Andrew, this is very similar to the way we do it, however we also sell a
dial up service that will not disconnect the user but the pricing reflects
1/24 the cost of a T1 so about 35.00$ a month. Since our network is powered
by three DS3's @ $3500.00/month we still make a good profit.
Chris
Hi all,
Did I make my issue clear? Can any one give me a big hand?
Many thanks.
Newbie
On Apr 5, 2005 12:59 AM, VoIP Newbie [EMAIL PROTECTED] wrote:
Hi all,
When I made calls from SIP phones through a analog PSTN gateway to
PSTN phones, I could hear rings twice on my SIP phones. From
I have an Asterisk server that I am running PRI NI2 NET, I
would like to pass the caller-id name to the customers PBX. From what I
read I am going to checkout libpri from CVSHEAD, which should contain most of
Matts patches for sending caller-id name. My question is whether
or not I need to
Eric Wrote:
Digium has a hardware echo can?
Not shipping, according to their online store.
Crap!, I spend all my time reading emails from this list, now I have to
check Digium's online store twice a day so I can get my hands on one of
those cards!!
Chris.
Is it possible to run NETHDLC and PRI signaling on the same
4 port T405P card?
This is what I get when I try to start asterisk:
== Parsing '/etc/asterisk/zapata.conf': Found
Mar 24 10:48:53 ERROR[2368]: chan_zap.c:9435 setup_zap:
Unknown signalling method 'pri_cpe'
Mar 24 10:48:53
Here is a sample of our zaptel.conf config as it was handed to me (I
inherited this Asterisk project, btw). These configs are likely a train
wreck, so if anybody could possible either generate a config that would
work, or explain a somewhat laymens terms how I can go about making a
good
Dose anybody on the list use the Vovida Load Balancer? If
so how stable is it? How many simultaneous calls can it handle? I am looking
to load balance two local PSTN gateways each with a 4 port 410P T1 card. I
currently have 1 gateway that passes 11 to 12 thousand calls a day and I want
to
Do yourself a favor and get a Sipura SPA-2100 - much easier to
configure and the quality is better than the Mediatrix unit.
This was true with the earlier SIP and H323 software, however if you get
their latest (11.70) it seems to be one of the better IADS out there. I do
agree with you though
Problem: Zap Channels Disappear @ random intervals. (Channels
have disappeared on both gateways twice this week).
My Setup:
I have 2 Dell 1850 Power Edge Servers Configured as.
P4 2.8Ghz
512 ECC Memory
SCSI Array (2 Drives Mirror)
Configuration is really simple the boxes are
= 2181,11,Dial(SIP/${OPERATOR},45,m)
exten = 2181,12,Hangup
This example lights the lamp on a 7914 sidecar to show there is a call
parked at that extension..
You can just put the parking spots in different contexts to keep them
separate from different groups of users..
Chris Modesitt wrote
Question: I am PBX multi-hosting several customers on one of
my * servers, what the best way to setup call parking to prevent company A from
picking up Company Bs parked calls ?
Any basic examples would be greatly appreciated.
Thanks
Chris.
Lots.
juse valetparking. I can't give you an example right now, since I'm
working on implementing it. When I'm done I'll report
On Tue, 22 Feb 2005 11:51:45 -0700, Chris Modesitt [EMAIL PROTECTED]
wrote:
Question: I am PBX multi-hosting several customers on one of my * servers,
what the best
,Hangup
This example lights the lamp on a 7914 sidecar to show there is a call
parked at that extension..
You can just put the parking spots in different contexts to keep them
separate from different groups of users..
Chris Modesitt wrote:
Thank you any feedback would be greatly appreciated
I am terminating a PRI, NI2 signaling into a PBX (My companys
PBX not the PSTN) from an Asterisk server. Caller ID number appears to be
transmitted caller id name is not being transmitted is their a compile time
flag on libpri that I need to un-comment to enable this feature or is this is
via PRI NI2
On Fri, Feb 18, 2005 at 11:06:16AM -0700, Chris Modesitt wrote:
I am terminating a PRI, NI2 signaling into a PBX (My company's PBX not the
PSTN) from an Asterisk server. Caller ID number appears to be transmitted
caller id name is not being transmitted is their a compile time flag
Is it possible to disable a reinvite on a specific call in
the dial plan? Any help would be greatly appreciatedJ
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
ahh..american arrogance
Excuse me but I am an American, and while I have rightly been accused of
many things most of us aren't arrogant. You must have a superiority complex
to be making comments like that to a list of over 800 people.
Chris.
___
aussies who would prefer to sink piss (beer) rather
than expend our energies in more significant areas) but this has to be
said.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Modesitt
Sent: Wednesday, January 12, 2005 1:06 PM
I have 3 DL380 G4's in production, only difference that I can tell is that I
am running a 2.4.22 kernel.
Chris
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 03, 2005 11:35 AM
To: asterisk-users@lists.digium.com
I have about 40 of these in production with Asterisk, send me an email off
list with your sip.conf file and you extensions.conf file and I will help:)
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deepak
Malhotra
Sent: Wednesday,
You are definitely not getting calling name over facility information
element
from your telco. I do not see calling name information anywhere in that
dump.
Unless that's not a complete dump, or there's some other problem, I'd talk
to your telco about it.
Matthew Fredrickson
Your right, after
Okay, now I am really confused. I have two PRIs
coming in from two different Carriers (QWEST and ELI), both of them are
supposed to be setup to pass name and number on incoming calls. Problem
that I am having is that I am not receiving inbound caller id name on either PRI,
the only thing
Subject: Re: [Asterisk-Users] No Caller ID Name PRI NI2.
Chris Modesitt wrote:
Okay, now I am really confused. I have two PRI's coming in from two
different Carriers (QWEST and ELI), both of them are supposed to be setup
to
pass name and number on incoming calls. Problem that I am having
.
On Wed, Dec 15, 2004 at 10:52:07AM -0700, Chris Modesitt wrote:
Okay, now I am really confused. I have two PRI's coming in from two
different Carriers (QWEST and ELI), both of them are supposed to be setup
to
pass name and number on incoming calls. Problem that I am having is that
I
am
There was no SETUP message in that debug. You need to start the debug
before you place your inbound call from the telco so that I can see
the SETUP message that usually contains the IE that has this information
in it.
Matthew Fredrickson
Sorry Had the wrong debug options on, this shows all of
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Modesitt
Sent: Wednesday, December 15, 2004 12:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Caller ID Name PRI NI2.
Included is my debug.
-Original Message-
From: [EMAIL PROTECTED
Mark this config is a little nasty but it is how I got it to work.
zaptel.conf
span=1,0,0,esf,b8zs
em=1-24
zapata.conf
context=OutBound
signalling=em_w
immediate=yes
group=1
channel = 1-24
[OutBound]
exten = s,1,DigitTimeout(0)
exten = s,2,ResponseTimeout,10
exten = s,3,Playtones(350+440)
Do you want to stop a call in progress, or stop the call from ringing the
far end device and move to the next priority?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of michelle li
Sent: Wednesday, November 24, 2004 2:57 PM
To: [EMAIL PROTECTED]
Subject:
Is there a way to check if an extension exists?
This is the problem I ran into, I have exceeded the number
of extensions you can attempt to match in one pass (1500+ Extensions). I am
hoping that someone has discovered a clever way of checking if an extension
exists in a particular
On Wed, 17 Nov 2004, Steven Critchfield wrote:
On Tue, 2004-11-16 at 23:34 -0700, Chris Modesitt wrote:
Thanks for your feedback, after I restarted Asterisk the card came up
as
expected. However I am still seeing these WARNINGS when I reload *, to
be
clear I have not made any
Okay, I admit most of my T1 knowledge comes from the Cisco
world so please dont castrate meJ I have
several T1s running point-to-point between our Cooperate office and our
Branch offices, our service provider dose not provide Loop timing on any point
to point T1s. So typically when I would
Is it possible to run multiple signaling types on 1 card
aka, asterisk screams @ me when I try to do this:
Zaptel.conf
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
em=1-24
bchan=25-47
dchan=48
loadzone = us
defaultzone=us
Zapata.conf
context=ToVeracity
signalling=em_w
Critchfield
Sent: Tuesday, November 16, 2004 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T405P Mulitiple Signalling modes on 1 card.
On Tue, 2004-11-16 at 14:36 -0700, Chris Modesitt wrote:
Is it possible to run multiple signaling types on 1 card aka
I installed and brought online a new gateway last night,
calls coming into the system sound great and dont appear to be
generating any errors. However if I send calls out these T1s I get a
tone errors and I hear static popping and clicking on the lines.
Zapata.conf
switchtype=national
Thanks Matt, I will give that a shot tonight and will let
you knowJ
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Of Chris Modesitt
Sent: Thursday, November 11, 2004
12:12 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] No
Inbound CallerID Name Has me Stumped.
Thanks Matt, I will give that a shot tonight and will let
you knowJ
___
Asterisk-Users mailing
My Telco swears that I have Caller ID (Name and
Number) being sent to me over our PRI's (I have called them a half dozen times
to confirm). My gut feeling is that they are lying to me, this is
why.
First I decided to Look into my CDR records, they
all look like this for incomming calls
Is there a way to have the queue branch to +101 on timeout?
Bellow is a snip from my extensions.conf file that shows how
I am joining the queue, currently if I try to join the queue but there are no
agents logged in it will transfer me to voicemail this works as expected.
What I would
I have a question about how Asterisk Parses the Dial Plan.
To create a hunt-group which would be the appropriate dial plan:
[CompanyABC]
exten = 722,1,Dial(SIP/801722,60,r)
exten = 722,102,Dial(SIP/8014361234,60,r)
exten = 722,103,Dial(SIP/8014362345,60,r)
exten =
55 matches
Mail list logo
