I am having a strange issue with setting the incoming caller id on the latest
version of TrixBoxCE. Right now I have it setup with a cross-over T1 cable to
our production Asterisk (1.0.9) box and from the Trixbox we can send and
receive calls just fine. The problem I am having is that if a
List,
I have begun to experience a strange echo problem on our
internal network. The problem starts when User A calls User B,
User A puts User B on hold. User B heres the on hold music.
User A returns and User B has trouble echo. I am using FC1,
Asterisk 1.0.9.
This electronic
List,
I just tried to swap out our 410 for a 411 and we started have
problems with on of our T1's.
Setup:
Span 1 - Dedicated PRI for long distance.
Span 2 - 12 channels fxs_gs outgoing local.
12 Channels em_w incoming DID's.
I didn't have any problems with the PRI. The
List,
I am having trouble with one of our IP600. Every five days or
so, the phone locks up. This is the third 600 I have put in place. I
am running asterisk 1.0.9. Has anyone had this problem with the IP600?
This electronic message transmission, including attachments, is for the
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX
sounds better then the land line.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julio
Arruda
Sent: Friday,
: [Asterisk-Users] IAX over HTTP
Eric Rees wrote:
We have been running IAX through OpenVPN with SSL for 6 months without
any trouble to Las Veags, and we are in Oklahoma. Most of the time,
IAX sounds better then the land line.
Using UDP or using TCP? Might want to confirm by using tcpdump
Could you pass along the information you used to get the Polycom lights
to work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden
Sent: Wednesday, July 20, 2005 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I have googled this and come up empty. Has anyone had any
problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am
getting when I run make.
app_nv_faxdetect.c: In function `nv_detectfax_exec':
app_nv_faxdetect.c:210: error: structure has no member named
`cid'
-Users] NVFaxdetect
What Linux version are you using?
There is an ebuild on Gentoo
--
#Joseph
On Tue, 2005-06-21 at 16:15 -0500, Eric Rees wrote:
I have googled this and come up empty. Has anyone had any problems
compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am
getting when I
I would also donate some bandwidth.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, June 21, 2005 9:34
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
voip-info.org unreliable lately?
I
Correct me if I am wrong. I can remember installing a T1's with a HDSL
unit at the last CO, in which the T1 was delivered to the customer's
prem in two wires. I think they called this fast half-duplex.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Can we get this guy kicked off of the
list.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey Richey
Sent: Thursday, May 19, 2005 1:11
PM
To: asterisk-biz@lists.digium.com;
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] LOOKING
TO HIRE
We
I am having problems with Broadvoice. I am not getting any audio,
either in or out, but the phone will ring. Could someone double check
my config.
[general]
context=default ; Default context for incoming calls
port=5060
bindaddr=0.0.0.0; IP address to bind to
Has anyone seen the error below or knows how to fix this. Every time
this error occurs, I starting getting a 3 second delay on all internal
and external calls and the only why to stop it is to stop and start
asterisk. I am using a TE410 with Asterisk 1.0.7, Zaptel 1.0.7, and
Libpri 1.0.7.
Requirements
On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote:
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB
of memory. This is serving about 75 sip clients, Polycom500's and
600's. We are running into problems with the memory. Asterisk, right
now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7,
Zaptel 1.0.7 with
You need to upgrade these phones to the latest firmware for it to work
well with asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thore
Sent: Sunday, April 03, 2005 3:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
19:33:42 -0600, Eric Rees [EMAIL PROTECTED]
wrote:
I tried leastrecent. I did change the strategy, but didn't make a
difference.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Dennick
Sent: Wednesday, March 30, 2005 6:49 AM
To: 'Asterisk
come into play.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees
Sent: Tuesday, March 29, 2005 9:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ACD queue question
I have a simple 4 person ACD queue
I have a simple 4 person ACD queue using the AgentCallback function. No
matter what strategy I use, anytime someone calls into the queue
asterisk dials the agents in the order that they are listed in the
agents.conf file. This doesn't seem right to me, or am I wrong.
I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the
music on hold does not work.
More Detail:
While I was running asterisk 1.0.5, when someone called into an Polycom
IP500 and was put on hold via the Polycom Hold button, the hold music
would play. After upgrading to 1.0.6 that
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only
one will pick
, 2005 10:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom Auto-Answer
Eric Rees wrote:
I am having a problem with Polycom auto-answer. I have the
auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between
+7940-7960+auto-answer+config. With
a
few small modifications it should work like a champ on the Polycom
phones.
B. J.
-Original Message-
From: Eric Rees [mailto:[EMAIL PROTECTED]
Sent: Tuesday, March 01, 2005 10:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
Use the wiki luke.
-Original Message-
From: Bill Maidment [mailto:[EMAIL PROTECTED]
Sent: Thursday, February 10, 2005 5:08 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk and Fedora Core 3
Hi
Has anyone seen this message trying to install an TDM400.. spurious
8259A interrupt: IRQ7
This error happens after I do a modprobe wctdm and then the system
hangs. I am installing this in an Asus motherboard with a VIA P4M266
chipset.
___
Here is what I have done to get around the call waiting problem.
This is for a Polycom 500. This is kind of a pain, but it works.
Exten.conf
exten = 1051,1,Dial(SIP/1051,20,tTr)
exten = 1051,2,Voicemail(u${EXTEN})
exten = 1051,102,Dial(SIP/1051b,20,tTr)
exten = 1051,103,Dial(SIP/1051c,20,tTr)
: Polycom and call waiting again..
incominglimit is deprecated. It will be EOL'd.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+incominglimit
On Thu, 27 Jan 2005 10:21:25 -0600, Eric Rees [EMAIL PROTECTED]
wrote:
Here is what I have done to get around the call waiting problem
Has anyone been able to find a way to disable call-waiting
on Polycom phones?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
I have a channelized T1 with the first 12 channels set to FXS_GS. In my
extension.conf file, I have a variable in [globals] DIALOUT=ZAP/g1. The
problem is when I try to make an outbound call, the console tells me
that everything is busy, but is I change the variable to DAILOUT=ZAP/1,
I can dial
List:
I already have asterisks up and running on a PRI, but where we
are moving we cannot get a PRI so we are going to get T1. My question
is: We are going to us EM Wink for signaling with DTMF and caller id.
The channels are going to be setup like this, 12 channels for 2-way and
12
I don't know if this is possible, so I will let the collective decide.
Here is what I would like to do.
BossA calls BossB, BossB's admin assistant sees the call from BossA on
her phone. CallerID would look something like: BossA to BossB : on her
phone. And she would be able to pick if BossB
, December 06, 2004 1:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Is this possible
Eric Rees wrote:
I don't know if this is possible, so I will let the collective decide.
Here is what I would like to do.
BossA calls BossB, BossB's admin assistant
I have a similar setup, and when get the same thing displayed on our
6408D+ phones.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miedema,
Bud
Sent: Friday, December 03, 2004 1:40 PM
To: '[EMAIL PROTECTED]'
Subject: [Asterisk-Users] Umlaut over I on
I don't know if the is possible on not. I would like to know the
easiest way to ring all extensions in the sip.conf file for intercoms.
I have phone to phone intercom working.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Configured Extension
exten = 4000,1,Dial(SIP/3001SIP/3002SIP/3003...on and on, 30,
t)
Matthew
- Original Message -
From: Eric Rees [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004 8:56 AM
Subject: [Asterisk
We are in the process of testing * for company wide
deployment. We are using Polycom 300 phones, the only problem that I am
running into is when I call an 800 number that has an IVR I get disconnected
after about 60 seconds. Here are the logs from asterisk. I am not
sure if this is a
: [Asterisk-Users] Ring all Configured Extension
Why are you afraid of that suggestion?
Matthew
- Original Message -
From: Eric Rees [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Thursday, December 02, 2004 10:56 AM
Subject: RE
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Problem or Polycom Problem
On Fri, 2004-12-03 at 05:35, Eric Rees wrote:
We are in the process of testing * for company wide deployment. We
are using Polycom 300 phones, the only problem that I am running
or Polycom Problem
Eric Rees wrote:
Thanks for you suggestion, but the last time I tried this I was
talking
to a person and it cut me off. But I will try what you suggested.
If you have busydetecr or callprogress in zapata.conf, turn them off.
--Eric
--
I am seeking part or full time employment
I have spandsp installed and working, but when it emails using Scotts
mailfax, the attachment is a dat file. I tried to rename the file to
.tiff or .pdf, but it will not open. In the /var/spool/asterisk/fax
folder, that faxes are there as tiffs, and I can open those without any
trouble. The
I was finaly able to patch the Makefile in the apps dir. I used 2pre4
version.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Seth
Remington
Sent: Tuesday, November 23, 2004 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
When I try to patch the Makefile for asterisk with the
Apps_makefile.patch from Spandsp I get the following error.
patching file Makefile
Hunk #1 FAILED at 47.
Hunk #2 FAILED at 76.
2 out of 2 hunks FAILED
Has anybody seen this.
___
Asterisk-Users
, 2004 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Patching asterisk for spandsp
On Mon, 2004-11-22 at 14:38, Eric Rees wrote:
When I try to patch the Makefile for asterisk with the
Apps_makefile.patch from Spandsp I get the following error
I realized that after this first two times I tried that, but I still
will not patch. I tried to path the file manually. This is where make
clean dies at.
app_rxfax.so : app_rxfax.o
$(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
app_txfax.so : app_txfax.o
$(CC) $(SOLINK) -o $@ $
We are having a problem with the Polycom 300. For some reason, it will
deregister and not register back. I have looked the config files for
the Polycom, but since it is all XML I might be missing something.
Thanks.
___
Asterisk-Users mailing list
Polycom phones are nice and are about half the cost of Cisco phone.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kavit
Munshi
Sent: Thursday, November 18, 2004 9:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
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