I recently switched from BroadVoice to VoicePulse Connect on my Asterisk
box. Too many Meetme quality complaints (whether real or perceived).
I had to make a choice to use IAX2 or SIP with VoicePulse. I first
tried to go with SIP because I already had it working and all of our
devices are SIP.
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Wednesday, March 23, 2005 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoicePulse Issues
Adam Robins wrote:
So, I switched to IAX2
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
Thanks,
Adam
The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally privileged.
This transmission is
I think that the best solution I found is to use a standard ATX power
supply externally and jumper/switch the green and black connectors on
the 20-pin Molex connector to simulate the case/motherboard power
switch. I can then snake in the output power cable with the 4-pin
connector to the TDM400P.
Hello,
I've been using VoicePulse Connect with Asterisk for several weeks now.
From time to time, I've had both inbound and outbound calls drop in mid
conversation. Following are the pertinent portions of my iax.conf.
Note, that most settings are default. Has anyone put together an
iax.conf
They can do PoE with the additional cable.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Trauring
Sent: Tuesday, April 05, 2005 9:51 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users]
I am issuing an AGI
call in extensions.conf as follows:
exten =
2000,1,Answer
exten =
2000,n,AGI(script.pl)
exten =
2000,n,Hangup
Asterisk perl is
installed. "script.pl" is a valid perl script in
/var/lib/asterisk/agi-bin
Output
is:
-- Executing
Answer("SIP/2034-e908", "") in new
On Tuesday 05 April 2005 3:24 pm, Adam Robins wrote:
I am issuing an AGI call in extensions.conf as follows:
exten = 2000,1,Answer
exten = 2000,n,AGI(script.pl)
exten = 2000,n,Hangup
Asterisk perl is installed. script.pl is a valid perl script in
/var/lib/asterisk/agi-bin
Output
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten = _2XXX,Dial(SIP/${EXTEN})
;
exten = i,1,Playback,invalid
I would like to apply the app_dial macro patch referenced in:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=2905
To my stable version of Asterisk:
Asterisk CVS-v1-0-12/21/04-14:14:46 built by [EMAIL PROTECTED] on a i686
running Linux.
Mantis has 5 attached patch files. It looks
This means that you do not have the version of sip.c that
the patch is looking for.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
ShawSent: Tuesday, January 18, 2005 11:55 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject:
[Asterisk-Users]
I just installed 1.0.4. When I do a show version, it still says:
Asterisk CVS-v1-0-12/21/04-14:14:46 built by [EMAIL PROTECTED] on a i686
running Linux
Which is exactly what is said before the upgrade. Is this right?
-Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Last week, when
1.0.4 was released, I obtained the latest stable version from CVS and applied
it. Today, upon hearing that 1.0.5 was out, I did it again. When I
go to the CLI and do "show version", it shows me the same information it did
before the upgrade:
Connected to Asterisk
Just tried it. Show version still shows:
Connected to Asterisk CVS-v1-0-12/21/04-14:14:46
Which is exactly what it said prior to the upgrade.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, January 24, 2005 10:31 PM
And I just checked the ChangeLog file in /usr/src/asterisk and it
show 1.0.5
-Original Message-
From: Adam Robins
Sent: Tuesday, January 25, 2005 10:13 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Updating Asterisk
1. I wiped out
-Commercial Discussion
Subject: Re: [Asterisk-Users] Updating Asterisk
Adam Robins wrote:
1. I wiped out the /usr/src/asterisk directory structure 2. I
followed the instructions below for re-downloading, installing and
restarting Asterisk 3. The Asterisk module in /usr/sbin/asterisk
reflects
I am
looking at running Asterisk on a Dell PowerEdge 1850 server.
My
choices for expansion slots are:
PCI-X riser
with 1 x 64-bit/133MHz and 1 x 64-bit/100MHz
PCI Express
riser with 2 PCI Express slots, one x4 lane and one x8
lane
Will a Digium
TDM400P work in either of these?
: Tuesday, January 25, 2005 1:01 PM
To: Adam Robins
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Updating Asterisk
And you are sure that the old asterisk processes is completely dead?
have you verified this with ps ax? If ZAP resources are tied up
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM400P Dell 1850 Server
Adam Robins wrote:
I am looking at running Asterisk on a Dell PowerEdge 1850 server.
My choices for expansion slots are:
BulletPCI-X riser with 1 x 64-bit/133MHz and 1 x 64-bit/100MHz
] Updating Asterisk
Adam Robins wrote:
1. I rebooted the server. Still NG
2. I manually deleted the asterisk executable in /usr/sbin/. I then
did a make clean, make, make install. The executable was replaced,
but STILL shows version 1-0 12/21/04.
delete the .version file in the Asterisk
1. I wiped out the /usr/src/asterisk directory structure
2. I followed the instructions below for re-downloading, installing
and restarting Asterisk
3. The Asterisk module in /usr/sbin/asterisk reflects the new
date/time
Still shows version 1-0 12/21/2004.
I can not find a .version file in
Hello,
I've set up a dial plan so that outside callers hear a Welcome message which
asks them to enter an extension or press * to dial by name. This works great.
I also want to allow a remote employee to interrupt the message by pressing #,
which will direct them to voicemail.
The issue I am
I would like to allow my remote users to dial in from their homes,
cells, etc., and instruct Asterisk to forward calls made to their office
extension to a number of their choosing. The wiki entry on Asterisk
call forwarding shows how to do this. For security purposes, I would
like to front-end
I've written a macro that allows users to dynamically change their call
forwarding destination. The purpose is to set up a follow me process
where a user can get calls on their cell, at home, etc., based on the
forwarding number they enter. Using the CFIM database, I have the setup
portion
Works beautifully! Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Friday, February 04, 2005 10:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Call forwarding
Adam Robins
Here is the dial plan. Uses CVS-HEAD features for call screening. This
is a pre-production system. We will be replacing our existing Comdial
within the next few weeks.
;=
; extensions.conf file
I'm trying to develop a company phone list accessible via the
minibrowser feature on the phone.
The pertinent section of ipmid.cfg is as follows:
microbrowser mb.proxy=
idleDisplay
mb.idleDisplay.home=http://server/polycom/index.html;
mb.idleDisplay.refresh=300/
main
Yes.
http://lists.digium.com/pipermail/asterisk-users/2005-February/087538.ht
ml
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Wednesday, March 02, 2005 2:56 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject:
Until recently, I was using Broadvoice for my in/out calling thru
Asterisk. I was extremely pleased to see that Broadvoice was actually
passing the callerid info (number and text) that I had set up on each
device in my SIP.CONF file. I had PSTN users tell me that they were
actually seeing name
http://www.freedomphones.net/polycom/files/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Sent: Sunday, December 05, 2004 4:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Polycom IP500
Does anyone
I just set up a
Polycom 500 on *. Every few calls I make, the call connects and the
receiving party can hear me (thru Broadvoice), but I still get ringing on my
end, as if they never picked up. * logs look just fine. Does any one
have any suggestions? Thanks.
Adam S. RobinsExecutive
As I am not a Polycom dealer, I cannot download the software from their
site. Any alternative locations you know of? Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tor Setane
Sent: Thursday, December 09, 2004 7:47 AM
To: Asterisk Users Mailing
Could someone please
direct me (via personal email) to a provider with good prices on Polycom
Soundpoint IP 500's with POE cables? I need 14 of
them.
Thanks,
Adam
Adam S. RobinsExecutive Vice President CIO
PHARMACENTRA,LLP 5901B Peachtree Dunwoody
Road, Suite 380Atlanta, GA 30328
I am attempting to
update my Asterisk installation from 1.0 to the latest stable version.
When I use CVS checkout, I am receiving the following messages on
chan_sip.c:
RCS file:
/usr/cvsroot/asterisk/channels/chan_sip.c,vretrieving revision
1.510.2.25retrieving revision 1.510.2.27Merging
!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Cotton
Sent: Monday, December 20, 2004 10:49 AM
To: Asterisk List
Subject: Re: [Asterisk-Users] Updating Asterisk
On Mon, 2004-12-20 at 10:30 -0500, Adam Robins wrote:
I am attempting to update my Asterisk
I unloaded essential?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Monday, December 20, 2004 11:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Updating Asterisk
Deleting chan_sip.c
Has anyone else been
experiencing dropped calls with Broadvoice? I've lost every single call
during the past 2 days.
Adam S. RobinsExecutive Vice President CIO
PHARMACENTRA,LLP 5901B Peachtree Dunwoody
Road, Suite 380Atlanta, GA 30328
Office: 770-395-0088 x34Fax:
770-395-0989Mobile:
I signed up for an
IAXTEL account and have been trying, unsuccessfully, to get it working. In
IAX.CONF I have:
[iaxtel_out]type=peerhost=iaxtel.comusername=USERNAMEsecret=SECRETauth=rsainkeys=iaxtel
[iaxtel]type=friendcontext=incominghost=iaxtel.comauth=rsainkeys=iaxtel
However, when I
]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: den 21 december 2004 22:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAXTEL Configuration
I signed up for an IAXTEL account and have been trying,
unsuccessfully, to get it working. In IAX.CONF
I have done this successfully with Asterisk 1.07 and Zaptel 1.09 and
1.2.1 for the same reasons as you.
However, if you ever need to go recompile Asterisk, then you will first
need to recompile the old Zaptel, compile Asterisk and the new Zaptel
again.
-Original Message-
From: [EMAIL
What would cause the message:
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 up
== Primary D-Channel on span 1 up
To keep appearing on CLI about once every second?
If I do a zap show status:
Description Alarms IRQbpviol
CRC4
We have (had) two identical Asterisk servers for our outbound call
center. Both were running Linux 2.4 kernel, Asterisk 1.0.7, Libpri
1.0.7 and Zaptel 1.2.1. Each server has a TE410P card with two PRIs.
Last week, we upgraded one of them to Asterisk 1.2.4, Zaptel 1.2.3,
Libpri 1.2.2.
The
After many days of playing with the new jitterbuffer and trunking options for
IAX2, I have finally received almost acceptable quality. I am receiving 5-8
complaints a day of calls breaking up from both the customer and agent sides.
What I have discovered is that in most of these cases, the
: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning
Adam Robins wrote:
Hi Adam
After many days of playing with the new jitterbuffer and trunking
options for IAX2, I have finally received almost acceptable quality. I
am receiving 5-8 complaints a day of calls breaking up from both
] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning
Adam Robins wrote:
Hi Adam
After many days of playing with the new jitterbuffer and trunking
options
Message-From: Adam Robins [mailto:[EMAIL PROTECTED]]Sent:
Monday, February 20, 2006 14:43To: Asterisk Users Mailing List -
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2
New JitterbufferTuningI have now set the "resyncthreshold" to
-1, to turn it off. I ha
Jitterbuffer
Tuning
Adam Robins wrote:
This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected via the same
network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes
|reliabilityRegards,Jesus-Original
Message-From: Adam Robins [mailto:[EMAIL PROTECTED]]Sent:
Monday, February 20, 2006 14:43To: Asterisk Users Mailing List -
Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2
New JitterbufferTuningI have now set the "resyncthreshold&qu
implementation. none of which made any difference. I
also tried with and without trunking enabled.
SIP is running much more acceptably now.
Adam Robins wrote:
After many days of playing with the new jitterbuffer and trunking
options for IAX2, I have finally received almost acceptable quality. I
am
IAX2 New Jitterbuffer
Tuning
Adam Robins ha scritto:
Thanks, but we already have the TOS bits set to 0xB8, which matches
the QoS settings in our switches and routers.
This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected
Jitterbuffer
Tuning
On Feb 23, 2006, at 4:58 AM, Adam Robins wrote:
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak
issues.
??? What do you think this is windows 95??? I had a problem like that I
would
I was using IAX2 with ILBC and no trunking. I also set the
resyncthreshold=-1 to turn it off. Still had major jitter problems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
Adamson
Sent: Thursday, February 23, 2006 6:44 PM
To: Asterisk Users
I'm trying to
compile Asterisk 1.2.4 on a Redhat Enterprise system, kernel
2.4.21-27.0.2.ELsmp
I'm getting the
following errors and then the compile stops.
/usr/kerberos/lib/libgssapi_krb5.so.2: undefined
reference to `add_error_table'/usr/kerberos/lib/libgssapi_krb5.so.2:
undefined
Try Allison at theivrvoice.com. She is the voice of Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, March 08, 2006 11:06 PM
To: Commercial and Business-Oriented Asterisk Discussion
Cc: Asterisk Users Mailing
Can someone tell me what I'm doing wrong here? I'm trying this from the
command prompt.
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 000 -o
/tmp/1141915933.wav
rateconv: failed to convert from 16000 to 0
doing v
#
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
No, I did not install Festival, but I saw that the text2wave module is
in the usr/bin directory.
I'm running RH Ent 2.4 kernel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert La
Ferla
Sent: Thursday, March 09, 2006 10:17 AM
To:
I figured it out. It should read:
# echo Hello World | /usr/bin/text2wave -scale 1.5 -F 8000 -o
/tmp/1141915933.wav
The 8 was missing in front of the 000'.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, March 09, 2006 12:04
I have two Asterisk 1.2 boxes across a WAN. Calls between them are sent
via SIP g729a. The issue is that the original calleridnum is
overwritten by the value of the fromuser parameter in sip.conf on the
originating server. Is there any way to preserve the original
calleridnum value?
Nevermind, I just answered my own question. Used username instead of
fromuser.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Friday, August 15, 2008 3:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP Callerid
We are running Asterisk on native CentOS. We then install VMWare on
CentOS with Windows 2003 in the VMWare partition for AD services. We
have 50+ users in a call center environment with no issues.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
/msg180825.html
Good luck,
François.
Adam Robins wrote:
We are running Asterisk on native CentOS. We then install VMWare on
CentOS with Windows 2003 in the VMWare partition for AD services. We
have 50+ users in a call center environment with no issues.
-Original Message-
From
or slower then normal on multi core systems and on systems with
power stepping.
In my case i'm getting those timing issues on two dual core amd machines and
i'm not getting timing issues on three dual-core intel machines.
--
Cosmin Prund
-Original Message-
From: Adam Robins [EMAIL
I am using a Polycom SIP phone (ext 2042) to call an analog phone
connected via an IAXY (ext 2120). The analog phone rings, and when I
answer, I can hear the person speaking on the SIP phone, but they cannot
hear me. However, if I originate the call from the analog phone to the
SIP phone, it
in the iax.conf for the IAXY device. It doesn't support
it.
Regards,
Steve
Adam Robins wrote:
I am using a Polycom SIP phone (ext 2042) to call an analog phone
connected via an IAXY (ext 2120). The analog phone rings, and when I
answer, I can hear the person speaking on the SIP phone, but they
cannot
I have an application where a caller leaves a voicemail message and then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature in voicemail.conf.
My script has no knowledge of the name of the
, Feb 05, 2009 at 05:04:11PM -0500, Adam Robins wrote:
I have an application where a caller leaves a voicemail message and
then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature
I have five Asterisk servers running 1.2.14, and am planning to upgrade
to 1.4 this weekend. In preparation, to use the most efficient g729
codec, I am running the new benchg729 program. It works great on two
systems, but on the other three it says it cannot locate a valid g729
license. I have
: [asterisk-users] benchg729 - no valid g729 license
Adam Robins wrote:
I have five Asterisk servers running 1.2.14, and am planning to
upgrade
to 1.4 this weekend. In preparation, to use the most efficient g729
codec, I am running the new benchg729 program. It works great on two
systems
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins arob...@pharmacentra.com
wrote:
I am switching from IAX2 to SIP for my inter-Asterisk transport due to
assorted quality issues following the 1.2.4 upgrade.
On the server that SENDS the call, I have the following in SIP.CONF:
[192.168.1.2_OB]
type
Title: Re: [Asterisk-Users] editing ring time
I am using the auto-dial-out
feature to play recordings. I create the call files, place them in the
outgoing directory and off they go.
The problem is that the number I am dialing
does not get stored in CDR. One suggestion was to put this
Title: [Asterisk-Users] Running Asterisk on a Dell PowerEdge 2850 ServerRe: Dell Hardware
It's Digium, not Dell.
I have two identical Dell 1850s, each with the allegedly offensive
built-in E100 Ethernet ports. I placed a TE410P card in each. One
worked great, the other would not
Title: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects
The Changelog for Zaptel 1.0.9.1 has
only one fix listed:
-- continue fxo operation after the magical 25
days
Could someone please translate this highly technical
explanation into something more meaningful?I already
Basically, we have a multi-site Asterisk call center application we tried
to bring up last week.When the agent places an outbound call ( or
takes an inbound call), the agent can hear the customer just fine, but the
customer has issues hearing the agent. This does not happen every time and
I have configured my phone following your example, but it does not work
for me. Can you also please share your sip.cfg settings?
Thanks,
Adam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Tuesday, August 02, 2005 3:44 PM
To:
This is in the -app.log file:
0804194926|sip |4|00|Registration failed User: 1800, Error Code:403
Forbidden
Where '1800' is the extension I am attempting to register. SIP.conf is
set up properly, and there is nothing in Asterisk showing a denied
registration attempt.
Could it be because
information
in sip.conf, so it was always going to one server.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adam Robins
Sent: Thursday, August 04, 2005 2:41 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE
Can anyone out there please tell me what ports Speex uses? I want to
set up QoS on switches but I can't seem to find this information
anywhere.
The contents of this email message and any attachments are confidential and are
intended solely for addressee. The information may also be legally
QoS
speex is a codec.
it's not a network protocol or a service.
you need to be looking to be providing QOS for RTP data, over which the
speex encoded data is sent.
cheers,
Mark
On 8/8/05, Adam Robins [EMAIL PROTECTED] wrote:
Can anyone out there please tell me what ports Speex uses? I want
I drop every 3-4 call with VoicePulse Connect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean
Kennedy
Sent: Wednesday, April 20, 2005 6:21 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Trevor
Harrison
Sent: Thursday, April 21, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] BYOD provider other than broadvoice
On 4/21/05, Adam Robins [EMAIL PROTECTED] wrote:
I drop every 3-4 call
Why would you use gateways and PRI's when several of the major carriers
(ATT, Global Crossing, etc.) also have products that can interface
directly with SIP for the same per minute cost?
We have a multisite Asterisk call center application and are routing all
calls over private VPN to one central
Hello,
Being totally fed up with the lack of quality and reliability from both
VoicePulse and BroadVoice,
We are switching to a direct IP connection to Global Crossing. We've
installed a local point-to-point T1 into their CO, and they will
give/take SIP g729a directly and act as the gateway for
Hello,
We are attempting to install a TDM400P card in a Dell Poweredge 1850
server. We are running Red Hat Linux kernel 2.4.21-15.
We can compile zaptel and asterisk without incident. When we try to
modprobe zaptel, it produces pages of:
/lib/modules/2.4.21-15.EL/misc/zaptel.o: Relocation
If anyone out there is running Asterisk with Zaptel and a TDM400P card
on a Dell Poweredge 1850 server, please let me know what OS and kernel
version you are running.
I keep getting errors when modprobing zaptel and am running out of
possibilities, other than motherboard incompatibility.
Thanks,
Title: Message
Try DIAX. Works just fine!
http://www.laser.com/dante/diax/diax.html
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeromy
GrimmettSent: Monday, May 23, 2005 12:09 PMTo: 'Asterisk
Users Mailing List - Non-Commercial Discussion'Subject:
[Asterisk-Users]
I have Broadvoice set up with dtmfmode=inband. All was working just
fine. Suddenly today I noticed that if someone calls in to my Asterisk
box thru the Broadvoice number, the system no longer recognizes the DTMF
tones. I also tried rfc2833 and info. Any ideas?
Thanks,
Adam
The contents of
Nevermind. It is now working. Must be Broadvoice. Surprise!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Wednesday, June 15, 2005 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
I am having this exact problem today.
I have two Dell 1850's running Asterisk 1.07. Both had TDM400P cards
running just fine. I replaced the TDM400P in both machines with TE410P.
Server One works just fine with just a new modprobe. Server 2 does not
even see the card upon reboot.
Swapped
I installed a new Digium TDM400P in a Dell 1750 server. The system
would not recognize the card. I took the FXS modules off of it and put
them on another TDM400P card I already had. Old card worked fine with
new modules. Old card is Rev. H and new card is Rev. I. Anyone else
having any issues
I installed a TDM400P with 4 FXO modules. Before moving all of my
office phone lines to it, I decided to move only one for testing. I
plugged it into port 4 on the card.
In zaptel.conf I have:
fxsks=1-4
And zapata.conf:
context=incoming
signalling=fxs_ks
busydetect=yes
callprogress=no
I guess that my definition of first available trunk (either forward or
backward) differs from Digium. I would think that the card should know
which ports had an electical signal attached.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Kawakami
I have one POTS line going into a TDM400P. Here in Atlanta, we have 10
digit local dialing. I launch a call Zap/1/7705551212 and it goes
thru just fine. The next time I try it, without any modifications, I
get a Bell recording telling me that I must dial the area code and seven
digit number
Hello,
I saw some conversation about this in the archives, but nothing
definitive.
If a call comes in over a CO line via the TDM400P, the Comedian Mail
recording volume is so low it's inaudible. Calls coming in via SIP or
IAX do not have this problem.
Does anyone have any information on this
I have a user who goes into Comedian Mail for the first time and goes
thru the initial setup, changes password, records name, etc. Problem is
that every time he calls in, it thinks that it's his first time and
keeps reprompting him. His password change is reflected in
voicemail.conf. Others do
I was able to raise the volume from inaudible to acceptable by
increasing the RxGain in zapata.conf by 5db. I'd rather not go the
uncomressed wav route, as it will chew up storage in my email system.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich
I am using
rawplayer:
default =
custom:/var/lib/asterisk/mohmp3/raw,usr/bin/rawplayer
as in: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it
However, the music is too loud.
Without having to rerecord it, is there a parameter like quietmp3 that can be
used with the above to
No, I am not using mpg123 at all.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Giordano
GrandisSent: Thursday, June 30, 2005 9:35 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: R:
[Asterisk-Users] Music oh hold
Did u installed mpg123 0.59r ?
I do not want to use the default key of '#' for call transfer, because
as we all know, it interferes with many IVRs that require # as a
termination character. I modified features.conf and added:
[featuremap]
atxfer = **
The double-star now works great. If I press it while on a call, I go
into
We have three remote call center Asterisk servers communicating with two
central Asterisk boxes over a private IP-VPN with QoS. All systems were
running Asterisk 1.0.7 communicating via IAX2 with little or no quality
issues at all.
Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was
Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 10:21 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problem with chan_iax.c implimentation
causesbadaudio?
On Tuesday 21 March 2006 09:47, Adam Robins wrote:
We have three remote call center Asterisk servers communicating
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