I'm working that angle. I tried to use Dameware to get into her router via her
home PC, but the screens weren't drawing correctly. I'll need to try LogmeIn.
Also the IP address she read me directly off the phone is dubious. I cant ping
it nor can I bring up the web interface.
To be continue
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone
Un-top-posting and trimming cruft...
On Fri, 14 Oct 2011, Adam Robins wrote:
> Thanks I will do that. The user is remote, so I must first RDP into
&g
address of
the phone. If you can do that, perhaps something there will be of use to you.
From:
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
[mailto:asterisk-users-boun...@lists.digium.com]<mailto:[mailto:asterisk-users-boun...@lists.digium.c
ug and Eric said. Sometimes in asterisk console I don't see anything in
logs if the Sip extensions' context don't contain the number that is being
dialled
Do you've access to any phone debugging console?
Sounds like problem is somewhere around "She" :p j/k .
--
Rega
r you can access the phone web
interface and confirm the dialplan active on the phone is the same as what you
set in the config file on the server.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
S
PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone
Adam Robins wrote:
> No change, thanks
Well,
In the long run, it may just be easier to send her out a replacement phone and
ask for that one back, so you
[asterisk-users] Problem with outbound dialing from remote phone
What happens if she keys in the number+# then presses dial?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Friday, October 14, 201
-Commercial Discussion
Subject: Re: [asterisk-users] Problem with outbound dialing from remote phone
Adam Robins wrote:
> Any ideas, suggestions, etc., would be greatly appreciated
My guess that the Polycom digitmap isn't being loaded (sip.cfg). I'm sure if
she were to dial the phone nu
I have a real head scratcher . . .
We have several employees who work from home. All have Polycom 501's that
register to our office Asterisk 1.6.x server and communicate using SIP g729a.
About two weeks ago, one of these remote users starting experiencing a problem
with a previously working p
This did the trick! Masks the busy signal. Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, October 21, 2010 1:22 PM
To: 'Asterisk Users Mailing List - Non-Commercial Dis
risk-users-boun...@lists.digium.com] On Behalf Of Adam Robins
Sent: Thursday, October 21, 2010 7:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Busy detection in dialplan - Asterisk 1.6
We have an employee who works from home. We sent her a SIP phone to work as an
We have an employee who works from home. We sent her a SIP phone to work as an
extension off our Asterisk 1.6 system, but her DSL service is so bad she was
dropping calls all the time. It's not just a tuning or QoS issue. Her service
is simply unreliable.
She had a POTS line installed and I
Have you tried replacing the "s" extension with "_x."?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alan Lord (News)
Sent: Friday, July 17, 2009 11:12 AM
To: asterisk-users@lists.digium.com
Subject: Re: [ast
On Wed, Mar 29, 2006 at 2:12 PM, Adam Robins
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=peer
fromuser=OB
host
: [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
> sys
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 v
ng
On Thu, 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 th
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
n 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
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 work
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:
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? Callerid
aster 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: "
/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.
>
> -O
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
I just installed the Digium asterisk-gui from svn on to an asterisk 1.4
beta3 configuration.
I can get to the main page, cfgbasic.html, and then log in OK, however
after I log in and then
each time I click on a new menu item I receive "Stack overflow at line:
0". None of the data
Fields on the s
We have a
centralized infrastructure where we deploy Asterisk servers in remote call
centers for authentication and transcoding. SIP g729a calls are then sent
over an MPLS VPN to a central Asterisk farm, from which calls
are sent/received via PRI.
To avoid placing two
servers in each ca
This works great, however, when I look at the "full" log, it says that
the sendmail is executing prior to vm-audio. Any way to change this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cullin J.
Wible
Sent: Tuesday, June 27, 2006 8:41 PM
To: [EMAIL PR
I have an application where I need to send outbound prerecorded
messages. The Asterisk "call file" process works fine if I am sending
the call via SIP or IAX, but not via ZAP over a PRI channel. The
destination device (my cell phone) never rings. The only unusual thing
I see is on the fifth li
Anyone out there have a functional DUNDi configuration using SIP for the
inter-Asterisk transport? I've gotten it to work with IAX2, but if I
change it to SIP it does not pass the call over even though it knows
where to send it. Thanks.
The contents of this email message and any attachments are
When doing an inter-Asterisk call transfer using SIP, I am using the
"fromuser" parameter to route the call into the proper context on the
receiving server. This causes the original callerid to be lost.
Does anyone have any ideas how to preserve the original callerid in this
scenario?
Thanks,
Ad
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=peer
fromuser=OB
host=192.168.1.2
And in EXTENSIONS.CONF
exten => 91NXXNXXX
: [Asterisk-Users] Problem with
chan_iax.cimplimentationcausesbadaudio?
On 21 Mar 2006, at 16:19, Adam Robins wrote:
> All switches and routers give highest priority to traffic on IAX2 port
> 4569. We use DSCB values over the IP-VPN to prioritize it as well.
> This did not change
:19, Adam Robins wrote:
> All switches and routers give highest priority to traffic on IAX2 port
> 4569. We use DSCB values over the IP-VPN to prioritize it as well.
> This did not change with the upgrade, as we can still see proper
> packet coding.
Right, I wouldn't suspect
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, March 21, 2006 11:08 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Problem with chan_iax.c
implimentationcausesbadaudio?
On Tuesday 21 March 2006 10:55, Adam Robins wrote:
> End users
ehalf 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 commu
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 hor
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
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: asterisk-users@lists
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 PROT
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 L
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 refe
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 Mail
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
1.2.4 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 id
ld jitterbuffer 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 acce
elay|throughput|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 "
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
tyRegards,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" to
-1
CTED]
[mailto:[EMAIL PROTECTED] 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
ion
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 for IAX2, I have finally received almost acceptable quality. I
am receiving 5-8 complaints a day of calls &q
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
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 a
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
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 PRO
Anyone out there using small-midsized (2-4 TB) SAN solution among
multiple Asterisk systems? I don't have the budget for an EMC-caliber
solution, and can't seem to find much else out there.
Thanks,
Adam
The contents of this email message and any attachments are confidential and are
intended sol
l Failover
Adam,
An Audicodes Mediant 2000 gateway with a couple of PRI's.
Why?
Doug.
-Original Message-----
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asteris
What are you using to terminate the PSTN calls and do the SIP
transcoding?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
We are looking for a high density PRI-to-SIP gateway for
our call center and IVR applications. The device must take in a
channelized DS3 and output SIP g729a to multiple Asterisk servers. We have
looked at the Cisco AS5400XM, Lucent APX 1000 and Quintum Tenor CMS (fronted by
an Adtran M13)
I am trying to test whether a callerid number is a valid ten digit
number. I'm a total novice with regular expressions.
I've tried:
exten => s,n,GotoIf($[${CALLERIDNUM} : \d{10,10}]?label)
But CLI gives an error. Can someone please show me what the correct
syntax would be to do this?
Thanks,
A
We had a Rev I card that did not work. We sent it
back to Digium and had it reflashed back to H.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob
LithSent: Friday, November 11, 2005 1:40 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk
Thank you all for your input on this subject. I think I'll pass for
now!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Pyeron
Sent: Wednesday, November 02, 2005 2:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [A
We have built an Asterisk network using an MPLS-based IP VPN. We have
one location in New Brunswick Canada that consistently gives us major
quality problems, whereas the others are flawless. Quality problems
take the form of static, poor voice tonality, popping & clicking, drops,
sporadic echo,
else? and where do I look
for it?MATT---
On 10/5/05, Adam
Robins <[EMAIL PROTECTED]>
wrote:
It's
already built in. AMD.On Wed, 5 Oct 2005, Cory Andrews
wrote:> Anyone aware if Digium or Sangoma, or possibly a function
of Asterisk,> supports answering machine detec
It's already built in. AMD.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, October 05, 2005 4:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Answering Machine Detection
I have two Asterisk boxes that I thought were trunked, but based on not
seeing the (T) in iax2 show peers, now I'm not sure.
Server 192.168.xxx.1 extensions.conf has:
Exten => _2XXX,1,Dial(IAX2/interoffice:[EMAIL PROTECTED]/${EXTEN})
Server 192.168.xxx.1 iax.conf has:
[general]
trunk=yes
[interof
Does anyone know how to use ztmonitor to set gain on a PRI circuit via a
TE410P card, or is it just for FXO?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marek
Zachara
Sent: Friday, September 09, 2005 2:55 PM
To: Asterisk Users Mailing List - Non-Com
Softphone Quality & Network Cards
Matt Riddell wrote:
>Adam Robins wrote:
>
>
>>Should it be in half duplex or full duplex?
>>
>>
>Full.
>
AFAIK, depends...
If you have your switches doing autonegotiation, you can't disable
autoneg in the NIC and hard
Quality & Network Cards
Adam Robins wrote:
> We are in the process of an Asterisk call center deployment using IAX2
> G711 ulaw softphones. Outbound sound quality is terrible.
Check if the network card is in half duplex mode.
--
Cheers,
Ma
We are using Plantronics H51N headset top with DA55 USB adapter which
has DSP built-in. Terrible means garbled, unintelligible,
underwater-sounding.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
von Klitzing
Sent: Friday, August 26, 2005 11:23
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
This week we rebuilt the entire LAN with Cisco 2950-EI switches and have
employed QoS on the switches and router. Still sounds terrible.
What we are now finding is
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
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 priv
erver 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
Subjec
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 th
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:
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
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
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 modpr
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 n
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
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 ?
Gior
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
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
A
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 no
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 is
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
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
Sen
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
musiconho
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 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 car
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 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 th
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] Win
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,
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 ove
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 u
Why would you use gateways and PRI's when several of the major carriers
(AT&T, 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
om: [EMAIL 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:
&g
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
Mich
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