Hi Alex,
Thank you for your help :) now it's work fine thank you very mutch :)
before in DID i put all a number that's why it doesn't work know i put
like as you tell me its work fine :)
Kind Regards
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Hi,
has someone here done a patch to use voicechanger with asterisk 1.4 and/or
trunk? The Bug in bugs.digium.com was closed with the note that there is a
version on http://www.lobstertech.com/code/voicechanger/ , but i cannot find
something for 1.4 there...
Greetings,
Andreas
Hi,
Is this server silent enough to be installed beside users in office
environment ?
http://www.dell.com/content/products/productdetails.aspx/pedge_860?c=uscs=04l=ens=bsd
Regards
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Hi every body,
can someone please tell me about a SIP phone that support more than 10
extension (free or not free ;) ) wich will be used in my company, i've
bought a SNOM but it just support 5 sip extension
Kind regards
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how much users do you have?
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On 15 Mar 2007, at 16:19, Trevor Peirce wrote:
Patrick wrote:
Thanks for the info Trevor. Was your proof of concept also with
Windows
PCs or *nix PCs? I haven't played with realtime yet so I might be
in for
a bit of a learning curve.
This was just on Linux user stations with a simple
All,
Am running asterisk on an Opteron 165 with 4GB RAM and 1x80GB and 1x320GB
SATA for a call center application (running VICIDIAL). Asterisk CLI
(accessed by screen logging asterisk on startup and entering the allocated
screen) gives me 'Warning LSP Low' and the voice quality goes down when
On 16 Mar 2007, at 09:53, Olivier wrote:
Hi,
Is this server silent enough to be installed beside users in office
environment ?
http://www.dell.com/content/products/productdetails.aspx/pedge_860?
c=uscs=04l=ens=bsd
No.
I've had one on my desk for a day. My co-workers insisted it went
Hi,
im trying to use bling transfer on asterisk 1.4.0. it doesnt work i have
configured # key for transfer when i press it nothing happens. also my
dynamic features do not get executed. everything works fine in asterisk
1.2.13 which is on another workstation. can somebody plz help? here is my
Matt wrote:
Yikes.. you'd think a server would be running RAID.
At any rate.. Please feel free to visit http://www.voip-wiki.us
I have set this up to be able to hold information for the Asterisk
community. I will also gladly allow others to mirror it.
It is sitting in a climate controlled
Stephen Bosch wrote:
RAID or no RAID, the site should have one or more mirrors.
As soon as wiki comes back, mirrors should be created.
--
Tomislav Parcina
[EMAIL PROTECTED]
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James H Thompson wrote:
I will definately be looking for an easy way to create a mirror site
once voip-info.org is back up.
This is made difficult by the dynamic nature of the site, but its been
on my list of things to do for a while now.
Hopefully this will happen before next crash :)
--
Did some more googling and grep-ping and I found that this message most
likely comes from codec_g729a.so.
Has anybody seen this before? Anything that we should be concerned about?
Thanks
rajeev
On 3/16/07, Rajeev Natarajan [EMAIL PROTECTED] wrote:
All,
Am running asterisk on an Opteron 165
Matt Putnam wrote:
I have 3 cisco 7912 that all stoped working at the same time on sunday.
There is nothing on the display and the menu and hold buttons are lit.
Resteing produces the same results the phone dosent respond. Anyone have
an idea how to fix this or if it can even be fixed. Ive
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
all the machines are in the same 192.168.0.x net
the routing Server in the middle has iaxusers realtime
snom320, snom360 and snom370 are supporting 12 different SIP identities.
Regards,
Sven
On Friday 16 March 2007 10:57, younss azzayani wrote:
Hi every body,
can someone please tell me about a SIP phone that support more than 10
extension (free or not free ;) ) wich will be used in my company,
Hi,
Contact me if you need a mirror,
I'll be glad to offer some space on my webserver for this.
Regards,
Tristan Mahé
Tomislav Parcina a écrit :
Stephen Bosch wrote:
RAID or no RAID, the site should have one or more mirrors.
As soon as wiki comes back, mirrors should be created.
I have 10 users in a small training-teaching room (20 m2).
There is a small cabinet in which a switch and modem are installed but you
can't count on it to filter any noise.
2007/3/16, younss azzayani [EMAIL PROTECTED]:
***
This message was sent to your KasMail disposable email address:
I'm really after 1U-2U silent servers as I've got the feeling most of them
are too noisy for offices and most of our clients don't have server rooms.
regards
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By the way, which type of CPU equiped you server (as it is distributed with
different options) ?
Xeon, Celeron ?
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On Fri, Mar 16, 2007 at 01:08:54PM +0100, Olivier said:
I'm really after 1U-2U silent servers as I've got the feeling most of them
are too noisy for offices and most of our clients don't have server rooms.
Herin lies the problem. Slimline rackmount servers require several small fans
operating
Kevin P. Fleming wrote:
There is no need for any 'map'; any Asterisk 1.2.x release should be
usable with any Zaptel 1.2.x release, but of course we'd suggest using
the latest releases of both. There are no API changes or feature
additions (generally) in release branches, so frequently you can
Self contained water cooled solutions maybe?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
bristuff-0.2.0-RC8s
two isdn lines plugged into first two ports
and like I said, also a digium tdm400 card in there for analog phones
this 'timer' error message it is something to do with the qozap driver
isn't it? not sure
Thanks for any ideas!
--
Chris
Tzafrir Cohen [EMAIL
Here it is (I hope thi is the one Steve was speaking of!) :)
http://groups.yahoo.com/group/astcallcenters/
Hope this helps
l.
In data Thu, 15 Mar 2007 21:35:06 +0100, nik600 [EMAIL PROTECTED] ha
scritto:
i haven't found any call center asterisk mailing list, but i've found
this:
what's the easiest way of using cepstral voices with asterisk ? On their
website, in the ssml page
(http://www.cepstral.com/cgi-bin/support?page=ssml), they say
Asterisk PBX
SSML can be used with Cepstral voices in Asterisk by simply embedding
the markup into the input text.
what input text
With Ioan suggestion it still doesn't work, because Asterisk still
thinks that the INVITE sent as consequence of the REFER message isn't
correlated with a transferred call coming from the secretary.
I've also tried to do it using different contexts, but it still doesn't
work. I've done like
Hi Steve,
this means that back-compatibility is to forget, right?
Giorgio
Steve Murphy wrote:
On Thu, 2007-03-15 at 15:33 +0100, Giorgio Incantalupo wrote:
Hi,
I'm testing Asterisk 1.4.1 with Snom phones but leds are not working to
show which devices are busy/not connected. The same
Has this issue been resolved? I'm having the problem now with the code base
downloaded yesterday.
On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote:
while compiling svn 53132 of asterisk branch 1.2
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g3 -Iinclude
I would check to see if you had the latest version of the zaptel library for
the version of asterisk you are trying to compile.
On 3/16/07, Wilson Pickett [EMAIL PROTECTED] wrote:
Has this issue been resolved? I'm having the problem now with the code
base downloaded yesterday.
On 2/3/07,
I have been working with a couple companies who are interested in
integrating Cisco VoIP (mostly call manager express) but utilizing Asterisk
for AA, VM, failover trunks etc. I have found some materials and guidance
out there but I think a list and/or wiki for general asterisk integration
with
ok thank you :) i'll look for this
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i going what Walt Reed says, i've a U1 witch make a lot of noise, but
i reconfigured the bios to slow down the fan speed, and know it's
work fine, look for a serial devices, for the CPU it's depend where
you are, for me in africa AMD goes hut, but Intel work fine, but i
work with AMD because i
On Fri, Mar 16, 2007 at 02:46:24PM +0100, Wilson Pickett wrote:
Has this issue been resolved? I'm having the problem now with the code base
downloaded yesterday.
On 2/3/07, Erick Perez [EMAIL PROTECTED] wrote:
while compiling svn 53132 of asterisk branch 1.2
codec_zap.o codec_zap.c
2007/3/16, younss azzayani [EMAIL PROTECTED]:
i agree with what Walt Reed says, i've a U1 witch make a lot of noise, but
i reconfigured the bios to slow down the fan speed, and now it's
work fine. look for a serial devices.for the CPU it's depend on where
you are, for me in africa AMD goes hut,
Try the Snom 360. The softphone version of it (a free demo) has 12 lines (I
presume the real thing has the same). You can find the softphone at
http://www.snom.com/download/snom360-5.3.exe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of younss
azzayani
Whats the non-workaround solution? Is there one?
On 3/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Fri, Mar 16, 2007 at 02:46:24PM +0100, Wilson Pickett wrote:
Has this issue been resolved? I'm having the problem now with the code
base
downloaded yesterday.
On 2/3/07, Erick Perez
Simone Cittadini wrote:
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
What version of Asterisk exactly?
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
This doesn't make any sense; the 'Server' is
Hi ALL,
I'm using this TNT to front-end an asterisk cluster, working pretty
well so far. Some T1's are inbound from PSTN PRI's and others are
Outbound to PSTN PRI's.
Specifying what traffic to send out what PRI is pretty easy, we have
unique trunk numbers assigned to specific T1's or groups of
Julian Lyndon-Smith wrote:
what's the easiest way of using cepstral voices with asterisk ? On their
website, in the ssml page
(http://www.cepstral.com/cgi-bin/support?page=ssml), they say
Asterisk PBX
SSML can be used with Cepstral voices in Asterisk by simply embedding
the markup into the
Kevin P. Fleming ha scritto:
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
What version of Asterisk exactly?
1.4.1
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
This doesn't make
I discussed this with Digium techs, who recommended using the 1.4 version of
Zaptel with the 1.2 version of Asterisk, at least with my hardware
(TDM400P). The 1.4 zaptel does not yet support the HPEC (which won't run on
my system anyways...), but does have a totally rewritten MG2 EC. I am
running
Hi all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
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On 16/03/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Here's a better question: why make everyone join another list when
this one already works perfectly well?
On Fri, 2007-03-16 at 17:35 +0100, Giorgio Incantalupo wrote:
Hi all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
An interesting idea, but... my bet would be that most stuff discussed
- Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
No, it would be yet another list that people would have to subscribe to, and
many of
There's also an app_swift available at http://www.loopfree.net/app_swift/
--
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On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote:
Whats the non-workaround solution? Is there one?
I haven't really tested, but it should be along the lines of:
Index: codecs/Makefile
===
--- codecs/Makefile
No, it would be yet another list that people would have to subscribe to, and
many
of the questions/answers for one version are quite relevant to the other.
Can't we just require everyone on this list to upgrade to v1.4? :)
I'm sure in due time they will anyway.
-Kenneth
On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote:
Whats the non-workaround solution? Is there one?
http://bugs.digium.com/view.php?id=9303
Please test.
Wasn't there an existing issue on this one?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL
Use the instructions posted on lobstertech, just replace voice changer with
the newest version on the site (not 1.4)
On 3/16/07, Andreas Anderson [EMAIL PROTECTED] wrote:
Hi,
has someone here done a patch to use voicechanger with asterisk 1.4 and/or
trunk? The Bug in bugs.digium.com was
Baruch,
I have tried million different combination. These asterisk guys change
commands every version. Our version - Asterisk 1.2.13
The only command that actually does anything is SetCallerID. Whether
we do it as, SetCallerID(4016261150) or SetCallerID(DIVON
4016261150) the number is being
http://deancollinsblog.blogspot.com/2007/03/microsoft-buys-tellme.html
I thought I would email this post I made on my blog from yesterday as a
way of stimulating discussion on this.
It looks like the Asterisk community is no closer to getting a Pre-Paid
'Offboard Speech Recognition
Julian Lyndon-Smith wrote:
what's the easiest way of using cepstral voices with asterisk ? On their
website, in the ssml page
(http://www.cepstral.com/cgi-bin/support?page=ssml), they say
Asterisk PBX
SSML can be used with Cepstral voices in Asterisk by simply embedding
the markup into the
Igor Shmukler wrote:
Baruch,
I have tried million different combination. These asterisk guys change
commands every version. Our version - Asterisk 1.2.13
The only command that actually does anything is SetCallerID. Whether
we do it as, SetCallerID(4016261150) or SetCallerID(DIVON
4016261150)
Hi all,
I decided the best way to get to know * well is to do it from
scratch. Having read the majority of the Asterisk: The Future of
Telephony I am now attempting to compile zaptel 1.4.0 and am receving
the very same series of errors mentioned in this post on the forums:
I have tried million different combination. These asterisk guys change
commands every version. Our version - Asterisk 1.2.13
The only command that actually does anything is SetCallerID. Whether
we do it as, SetCallerID(4016261150) or SetCallerID(DIVON
4016261150) the number is being proper
Has anyone been successful in getting a refund from SellVoip when you've
cancelled service?
Tom Lynn
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Hello,
I have a setup like this:
exten = s,1,Ringing
exten = s,n,Wait(3)
exten = s,n,Answer
exten = s,n,Set(TIMEOUT(digit)=6)
exten = s,n,Authenticate(1)
exten = s,n,DISA(no-password|my-context)
exten = i,1,Playback(invalid)
exten = i,n,Wait(1)
exten = i,n,Goto(s,5)
Gordon Henderson wrote:
On Wed, 14 Mar 2007, Jonathan k. Creasy wrote:
I would be willing to mirror it also?.
At the risk of sounding like an AOLer, Me Too ... (UK based mirror?)
The site is pingable, so I'd suggest it's either crashed in some awkward
way and just needs resetting, but
On Mar 15, 2007, at 10:34 PM, Vincent Tam wrote:
Hello,
We have an asterisk setup at our client's site using a TE205P. The
line to telco is a 23 channels T1 PRI, however the line has random
echo problems (about 5-10% of the calls)!
Can anybody tell me if echo cancellation is really needed
Bruce Reeves wrote:
Cory,
Are the Polycom phones able to detect that the headset is off-hook? We
have had problems with 2 different brands of headsets working fine but
the phones seem unaware of the headset's hook state.
We're working with a client who uses headsets on Polycom phones,
Chris Nighswonger wrote:
I am working with a fresh install of fc6.
Kernel 2.6.20 was released after Zaptel 1.4.0, so it will not build
against that kernel. Either use an older kernel, use the SVN version of
Zaptel branch-1.4, or wait for the release of Zaptel 1.4.1 which should
occur later
On 13 Mar 2007, at 15:58, Brandon Comouche wrote:
What I am most curious about at this time is the methods used to move
from server to server. *Ideally* I would like to sit down at a phone,
enter my extension/password and have that phone ring as my extension.
Essentially, I would like a log
It's been up since early morning for me. Just refreshed - still up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Bosch
Sent: Wednesday, March 14, 2007 12:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
This work with real time ?
nik600 escribió:
Hi
i just want to let you know that is available a new release of ccmanager.
I've added the possibility to import queue_log information in a mysql
database and to generate reports using this information.
The software is in a beta state and
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Igor
Shmukler
Sent: Friday, March 16, 2007 2:20 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.13 Caller ID problem
Simone Cittadini wrote:
So the config is :
realtime mysql users on the server to auth the customers (Input) and one
user entry in iax.conf on the Termination to auth the Server
transfer=mediaonly is set in [general]
OK, then you'll need to get a verbose/debug console trace, and
preferably a
How about this?
We will assume the boss is named Andrew and is 101 and his secretary is 116.
;Boss Extension (Custom)
exten = 101/116,1,Dial(SIP/101,30,tr)
exten = 101/116,2,Goto(andrewvm,1)
exten = 101,1,SetCallerID(Andrew ${CALLERIDNAME} ${CALLERIDNUM})
exten = 101,2,Dial(SIP/116,30,tr)
exten
Hi List,
I'm trying to connect Asterisk with a Nortel Meridian using an E1 with
MRFR2 signaling.
I've connected both cards, and compiled all the required software...
The problem is every call (outgoing or incoming) got dropped,
complaining about some T1 timed out
Only for testing purposes
Tzafrir Cohen wrote:
On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote:
Whats the non-workaround solution? Is there one?
http://bugs.digium.com/view.php?id=9303
Please test.
Wasn't there an existing issue on this one?
The bug traker says, to me at least, that this
Have you looked at Voxeo Prophecy?
http://www.voxeo.com/prophecy/
cheers,
Wendell
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean
Collins
Sent: Friday, March 16, 2007 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users]
I am having an issue with the follow me application in 1.4
The application description (below) indicates that if the specified
followmeid profile doesn't exist in followme.conf, execution will be
returned to the dialplan and call execution will continue at the next
priority.
That's not happening
On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote:
Another interesting (from an American's perspective anyways) is that
inbound calls on cell phones are free. Even if you buy a SIM with a
little pre-paid time and use up the time, you can still receive inbound
calls for free for a
I'm interested. I turned my last call manager off last month, but I still
use the handsets and a Cisco router for PSTN access.
On 3/16/07, Curt Shaffer [EMAIL PROTECTED] wrote:
I have been working with a couple companies who are interested in
integrating Cisco VoIP (mostly call manager
tim, patrick,
SSO is a hot button for large orgs/corps... have heard it bantered for
years but no solution. I have seen a product that had a small utility
on windows that transmitted login info to a linux box. it was in every
users profile so upon login, the transaction was completed. I have
Igor Shmukler wrote:
I have tried million different combination. These asterisk guys change
commands every version. Our version - Asterisk 1.2.13
The only command that actually does anything is SetCallerID. Whether
we do it as, SetCallerID(4016261150) or SetCallerID(DIVON
4016261150) the
Hi Wendel,
Voxeo got in touch with me yesterday however from the initial
discussions it appears they are still looking for a minimum $500+ per
month in billing before they are interested in a relationship.
Lets just say I'm in discussions and will keep the list posted as
discussions evolve
Another idea that has just come to me regarding bluetooth and a PBX is
like this.
Many people would like to use headsets with their IP phones. Some
support wired headsets, but bluetooth headsets can be a good choice
for a headset -- no wires, many people often have one, and there is
a rich
Not their hosted service, their prophecy application (windows based).
It's free for a single port, and reasonably priced for more. No monthly
recurring, just the per port licensing charge.
From: Dean Collins [mailto:[EMAIL PROTECTED]
Sent: Friday, March 16, 2007 2:03 PM
To: Asterisk Users
here! here! they are different beasts...
Giorgio Incantalupo wrote:
Hi
all,
since Asterisk 1.4 seems to have too many differences from previous
versions, wouldn't be nice to have a new mailing list?
Giorgio Incantalupo
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Question:
Is it possible to pickup someone else's call who didn't park a call?
My boss would like to see a way to pick up some one else's incoming call
if they aren't at their desk and it's not forwarded. So if my phone were
ringing and he knew i ran down the hall, he could press some key combo
Steve Prior wrote:
Julian Lyndon-Smith wrote:
what input text ? To what application ?
I agree completely with the app_swift suggestion from loopfree as Kai
suggested. It provides the app_Swift which you can use from within a
dialplan. In fact, if you're getting fancy by using a fastAGI
An equally unrealistic expectation would be to require that people write
RELEVANT and specific Subjects.
If your question relates to 1.4, put 1.4 somewhere in the Subject, or
if it relates to unreleased trunk, specify it.
So you can quickly filter out/in whatever your interests are.
But as I
On Fri, Mar 16, 2007 at 11:32:31AM -0700, Tom Lynn wrote:
Has anyone been successful in getting a refund from SellVoip when you've
cancelled service?
You were able to cancel service with Sellvoip? That's impressive, that
implies they actually responded to a request you made to cancel
service.
At 11:32 AM 3/16/2007, you wrote:
Has anyone been successful in getting a refund from SellVoip when
you've cancelled service?
No, I'm just using the credit up slowly whenever their network works.
Ira
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yes, queue information are parsed directly from queue status asterisk manager.
On 3/16/07, Germán Aracil Boned [EMAIL PROTECTED] wrote:
This work with real time ?
nik600 escribió:
Hi
i just want to let you know that is available a new release of ccmanager.
I've added the possibility to
Group pickup / call pickup is the feature you want.You put everybody
in a group and if you want to grab a ringing phone, you just hit the
group pickup code.
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
Rob Schall wrote:
Question:
Is it possible to pickup
On 3/16/07, Kevin Kiely [EMAIL PROTECTED] wrote:
I am having an issue with the follow me application in 1.4
The application description (below) indicates that if the specified
followmeid profile doesn't exist in followme.conf, execution will be
returned to the dialplan and call execution will
From: Ricardo Carvalho [EMAIL PROTECTED]
Date: Fri, 16 Mar 2007 13:41:49 +
With Ioan suggestion it still doesn't work, because Asterisk still thinks
that the INVITE sent as consequence of the REFER message isn't correlated
with a transferred call coming from the secretary.
Your
From: Brad Templeton [EMAIL PROTECTED]
Date: Fri, 16 Mar 2007 13:37:55 -0700
On Wed, Mar 14, 2007 at 09:37:45AM -0500, Steve Totaro wrote:
Another interesting (from an American's perspective anyways) is that
inbound calls on cell phones are free. Even if you buy a SIM with a
little pre-paid
At this point, I'm simply contacting the State of Washington Attorney
General's office. They're ignoring my e-mails and I'm done monkeying
around.
On 3/16/07, Ira [EMAIL PROTECTED] wrote:
At 11:32 AM 3/16/2007, you wrote:
Has anyone been successful in getting a refund from SellVoip when
On 3/16/07, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Chris Nighswonger wrote:
I am working with a fresh install of fc6.
Kernel 2.6.20 was released after Zaptel 1.4.0, so it will not build
against that kernel. Either use an older kernel, use the SVN version of
Zaptel branch-1.4, or wait for
On Fri, Mar 16, 2007 at 03:55:56PM -0400, John Novack wrote:
Tzafrir Cohen wrote:
On Fri, Mar 16, 2007 at 11:28:43AM -0400, Sean Bright wrote:
Whats the non-workaround solution? Is there one?
http://bugs.digium.com/view.php?id=9303
Please test.
Wasn't there an existing
Arturo, the error does not says much really, just that either the
other end timed out expecting a response from you, or your end timed
out expecting a response from the other end :)
However, from my experience, it may be an error in your DNIS/ANI
configuration and/or an mfcr2 library error (
Hi Folks,
I am planning to create an internal portal where the users can enter two
phone numbers (theirs and the party they are trying to reach) and connect
the two of them by initiating two calls from Asterisk. Any pointers on how
to initiate two calls and then bridge them (without using
Hi Folks,
I want to setup a follow me routine so that asterisk can call me on the
multiple numbers.
I tried some of the samples at voip-info but there is a problem with those
examples.
I dont have coverage in my home area and my cell phone answering machine
picks up the phone right away so my
Anyone noticed a problem with 1.4 where the channels are getting permanently
stuck until the reload/restart event?
R.
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On Fri, 16 Mar 2007, Ritesh Agrawal wrote:
Hi Folks,
I am planning to create an internal portal where the users can enter two
phone numbers (theirs and the party they are trying to reach) and connect
the two of them by initiating two calls from
On Fri, Mar 16, 2007 at 04:16:21PM -0700, Tom Lynn wrote:
At this point, I'm simply contacting the State of Washington Attorney
General's office. They're ignoring my e-mails and I'm done monkeying
around.
It makes no sense. The put together a good system on the tech end,
Asterisk based,
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