Hi,
I can confirm that Michael's patch solve this issue and some others even if
you have Asterisk installed over VMWare, where all other 'dummy' devices
does not work.
BR,
Dan
- Original Message -
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05,
Stefano,
I've come across this problem as well using SIP devices and asterisk.
As far as I can tell, the IVR systems are deliberately not answering in
order to not pay the telco for call charges. Ironically, they are
sending audio before they answer the call. Depending on what device you
are
On Mon, 2003-08-11 at 11:28, Julien wrote:
Just a last question, if i configure G723 in my ATA, i can't call the
voicemail for exemple. I've seen that messages were in GSM format. Is there
a way to be able to acces to the voice mail in G723 (for remote users) and
in G711 for local users ?
In
It's easier to check that up ...
On Wed, 6 Aug 2003, Jim Mercer wrote:
On Wed, Aug 06, 2003 at 09:59:18AM -0500, Martin Pycko wrote:
You're looking for libncurses-dev and in libpri you can remove -Werror
from libpri/Makefile or cvs update libpri (it should be fixed)
fixed, in that they
Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q..
Anyone know why?
--
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr
Powered by Outblaze
___
Has anyone experienced an unstable Asterisk server when
using CDR MySQL. Asterisk keeps
crashing or answering the incoming lines with a fax/modem sound when I use the cdr_mysql module.
When I remove it, I have no problems.
Tim
Ahhh
Thanks, that's the final answer
Lee
- Original Message -
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, August 07, 2003 11:26 AM
Subject: Re: [Asterisk-Users] MWI bug ?
Hi Lee,
You need to specify the VM context that you are using..
so using your
have
you looked at digiums site? there are few simple sample
there.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Adelino
BaenaSent: 09 August 2003 21:47To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] H323 and
SIP
Dear
Any suggestions would be great.
No config files changed. I have been trying the latest cvs every
few days for a few weeks now. I currently am using 06/06/2003.
[EMAIL PROTECTED] asterisk]# asterisk -c
Asterisk CVS-08/07/03-15:43:33, Copyright (C) 1999-2001 Linux Support
Services, Inc.
Written
On Tue, 2003-08-05 at 02:45, Kent Williams wrote:
Mine has been working well, but the only problem is that it doesn't
support callerid (from the POTS side).
I didn't say they didn't work. Mine has been in production use for over
a year now with only a couple hiccups related to sync source and
Just wondering if anyone knows of a workaround for the BudgeTone being
signalled while off-hook, and then instead of playing nice with call
waiting allowing the conversation to be ruined with ringing tones.
I am trying to figure out how to test this phone fully, including
allowing incoming
Yes, the voice mail is at 2999 , but it doesnt work when i call it from the
ata .I talked about the 600 (echo test) but i removed it from the
extension.conf, sorry.
In sjphone , i've got this error
15:06:42 INFO Session rejected. Reason: 404 Not Found
15:06:42 INFO Call 153 ended: Session
And remember to use Dial with t option
Martin
On Thu, 14 Aug 2003, Brian West wrote:
http://www.bkw.org/~brian/ata.html
Pay attention to connectmode and audiomode Its how I set it up and it
works.
bkw
On Thu, 14 Aug 2003, Dan wrote:
Hi Brian,
ATA is in SIP mode, and RFC2833
Hi Matthew,
That argument doesn't seem to work. I don't hear many complaints here
about the cost of the VoiceAge codec. It's the clunkiness of the
protection scheme people don't like. It's only the protection scheme
that seems to be making people want to dump the VoiceAge code.
Remember how
Helo,
Try the www suggestion, but it does't work on H323 channel I guess, cause it
just send the whole string to gatekeeper including the 'www'. I got called
party not registered on gatekeeper. I think '' only work on Zap channel.
:(
thanks anyway mate.
So WipeOut your project is
On Monday 11 August 2003 02:46 pm, Eduardo Goncalves wrote:
On Sun, 10 Aug 2003 01:50:33 -0500
Tilghman Lesher [EMAIL PROTECTED] wrote:
Are the faxes all being sent by the same fax machine? If not,
are the faxes being detected consistently from each source? If
the remote fax
Eric Wieling wrote:
On Tue, 2003-08-12 at 15:37, Mark Spencer wrote:
Couldn't agree more. The G.729 codec is so unDigium-like... don't buy
it is my recommendation.
I don't think anybody buys G.729 just to have it. They buy it because
they *have* to have it. And we sell it because they
On Mon, 2003-08-11 at 10:19, Jim Friedeck wrote:
Our CSR people need to be informed when a call is ringing in when they
are on the phone. Is there a mechanism for informing an off-hook target
channel of an incoming call? We have a guy who should get first shot at
all incoming calls on our
I was speaking in a logic related to real call routing and queueing. In
House policy can be built on top of your call strategy. What we are
needing is input on logic only ..
bkw
On Tue, 12 Aug 2003, Richard Lyman wrote:
translation: manager gets off thier fat ass and actually talks to
I see. :P
On Tue, 12 Aug 2003, Richard Lyman wrote:
my point was your logic regarding 'calculating magic/metric' for
extremely long call times shouldn't be part of the 'logic' it
SHOULD be 'inhouse' policy where the mgr gives agentA a nice long
chat about how to sell/service the client more
Title: ANI/DNIS call routing
Someone please double check me then forgive me for answering my own
question.
If we have PRI service and multiple 800 numbers come in over the circuit
the DNIS will be reference as the extension in the dial plan? So for example if
we had 1-800-745-8765 and DNIS
Hi all,
I can't seem to find any info on this anywhere on the web, except that BT
caller ID doesnt use the standard bellcore system in use in the US. So, if
anyone here in the UK is onlist and using the x100p successfully, please let
me know.
TIA,
Dave
Hi ;)
I'm a french newbie and i installed asterisk 1 day ago.
I've got an ATA186 and a computer with Sjphone installed.
If i want to call the sjphone from the ata or call the ata from de sjphone
everything is ok.
My problem is ,that i can't call the voicemail or any other phone number
..as 600
The unique id is generated when a channel is requested, so the uniqueid
will be unique for any call or channel. You should never see the same
uniqueid more than once. The uniqueid is available as a channel var
${UNIQUEID}, as an agi environment var (agi_uniqueid), as an entry in
(hopefully)
Quoting WipeOut:
Hi,
I have just been playing with the latest X-Lite.. It works fine
with Asterisk..
As for codecs I tested G.711a/u, GSM and iLBC... iLBC is the only
one that didn't work.. not sure why..
Did you get Speex working? I've tried, but although I can get it to connect,
there is
Hi,
On Thu, 2003-08-07 at 14:23, The Traveller wrote:
The correct way to dial Dutch toll-free numbers using the gateway-prefix is:
1010-666-0800-rest of number
I haven't tried the *31(800)... they mention on their site yet, but
I don't have special provisioning on the gateway for it and
How do I stop asterisk when it is in a bad mood? It keeps dialing
extensions and won't listen! I tried kill PID. No go. I don't want to
have to reboot again. Thanks.
Jim Friedeck
P.S. I love it when my boss looks over my shoulder and I don't have an
answer when he says: 'So, what are you
http://voipstore.atacomm.com/shops/Browse.aspx/27934028032-27934031360.htm
Thats a good start..
bkw
On Tue, 12 Aug 2003, Fabrice Tereszkiewicz wrote:
Hello,
I would like to buy a SIP IP phone, but I don't know wich one to
choose... Can you tell me wich IP Phone is known to work with
I won't as soont as I get asterisk configured. I'm only using X while
I'm configuring and building *. It's easier that switching console screens.
Jim
Tilghman Lesher wrote:
On Friday 08 August 2003 17:07, Jim Friedeck wrote:
Martin,
Tried that over and over.
Ya what he said! :P
On Thu, 14 Aug 2003, Martin Pycko wrote:
And remember to use Dial with t option
Martin
On Thu, 14 Aug 2003, Brian West wrote:
http://www.bkw.org/~brian/ata.html
Pay attention to connectmode and audiomode Its how I set it up and it
works.
bkw
On Thu,
On Thu, 2003-08-07 at 04:14, Jamie Carl wrote:
Dunno what I'm doing wrong here but I just did an
upgrade to the latest
version and now I get no audio at all!
I havn't changed a single thing. Is there anything
special I need to do
to get this to work again?
I get a quick 'chirp'
Ok just had my boss point something out:
I'd think dumping calls on most-idle would be fairly straightforward, but
could be skewed if agentA is on a 40 minute call, agentB has a bunch of 5
minute calls
So total call time should be counted in the logic somewhere.
bkw
On Sun, 10 Aug 2003, Brian
http://www.nanpa.com/number_resource_info/vsc_assignments.html
http://www.nanpa.com/number_resource_info/vsc_definitions.html
On Wed, 2003-08-13 at 14:16, John Todd wrote:
I'm looking to implement some basic CLASS features, using my own
dialplans as well as those so thoughtfully contributed by
Done ...
[1945]
type=friend
username=1945
secret=1945
host=dynamic
context=from-sip
mailbox=1945
defaultip=192.168.0.6 (and if i use DHCP ???)
But it doesn't work well. It's strange, because calls between ip phones or
soft phones work well ...but the voice mail is unavailable..
I'm searching
On Fri, 8 Aug 2003, Adams, Gavin wrote:
Now it's back to tweaking the configuration on our SIP phones (7960s).
The message_uri parameter in the phone's configuration file is working
great. Dials comedian mail directly. Is there a way to let voicemail2
know what the incoming extension is, and use
For testing purposes, my dial line is:
Dial(${ARG2},20,tT)
When I call from one machine through asterisk to
another, I can press # from either side and hear "Transfer."
However, from the caller side I can continue on and
put people on hold by dialing '700'.
From the callee side, I can
Hello all,
Does anybody has a problem where asterisk only grap
the first digit of a string of multiple DTMF digits?
My setup is:
PSTN --- AS5300 ---Gatekeeper ---
Asterisk
When call coming from PSTN all the way
toAsterisk to access a conference room,I press the conference room number
On Thu, 2003-08-07 at 17:06, Jayson Vantuyl wrote:
I've taken the liberty to edit your patch, to put back in the
'adsi_logo' and the values for adapp and adsec as they are in CVS. As
far as I can tell those changes have no relation to problem this patch
solves, they're just local changes
Hello All,
I wonder is there a way where I reload asterisk on
CLI without disconnect any call that is currently taken place.
Foong
I used the exact versions listed in the readme for chan_h323 and it works
fine. Slackware and RH8 and 9.
bkw
On Fri, 8 Aug 2003, Kelvin Chua wrote:
hi guys,
i'm encountering one way audio on cvs using netmeeting and chan_h323.so
is there a quick fix or workaround for this?
compiled using
Title: Message
Simple Q but I can't
find the answer in the archives (and am too lazy to look in the source, but then
its 32 Celcius here...
Do all digium cards
provide the zapata timing? e.g.also the analogs (including the
X100P)or only the E1/T1 -ones or do I need to use ztdummy on the
In my recent new Asterisk installation I'm having users complain that if
they answer a call on call waiting while talking on an existing line they
are then unable to park a call without one of the two parties hanging up.
Is there anyway whatsoever to be on a call, answer a call on call
Hi,
I have two Philips DECT phones: Onis2 Memo (TD6511/BS6511) - british
standard and one Onis TD6211/BB191P (BS6211) - french standard. Both of
them does not show nothing as callerid connected to an ATA186 or to the PSTN
line. A cheap analog cordless (GE) can get the callerid (both name and
I have been using chan_capi and a Fritz AVM (passive) card for a little while now.. So
far so good.. There are still a couple of things I am not totally sure about..
First..
What is the difference between softdtmf=0 and softdtmf=1??
Which is the preffered/recommended option??
Second..
What is
For some reason my Voice-mail is not sending E-mails with the voice
attachment anymore.
It just stopped working.
any suggestions on how to debug?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Scott Stingel wrote:
Hi all-
This question is for those familiar with EuroISDN setup.
I have a customer in Europe where I'm going to install an asterisk based
system with 4 E1's. The customer will configure them all in one large hunt
group.
My question is about the E1 channel configuration. I
While we are on this subject. For testing and such I have been trying to
get one asterisk server to register with another via sip(i know i know use
IAX) but it doesn't work It should... I can't see any reason it
shouldn't Any pointers? All I get is proxy auth and * crashes a
bloody
OK, it's chan_sip. I'd recommend updating asterisk since it might have
been fixed already.
regards
Martin
On Tue, 5 Aug 2003, Ricardo Villa wrote:
I have attached the output. It is just one test call that goes to
voicemail. You can see the NOTICE message several times.
There is one thing
Hi List,
I am facing the reverse problem as stated here.I am
using ATA 186 to make
and recieve call to * through OH323 driver.
When I use G711 codec in the ATA to make call then
then as soon as i dial an
extension the * crashes with 'segmentation fault'.
But the same scenerio works fine when i
On 11 Aug 2003 15:34:38 -0500
Eric Wieling [EMAIL PROTECTED] wrote:
Try adding:
exten = fax,1,Dial(blah)
Where Blah is the zap or SIP port your fax machine is connected to.
But I want to send a fax, if I put Dial(blah), and blah is my fax machine, how
could I send the fax over
Thanks.
I will move * on one of our internet servers. That should take away
all NAT/PAT issues.
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: 09 August 2003 08:37
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] X-Lite - No sound
Thanks Alistair,
I'm a relative linux newbie and haven't worked with perl in over 5 years but
your info makes sense (will probably be much clearer when I'm in the middle
of configuring stuff).
I'll take a look at the links you provided, in the meantime just to clarify
on using a bash script - Am
Fabia,
The only numbers you should be able to dial from that config are
1945
1943
2999
and nothing else...
The entry under bogon-calls (isn't it bogus calls?) should read
exten = s,1,Congestion
rather that using the _. ...
HTH
Andy
*** REPLY SEPARATOR ***
On 10/08/2003
Hello,
I would like to buy a SIP IP phone, but I don't know wich one to
choose... Can you tell me wich IP Phone is known to work with Asterisk
please.
I've seen the Cisco 7940, but I don't know if it works, and how
expensive is it ?
I'm french, so if you know some french resellers, tell me.
Nope.. sure doesn't.. You call the AgentLoginCallback extension from any
phone.. Enter you agent ID.. and Password... then enter the extension your
calls should go to and its done.
bkw
On Wed, 13 Aug 2003, Devon Henderson wrote:
Being a relative Asterisk newbie, I may be wrong.. but as far as
From: Sean Figgins [mailto:[EMAIL PROTECTED]
On Fri, 8 Aug 2003, Adams, Gavin wrote:
Also, we decided to go with actual extension numbers on the phones
instead of usernames per extension. On the Cisco phones, is there a
way
to change the name/number on the top line (white text on black)
Just updated asterisk from cvs. It's now CVS-08/01/03-18:27:08.
Also I've removed ztdummy module. It seems to be better now.
Not perfect - some sound glitch still happens, but much better.
Also I've noticed that it somehow depends on the mp3 itself.
Some songs are played out almost perfectly,
Hello,
Is there any configuration in zapata.conf for fax detection (or transmission)?
When I try to send a fax trought asterisk, the line 'Fax Handled:' is always
set to no. The scenario is:
[ata186]---sip---[asterisk]---e1 EM---[pstn]
Fax sometimes goes without problem
Why not ask them??
Christian Stredicke is the man to talk to.
Although, i just had a look at the SNOM website and 1.16w
is still there to download.
J
On Sun, 10 Aug 2003 16:59:37 +
WipeOut . [EMAIL PROTECTED] wrote:
*This message was transferred with a trial version of
CommuniGate(tm)
Hi Brian,
Thanks for the explanation.
In my second scenario, agent 1002 wouldn't be sat idle as the call would
rotate around all them (like round robin) but in the order they stand in the
league of fewestcalls / least recent calls. They'd all ring but the order
would change according to
On Fri, 2003-08-08 at 12:25, Steven Critchfield wrote:
With the increased traffic as of late, I'm wondering if it is time to
split the list again. Specifically I am wondering if it should be split
along the various VoIP protocols and zap hardware, then leave a general
list that does
On Wed, 2003-08-13 at 11:28, Devon Henderson wrote:
We're still in the planning stages of our Asterisk implementation, but we
have a requirement that the extension be mapped to a user, with the phone as
a variable (we have hot seats in our contact center, and we also have agents
that work both
On Thu, 2003-08-07 at 22:17, Wade Weppler wrote:
I've taken the liberty to edit your patch, to put back in the
'adsi_logo' and the values for adapp and adsec as they are in CVS. As
far as I can tell those changes have no relation to problem this patch
solves, they're just local
Hi,
Just wondering if anybody has encountered a similar problem as I have
with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have
dtmf relay configured on the AS, however, when someone calls in from the
PSTN sometimes their digits are inputted twice, which messes up the
At 14:26 -0700 6/8/03, Steve Haehnichen wrote:
Since sipphone.com appears to make no money at all off the free SIP
directory services, they might not be inclined to lock down the
phones.
on the Free World Dialup list he has stated that the phones are
preconfigured but absolutely not locked down.
Hi,
when using multiple * boxes, there appear to be 2 choices as to how to go about
sharing cards and dialplans
1) using switch
2) using dial fail fall-through ie
exten = 1,1, dial(xyz)
exten = 1,2, dial(otherpbx/xyz)
As i see it switch could end up being recurrsive resulting in a wild ooc
hi Marian,
you cannot use the AVM Fritz in P2P mode (at least not with
the capi drivers), maybe I4L will work (hisaxctrl driverid
7 1) IIRC.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30
So its sounds like I do have a clue then...can analog devices have
their own extension and do call parking, and paging and all that? I
assume the caller id gets passed from the POTS CO to the internal
phones?
So from my understanding I can get a TDM400P with one port now and
upgrade to
Hi Tilghman,
I am trying to use the Windows iax client.
My iax.conf looks like this:
-snip-
[pos|
^
I'm hoping this is a typo in the email. If it isn't, that might
explain everything.
Thanks eagle eye. That was the error.
Thanks again.
rgds
pos
Any idea if these fixes will get added to CVS?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jayson Vantuyl
Sent: Thursday, August 07, 2003 11:07 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ADSI and SoftKeys
I've
It's just a proxy service like fwd it will work with asterisk... The phones they are
selling
with the deal are Grandstreams.
It's very likely that they just have been preloaded with their settings, and probably
point to their
own tftp server. simply create fake dns entries and a static route
Can you try iax2?
mark
On Thu, 14 Aug 2003, Dave Wilson wrote:
Hi all,
I'm setting up my first * install and have it peering with another * machine
using IAX across the internet which provides our pstn gateway.
So far I have the IAX friend set up correctly but when I make a test call
anything that supplies a reasonably straight 5v should work ..
do not send *more* than that into the phone which is what
most unregulated supplies will do ..
the supply which comes with the grandstream seems to
be a nice switchmode one.
cheers
Dave
- Original Message -
From: WipeOut .
Hi Andrew,
On Sun, 2003-08-10 at 19:39, Andrew Joakimsen wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Armand A.
Verstappen
Sent: Sunday, August 10, 2003 4:57 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Iconnecthere
On Sun,
yep... iaxclient.sf.net
- wasim
On Wed, 6 Aug 2003, Dan wrote:
There is any Windows IAX based soft phone available?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
I have two questions about asterisk's vm:
Is there a way to do group voicemail in asterisk? I can't seem to find
any references on this.
What we have with the current phone system is the user can dial a group
mailbox number, leave a voice mail, then every one in the group gets a
copy of
Steve Underwood wrote:
The ITU G.729 code is pretty much useless for real world use. It is
very slow. It gets the right answers, but not by efficient means. All
the voice codec reference code I have seen is like this. The people
who develop these things *have* to write an efficient version, as
AgentLoginCallback does this doesn't it?
On Wed, 13 Aug 2003, Steven Critchfield wrote:
On Wed, 2003-08-13 at 11:28, Devon Henderson wrote:
We're still in the planning stages of our Asterisk implementation, but we
have a requirement that the extension be mapped to a user, with the phone as
Very good. We need leastrecent before deployment. Just a basic sorting
of 'leastrecent-ness' should work for us. For now, anyway.
Jim
---
Brian West wrote:
It just rings the fewestcalls or leastrecent over and over.. it doesn't
hunt one bit right now. Thats
Dan wrote:
Hi Steve
Steve Underwood wrote:
06.10 isn't that great a codec,
though. I don't think it is used very much on the GSM networks these
days. Most of the time they use the enhanced full rate (EFR) or half
rate codecs.
What do you mean by isn't a great codec?
06.10 should be
This Windows binary is probably fairly easy to convert for someone with
sufficient skills. It's a simple library, COFF format. It's probably
sufficient to split it into .o files (using ar), then convert the .o
files (using objcopy --target=elf32-i386, objcopy from cygwin has both
elf32 and
Why not make this a client that can be used by all of the people within
the office. This would allow them to control their extension from within
the app and dependant on passwords allow them to manage their own team
or group of extensions.
This could also act as a CTI interface to * if it was
At 11:50 13-8-2003 -0500, you wrote:
Correct me if i'm wrong but doesn't the cdr modules log the call duration?
True, but in some cases you would want to do more. I have built IVR's that
require some after-call handling. For this I tried to use variables as
well, but failed. The solution in my
Another approach would be...
Just modify the mod_g729b.so such that the licensing constraints aren't
so problematic...
A little bird said it shouldn't be hard to do so...
Matt Hardeman
PaperSoft
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jan
try in zapata.conf
usecallerid=no before the definition of channel = a
regards
Martin
On 13 Aug 2003, Mark Farver wrote:
On Tue, 2003-08-12 at 21:59, Steven Critchfield wrote:
To answer on the first ring turn off callerid support. If you need
callerid support and answering on the first
Can I use the Xten-Lite with
Asterisk? I am experimenting with a total soft solution for a carrier that uses
the Vocal Data platform (Hosted PBX). I would also like to know how I could use
asterisk to connect the carrier without using any gateways. I would like to do
something that Avaya
Hi listers,
is my question about stability of * a wrong question to ask here?
Nguyen
Date: Wed, 13 Aug 2003 10:39:43 +0700
To: [EMAIL PROTECTED]
From: Nguyen Nam [EMAIL PROTECTED]
Subject: Stable versions of Asterisk (Was: Re: Fair comparison (John Todd))
Hi,
It's really a problem for new Asterisk
On Thursday 14 August 2003 08:35, Jerk Face wrote:
I am running Mandrake 9.1, and MySQL 3.23.57-1; and yes, I would
think that /usr/src/usr/include/mysql is not the right place for
errmsg.h. What can I do to get around this?
I don't know what you did, because I just installed the MySQL
One variance to the configuration that was described is that I am using
a Cisco ATA186 rather than a 7960 IP phone. I have tried configuring the
ATA with in-band DTMF and out-of-band DTMF both where unsuccessful.
Jay
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
All -
If I supply the box and the OS, what's a price for someone to set it all
up for us and include 4 SNOM-200 or Cisco 7940 phones? We'd like to
handle an 8x16 system (from what I've read, we can use the Zhone for this)
I'd like to see what kind of pricing I can get for this setup, as well
We're moving a somewhat complicated call center over to an Asterisk system,
and I'm looking for documentation on queue/agent configuration. So far I
haven't found anything on the Digium or Asterisk websites, and I was hoping
that someone could point me in the right direction.
Thanks,
Devon
On Wed, 2003-08-13 at 10:59, John Todd wrote:
This is starting to sound like a feature request,
Absolutely, that would be the real icing and cherry on the cake all in
one go.
Totally seamless coms behind a NAT firewall.
IAX, Analog, ISDN, SIP and H323, etc..., if Pavlov could see me now.
--
I have successfully built and made asterisk talk SIP extension
to SIP extension, read all the docs, and about 1000 emails from
the archive.
The trunk side of Asterisk, from the docs perspective, is a
smidgin TDM-centric, Analogue, T1, zaptel.conf etc.
Asterisk cares not about the externally
I second that, I have also started using iLBC and have been very impressed with the quality.. Pity there isn't any support for it in the IP phones yet..
I have tried to get it to work with X-Lite but it wasn't a happy camper.. :)
Later..
Jan Rychter wrote:
Please try to find a better solution.
At 3:26 PM +0200 8/9/03, Siggi Langauf wrote:
On Fri, 8 Aug 2003, cwitte wrote:
There was a thread a few months ago that tossed around some ideas for
using a cisco phone for intercom or paging. I don't have any ip
phones, and wondered if anyone had any luck getting intercom or paging
to work
Hmmm. I never got one like that. (I haven't ever tried SIP firmware,
though...)
Which firmware version are you using?
The phone says:
Application Load ID
P0S3-05-0-00
Boot Load ID
PC03M030
I hope that's what you wanted to know :)
Have you ever had any XML services running on these phones?
On Thursday 14 August 2003 14:40, Ian Blenke wrote:
Brian West wrote:
Welcome to the club... I can't get it working either.
bkw
On Wed, 13 Aug 2003, Eric Wieling wrote:
Is there any way on the iaxtel.com web site to see if my asterisk
is registering and what 700 number is associated
It should work with the standard PSTN but you can get problems if you
connect a X100P to a PBX or ISDN TA. Try editing the wcfxo.c file and
enabling support for ZERO_BATT_RING (uncomment the #define) then rebuild
and reinstall the zaptel modules - you will need to unload and reload the
wcfxo
Does anyone know if it would be possible to play music on hold in the
background whilst playing IVR prompts. I am hoping that this would have
the effect of the background music being continuous when moving between
IVR levels. There maybe a break when moving between IVR levels but the
background
Sorry to say this Gary, but I think you're missing the
whole point behind Asterisk. Why merely try and
'recreate' what PABXs already do and have been doing for
the last quadzillion years. Asterisk, BY DESIGN, is
trying to do, not only the same stuff, but MORE and
BETTER.
So, keeping this
101 - 200 of 501 matches
Mail list logo
