One thing to keep in mind with some of these phones is that physical quality can not be changed easily, why software is a different matter, though no guarantees.
I have been working with Aastra 480i's for about 14 months and at first they were fairly limited and had many issues. This all turned
Basically you just plug it into an analog interface after installing the
GSM chip.
The voice quality is good even in my office; a sort of radio waves-black
hole. Normally most cellphones just disappear when they are there..
The only problem I have so far is that the TDM400 FXO module does
On 19 Feb 2006, at 06:04, Lee Howard wrote:J Poz wrote: Using an analog line is not an option for my service. My application runs on a ROOT SERVER of an ASP. So I can do anything I want to the server but I can't connect to or get external analog lines. So my options are doing faxing via the
The GXP2000 firmware is not bad for features and ease of use
but still buggy. The hardware is junk to be quite honest and
I don't think firmware will ever fix that. The Aastra 9133i
hardware is 10x better.
I have a few of both here at the moment, and I'm not sure I'd agree with
that. The
I know it's still beta, but don't use the latest firmware in
production unless you can live with an empty display after
transferring a call.
Only a reboot of the phone will give you text on the display again.
I tested and confirmed this with 5 phones.
What firmware are you running? I've
Hi allI have a problem to register a cisco 7960 to an asterisk 1.2.2I defined in sip.conf the next :["phonenumber"]type=friendusername="username"secret="password"host=dynamiccontext=workI am trying to catch the register requests with sip debugwith no success (empty screen).I can only catch the
On 11:07, Sun 19 Feb 06, Chris Bagnall wrote:
I know it's still beta, but don't use the latest firmware in
production unless you can live with an empty display after
transferring a call.
Only a reboot of the phone will give you text on the display again.
I tested and confirmed this
On Sat, 2006-02-18 at 22:04 -0800, Lee Howard wrote:
Traditional faxing (not T.38) pretty much requires a lossless audio
channel. Normally the best way to get this is with PSTN channels/lines
through a Zap device. That said, VoIP channels can be configured such
that they are also
You can use Sprint (Group Telecom) and/or Magma.
Keep us posted about the group meetings..
Thanks,Wojtek
- Original Message -
From:
Richard
OSS
To: asterisk-users@lists.digium.com
Sent: Sunday, February 19, 2006 12:03
AM
Subject: [Asterisk-Users] co-location
On Sun, 19 Feb 2006, Michiel van Baak wrote:
I tried with one phone on both * svn head, *1.2 and *1.0.9
The exact fw version for the phone is something I cannot get
for you now as the phone is back to the shelve.
What is the MAC address of your phones? There are hardware revisions of
the
On Sat, 18 Feb 2006, Michael J. Liberatore wrote:
Well the gxp-2000 has BLF, the polycom 501 does not correct? I had an
astra 480i and it was prety bad, but I was going to test the 9133i for
an inexpensive phone to compete with the gxp2000. The gxp2000 is not
bad though, the new firmware helps
On Sun, 2006-01-29 at 19:15 +, Phil Blundell wrote:
Our HT386s are also a little bit prone to locking up and needing to be
rebooted, but that seems to be a different problem: it occurs less often
than on the HT488, and seems to be triggered by something to do with
call transfers (which we
http://www.jivesoftware.org/
Is anyone running a wildfire messaging server on the same pc
as their asterisk server?
Is anyone specifically running it on an [EMAIL PROTECTED]
installation?
TIA,
Dean
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Traditional faxing (not T.38) pretty much requires a lossless audio
channel. Normally the best way to get this is with PSTN channels/lines
through a Zap device. That said, VoIP channels can be configured such
that they are also lossless. IAXmodem, for example, functions on the
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
On the new Asterisk box my sip.conf
I have a problem to register a cisco 7960 to an asterisk 1.2.2
I defined in sip.conf the next :
[phonenumber]
type=friend
username=username
secret=password
host=dynamic
context=work
I am trying to catch the register requests with
sip debug
with no success (empty screen).
The
I think Unlimitel.ca(Embrun) offer this service.
On 2/19/06, Richard OSS [EMAIL PROTECTED] wrote:
Anybody know ifthere are co-location providers in Ottawa,
Canada? We are planning on co-locating our Asterisk conferencing server.One
more thing, is there an interest in reviving the Ottawa
Hi everyone,
Can anyone give me suggestions on any equipment that can connect from VOIP to a GSMgateway(channelbank that can load up to 30 sim cards and make 30 VOIP-to-GSM calls simultaneously), i hav looked at 2z's Stargate (which has a VOIP card) but the device is really a GSM-to-ISDN device. I
Yes to both.
It works perfectly fine - followed the instructions to the t; a few
things you learn as you install :)
let me know if you need any assistance or if you want us to install it
for you
rajeev
Dean Collins wrote:
http://www.jivesoftware.org/
Is anyone running a wildfire
testing first email to list
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is there any documentation or simple example around for app_sms.so
to get started with it and do two simple tasks:
1. send a message to an sms-capable phone connected to an ATA
2. receive a message from an sms-capable phone and so something
simple with it, even just write it to the debug
If you are sharing a box at an ASP, you might have just
identified the cause of your problems. Faxing is very time
sensitive. With voice, you won't notice or care if there are brief
dropouts of audio. With fax, these will cause resend of the raster line
(hence the long delays). If your box
Any issues with timing etc? up until now I've been very careful to run
my asterisk as a standalone solution.
Any issues with interfacing into other IM platforms? I'm happy to try
this out but I have one client who uses MSN IM for voice a lot (I keep
trying to get him to adopt skype but not
Sorry, I didn't intend to imply I was sharing the server. It's a root server and I control everything on it. The only thing running on it is my application - it's not shared with anyone or anything else.Technical Support [EMAIL PROTECTED] wrote: If you are sharing a box at an ASP, you might
There are still seats open in our March 21st to 23rd Introduction to
Asterisk and VoIP telephony course. More information is available at
www.signate.com.
Paul Mahler
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Asterisk-Users mailing
Phil Blundell wrote:
1. Connect the fax machine to an ATA and have it speak SIP or IAX to *
This will work provided that you can create a near-lossless
communication path between the ATA and the PSTN gateway (which is the
Asterisk box, I assume).
One way of creating that, I would
Why not get 30 GSM Gateway from us at £60
each and then get an asterisk or some voip gateway like A800 and then link it
all up
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid Exeplish
Sent: Sunday, February 19, 2006
10:54 PM
To:
This seems pretty commercial for a non-commercial list!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: Sunday, February 19, 2006 10:33 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Intro to
On Sun, 2006-02-19 at 07:36 -0800, Lee Howard wrote:
This will work provided that you can create a near-lossless
communication path between the ATA and the PSTN gateway (which is the
Asterisk box, I assume).
One way of creating that, I would expect, would be to add another
ethernet card
My 2 cents worth:
I really think that a suggestion that someone in this thread gave
regarding asterisk not being the right medium for this is correct.
Check out http://www.tpc.int/ which implements a email2fax gateway, one
you can implement for yourself, instead of providing to the public.
Make sure you have the correct codec for your platform. If you use an
optimized codec intended for another platform it would have this
problem.
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The short answer is all the officially supported configuration parameters are
in the admin guide and release notes. Options that aren't documented aren't
guaranteed to work between releases.
So, sorry but the current documentation contains all the config options.
Gareth
-Original
hi
after some testing with [EMAIL PROTECTED], i've decided to install my asterisk
server on a slackware (because it's my favourite distro and it is
still suggested here
http://www.voip-info.org/wiki-Asterisk+Linux+Slackware)
, so, i've installed the last 10.2 release, and i've recompiled the
Hello,
Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or
1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax,
and it builds, but I'm not having any luck getting it working. 99% of my test
faxes fail. Reverting to 0.0.2pre20 yields a much higher
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
...
Can anyone offer any advice?
TIA
--
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of BJ Weschke
Sent: 17 February 2006 22:17
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MixMonitor and command
On 2/17/06, Alex Barnes [EMAIL
Because there are cheaper solutions than purchasing 30
gateways that have an RJ11. S/He (sorry abd with
names) would then have to get a channel banker. This
is a lot more costly than some solutions out there.
--- Sam Tam [EMAIL PROTECTED] wrote:
Why not get 30 GSM Gateway from us at £60 each
My GXP-2000 is currently collecting dust. I had
several issues with it. Mainly echo while on speaker.
The other person can barely mae out what you are
saying. Another issue was if the phone recieved to
many calls it would just freeze up and I had to pull
out the plug. Again I have not used it in a
I have both as well,
I mostly agree with you about the display. The buttons are ok but not great
on either.
At the end of the day a phone is for talking and listening and the 9133i is
far superior in that regard. Both the handset and speaker phone on the
9133i are the same as the 480i and
Some people have to stap on others to make them selves
feel good. Very unfortunate.
--- Rusty Dekema [EMAIL PROTECTED] wrote:
I don't think it takes a great leap of the
imagination to infer that
Mr. Kennedy is in fact having the problem he
describes and that,
although it may not be 100%
Interesting discovery on the salesforce appexchange
https://www.salesforce.com/appexchange/detail_overview.jsp?NavCode__c=MF3fid=a033000NIcAAAW
FEATURES
VoIP w/integrated
PBX/ACD/Contact Center:
On Feb 19, 2006, at 6:06 AM, Phil Blundell wrote:
Overall, I'm happier with the SPAs than the handytones, though neither
of them are entirely perfect. Oh well.
Thanks for the update...
I am being told by the freaks at Grandstream that there will be a
firmware update forthcoming to try to
On Feb 19, 2006, at 9:41 AM, Dovid Bender wrote:
Some people have to stap on others to make them selves
feel good. Very unfortunate.
Some people have no sense of humor. Very unfortunate.
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nik600 wrote:
after some testing with [EMAIL PROTECTED], i've decided to install my asterisk
server on a slackware (because it's my favourite distro and it is
still suggested here
http://www.voip-info.org/wiki-Asterisk+Linux+Slackware)
, so, i've installed the last 10.2 release, and i've
On an Asterisk server- yes.
[EMAIL PROTECTED] -
not me.
PaulH
- Original Message -
From:
Dean
Collins
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, February 20, 2006 1:17
AM
Subject: [Asterisk-Users] Wildfire
messsaging
You are wonderful !!!
for this bug, I noticed later on that by removing the second path in the monitor folder ...
I didn't get any error ...
the script was searching inside a file, thinking that it could be a directory where recordings were.
Anyway, Again, Thanks a lot,
JM
On 2/17/06, Dan
So lets pool our knowledge so next time we all get a perfect phone :)
Phones I have used:
GXP2000: We all know about this one, lots of features but you get what
you pay for Echo, hums, old hardware revisions have lots of
problems (screen, etc). The upside includes lots of features, BLF, 4
We had to stop offering the GXP-2000 due to all the same issues
mentioned above. Really not for business use. Have had good
results with Linksys SPA-941.On 2/19/06, mustardman29 [EMAIL PROTECTED] wrote:
I have both as well,I mostly agree with you about the display.The buttons are ok but not
Are the PAP2's you can get branded vonage at staples for free after
rebate still hackable? I read that you cant do it beyond a certain
firmware but wasn't sure if it had to be connected to the internet for
that download or if it ships with that now
-Original Message-
From: [EMAIL
Stutter tone has been used for years, you can dial whenever you want
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Meredith
Sent: Friday, February 17, 2006 3:20 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: SPA-941
Where would it display the status? There are no BLF buttons...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Piazza
Sent: Friday, February 17, 2006 10:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users]
So you have 2 asterisk systems connected, I am doing this for the first
time. Any tips you can give me besides whats on the wiki? I am not
sure the best way to set it up, I want to be able to have the 2
locations act as 1 over their internet connection to each other, I was
planning to use vpn...
Hi,
We have a Ericsson BP250 Phone system setup witht
he following configuration
Telco - Asterisk E405P -
BP250
The system seem to work perfectly on 1.0.9 for a
very long time but there is some functionality we wanted to take advantage of in
the 1.2 version branch so we upgraded.
Currently
I did some testing - More
Information - Hope this helps...
Next thing to try is to maybe move the port
that the Asterisk - BP250 (Group 1/D-Channel 16) resides on and see if that
makes a difference.
Ok here it is, just remember who hooked you up :)
But I don't see anything about fixing a crashing problem that you
described in 5.3
I am running this on 1 phone and 5.3 on 3 others, the ones with 5.3 seem
perfect, the one with 5.3.3 actually locked up once doing a transfer.
Release 5.3.3:
o
I had voicepulse connect but had to transfer IAX2 had non
stop drop outs in audio all the time. Tried everything to fix it, even
with 14ms ping times it just didnt want to work right. I never figured out
why, just canceled. Although i didnt like the no-name on incoming caller
id either
Do you have
allow=speex
in your codecs list in either sip.conf or iax.conf?
if not this this could be the reason.
Also, Speex won't get selected if its not the primary codec on either
side's call initiation. In other words you allow list should look like this
disallow=all
allow=speex
Still beta, but we could not make it crash any more...: We would be
happy about the feedback from volunteers:-)
http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin
http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin
http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin
On 2/19/06, Kevin Bockman [EMAIL PROTECTED] wrote:
nik600 wrote:
after some testing with [EMAIL PROTECTED], i've decided to install my
asterisk
server on a slackware (because it's my favourite distro and it is
still suggested here
http://www.voip-info.org/wiki-Asterisk+Linux+Slackware)
Are you from snom?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Stredicke
Sent: Sunday, February 19, 2006 6:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk and Snom 360
Still beta,
i would assume so, since his address is [EMAIL PROTECTED]
-Dan
On Sun, 19 Feb 2006, Michael J. Liberatore wrote:
Are you from snom?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Stredicke
Sent: Sunday, February 19, 2006 6:19 PM
To:
I have successfully used the Grandstream ATA286 and Linksys PAP2NA. I would
recommend the Grandstream over the Linksys as there is less configuration to
do and it is IMHO more reliable for faxes. I have been able to get analog
data modem connect at 48k on the grandstream whilst cannot get
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4
to receive from analog fax machines. I have never yet been able to get
rxfax working with txfax - my debugs when I try look like the logs in your
email.
Craig
- Original Message -
From: Jesse Guardiani
Hi Guys
I have a problem compiling Asterisk 1.2.4. I am getting this
error
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
Has anyone come across this?
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Please trim your responses; there is no need to quote the entire message
including irrelevant text and signature lines.
On Sunday 19 February 2006 18:25, nik600 wrote:
sorry, i forget to say that i have installed libpri1.2.2 too
installed with make make install
Before or after you compiled
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
MBIT Technologies wrote:
Hi Guys
I have a problem compiling Asterisk 1.2.4. I am getting this error
make[1]: Leaving directory `/usr/src/asterisk/apps'
make: *** [subdirs] Error 1
Has anyone come across this?
I used yum to install speex (although you could quite easily build your own
from source).
# yum install speex
# yum install speex-devel
# cd /usr/src/asterisk
# make clean
# make
# service asterisk stop
# make install
# service asterisk start
This sequence of commands may require variation
Someone on the list a while back suggested that if you were having
problems with call disconnects, to look into a product from Viking
TellecomSolutions called cpc-disconnect:
http://www.vikingtelecomsolutions.com/catalog/model_CPC-1.htm
I received my unit on Friday and put it into place
I've been using voicepulce connect for several months with
very few problems. Occasionally I get "all circuits are busy" messages
when trying to dial out but no too often.
d
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael J.
LiberatoreSent: Sunday, February 19,
I have the following in my dialplan, counts the number of loops and when
it hits greater then 5, exit. It works, but errors initially with,
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or
tolken; Input: +1.
Could somebody tell me why?
Thanks:
;
I have voicepulse connect too. I had occassional problems with
incoming calls, but not many and not recently. Have had more problems
with outgoing calls which is fine for me, as I have more than one
backup (I use voxee as my primary due to lowest price, then
voicepulse, and failing that I can
On Sun, 2006-02-19 at 20:30 -0500, Doug Lytle wrote:
I have the following in my dialplan, counts the number of loops and when
it hits greater then 5, exit. It works, but errors initially with,
syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or
tolken; Input: +1.
Could
trixter aka Bret McDanel wrote:
Could somebody tell me why?
is count defined before it tries to do count + 1?
No it isn't, thank you for the clue. I'll define it.
Doug
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Mike Newton wrote:
My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k
on Debian stable). It could, however, register with another
installation of Asterisk and the settings on the phone (apart from the
SIP proxy address) haven't changed since then.
I was having this very
No timing issues. but we don't have much load anyway - it's sort of a
small biz setup. but it sure looks like it can scale well.
haven't used other IM platforms...
Dean Collins wrote:
Any issues with timing etc? up until now I've been very careful to run
my asterisk as a standalone solution.
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
version = chan_capi-cm-0.6.3
example;
exten = _9555XX,1,NoOp, ${EXTEN}, ${DNIS}
== ISDN1: Incoming call '04' - '9555'
-- Executing
On Sun, 2006-02-19 at 21:05 -0500, Doug Lytle wrote:
trixter aka Bret McDanel wrote:
Could somebody tell me why?
is count defined before it tries to do count + 1?
No it isn't, thank you for the clue. I'll define it.
since you have had a little time to play with this, was
I get the following errors when starting
Asterisk.
== Parsing
'/etc/asterisk/zapata.conf': Found
Feb 19 21:14:35
WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No
such device
Feb 19 21:14:35
ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such
I have a pretty standard
setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case,
SwissVoice IP10S).
My users are complaining
that when they forward their phones to their cellphones on unavailable (i.e.
forward when no-answer), their cellphone only rings once or
On 2/20/06, Nathan Alberti [EMAIL PROTECTED] wrote:
Is there a reason the variable ${DNIS} does not get set with incoming
calls via chan_capi ?
Is it related to the MSN=X in capi.conf ?
Just a guess, are you thinking of ${DNID} instead? There's no direct
mention of ${DNIS} on the wiki
Try this:
Use login ID: 0
Clear the Login ID Field so it's blank
lawyer, IT consultant and actor
Versatile us Aussies :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Malcolm
Sent: Monday, 20 February 2006 1:57 PM
To: Mike Newton
Cc:
Craig Guy cguy at bigpond.net.au writes:
Yes, I have compiled and used rxfax succesfully on 1.0.9, 1.0.10 and 1.2.4
to receive from analog fax machines. I have never yet been able to get
rxfax working with txfax - my debugs when I try look like the logs in your
email.
Craig
Perhaps
Hi,
I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's
running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and
copied all config files from original to the new server. But when a caller lands inside
the queue no queue message
Hello, world!
I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire
X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently
this works. I've read that the Zaptel package won't work on anything
other than Linux, since it's intended to hook into the Linux kernel
in the
Don't you need an
exten = s,1,Answer
???
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rajkumar S
Sent: Monday, 20 February 2006 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Queue Messages now playing
Michael J. Liberatore wrote:
Are the PAP2's you can get branded vonage at staples for free after
rebate still hackable? I read that you cant do it beyond a certain
firmware but wasn't sure if it had to be connected to the internet for
that download or if it ships with that now
My
David Ankers wrote:
Don't you need an
exten = s,1,Answer
The full sequence is:
[ivr] ; Voice Menu
exten = s, 1, wait(2)
exten = s, 2, Answer
exten = s, 3,Goto,MainMenu|s|1
[MainMenu]
exten = s,1,Background(Welcome)
exten = s,2,Queue(callcenter|tT|||600)
extern = s,3,Hangup
I am sorry that
In queues.conf:
[queuename]
announce-frequency = XX ; where XX = number of seconds
Rajkumar S wrote:
David Ankers wrote:
Don't you need an
exten = s,1,Answer
The full sequence is:
[ivr] ; Voice Menu
exten = s, 1, wait(2)
exten = s, 2, Answer
exten = s, 3,Goto,MainMenu|s|1
[MainMenu]
pre25 is working fine for me.
On 2/19/06, Jesse Guardiani [EMAIL PROTECTED] wrote:
Hello,
Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or
1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax,
and it builds, but I'm not having any luck
Title: Message
Hi,
I
believe that you have inverted fxo_ks and fxs_ks into your zapata.cong file
"signaling=" declaration...
Invert
and redo the tests.
Good
Luck !
Francois BERGERET,
France.
-Message d'origine-De:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part
Peter Fern wrote:
In queues.conf:
[queuename]
announce-frequency = XX ; where XX = number of seconds
I had already given it. From my orig mail:
[callcenter]
music=default
leavewhenempty = yes
monitor-format = wav
strategy=rrmemory
timeout=15
retry=5
servicelevel = 60
wrapuptime=5
David Ankers wrote:
Try this:
Use login ID: 0
Clear the Login ID Field so it's blank
Thanks but no, already tried that and no difference. I also tried three
other different versions of Asterisk: 1.2.1, 1.2.4 and on a whim
downgrading to 1.0.2, and tried the conf files from the old
Has anyone have interfaced this successfully? I came to know
from M$ that Genesys GETS can be used to interface asterisk. I have
interfaced Cisco call manager to asterisk/ser but for my final setup I would like
to have a LCS talking to a CCM, without having the Genesys GETS is I dont
have
Do you know what the problem was? It was that I had three default
routes (all identical). This affected nothing adversely except for
Asterisk.
So, there you go.
--
JEREMY MALCOLM [EMAIL PROTECTED] - lawyer, IT consultant and actor.
Internet and Open Source specialist. Web site:
Anyone have a great reference for configuring the PAP2-NA with Asterisk?
-Original Message-
From: Ed Greenberg [mailto:[EMAIL PROTECTED]
Sent: Sunday, February 19, 2006 11:57 PM
To: Michael J. Liberatore
Cc: Asterisk User List
Subject: Re: [Asterisk-Users] Fwd: Which ATA device do you
On Mon, 20 Feb 2006, Asterisk - Mailing List wrote:
We have a Ericsson BP250 Phone system setup witht he following configuration
Telco - Asterisk E405P - BP250
The system seem to work perfectly on 1.0.9 for a very long time but there is
some functionality we wanted to take advantage
Hey Alex,
Please forgive the question, but what is the rationale behind using Solaris
over Linux as an asterisk hosting platform?
Cheers,
Mark
-Original Message-
From: Alexander Burke [mailto:[EMAIL PROTECTED]
Sent: Monday, 20 February 2006 3:45 PM
To:
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