But I don't think Digium is in a hurry to implement such a
feature since it forces people to buy more licenses than they really
need to avoid dead calls.
I don't think they're in ahurry either, but I doubt that whatever
their commission on the $10/channel fee is has a big impact on their
On 2/3/06, nik600 [EMAIL PROTECTED] wrote:
On 2/3/06, Script Head [EMAIL PROTECTED] wrote:
Yes, it is possible. You need to track the queue log and channels via
manager console or by tailing logs in real time and then match the
destination of the caller by the callerid. Then make the
On Sat, 2006-02-04 at 10:32 +0100, Wilson Pickett wrote:
I don't think they're in ahurry either, but I doubt that whatever
their commission on the $10/channel fee is has a big impact on their
annual sales :)
Their commission is about $9/channel according to pricing available at
the registrar.
This is probably a stupid question, but how do you specify multiple
fallovers? I.e., if provider1 is not reachable/busy, try provider2.
If provider2 is down, try provider3. If provider3 is down...etc. I
understand how to do it the old way, just keep adding 101 to the
extension. What would you
This is what I use, more or less:
http://mundy.org/blog/index.php?p=73 , go down to Incoming Call
Context (about 1/3 down). I had to modify it a bit, as I actually
need Asterisk to pick up and listen to some DTMF digits before hanging
up and calling me back, but it works great for me, and
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote:
This is probably a stupid question, but how do you specify multiple
fallovers? I.e., if provider1 is not reachable/busy, try provider2.
If provider2 is down, try provider3. If provider3 is down...etc. I
understand how to do it the old way, just
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the load balancing.
Regards,
Umair Bari
On 2/4/06, Michiel van Baak [EMAIL PROTECTED] wrote:
On 04:47, Sat 04 Feb 06, Joseph Tanner wrote: This is probably a stupid question, but how do you specify multiple
On 18:02, Sat 04 Feb 06, Umair Bari wrote:
Dear Michiel,
Would you be kind enough to put more light on RAND stuff. How you do the
load balancing.
Regards,
Umair Bari
Umair,
Here is a actual copy/pasted block from my
[outgoing-speakup]
I have a block like this for dutch numbers,
On 2/4/06, Eduard B. Cleofe [EMAIL PROTECTED] wrote:
Hi Guys,
Im planning to setup a server that has a 64bit processor and
32bit digium card using 64bit kernel of Linux.Id like to know if il be
having a problem later on its compatibility and the availability of drivers
or patches
Hi,
I've just noticed my Asterisk setup is having a small issue.
- Whenever I get a call (from VoIP provider to my Asterisk box, forwarded to
my GXP-2000 phone through SIP registration) I get perfectly clear audio,
both ways.
- When I call out with the phone (Phone to asterisk box through SIP
Hello asterisk-users,
I recently set up an asterisk server using Debian Sarge. I also added
an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed
a new kernel (2.6.15.1) with modular support for the CAPI stuff and
also integrated the FritzCard driver available from AVM.
capiinfo
On Fri, Feb 03, 2006 at 11:32:32PM -0500, Wai Wu wrote:
A better solution is write special modules for different language
to say 1) a string of digits 2) numbers 3) currencies
Translated into Asterisk jargon: patches adding support for Chineese
into say.c would be welcomed.
Luckily, HEAD
Hi Christian,
difficult to say for me. I would just recommend another config which runs
stable on my i586-based embedded system:
mISDN (latest CVS) and chan_mISDN (latest CVS as well) . I used this with a
FRITZCard PCI and now switched to a HFC-S card and have tested that
sucessfully in TE
Christian Schmidt schrieb:
[..]
- asterisk 1.0.7.dfsg.1-2
- chan_capi 0.3.5-11
Do your self a favour and get chan-capi_cm of Sourceforge
http://sourceforge.net/projects/chan-capi
--
Best regards
Peer Oliver Schmidt
PGP Key ID: 0x83E1C2EA
___
On Sat, 4 Feb 2006, Christian Schmidt wrote:
Hello asterisk-users,
I recently set up an asterisk server using Debian Sarge. I also added
an ISDN card (AVM FritzCard PCI) to the machine. I built amd installed
a new kernel (2.6.15.1) with modular support for the CAPI stuff and
also integrated
Good question, i would like to know the same. Im using MAGI patch to
execute AGI commands via the Manager. I have a PHP proxy connected to
the CallManager PHP server that do the routing stuff and decide to
execute Dial, Voicemail, Playtones, receive DTMF or some other stuff in
the channel, i have
Not a chance, they sell SPA3000's by the truckload. If you only need one
line, then go with the SPA3000, if you need more, I would go with the
Mediatrix 1204.
Kerry Garrison
Director of Technical Services
Tech Data Pros - Orange County's Mobile IT Service Provider
(949) 502-7819 x200 - [EMAIL
http://www.sipura.com/products/spa3000.htm
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dovid Bender
Sent: Saturday, February 04, 2006 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
Does anyone have any experience (good or bad) with ArtDio gateways?
I am having two problems, the configuration does not seem to be sticking
(part does, part does not) and it ignores * commands from the phone. I
checked and the phone is definitely sending the *.
Thanks for you help
[EMAIL
It's something like exten = 15,1,Dial(Console/DSP)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony
Azzopardi
Sent: Saturday, February 04, 2006 2:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] How
Hello Armin,
Armin Schindler, 04.02.2006 (d.m.y):
You really should update to new chan_capi-cm version (you can find it
on sourceforge.net).
OK, I gave that a try.
Now, my server is running asterisk 1.0.10 with chan_capi-cm from
SourceForge.
When calling asterisk from my phone, it rings and
Well, I posted this question a few days back and got no answers, I
just figured it out, so here's the answer to part of it and maybe
someone can still answer the how to I get the SIP extension that
called this macro part.
exten = _6[0-2][0-4],1,Flash()
exten =
I was wondering if anyone knows whether or not there is an accepted icon
for a telephony server for use in diagramming programs like Visio/Dia/etc.
The Cisco set has an icon for an IP phone, but I can't find one for a
telephony server.
I'm sure there must be such for telephone switches too,
Do people not use the Grandstream ATA's because they are cheap or
because there is actually a problem with them?
They have a 2 line version for around $50 that I have used in various
locations. I have about 8 or so. They seem to do an excellent job.
-Jonathan
-Original Message-
From:
On Sat, 4 Feb 2006, Christian Schmidt wrote:
Hello Armin,
Armin Schindler, 04.02.2006 (d.m.y):
You really should update to new chan_capi-cm version (you can find it
on sourceforge.net).
OK, I gave that a try.
Now, my server is running asterisk 1.0.10 with chan_capi-cm from
On Thu, 2 Feb 2006, John Todd wrote:
[SNIP]
3) Nobody else has thus far taken the bait and made any comments
about their systems. I appreciate Signate's comments; they seem to be
the only ones to publicly claim large-scale throughput using Asterisk
in a public forum. Most other people
Victor Alvarez wrote:
Hello,
My question is.. How does cdr_manager work? Does it suppose to populate
cdr-csv/Master.csv? What about the cdr table on the database? What is the
event some people talk about?
the cdr_manager.conf file control weather the Asterisk manager should include
the cdr
Please let me know when you are going to do it. My clients typical requirement
is a few hundred license.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of trixter aka
Bret McDanel
Sent: Saturday, February 04, 2006 4:47 AM
To: Asterisk Users Mailing List -
I am confused due a side effect produced in my * installation.
It consists of 1 Sangoma A101 E1/lSDN PRI card connected to Telmex
service
16 analog phones thru SIP enabled SP5004 Micronet gateways 4 SIP hard
phones.
Everything in a local network/no natting.
We are processing nearly 2000 calls/day
It sounds like you both need a Zap card. You can ring the analog phone
and/or the Sip phones when a call comes in on the POTS line that is
connected to the card.
MARK.
Brian J. Murrell wrote:
On Fri, 2006-02-03 at 07:37 -0700, Bromont Quebec wrote:
Well in my setup I have a few IP
Hello Armin,
Armin Schindler, 04.02.2006 (d.m.y):
On Sat, 4 Feb 2006, Christian Schmidt wrote:
OK, I gave that a try.
Now, my server is running asterisk 1.0.10 with chan_capi-cm from
SourceForge.
When calling asterisk from my phone, it rings and rings and rings.
Asterisk
You need to get BT to agree and allocate or port the numbers.
You need to agree how many digits BT will pass on to you
(probably 1925838395 but possibly just the last 2)
I don't know the number of digits that BT pass through on a PRI, but on a
set of BRIs with a range of DDIs, they're passing
i'm planning to migrate a callcenter to asterisk and VOIP,
the call center can have up to 25 cuncurrents agents logged in.
Can a normal server with
1 GB ram
100 GB HDD
Pentium 4 3.6 Ghz CPU
Ethernet 10/100/1000
One of our clients has a similar sized setup running on an Athlon64 2800+
On Sat, 2006-02-04 at 23:33 +, Chris Bagnall wrote:
You need to get BT to agree and allocate or port the numbers.
You need to agree how many digits BT will pass on to you
(probably 1925838395 but possibly just the last 2)
I don't know the number of digits that BT pass through on a
Roger Hill wrote:
I'm picking up the tail end of a thread, so apologies if this is
offtrack...
Have you perhaps got an old set of EXECUTABLES in your path, that are
being picked up before your newly compiled ones?
If you are under linux
rm /usr/lib/asterisk/modules/*
rm
I've been having horrible DTMF problems lately on from Sipura ATAs to
ZAP and IAX. It's primarily with repeated digits. I'm starting to move
my connections to SIP until I can get it all figured out. Other than
updating to the newest SVN trunk I haven't made changes on my end that
should
To quote Kevin:
DTMF handling in the trunk is in a state of flux right now. It won't be
resolved until this weekend.
Don't use SVN for a production system, it's lots broken right now. If
you really must, stick with r8786 for a while.
--Rob
-Original Message-
From: [EMAIL PROTECTED]
Good to know.
I was able to play around and get it mostly working but I'm still not
able to get DTMF working with Jitterbuffer ON for IAX although I
previously could at least with some providers.
I had to define my SIP extensions to use INBAND and set the Sipura
devices to also use INBAND
--- Graziano Poretti [EMAIL PROTECTED] wrote:
any idea where i can find the sip client to embed in my website ? (c# - java
or whatever)
SIP:
http://www.vaxvoip.com/WebDemo/Softphone.HTM
http://www.microappliances.com/site/html/index.php
http://www.etntalk.com/callto/loginany/
Can someone explain the difference between VoiceMail and VoiceMail2?
Thanks!
___
--Bandwidth and Colocation provided by Easynews.com --
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Digium confirmend that this was still the case but trixter may have a
way to at least make things much more efficient and save alot of money,
especially in a recording situation. See his announcemnt here.
http://www.trxtel.com/index.php?page=G_729_Codec
Thanks,
Steve Totaro
You could use big brother or something.
-Original Message-
From: [EMAIL PROTECTED]
Sent: Fri 1/27/2006 7:04 AM
To: [EMAIL PROTECTED]
Cc: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitoring
Hi
On Sat, 2006-02-04 at 22:01 -0500, Steve Totaro wrote:
Digium confirmend that this was still the case but trixter may have a
way to at least make things much more efficient and save alot of money,
especially in a recording situation. See his announcemnt here.
The original quesiton was that if you had a server performing G729
passthrough, could you do recording without licensing. Digium confirmed
that the server doing the passthrough would also need a license in order
to record the conversation. Using the Erlag formula you can pretty much
figure out
On Sat, 2006-02-04 at 22:44 -0500, Steve Totaro wrote:
The original quesiton was that if you had a server performing G729
passthrough, could you do recording without licensing. Digium confirmed
that the server doing the passthrough would also need a license in order
to record the
Have a customer running some 25-28 concurrents calls (with about 35 agents logged in)without problems with a P4 2.X Ghz, 1GB RAM,I'm doing no transcoding btw.Alyed Return-Path: [EMAIL PROTECTED] Sat Feb 04 16:59:29 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
Hi,all
Does asterisk support sip early media?
I have a setup asterisk for sip ATA boxs and a SIP trunk (SIP
GATEWAY) for PSTN access. The ATA can call PSTN phone, cell phone, BUT it cant
receive early media. I am sure the SIP GATEWAY support early media. If
use the ATA connect to
Hi
I am trying to do some simple experiment with
Asterisk . my intention is to communicated two PC in
my lan to voice -communicate with each other with out
extra hardware
I searched the FAQ and wiki for any links
for this , so far I have not found one , It would be
much help
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