I have an asterisk server at home. I'd like to test one just installed
elsewhere.
Both servers are behind firewalls. I can see the session start in CLI, my
congratulations is apparently playing and RTP is being sent.
Hearing no audio. Can send key presses and see audio playing changed. Peer
On Mon, 16 Mar 2009 23:00:32 -0700
Michael Higgins li...@evolone.org wrote:
I have an asterisk server at home. I'd like to test one just
installed elsewhere.
And did succeed just after emailing, of course. :(
Sorry for the noise!
--
|\ /|| | ~ ~
| \/ |
Vincent Li wrote:
Hello,
I just had a meeting about a pilot project going on in our University, The
project manager has done some research in the past year and concluded that
Asterisk can not scale well to large user base like 10,000 users, thus
Asterisk is not fit for large University
Hi all,
i know this question is not directly asterisk related - but i have no
idea where else to ask.
We do have around 50 pieces of LinkSys SPA941 - these phones do have a
2.5mm plug connection - and we do have many many headsets we used with
normal PC's before (so 2x3.5mm plug connection).
Hello Giorgio,
you simply pass that parameter along so that from the QueueMetrics agent
page you get that URL opened automagically when you get a call. It's for
interfacing to external CRM apps, usually passing the agent code that
handles the call, the Asterisk unique-id and the caller-id for
Hi,
Sorry for following on this off-topic, but,
On Mon, Mar 16, 2009 at 08:49:53PM -0600, drew einhorn wrote:
The plastics industry says polycarbonate bottles are safe.
http://www.bisphenol-a.org/about/faq.html#g
I'm sure Maggie and here friends would say ALL plastic bottles are
very
On Mon, 16 Mar 2009, Gavin Henry wrote:
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system there will have a max of 12 concurrent calls to PSTN
provided
17 mar 2009 kl. 07.26 skrev zoach...@securax.org:
Vincent Li wrote:
Hello,
I just had a meeting about a pilot project going on in our
University, The
project manager has done some research in the past year and
concluded that
Asterisk can not scale well to large user base like 10,000
On Tue, Mar 17, 2009 at 12:28:25AM +0100, Olivier wrote:
Hi,
Is the following behaviour a bug or a feature ?
A bug. Those two fields should be optional.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406
anyone know of a solution where mobile handsets out roaming the pstn
cellular network can be used and treated as full fleged centrex
extentions, i.e. I can transfer a call that comes in on a wired
centrex copper pair out to a cell phone and the cell phone can
transfer the call back or vice versa
2009/3/17 zoach...@securax.org zoach...@securax.org
Vincent Li wrote:
Hello,
I just had a meeting about a pilot project going on in our University,
The
project manager has done some research in the past year and concluded
that
Asterisk can not scale well to large user base like
Hi all,
maybe I find the problem and the solution.
I move the following parameters on section [general]:
[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
notifyringing=yes
and then on SIP account I put this:
[intphones](!)
type=friend
qualify=yes
host=dynamic
2009/3/16 Alex Balashov abalas...@evaristesys.com
I don't know how good Asterisk's GR.303 support, but you could use DLCs as
well. However, that's a lot of complexity and (seemingly) immature
functionality liability to achieve the same end you'd get with a channel
bank. The only benefit is
How are SipX solutions sold to Universities ?
Are those solutions directly sold by the company mostly contributing to SipX
development, by licenced partners or by local integrator, not having much
commercial link with SipX editor ?
___
-- Bandwidth and
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice the
quality starting to drop, but you could always get two connections and
bond them together into one using openvpn or some other method if you
wanted
Olivier wrote:
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the calling endpoint that
reINVITE the other party to drop current SIP/G711 session and start a
new T.38.
But sometimes, it's also the callee party that reINVITE the calling party.
Which is
What is the status of asterisk now and switchvox now that digium owns
both? Is it expected that both will stay in continued development for
the long term? why would someone use one over the other?
From what I've seen both seem easier to use than trixbox/freepbx which
I found so confusing as to
2009/3/17 Olivier oza-4...@myamail.com
2009/3/16 Olivier oza-4...@myamail.com
Hi,
I'm rather new to this domain so I may be doing stupid things without
being concious of that.
I've got a Patton MATA I'm trying to setup as T.38 fax adapter.
Whenever I connect a fax machine (Dell
Hi,
We have a PRI Trunk (physical E1) and we are getting
some rather weird and very isolocated problems. On outbound calls to
specific numbers, it would seem to me that DTMF from the remote side is
affecting the local asterisk system. Basically what happens:
- We make a OUTBOUND call via the
On Tue, 17 Mar 2009, Geraint Lee wrote:
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice the
quality starting to drop, but you could always get two connections and
bond them together into one using
Hi,
as German phone numbers are variable_length, I need to use direct dial-out.
The problem is that only the part which appears in extensions.ael (and thus
in the argument to Dial()) is logged to the call data record.
What I want, obviously, is for the Dial() app to append the additional
digits
Thanks for the reply..
You should be able to get support from the people who sold you the card.
You need to configure 2 files (I'm looking at an old system, so they
have
the zaptel style names).
My files are below - the thing to note is the span 1,1,0,
the second 1 tells you that the span
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Mon, 16 Mar 2009, Gavin Henry wrote:
Dear all,
I'm currently researching options for a MT asterisk gui/system for a
small business centre that will have 12 units in it. Each unit will be
configured for one extension.
The system
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
We can put about 9/10 calls using SIP/gsm through our BT Business Network
ADSL package connection (832kbit upstream, £65/month) before you notice
the
quality starting to drop, but you could always
I use Polycom 320s.
They have PoE, 2 lines, great sound quality and they work very well with
Asterisk.
They are also about $85 each.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles
Sent:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Mon, 16 Mar 2009, Gavin Henry wrote:
When budgets tight - I've deployed a lot of Grandstream phones - might give
you a bit more breathing space if you use (eg) GXP280's for the client
phones
Vlasis Hatzistavrou (KTI) wrote:
Olivier wrote:
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the calling endpoint that
reINVITE the other party to drop current SIP/G711 session and start a
new T.38.
But sometimes, it's also the callee party that
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
I know of a local company who're regularly putting 20 concurrent calls over
the same broadband setup using G729...
Yeah, we use g.729 ourselves too.
The
We have a system running SNOM 360s, and BLF works fine.
We are trying Grandstream GXP2000s and like the phones for what they are,
but can't get the BLF to work.
The IB just says to set to BLF and put in the phone number. We have tried
variations like adding @xxx.xxx.yyy.zzz, but no lights
I've been using one of the popular asterisk ISO distributions for a
couple of years and DTMF had always worked.
I recently switched to another asterisk ISO distribution, and outbound
DTMF is no longer working.
After doing a bit of digging, I noticed that the new distribution
wasn't setting
Never mind, found magic. We have to set account to the line that
represents that context in Asterisk.
Phone works pretty well for a POE, dual Ethernet, 4 line phone that accepts
a 2.5 mm headset, has 6 line display, and all the expected features for
$79.95. Speaker phone is clear, $9.95
Vlasis Hatzistavrou (KTI) wrote:
Fax transmission can be initiated from any one of the parties. AFAIK
T.38 as well as the PSTN fax standards do not show any preference
whether fax transmission is requested from a or b party.
In practice, the caller usually initiates a fax transmission, but
Hi!
has anyone seen specifications of the codec g711-HD? This is right now
spreading fast in the wake up CATiq (the DECT successor), for example in
the AVM products (www.avm.de).
Googling for G.711-HD only produces hits about AVM. The AVM web site is
very vague.
AVM support
On Tue, 17 Mar 2009, Cary Fitch wrote:
Never mind, found magic. We have to set account to the line that
represents that context in Asterisk.
thread hijack, but never mind...
Phone works pretty well for a POE, dual Ethernet, 4 line phone that accepts
a 2.5 mm headset, has 6 line display,
2009/3/17 Steve Underwood ste...@coppice.org
Vlasis Hatzistavrou (KTI) wrote:
Olivier wrote:
Hi,
I've been playing with T.38.
I observed that mostly but not always, it's the calling endpoint that
reINVITE the other party to drop current SIP/G711 session and start a
new T.38.
2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr
Vlasis Hatzistavrou (KTI) wrote:
Fax transmission can be initiated from any one of the parties. AFAIK
T.38 as well as the PSTN fax standards do not show any preference
whether fax transmission is requested from a or b party.
In
Olivier wrote:
2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr
mailto:vh...@kinetix.gr
Vlasis Hatzistavrou (KTI) wrote:
Fax transmission can be initiated from any one of the parties. AFAIK
T.38 as well as the PSTN fax standards do not show any preference
Hi,
I've read in this mailinglist archives some notes related to Linksys SPA3102
provisioning but I couldn't find there the answer I'm looking for.
Is it possible with this box (mine is unlocked) to store its config file(s)
in a TFTP server, and have this(these) file(s) reloaded at boot time,
Well yeah, even the SNOMs are Engineered in Germany, made in China.
And thanks for the tip on Speaker drop. It actually is Line drop rather
than only Speaker drop, but it works fine.
Cary Fitch
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I'm not sure how this work with Linksys, but with Polycom, you just touch
a file in the TFTP directory (syncinfo.xml), and this causes the phone to do
it's file transfers on reboot. Could be a Polycom thing, but I'd bet
there's a fair chance that they work similarly.
_
From:
Has anyone found a good wayt o do a gate intercom using Asterisk? I am
looking at a Xorcom PBX with programmable contact, so I have no issue
with opening the gate, but the interface at the gate is a bit tricky. I
thought about a weather proof housing containing a phone but it seems a
bit
Olivier wrote:
2009/3/17 Vlasis Hatzistavrou (KTI) vh...@kinetix.gr
mailto:vh...@kinetix.gr
Vlasis Hatzistavrou (KTI) wrote:
Fax transmission can be initiated from any one of the parties.
AFAIK
T.38 as well as the PSTN fax standards do not show any preference
Hi,
The format of the file for the provisioning is xml. You create a file with the
configuration you want and put it on your provisioning server. Then, you put a
rule in the spa3102 to retrieve the file when the unit boot up.
Jimmy
-Original Message-
From: oza-4...@myamail.com
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
i think?
who knows, i might be wrong with all of this but i guess it will work...
2009/3/17 Matthias Urlichs matth...@urlichs.de
Hi,
as German phone numbers are variable_length, I need to use direct
Hi Olivier,
Olivier wrote:
T.38 says that if the call starts in audio mode it is the called end
which should initiate a re-invite to change from audio to T.38. This
makes sense, as that is the end which has the best chance of figuring
out if a FAX machine answers the call. In
2009/3/17 Danny Nicholas da...@debsinc.com
I’m not sure how this work with Linksys, but with Polycom, you just
“touch” a file in the TFTP directory (syncinfo.xml), and this causes the
phone to do it’s file transfers on reboot. Could be a Polycom thing, but
I’d bet there’s a fair chance
Philipp von Klitzing wrote:
Hi!
has anyone seen specifications of the codec g711-HD? This is right now
spreading fast in the wake up CATiq (the DECT successor), for example in
the AVM products (www.avm.de).
Googling for G.711-HD only produces hits about AVM. The AVM web site
Marco Sambo schrieb:
Anyone know how to use busy-level parameter or some other useful parameters?
call-limit=2
busy-level=1
?
busy-level is not in Asterisk 1.4 of course.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany - http://www.amoocon.de
Asterisk:
Tzafrir Cohen schrieb:
Sorry for following on this off-topic, but,
On Mon, Mar 16, 2009 at 08:49:53PM -0600, drew einhorn wrote:
The plastics industry says polycarbonate bottles are safe.
http://www.bisphenol-a.org/about/faq.html#g
I'm sure Maggie and here friends would say ALL plastic
Jimmy Godbout wrote:
Hi,
The format of the file for the provisioning is xml. You create a file
with the configuration you want and put it on your provisioning
server. Then, you put a rule in the spa3102 to retrieve the file when
the unit boot up.
Well, with the other Linksys devices
Olivier schrieb:
Hi,
I've read in this mailinglist archives some notes related to Linksys
SPA3102 provisioning but I couldn't find there the answer I'm looking for.
Is it possible with this box (mine is unlocked) to store its config
file(s) in a TFTP server, and have this(these) file(s)
Hello all.
I have Asterisk connected to an Adit 600 channel bank with a TE110P and
the channel bank is connected to a PBX providing dialtone to the PBX
with fxo_ks signalling. When a call between the PBX and Asterisk
completes there is a momentary battery drop/reversal or something that
The 1.x firmware for Aastra's (for the 9112i / 9133i / 480i) do support some
of the XML functionality that you see in the newer 2.x firmware (for the
more recent models).
I;m not sure if controlling LED status of the keys is supported by 1.x - but
you should be able to find that out by taking a
When I was first looking at Aastra, over a year ago, I thought there was
some talk that Aastra was more supportive of asterisk then most vendors. Is
this still true?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high success rate. Is there any
benefit in moving up to a newer library? I looked at the Changelog in
the source, but it stopped at
Hi,
I stilll continue with the problem but I have noticed something new that
maybe a clue. The noise during the call progress is made by the
appearance of the different lines in the asterisk CLI, I mean, each line
is posted in the CLI generates a noise in the call's signallling tone.
For
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote:
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high success rate. Is there any
benefit in moving up to
2009/3/17 Stefan Schmidt s...@sil.at
Olivier schrieb:
Hi,
I've read in this mailinglist archives some notes related to Linksys
SPA3102 provisioning but I couldn't find there the answer I'm looking
for.
Is it possible with this box (mine is unlocked) to store its config
file(s) in
Ok, I read it.
Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with
field CURCALLS.
2009/3/17 Philipp Kempgen philipp.kemp...@amooma.de
Marco Sambo schrieb:
Anyone know how to use busy-level parameter or some other useful
parameters?
call-limit=2
busy-level=1
David Backeberg wrote:
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote:
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high success rate. Is there
2009/3/17 David Backeberg dbackeb...@gmail.com:
On Tue, Mar 17, 2009 at 10:34 AM, Steve Davies davies...@gmail.com wrote:
While we have your attention Steve (Underwood) do you have a
high-level changelog available for spandsp-0.0.4 to 0.0.5 to 0.0.6? We
currently use 0.0.4 with a very high
At 01:29 AM 3/17/2009, you wrote:
But there is another little problem. On Aastra phone (on other
phones I don't try yet), the xfer button doesn't work, until I set
call-limit=2, but making this I find the phone not busy.
As far as I can tell on my Aastra phones it takes 2 links to complete
a
On Tue, 17 Mar 2009, Yehavi Bourvine wrote:
Hello'
I am at the same situation as you. I also work at a university and we have
over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot.
I am using a realtime users database and the main problem is that Aaterisk
does too
Polycom IP 430 or 330.
asterisk-users-requ...@lists.digium.com wrote:
Date: Mon, 16 Mar 2009 18:24:33 -0400
From: David Ruggles da...@safedatausa.com
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125
On Tue, 17 Mar 2009 10:00:56 -0700 (PDT), Vincent Li wrote:
On Tue, 17 Mar 2009, Yehavi Bourvine wrote:
Hello'
I am at the same situation as you. I also work at a university and we have
over 8.000 extensions on a Nortel PBX. I also run a small Asterisk pilot.
I am using a realtime
2009/3/17 Michael Graves mgra...@mstvp.com
On Tue, 17 Mar 2009 10:00:56 -0700 (PDT), Vincent Li wrote:
On Tue, 17 Mar 2009, Yehavi Bourvine wrote:
Hello'
I am at the same situation as you. I also work at a university and we
have
over 8.000 extensions on a Nortel PBX. I also run
On Mon, Mar 16, 2009 at 5:34 PM, Vincent Li vincent.mc...@gmail.com wrote:
Hello,
I just had a meeting about a pilot project going on in our University, The
project manager has done some research in the past year and concluded that
Asterisk can not scale well to large user base like 10,000
Hi Visit, that's not correct - google Sam Houston University
It's a pretty well known asterisk installation.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original
Very complex installation,
so try to star with:
1) Compatibility of current phone platform + asterisk. For example,
you can convert current extension as sip extension using fxs ports.
This reduces your cost, you don't need buy 8.000 ip phones and install
an new wired network.
2) Planning and do
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C /lib/modules/2.6.18-custom.2/build ARCH=i386
SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi
Two of the wireless carriers have a Centrex-like solution:
http://www.networkcomputing.com/channels/wireless/showArticle.jhtml?articleI
D=202200832pgno=5
Frank
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
make[1]: entrant dans le répertoire « /usr/src/linux-source-2.6.18 »
WARNING: Symbol version dump /usr/src/linux-source-2.6.18/Module.symvers
is missing; modules will have no dependencies and modversions.
specifically Symbol version dump
See too:
http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1
Jorge Mendoza
Dean Collins wrote:
Hi Visit, that's not correct - google Sam Houston University
It's a pretty well known asterisk installation.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
A2billing is a good fit for that then. Yeah, voipon. Thanks for the
input Gordon. Maybe worth hooking up offline if we're doing similar
stuff.
Gavin.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson
Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.
On 17/03/2009, Gordon Henderson gordon+aster...@drogon.net wrote:
On Tue, 17 Mar 2009, Gavin Henry wrote:
2009/3/17 Gordon Henderson gordon+aster...@drogon.net:
On Tue, 17 Mar 2009, Geraint Lee wrote:
I know
If for whatever reason your kernel headers have been corrupted or there
is a new version for your particular kernel version, I would suggest
purging the package and pulling in the package from the repo
-John Knight
John Knight wrote:
make[1]: entrant dans le répertoire «
Steve Underwood wrote:
Oh, it was meant for him. In the time it took him to write his wrong
e-mail he could have gone to the ITU web site, downloaded a free copy of
the T.38 spec, looked up the annex where it described the negotiation
process, and found a clear statement of what is
On Tue, Mar 17, 2009 at 07:16:26PM +0100, Administrator TOOTAI wrote:
Hi,
We installed the latest 1.4.24 on a test machine and can't get zaptel
nor dahdi compile. It's a Linux Debian Etch. Errors we have:
keewi:/usr/src/dahdi-linux-2.1.0.4# make
make -C /lib/modules/2.6.18-custom.2/build
I thought I should also share this :
http://www.opensky.ca/~jdhildeb/software/spaconf/
Has anyone tried ?
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
For MT check out Thirdlane's MT PBX:
http://www.thirdlane.com/products/thirdlane-pbx-mte
I use the PBX Manager which it's based on and it works very well.
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asterisk-users mailing
I finally got around to updating my dialplan to use the new way of doing
callback queues. It seems to me that if one used something like
${CUT(CHANNEL,-,1)} instead of SIP/${EXTEN:3} in the AddQueueMemeber then
the device state of the device the agent logged in from, likely where you
want to call
On Tue, 17 Mar 2009, Gavin Henry wrote:
Yeah, I've experienced that. But what can you do other than stick woth
a fat codec.
It's tricky. I've been experimenting looking at the possibilitys of
using different codecs based on destination, so UK landlines stick to g729
as teh transcode to alaw
Dear all,
I am doing an interop testing with asterisk-1.6.0.5 now, and I have a
question about the G.726 codec on asterisk.
While my IAD supportes G.726-16,24,32 and 40 codecs, when doing a testing
about G.726-40, I found that asterisk removed the G.72-40 sdp attrib when
transmitting the INVITE
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