The PBX is a Vodavi. I do not
believe it is digital. No T1 interface. I wasnt sure if you
could use an ATA on one end and another ATA on the other end to create the OPX
off of the legacy PBX.
Paul Conn
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Hi, Scott
This is Michael Jia. So far, I searched the lists and with the following email threads
http://lists.digium.com/pipermail/asterisk-dev/2004-December/008295.html
I don't know what is the current working status now. Maybe somone in the lists
knows.
Thanks
Michael
On 9/27/05, Scott Huang
On 9/27/05, Matt Florell [EMAIL PROTECTED] wrote:
We have finished our tests of the new Digium firmware on the quad T1
cards(TE405P/TE410P). Overall it is a big improvement over the version 1
firmware.
Thanks much Matt, very interesting read. And I love the graphs. Look
forward to your echo
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An interesting read if you still have an analogue phone that does not speak
DTMF tones (yet): http://www.geisterstunde.org/drupal/?q=w48_asterisk
On Tuesday 27 September 2005 19:10, Rajesh Bhairampally wrote:
I am a newbee to asterisk. I recently installed [EMAIL PROTECTED] Everything
went
M. Loiselle:
Bonjour!
We are a small VSP in Vancouver specialising in business VoIP services
(only -- no residential/no wholesale) hosted on Asterisk with all the
features normally associated with a customer-owned PBX.
Most of our customers are outside the VoIP world and therefore not
Hi Jacky,
thx for the feedback
rich.
--- Jacky [EMAIL PROTECTED] wrote:
Hi, Richard,
I still try, but fail with rtp transfer.
2005/9/27, richard Coco [EMAIL PROTECTED]:
I still find out how to let LCS 2005 accept SIP
invite from Asterisk,
Need more help.
Hi jacky,
None else has had a similar problem ?
I suspect it is a simple configuration/setup of the mISDN and pci cards?
Still experiencing the problem, can not use mISDN/g:TEmode/${EXTEN}
syntax.
On Sat, 2005-09-24 at 10:42 -0700, Dias Badekas wrote:
I have set up a * box with two hfc ISDN pci
Recently I put callwaiting=yes in zapata.conf because customers want to
speak to the operator in person, not leave her a voicemail, when she's
busy with another caller. But now she can't transfer either of the
calls (which she can do when there's only a single call).
The operator has an
I investigated further the problem
I tried to disable vm from the extensions that join the queue and the
problem became better.
Now when remote extension is busy, the ring strategy goes to the following
extensions.
But when ALL the remote extensions belonging to the queue are busy, then
the
This is the only information Ive
found. From zapata.conf.sample
; For fax detection, uncomment one of the
following lines. The default is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension (conferences, 101, 1)
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On Wed, Sep 28, 2005 at 06:33:23AM +0100, Ade Agbero wrote:
I have successfully installed oH323, but when I make a call the recipient can
hear me but I can't hear anything at my end, so voice is only travelling in
one direction.
Client Softphone (SIP) = Asterisk = H323 Gateway.
All
On Mon, 2005-08-01 at 18:31 -0400, Joseph wrote:
Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on
open /etc/asterisk/modules.conf and add the following:
load app_meetme.so
save and close file; reload asterisk
Fabio Montemaggiore wrote:
I have install Flash Operator Panel but Asterisk show
this message:
WARNING[3564]: pbx.c:1650 pbx_extension_helper: No
application 'Meetme' for extension
why can't we compile the asterisk coading in windows, it's done in c++ so it
should work in windows as well
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Sorry?
I have used Bicom for a while and the support I get is top notch.
And the end product is working very well and does everything we have
asked of it.
Bicom have even added features for us with no charge and have
implemented them very fast.
Sure its not as flexible as plain asterisk but its
View the source, Luke.
Michael Jia wrote:
Hi,
I am reading Wiki page on IAX2 at voip.org http://voip.org
Besides peer authentication, does IAX2 also encrypts data packets?
What kind of encryption algorithm it is using?
How the session encryption is neogiated?
Is there any information I can
Hello,
Configuration:
Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1
I am trying to sort out the problem:
1. Provider's switch sends SETUP;
2. Asterisk receives SETUP, rings allocated extension but does not
send Setup acknowledge (or any other messages) to switch;
3.
I was reading on the wiki different possibilities of automatically
restarting asterisk every so often. In some places, people mention
they restart it once a day other on shorter or longer intervals. I
believe the main reason people are doing this is because of possible
memory leaks.
I'm
On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote:
why can't we compile the asterisk coading in windows, it's done in c++ so
it's written in C... have you bothered to look at the source code?
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Why on earth would you want to run it on Windows? First off, your
performance is going to go down because of the GUI... oh your call
quality just went down the toilet? Yeah sorry the screen saver just
kicked in. Having issues making calls? Oh sorry we had to reboot
for a critical update.
QueueMetrics does that, together with a bunch of other things (like
showing pauses, logons, calls waiting and more) - see
http://queuemetrics.loway.it
Or you can use the managemente interface and poll it every once in a while.
Bye
l.
In data Wed, 28 Sep 2005 13:02:31 +0200, Mark Elkins
Just press Ctrl-Alt-Del
Usual on windows ;)
Olivier
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Matt
Envoyé : mercredi 28 septembre 2005 15:22
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] Asterisk on
On Wed, Sep 28, 2005 at 12:09:46PM +1200, Andrew Thrift arranged a set of bits
into the following:
Hello,
We have an Asterisk 1.0.9 machine with a Sangoma A101 card fitted, it is
connected to a Telstra OnRamp E1 in Melbourne, Australia. The problem
we are experiencing is extreme echo and
On Wednesday 28 September 2005 14:14, Kanishka Somaratne wrote:
why can't we compile the asterisk coading in windows, it's done in c++ so
it should work in windows as well
oh, and did you try google? how about this:
http://www.digium.com/index.php?menu=astwind
it's a bit of a cheat though
Hi,
I am trying to dial # or *411,
in order to understand what the * box should answer me.
In both cases, I only ear Good-Bye (italian , arrivederci)
dialing #
-- Executing Wait(SIP/555-a2e5, 1) in new stack
-- Executing AGI(SIP/555-a2e5, directory||ext-local|lo) in new
stack
--
hi
is it possible to use asterisk as an sms central to send SMSes
directly to clients on PSTN instead of just communicating with a
central? the telco to which we're currently connected doesn't have a
central
roy
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Hello,
I have just installed asterisk and I would like to connect it to the PSTN.
I have a gateway Cisco 2600, how must I declare it in the file
of configuration (extensions.conf, sip.conf).
thanks for your helping
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Well the simplest is to make the connection insecure with a static ip.
sip.conf
[cisco2600]
host=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
insecure=yes
type=friend
disallow=all
allow= (your codecs)
extensions.conf
[default]
;dial out cisco
exten = _1X.,1,Dial([EMAIL PROTECTED])
As far as
Well the simplest is to make the connection insecure with a static ip.
sip.conf
[cisco2600]
host=xxx.xxx.xxx.xxx
defaultip=xxx.xxx.xxx.xxx
insecure=yes
type=friend
disallow=all
allow= (your codecs)
extensions.conf
[default]
;dial out cisco
exten = _1X.,1,Dial(SIP/[EMAIL PROTECTED])
You're going to need to explain a little more. When you say central
are you talking about an SMSC?
--
Tom
On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
hi
is it possible to use asterisk as an sms central to send SMSes
directly to clients on PSTN instead of just communicating
somesh s wrote:
Hi,
I didn't get any solution in the mailing list.
[http://asterisk.linkx.net/asteriskusers/200409/msg01167]
What should be the next step?
Changing the machine???
Is it machine dependent?...
Regards,
Somesh S. Shanbhag
Have you talked with Digium support?
Their
Or even . http://www.asteriskwin32.com/
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Hi all,
I'd like to know how Asterisk process a RTP data flow.
Is there any clue to find out about this? The rtp.c?
Thanks.
Regards,
Sam
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Hi all,I'd like to know how Asterisk process a RTP data flow.Is there any clue to find out about this? The rtp.c? Thanks.Regards,
Sam
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Hi all,I'd like to know how Asterisk process a RTP data flow.Is there any clue to find out about this? The rtp.c? Thanks.Regards,Sam
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Dinesh,
Yes, that was the problem. I had everything setup as None and apparently for
inbound calls Calling search space was necessary.
Thanks for your help,
Brian
-Original Message-
From: Dinesh Birlasekaran [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 27, 2005 11:04 PM
To:
Hello everyone,
Ok. I am at a bit of a loss and would like someone to point me in
the right direction...(btw www.google.co.za did not give me ANY solutions).
The issue at hand is simple, I get asterisk (1.0.9) to answer the
incoming call with no problems... it does the fax detection
On 9/28/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
The machine has 1 GB RAM. When I boot themachine fresh with asterisk, it uses approximately 80 MB of RAM whenI run top. After 10 days, top shows that it's using 730 MB of RAM.
Are you sure it is asterisk that is using all of that memory?
Are
I haven't app_meetme.so file...
Where I can search?
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Matt wrote:
Why on earth would you want to run it on Windows? First off, your
performance is going to go down because of the GUI... oh your call
quality just went down the toilet? Yeah sorry the screen saver just
kicked in. Having issues making calls? Oh sorry we had to reboot
for a
I have the digital receptionist answering when an incoming call comes in to the main trunk. How do i get it to answer after so many rings or seconds. It seems to pick up on the first ring. I know how to modify the time it takes for a call to go to voice mail but I do not see an obvious setting for
Why on earth would you want to run it on Windows? First off, your
performance is going to go down because of the GUI... oh your call
quality just went down the toilet? Yeah sorry the screen saver just
kicked in. Having issues making calls? Oh sorry we had to reboot
for a critical
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Does anyone know what I need to put in the mgcp.conf to connect to an
adit 600? Also if you know what I need to configure on the Adit600
itself, that would help too.
- --Tod
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment:
Personally, I could care less which O/S the stuff runs on as long as
it runs reliably, and the sys admin understands how to manage whatever
sytem he/she is responsible for.
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so).
Could not agree more with Matt. I have been a linux geek for a long time
and I would think twice before calling Windows a crap o/s as linux feels
crappier when it comes to usability, administration and the pain in
making it work the first time, with due respect to all those who are
contributing to
/usr/src/asterisk/apps/app_meetme.so
/usr/lib/asterisk/modules/app_meetme.so
Fabio Montemaggiore wrote:
I haven't app_meetme.so file...
Where I can search?
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Matt wrote:
Personally, I could care less which O/S the stuff runs on as long as
it runs reliably, and the sys admin understands how to manage whatever
sytem he/she is responsible for.
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several
You are probably the guy whom they are using as reference point, to
screw others.
Why are they not delivering the software and support to several guys who
paid for the software.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Morgan
Gilroy
Hi group:
I would like contact somebody who has experiences connecting an
Aterisk-PBX with Manager API. Thanks. Ezequiel
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Hi Folks,
OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back
to V1.09. Other than downloading the code, how do I do it? I thought
someone once said that I have to delete all my modules or something?
Thanks
Mark
--
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Hi,
I have * 1.0.7 and I have your same problem.
I dunno what version you have but till 1.0.7 simply you cannot.
We have to wait new * versions
Giorgio
amaury BOSSE wrote:
Hi all,
I don't find where you can setup the date (${VM_DATE}) in french for the mail.
Is anybody can help me?
Sounds like an IRQ conflict!
On 9/28/05, Steve Underwood [EMAIL PROTECTED] wrote:
Matt wrote:
Personally, I could care less which O/S the stuff runs on as long as
it runs reliably, and the sys admin understands how to manage whatever
sytem he/she is responsible for.
Extremely good
Personally, I could care less which O/S the stuff runs on as long as
it runs reliably, and the sys admin understands how to manage whatever
sytem he/she is responsible for.
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or
Any ideas?
51] logger.c: [chan_zap.so] = (Zapata
Telephony)Sep 28 00:42:35 VERBOSE[5151] logger.c: == Parsing
'/etc/asterisk/zapata.conf': Sep 28 00:42:35 VERBOSE[5151] logger.c:
== Parsing '/etc/asterisk/zapata.conf': FoundSep 28 00:42:35 VERBOSE[5151]
logger.c: == Parsing
Personally, I could care less which O/S the stuff runs on as long as
it runs reliably, and the sys admin understands how to manage whatever
sytem he/she is responsible for.
Extremely good point... I myself am a Linux person, but manage several
Windows machines (several meaning 25 or so).
[me shrugs]
I read an interesting quote the other day, can't remember where:
A religious zealot subconsiously realizes his position is fundamentally
irrational, so he tries to convert
other people to religion in order to validate that position
:%s/religion/linux/g
Far as I'm concerned, right
-Original Message-
From: Julien Goodwin [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 6:28 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] BAD echo problems with Sangoma and
Telstra
On Wed, Sep 28, 2005 at 12:09:46PM +1200, Andrew
I think that you probably can just use two Sipura ATAs to connect a remote extension. The key question is whether you can just plug a regular analog phone (Like you would have at home) into an extension port on the Vodavi and have it work. If so, you're in business. If not, more
Ignore me.
When I ran make install a rather usefull banner appeared at the end
telling me what to do.
Thanks
Mark
Mark Phillips wrote:
Hi Folks,
OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back
to V1.09. Other than downloading the code, how do I do it? I thought
Mark Phillips wrote:
OK, I'm running CVS Head as of about 3 weeks ago. I want to roll back
to V1.09. Other than downloading the code, how do I do it? I thought
someone once said that I have to delete all my modules or something?
rm -rf /usr/include/asterisk/*
rm -rf
Rm -rf /usr/lib/asterisk/modules/*
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Mark Phillips
Sent: Wednesday, September 28, 2005 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Roll back from
Hi,
Is there a way to reduce the ring time to answer on the Asterisk @ Home
platform.
Currently it sometimes takes 3 UK Rings before asterisk picks up the call.
In AMP I have Setup-General Settings- Extension of Fax machine DISABLED
Below is my ZAPATA.conf file if this is any help.
Thanks in
I have made a simple click to dial using the manager API. What do you need?On 9/28/05, Ezequiel A. Sculli [EMAIL PROTECTED]
wrote:Hi group:I would like contact somebody who has experiences connecting an
Aterisk-PBX with Manager API. Thanks.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk on windows
Why on earth would you want to run it on Windows? First
Hi all,
I am using Asterisk CVS, and I am getting a huge delay in dialing SIP.
This Asterisk box is taking calls from a PABX over ZAP, then dialing SIP
users.
So, a user '0251' dials from his phone, the PABX sends it the my
Asterisk box, no delay, then I get a 15 sec delay, before it actually
What exactly are you trying to do? Get it to say the date in French?
Nathan
gincantalupo wrote:
Hi,
I have * 1.0.7 and I have your same problem.
I dunno what version you have but till 1.0.7 simply you cannot.
We have to wait new * versions
I don't find where you can setup the date
Did
you compile and install libpri *before* Asterisk? I had same problem (among
others) b/c I didn't install in the correct order. Try the awesome
asterisk_update.sh shell script.
Are
you trying to emulate CPE or NET? Try signalling=pri_cpe
Check
for whitespace behind the statement,
We have an Asterisk 1.0.9 machine with a Sangoma A101 card fitted, it is
connected to a Telstra OnRamp E1 in Melbourne, Australia. The problem
we are experiencing is extreme echo and clicking noises.
These are only audible to the calling party, e.g. the person calling in
from the PSTN to
Rich Adamson wrote:
Both probably resulted from some untested/unexpected activity the
developer never addressed for whatever reason.
Moving the mouse?? lol.
Actually I remember this problem on NT4.. the mouse driver used to drag
the system down completely.. it was a complete resource
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a telephone call, but for
Sorry -- I goofed on the sample rates! Apologies!
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's
I am looking for only a few DID's in the above areas using IAX.
If the solution works to my customers satisfaction,
I will be interested in getting whole sale prices.
CA - California, mostly LA.
WA -Washington, mostly Seattle
BC - British Columbia, Canada, mostly Vancouver
FL - Florida, mostly
Hello:
What is the maximum *audio* bandwidth that a TDM-400 card can support?
Can I record a signal at 16 kHz instead of 8 kHz? Can I record at 16 bit
resolution?
If not, is there a card on the market that works with Asterisk that can?
Cheers,
Stephen Bosch
Well I guess im the lucky one then.
I have had no problems with them at all and have been treated very well.
Anyone else had a good/bad experience with them?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT)
Hello
Does anyone have an example of how to use the MONITOR command from an
AGI-script ?
I have tried different methods, but none of them worked :-(
I'm using Python
MIR
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've made some progress. I've turned on the mgcp debugging, and I've
configured the adit to use my asterisk server as a callagent. I keep
getting these errors:
MGCP read:
RSIP 439 [EMAIL PROTECTED] MGCP 1.0
RM:restart
from ***.***.***.***:2427
Verb:
I download and installed ASTCC over the weekend and
I am having an issue where the INUSE flag will not get set back to 0 if the user
drops a call while the balance is being played. All other times it seems to
reset the flag correctly.
I have tried both AGI and DeadAGI with the same
Matt Love wrote:
Is there a way to reduce the ring time to answer on the Asterisk @ Home
platform.
Currently it sometimes takes 3 UK Rings before asterisk picks up the call.
In AMP I have Setup-General Settings- Extension of Fax machine DISABLED
Below is my ZAPATA.conf file if this is any help.
Ric Moseley wrote:
I have the digital receptionist answering when an incoming call comes
in to the main trunk. How do i get it to answer after so many rings
or seconds. It seems to pick up on the first ring. I know how to
modify the time it takes to go to voice mail but I do not see an
Why on earth would you want to run it on Windows? First off, your
performance is going to go down because of the GUI... oh your call
quality just went down the toilet? Yeah sorry the screen saver just
kicked in. Having issues making calls? Oh sorry we had to reboot
for a
Not to mention NT on Alpha and CHRP was a joke, the GUI was not native code
and proper drivers were non existient. At the time MS was hedging their bets
because it looked like CHRP / Alpha might be going somewhere. I had for a
while a Motorola CHRP machine with Daytona on it and it was utter crap
Hello,
I have a problem with the following: When I dial a PSTN number from a
UAC, the call is made through a SIP Trunk (which has a connection to the
PSTN) in Asterisk. The PSTN Gateway returns a 100 Ringing WITH SDP, but
Asterisk forwards the 100 Ringing WITHOUT SDP:
As you can see below, the
I'm not sure if it matters, but I am running Asterisk 1.0.9. I used the
AAH distribution to do the build.
Jason
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Stephen Bosch wrote:
Hello, everybody:
I'm developing an application using Asterisk and a TDM-400 card.
I understand the concept of the difference between GSM and WAV files
when using Asterisk, but I'm not happy with the sound quality with the
GSM compression. It's merely *acceptable* for a
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 28, 2005 8:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk on windows
[me shrugs]
I read an interesting quote the other
No, I have my cisco gateway setup as a SIP trunk in CCM. The calling search
area was my problem on inbound call from Asterisk to CCM. The problem on calls
from CCM to Asterisk was that I had the inbound context setup incorrectly in
sip.conf.
-Original Message-
From: Greg Oliver
One way audio in my experience is always a firewall issue.
-Original Message-
From: Arnaldo M. Pereira [mailto:[EMAIL PROTECTED]
Sent: Monday, September 26, 2005 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk to CCM
Have you
Morgan,
For the most part people have either ignored or dismissed this company
Bicom Systems and it's products Pbxware and Switchware as they don't fit
into the same mould as Asterisk / Linux / APACHE/ Mysql /PHP /SugarCRM.
Bicom Systems want to ride on the Open source systems without giving
back
I found out Voice in one direction only is a bug within oH323, which has been corrected in Version 0.6.1 (see http://www.inaccessnetworks.com/projects/asterisk-oh323)
However, I am using [EMAIL PROTECTED], I don't know if I can install the latest oH323 version on [EMAIL PROTECTED], if so, which
This is an [EMAIL PROTECTED]
installation. I tried both pri_net and pri_cpe.
Checking for whitespace.
- Original Message -
From:
Colin Anderson
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Sent: Wednesday, September 28, 2005 8:57
AM
Subject:
It was the whitespace i think. I did the
asterisk-update script and eliminated any whitespace in or after a line and it
works now.
- Original Message -
From:
Steve Totaro
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, September 28, 2005
Matthew T. O'Connor wrote:
I'm not sure I know about 3, where can I read more about the Polycom's
known issues. Are you talking about the problems with type=friend?
Thanks for you help.
Matt
Hey Matt,
This is what I was referring to from the wiki:
Matthew T. O'Connor wrote:
Here is the entry in sip.conf for one of the problem phones.
[124]
callerid=24 East Sales Office 124
supposed to be?
callerid=24 East Sales Office 124
but I doubt that's the issue :)
am
___
--Bandwidth and Colocation
On Wed, Sep 28, 2005 at 01:02:31PM +0200, Mark Elkins wrote:
On Mon, 2005-08-01 at 18:31 -0400, Joseph wrote:
Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the
We're having issues with the FXO/FXS ports on our Digium TDM cards
sporadically. I'm wondering if anyone else has had these problems, or if
anyone can provide guidance diagnosing or fixing the issue?
The symptoms are that the FXO and FXS ports stop working, usually
after 2-4 weeks of server
Does anyone know how to maximize music on hold quality on
calls inbound from PSTN? I know that it is common to have choppy and static
sounding music on hold when connecting via PSTN but how can that be minimized?
I assume that the bitrates, type of music, etc can minimize the effects. Does
Hello Ronald,
A 180 Ringing is something that should not have SDP because it's out of band
signaling of the exact status of the call, ringing. The PSTN Gateway should
return a 183 Session Progress if it wants to deliver inband audio progress.
Their SIP implementation doesn't look the best
I disagree - I ran exchange 5.5 on a digital alpha using windows nt. At
the time it was the most reliable NT system I had ever seen and it ran
faster than any i386 system. Personally I wish MS would have continued
development on it.
-Justin
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