Hi,
spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2),
but rxfax doesn't work. After some FAX sounds, it hangup!
Could someone tell me how to debug?
The following is the * CLI> log
to 192.168.0.161:43222
-- Executing NoOp("SIP/4881-bde9", "") in new stack
-- Executing
Welcome to the Asterisk users community!
Asterisk is the leading Open Source Telephony platform,
with support both for classical telephony and IP telephony.
Asterisk.org is a fast moving project. New code is added every
day.
The next community meeting is A
On Tue, May 24, 2005 at 08:18:24AM -0400, Christopher Kenna wrote:
> Sorry about last posting, typo...
>
> I just added 2 Digium X100P cards. When my * box boots, it found them
> and configured them. When I enter genzaptelconf, it comes back with
> the following error:
>
genzaptelconf genera
Ivan Meic (Vox Mundi) wrote:
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the sil
>Actually G.729A is a reduced complexity version, and G.729B is a version
>with silence suppression. The data rate while sending voice is exactly
>the same, although the quality of G.729B should be a little higher.
>However the average rate for B can be lower if the silence suppression
>is used
Since many on this list use cisco ip phones I thought they may find this
information worthwhile to know
http://www.SecurityTracker.com/alerts/2005/May/1014043.html
--
Trixter http://www.0xdecafbad.com
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWo
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> trixter http://www.0xdecafbad.com
> Sent: Wednesday, 25 May 2005 3:22 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] echo problem
>
> On Wed, 2005
On Wed, 2005-05-25 at 14:50 +1000, Terry H. Gilsenan wrote:
> Have you tried Xten's softphone for the ipaq? I am using it on a 5550 and a
> 6325 and the quality is as good as a regular phone. It just worked!
>
> I had echo problems with sjphone on the 5550, and I never even tried it on
> the 6325
Anyone with any comments on DSS buttons and general phone
features?
Thanks,
Shane
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
While you have active calls, type at the cli prompt "iax2 trunk debug".
If trunking is working you should get a reply like:
IAX2 Trunk Debug Requested
Beginning trunk processing
Ending trunk processing with 1 peers and 3 calls processed
If you want to free up more bandwidth add "echocancel=no" t
On 5/25/05, trixter http://www.0xdecafbad.com <[EMAIL PROTECTED]> wrote:
> On Wed, 2005-05-25 at 00:14 -0400, Michael Stearne wrote:
> > Hi,
> >
> > I am a newbie and just discovered AGI (after learning a lot about
> > extensions.conf's language). Before putting in a lot of time on
> > AGI/Perl/PH
Just make sure that the Carier you use on the PSTN side will support
Modem comunications, some cariers use VoIP for the long haul
(surprise) This makes it VERY difficult to get a good connection.
Also make sure that yur clocking on the T1 is solid, Minute frameslips
cause a multitude of errors
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> trixter http://www.0xdecafbad.com
> Sent: Wednesday, 25 May 2005 2:39 PM
> To: Asterisk Users Mailing List
> Subject: [Asterisk-Users] echo problem
>
> I have searched for how to locate echo cance
I have searched for how to locate echo cancelation on SIP clients, but
cant find anything and echocancel=y doesnt seem to have any effect.
Configuration:
CVS-HEAD from last month
iPAQ h5500 with SJPhone (gsm/ulaw/alaw)
Problem description:
When I place or receive a call I hear a faint delayed ech
Aleichem shalom,
The process is completely dependent on the type of IP Phone you are
trying to install.
Are you able to provide more information?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SYED
ADEEL ALISent: Wednesday, 25 May 2005 2:31 PMTo:
asteri
On May 24, 2005, at 9:30 PM, SYED ADEEL ALI wrote:
Assalam Alaikum
I want to know how can i connect IP phone with asterisk... which config
files, i need to configure... plz tell me stepwise ... i m new to
asterisk n
i just used softphones with asterisk
http://www.asterisk.org/index.php?
Assalam Alaikum
I want to know how can i connect IP phone with asterisk... which config
files, i need to configure... plz tell me stepwise ... i m new to asterisk n
i just used softphones with asterisk
Don't just search. Find. MSN Search Check out the new MSN Search!
__
On Wed, 2005-05-25 at 00:14 -0400, Michael Stearne wrote:
> Hi,
>
> I am a newbie and just discovered AGI (after learning a lot about
> extensions.conf's language). Before putting in a lot of time on
> AGI/Perl/PHP I would like to know if its possible to do most of the
> functionality performed i
Hi,
I am a newbie and just discovered AGI (after learning a lot about
extensions.conf's language). Before putting in a lot of time on
AGI/Perl/PHP I would like to know if its possible to do most of the
functionality performed in extensions.conf through AGI. Can AGI be
used as a replacement for s
Title: My VistaPrint Electronic Business Card
Yes, I actually just got it working. I needed to change the
SIPRegOn to 1. I had it on 0. Thanks. It's always something so simple it gets
overlooked.
Adam Collard
General Manager, ER
Wireless
(800) 757-5669 x4861
(517) 242-1800 Cell
Nextel DC 13
Title: My VistaPrint Electronic Business Card
Hi,
If you make an outgoing call, doe sthat call "work"
properly? eg: Audio in both directions works?
I encoundered this once, and the problem was that I needed
to add a port forwarding to the firewall/router for the ports that the ATA was
usi
On 5/24/05, SYED ADEEL ALI <[EMAIL PROTECTED]> wrote:
> I want to know how can i connect IP phone with asterisk... which config
> files, i need to configure... plz tell me stepwise ... i m new to asterisk n
> i just used softphones with asterisk
We need to know what kind of phone you have, is it
Title: My VistaPrint Electronic Business Card
OK an update. I've got it so the ATA can call out, but it
can't receive calls.
Adam Collard
General Manager, ER
Wireless
(800) 757-5669 x4861
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAI
Title: My VistaPrint Electronic Business Card
I've updated my
firmware to 3.1. I've configured the ATA on the web config, but I still can't
call the ATA or Call out.
Adam Collard
General Manager, ER
Wireless
(800) 757-5669 x4861
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]
Title: Junction Networks
They charge 0.029 per minute on DID's, 0.039 on Toll free
DID's.
Adam Collard
General Manager, ER
Wireless
(800) 757-5669 x4861
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug
Har
Title: Junction Networks
These
guys seems to have Canada DIDs. They do not explicitly say that there is a per
minute charge for incoming on DIDs. Are their DIDs flat fee
?
-Original Message-From:
Adam Collard [mailto:[EMAIL PROTECTED]Sent: Monday, May 23,
2005 2:01 PMTo: Asterisk
Hi Michiel.
Could We do that? Could you help me disect your apis so I can better
understand what you are doing? I code in php so mustly what We need to do is
understand what you are doing to your DB, etc.
Let me know what to do.
Thx! And I sure think this is going to be a great addon to the list
hi, I'm with problems:
1- I'm a brazilian and dont speak english
hehehehehe
2- heheh my asterisk dont speak in micprohone in
virtual extension. while extension is local comunication is perfect, but
while a extesion is virtual is possible just listen.
Nevertheless Mozphone looks like a great softphone and the manager windows,
etc gave me some great ideas!
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Jean-Denis Girard
|Sent: Martes, 24 de Mayo de 2005 06:20 p.m.
|To: Asterisk Users Mailing List
Title: My VistaPrint Electronic Business Card
I need to get the
newest firmware for a Cisco ATA 186. The version says v3.1.1 atamgcp (Build 040629A). I'm assuming that it's
the MGCP Firmware, I need the SIP firmware. I do not have a Cisco service
contract and I do not really want to spend the
Steve
Thank you for the enlightenment, that explains allot and it even makes
sense.
So I take it that means that there is no other version as far as Asterisk
is concerned; it only is designed to use G.729A, correct?
Do you know if the data rate, kbps, is a constant or does it fluctuate?
For e
Yep, all easy stuff to do but I'm hoping that Asterisk users start to
post more code rather than each of us inventing the wheel each time we
want something done.
Cheers,
Dean
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Darren Wi
I can put you in touch with a used equipment dealer that sells 24 port
Adtran channel banks for less than $500. I got one and there was one module
with 2 UBRt1E channels, he fedexed the 4 x FXO module to me in Anguilla
right away and didn't require the old one back. Great service andreliable
equipm
IPswitch shows daily the message:
Adding the tip to the native ToolTip control did not succeed.
It pops up all the time, so I cannot even close the program orderly.
bye
Ronald
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://l
hey let me know when its fixed so I can upgrade mine to :)
take care
hank
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
- Original Message -
From: "Bob Goddard" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - N
I've been looking at a similar problem. Mine is slightly different but
it involves a customer phoning in, leaving a recording, and having that
recording delivered to a list of users. I hope to code at least some of
this in the next few weeks.
Darren Wiebe
[EMAIL PROTECTED]
Dean Collins wrot
On Tue, 2005-05-24 at 15:23 -0700, hank smith wrote:
> I missed the numbers can some one repost?
> thanks
> hank
Sure, I have no problems posting *business* numbers for a business, its
not like I am posting their home addresses, phone numbers, pets names,
credit headers, etc. Its nothing personal
> Asterisk 1.0.7 Sounds (sounds.txt) Transcriptions Listing:
>
> http://www.nathanpralle.com/software/ast_soundlist.html
>
> Asterisk-sounds (sound-extra.txt) 1.0.7 Transcriptions Listing:
>
> http://www.nathanpralle.com/software/ast_soundlist_extra.html
>
> Hope someone finds them handy.
WOW,
Actually G.729A is a reduced complexity version, and G.729B is a version
with silence suppression. The data rate while sending voice is exactly
the same, although the quality of G.729B should be a little higher.
However the average rate for B can be lower if the silence suppression
is used. Rig
Hi, I have a client who has asked me to look into the
delivery of 30 second audio messages to a list of opt-in customers. Probably
looking at about 5,000 messages a week over a 6 week period.
I know that this would be a piece of cake to have someone develop
but I thought I would ask her
Is anyone here familiar with configuring Cisco routers? I have a Cisco
3620 with 3x WIC-1DSU-T1, 1x 2FE-2W, and 1x 1E-2W. I have 2 T1 lines
being brought in by ACD.NET, a local CLEC. Within the next 3 months we
will be getting another line from them, as our bandwidth needs grow. I
am new to cisc
channel bank is cheaper per port.
after the faxes and phones are taken into acount don't for get the
multi zone paging and security systems needs.
On 5/24/05, Sean Cook <[EMAIL PROTECTED]> wrote:
> I am looking for a cost effective way to drop analog lines from our
> asterisk system to suppor
Hello All, After a wonderful conversation with Mr. O'Shield he asked me
to inform the Asterisk community that we run an SS7 network with Los
Angeles Area trunks and DID's that we can supply with Fixed
costs and NO per minute charges on our inbound trunks. Calls
include caller ID. Also we can
Anton Krall a écrit :
If I have a hardphone, can mozphone redirect the call to my hardphone
instead of using the softphone? For example, dial using the pc, see callerid
on the pc, etc but if I answer the call, redirect to my hardphone? Or when
making calls, send them to my hardphone?
Sorry I d
had the same problem never worked it out, however there is a script on
the wiki whidh takes ur old confs and ports into the db, i ran that, and
it took care of what should be where :-) but i never got voicemail to
work frm the db, so if u do, give me a shout
Iqbal
On 5/24/2005, "snacktime" <[EMA
Sean,
We setup a support department via just that way. In fact it's about the only
real way to get modems working correctly. We used T100p card attached to
Adtran 750 units. We got them on ebay for around $ 500.00 each. Which is
well worth the cost.
Good luck.
-Original Message-
From:
I would appreciate if people that have experience with stable Asterisk
and TE405P cards installations on Xeon servers can share what kind of
motherboards and chipset are using. I'm trying to configure a new
Asterisk server with one TE405P on an Intel board with Xeon 3.0Ghz but
not all Intel boa
I missed the numbers can some one repost?
thanks
hank
email:
[EMAIL PROTECTED]
gmail:
[EMAIL PROTECTED]
msn messenger:
[EMAIL PROTECTED]
aim:
hanksmith5
skype:
hanksmith5
- Original Message -
From: "Darren Nickerson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial D
Mike - If you don't mind Los Angeles Area DID's then I can supply you
with Fixed costs with no per minute charges on your inbound calls. If
This is what We sell. Please call Michael Schelin at Shelltel 626-814-2354.
Ed Greenberg wrote:
Hi Mike,
Understand that your supplier will be paying b
On Tuesday 24 May 2005 17:07, Daniel ANDRE wrote:
> Bob Goddard a écrit :
> >On Tuesday 24 May 2005 09:35, Daniel ANDRE wrote:
> >>Hello,
> >>
> >>I am trying latest stable Firmware for GS IP Phone BudgeTone 101 found
> >>at http://gs-firmware.gratissip.dk/firmwares/ and the phone doesn't send
> >>
Try
exten => 8661231234,1,agi,/some/directory/dtmf.php
Kyle
Hariharan Gopalan wrote:
Hi
Here is part of my extensions.conf
exten => 8661231234,1,agi,dtmf.php
When I dial this number, this is what I see in my asterisk console:
-- Accepting AUTHENTICATED call from 198.22.67.70:
> reque
Grandstream Handytone 486 or similar.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Sean Cook
> Sent: 24 May 2005 21:59
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Analog Lines
>
> I am looking fo
Never knew this list was for job seeking? Why don't you try to grow
into AT & T why just Vonage?
On 5/24/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> Thanks you for your interest in our project.
>
> Effectively we are looking to set up a small functioning "demo" site so
> that we can show s
On 23:44, Tue 24 May 05, Anton Krall wrote:
> I like you scenario Michiel!
>
> Ok, I have sugarcrm installed on the same box as asterisk...What do I need
> to do in order to, for example, start my softphone or hardphone and when an
> incoming calls comes in, make a window popup (web window that is
How do you use the tel: protocol :)? And the url passing?
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Gavin Hamill
|Sent: Martes, 24 de Mayo de 2005 03:22 p.m.
|To: asterisk-users@lists.digium.com
|Subject: Re: [Asterisk-Users] CTI
|
|On Tuesda
Somebody have a Ericsson WebSwith 100? I had 2 of them, but,
unfortunatelly,I have only one CD that broked inside the drive.
If somebody have the mencioned CD that can send me, I'll be grate.
Bernardino Campos
[EMAIL PROTECTED]
___
Ast
are these phones behind nat?
On 5/24/05, Arnd Vehling <[EMAIL PROTECTED]> wrote:
> Hi,
>
> some of my sip fones which have several external numbers assigned
> are not reachable after a certain timespan. Instead of the fone the
> Voicemailbox is trigger in "busy" mode. After a reboot if the sip-fo
Thats incorrect,
a and b are both the same, 8kbps.
a and b are just another way of calculating it, the result is the same
and a and b are compatible.
todd wrote:
- Original Message - From: "todd" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent
Try to run the script from the command line -- it should be hash-banged,
so you should be able to run it from the shell. When you run it like
that, you should see error messages.
> -Original Message-
> From: Hariharan Gopalan [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, May 24, 2005 3:54 P
On Tuesday 24 May 2005 18:16, hank smith wrote:
> they ever going to fix it?
I sure as hell hope so. Such a bug is a show stopper.
B
> - Original Message -
> From: "Bob Goddard" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, May 2
Title: Normal
I have a couple of Budgetones that I am playing with
trying to get them to work with * from a remote network over the Internet (yes
NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded
(UDP) to my public address (through a Cisco PIX). Internally, I
There is a step by step tutorial at:
http://www.asteriskguru.com/tutorials/firefly_softphone.html (as well as
a step by step for almost every other softphone out there.).
Zoa
Mojo with Horan & Company, LLC wrote:
my iax.conf:
-
[general]
bindport=4569
bandwidth=med
Hi all!
*Very* happy to report that it is working now :)
Indeed for a Dutch (KPN) PRI it seems that you must always connect the
TE110P to the HDSL desktop unit, not directly to the telco line!
For the connection HDSL -> TE110P a standard ethernet patch cable did the
trick.
Thanks for all t
First off.. Just do a: exten => 12345,1,AGI(dtmf)
And try running your php from the console and see if you get debug
issues.
.o---o.
Brian Fertig
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Anyone used the combination above? We are and it sounds like crap. The audio
drops out in regular intervals which suggests that someone's g729 isn't
doing its job correctly.
I'm blameing XTen cause when I make a ulaw call that gets converted to 729
using digium's 729, calls sound fine.
Anyone els
I am looking for a cost effective way to drop analog lines from our
asterisk system to support modems and faxes. More than would typically
be done with TDMxxB cards.
I have looked at going with a T1 interface to Channel Bank, but that
just seems like a very expensive way to solve this problem.
- Original Message -
From: "todd" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, May 24, 2005 12:34 PM
Subject: Re: [Asterisk-Users] G729 codec
Andrew
Thanks for the reply-
Forgive my ignorance in this area but from what I have been
Hi Mike,
Understand that your supplier will be paying by the minute.
What you want is your suppliers worst nightmare. Fixed income and variable
(increasing) costs, both to his upstream provider and also in bandwidth,
both network and computer.
Many of us will be happy to supply you with as m
Mike,
Martin O'Shield here from WindyCitySDR.
>From: mike castleman <[EMAIL PROTECTED]>
>Subject: [Asterisk-Users] origination providers
>To: Asterisk Users Mailing List
>
>Message-ID:
><[EMAIL PROTECTED]>
>Content-Type: text/plain; charset="us-ascii"
>hi folks,
>Has anyone found a good (and,
I guess I need to go over my college electronics again... it's been a while.
Ian
>>> [EMAIL PROTECTED] 24/05/2005 14:18 >>>
Yes, one might think that, IF one didn't understand the nature of
electricity and electrical components.
SOME people also puzzle over the fact that you can't boil eggs on a
If I have a hardphone, can mozphone redirect the call to my hardphone
instead of using the softphone? For example, dial using the pc, see callerid
on the pc, etc but if I answer the call, redirect to my hardphone? Or when
making calls, send them to my hardphone?
|-Original Message-
|From:
Hi
This works for us:
In IAX.conf
[NNN]
notransfer=yes
secret=**
context=contextname
host=dynamic
type=friend
callerid="John Doe"<0>
mailbox=NNN
qualify=no
language=gb
The mailbox is for the mailbox number.
In extensions.conf we use a macro
[macro-iaxext]
;
; Standard extensi
I noticed the same problem back when they "fixed" their problems.
I open a ticket (CAS-23281) on 5/7 and as far as I can tell, it's still open.
I have my outbound CID blocked from the their webpage but it shows a number when I place calls. To make it even more a problem, the
number it shows (202-
Hi,
I've connected a two T100P from digium with a 2 Rhino channelBank.
Everything is working as expected. but I have occasional Falls,
asterisk take 99% of CPU resources, with the following report
May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8ÌFð·n**
May 24 14:52:41 WARNING[11441]: We'r
I am getting a 302 redirect from a SIP Proxy and what
I want Asterisk to do then is dial thru the Zap
Channel. But seemingly, Asterisk, upon receiving the
redirect tries to create a SIP channel and tries to
dial that SIP Channel.
So if the redirect information says that the "new"
contact info is
Hi
Here is part of my extensions.conf
exten => 8661231234,1,agi,dtmf.php
When I dial this number, this is what I see in my asterisk console:
-- Accepting AUTHENTICATED call from 198.22.67.70:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
>
On Tuesday 24 May 2005 20:51, Jean-Denis Girard wrote:
> Have you tried MozPhone ? It has rough corners but is usable, and as a
> Firefox extension, it will give you:
> . popup with callerid on incoming calls (you can accept or reject call),
> . open a web page when accepting call by using url
On Tue, 2005-05-24 at 15:31 -0400, Matthew Crocker wrote:
> > Odds are likely that its 1 carrier they use. Without seeing where the
> > number was ported (requires SS7 access to the carriers involved) its
> > hard to see which carrier is doing this.
> >
> > If it really bothers you call broadvoice
I can give you all the simulataneous calls you need for $.02 / min. in
the U.S. and Canada. Please call me at 626-814-2354. Michael Schelin
Shelltel
Kanuri, Seshu (Company IT) wrote:
Mike,
Many of the providers I've tried contacting either
won't call me back, or want me to s
On Tue, 24 May 2005, Remco Barende wrote:
> On Tue, 24 May 2005, Huddleston, Robert wrote:
> OK, but being from Europe I haven't got a clue what an American SmartJack
> is for :)
>
> Would that mean that I would have to hook up the TE110P to the HDSL
> device? If so, what sort of cable would be
I have set it up, but I get an error, to do with the keys, if I can get
past that part I will have no problems setting up the mappings, dial
rules etc.
Do u have any ideas, on this error?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
Se
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hamish Whittal wrote:
> Hi folks,
>
> I have an asuscom ISDN BRI card in my server and was wanting to know
> whether this would be good enough to use with Asterisk. I am VERY new to
> this, so have no idea how to config the software, etc. But I am ver
I'm not entirely sure what you're asking. The application in question
will involve setting up asterisk in a datacenter where we already have a
fair amount of bandwidth. As far as the DID provider's portion of the
bandwidth, I assume that they would account for this in the rate they
quote us.
I'm j
Well, that really changes things then. I'm not really sure what to tell you
because we've never done it that way. The ciscos are limited in how you can
have them send calls to different servers based upon specific parameters so
you will be limited there somewhat. Is there a specific reason you're n
"Matthew Crocker" <[EMAIL PROTECTED]>:
I know David Epstein and Dan Geopp personally, they are good guys,
posting their direct office numbers on the mailing list is extremely bad
form.
As someone who has been given unacceptably vague, meaningless and often
blatantly dishonest replies from
Michael Stearne wrote:
>> Why do you need a precompiled version? Mac OS X comes with the tools
>> necessary to compile it yourself.
>
> Laziness. I used a binary for Asterisk so I didn't want to try to
> recompile the whole thing.
The short answer is no. There aren't any precompiled versions of
Looking to install asterisk for a client and was shopping
around for prices for 6 POTS lines with or an integrated T1 with voice and
data. I called up Sprint and I told
the sales rep that there was going to be a Phone system she said that they
would have to install “key” rotary lines and t
my iax.conf:
-
[general]
bindport=4569
bandwidth=medium
jitterbuffer=no
tos=lowdelay
[mojatwork]
type=friend
username=mojatwork
secret=somesecret
host=dynamic
context=internal
allow=ulaw
allow=alaw
allow=gsm
allow=speex
mailbox=22
---
and my
Hello,
Thanks for your answer, but i was wondering why the number of threads grow
with time...
Don't a thread have to be stopped when it has accomplished its tasks?
If it hasn't accomplished its task it goes on running, but why do other
threads appear?
So I worried to have a "thread leak"..
Anton Krall a écrit :
Hi Guys!
After 1 week or looking for answers about CTI and Asterisk, I havent been
able to find the necessary applications to do what I want to do. Maybe you
guys have more insight on this.
I tried installing asttapi, and works great! Can make outbound calls from
outlook,
Mike,
> Many of the providers I've tried contacting either
> won't call me back, or want me to sign an NDA just
> to get a rate quote, or some other bullshit.
Assuming that you will need about 12 to 24 simulataneous calls on each
DID you want to run, and you are using Ulaw to get these calls,
I like you scenario Michiel!
Ok, I have sugarcrm installed on the same box as asterisk...What do I need
to do in order to, for example, start my softphone or hardphone and when an
incoming calls comes in, make a window popup (web window that is) and show
the contact window for that particular call
Noah Miller wrote:
1. How do you set the music on hold to work with asterisk. Right now
when I place a call on hold the caller hears nothing. MOH works with
all
my other IP phones.
I don't know the answer to this one (it just works for me), but I
remember it coming up on this list before
Odds are likely that its 1 carrier they use. Without seeing where the
number was ported (requires SS7 access to the carriers involved) its
hard to see which carrier is doing this.
If it really bothers you call broadvoice, and dont bother waiting
in the
queue for techsupprt, go direct :)
All
Ok I'm a little confused about realtime static. The wiki has the
database schema but no explanation of what fields are for what. I
would appreciate if someone can confirm or deny how I think the schema
works.
cat_metric = sort order for category
var_metric = sort order for vars
filename = equiva
Hi folks,
I have an asuscom ISDN BRI card in my server and was wanting to know
whether this would be good enough to use with Asterisk. I am VERY new to
this, so have no idea how to config the software, etc. But I am very
eager to learn.
Cheers
H
___
As
-Original Message-
Message: 18
Date: Tue, 24 May 2005 13:12:44 -0500
does anyone have a sample iax.conf and extensions.conf i could use to get
this working right?
--post your iax.conf entry for 3902999. That looks to be the problem.
Jason
On Tue, May 24, 2005 at 12:31:09PM +0200, daren pereira wrote:
> Hello
>
> I was wondering why the asterisk processes where growing, from one
> process at launch time to more than twenty after some weeks
And all of them happen to use exactly the same ammount of memory? They
are simply diffe
On 11:18, Tue 24 May 05, Anton Krall wrote:
> Anyway that works would be nice :) What I would like to do Is get the
> callerid and then pass that parameter in order to open a http url containing
> the callerid to pass or open a CRM webpage like SugarCRM.
>
Why is it that you want to do this clien
On Tue, 2005-05-24 at 13:27 -0500, Brian Roy wrote:
> On 5/24/05, Johnathan Corgan <[EMAIL PROTECTED]> wrote:
> > I can call my Broadvoice DID from a outbound caller-id blocked phone,
> > and BV happily delivers the CID to Asterisk (and then on to my IP phone
> > display.) I've tested with the *6
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