OK, looks like I made a mistake in the rev number -- its 67457 not
6745 from early Sunday.
The bug was fixed a while ago, but has reared its ugly head again.
on Sunday 07/01/2007 Russell Bryant([EMAIL PROTECTED]) wrote
John covici wrote:
OK, using zaptel 1.4 and asterisk 1.4 rev 6745, if
I will appreciate any ideas concerning this issue
Thanks
Ivoc [EMAIL PROTECTED] wrote:
Hello everybody,
Just was wondering if somebody can help for G711 fax passthrough w/ asterisk.
The issue I have is regarding codec upspeed when the call is already
connected using G729 for
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi List,
I wonder if someone else discovered that behavior and hopefully fixed it.
I've two asterisk boxes, both have a user 102. If 102 from Box A calls 105 on
Box B, Box B want's 102 from Box A to authenticate. But it's an incoming call,
there
Hello,
I have many machines working with asterisk and a Digium 4E1 card.
I build a new machine and a failure started like:
Got reject for frame 46, retransmitting frame 46 now, updating n_r!
This error i got every 2 minutes.
Do anybody has an idea?
Thanks
Nico
Looks to me like the law only targets intentionally deceptive spoofing.
Dovid B wrote:
Anyone know if this is only to bother some one ? I have a client
that has a consulting business. The clients call in and his asterisk
server call's his cell when he is out of the office. It passes along
Hi,
We have quite a large setup working just fine most of the time. We have
60 outgoing lines on PRI and we never use all of these lines. But
sometimes we get the all circuits busy now message, seemingly random.
Sometimes we get it before the call even goes through to PSTN. Sometimes
after 5 or
Alex,
This is not correct. The reason is the calling party will be billed as soon as
you use Answer.
You want to do something like this
Exten = s,1,Dial(SIP/phone,60m)
The calling party will not be billed untill you pick up (I just tested this on
VOIP).
- Original Message -
From:
Hi,
somtimes my Gigaset 450IP loses its registration.
Is there anybody who knows why and how to solve it?
TIA
Giorgio Incantalupo
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To UNSUBSCRIBE or
dear all
I am new in asterisk and i have now setup asterik for 40 phone
now i want to configure call transfer between phone so how it is possible and
what configuration part in asterisk will perfomed for this task give me
suggestion for my solution
Regards
Satish Patel
On Monday 02 July 2007 01:45:44 pm satish patel wrote:
dear all
I am new in asterisk and i have now setup asterik for 40
phone now i want to configure call transfer between phone so how it is
possible and what configuration part in asterisk will perfomed for this
task give
We record calls using Asterisk. We can listen to music-on-hold. We can
play pre-recorded messages to a caller. So is it possible to inject an
existing recording into a conversation? From time-to-time we want to be
able to review all or part of a past conversation with a third party
with them.
On Sun, 1 Jul 2007, Russell Bryant wrote:
Eric ManxPower Wieling wrote:
I believe that Asterisk's app_voicemail uses lock files for locking
mailboxes when creating a message. IIRC, Asterisk records the voicemail
message in a temp audio file, locks the mailbox, moves the file into the
Hi:
Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
Reagrds;
jonnyhashem
-
Don't get soaked. Take a quick peak at the forecast
with theYahoo! Search weather
On Mon, 2007-06-25 at 12:19 +0200, voip crazy wrote:
Hi all,
On one of our client, I must to install an asterisk over a hi ability
cluster. I have no experience with clusters an linux neither asterisk.
Someone has installed an asterisk in a hi-ability enbviroment?
How do you install the
Is it a S 450IP ou C 450IP ?
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satish patel wrote:
dear all
I am new in asterisk and i have now setup asterik for
40 phone now i want to configure call transfer between phone so how it
is possible and what configuration part in asterisk will perfomed for
this task give me suggestion for my solution
Reminder: A few weeks ago, Stephan and Stephen were with us on the
Asterisk Users Conference which you can download the MP3 here:
http://recordings.talkshoe.com/TC-22622/TS-23423.mp3
Or you can listen to it here:
http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
Good luck with the book,
Hi Russell
Again, thanks for info.
2007/7/2, Russell Bryant [EMAIL PROTECTED]:
I can assure you that once we have some plans in place, I will announce them.
I'll wait for it, but I won't hold my breath ;-)
$ svn co http://svn.digium.com/svn/asterisk/team/blanchet/v6 asterisk-ipv6
I
Hi all,
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c:2309: warning: implicit declaration of function
`pri_keypad_facility'
chan_zap.c: In function
Dear all
i have read that document but dont understand about function i
have include featuremap in extension.conf
[mysip]
include = featuremap
and reload extention.conf i got this error
*CLI extensions reload
Jul 2 19:23:04 WARNING[16320]: pbx.c:6444
On Mon, 2007-07-02 at 15:38 +0200, randulo wrote:
Reminder: A few weeks ago, Stephan and Stephen were with us on the
Asterisk Users Conference which you can download the MP3 here:
http://recordings.talkshoe.com/TC-22622/TS-23423.mp3
Or you can listen to it here:
Wai Wu wrote:
I need the zap channels going, but got the following error. What do I
need to change in my configuration? Thnx.
chan_zap.c: In function `zap_send_keypad_facility_exec':
chan_zap.c:2309: warning: implicit declaration of function
`pri_keypad_facility'
chan_zap.c: In function
On Mon, 2 Jul 2007, jonny hashem wrote:
Hi:
Iam looking for a sip softphone that supports g729 codec
Any one have an idea ?
eyeBeam - the commercial version of X-Lite:
http://www.counterpath.com/index.php?menu=Productssmenu=eyeBeam
Gordon
___
Thnx. It is working now. I though that I didn't have to that since I am
just using RobBit.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Colp
Sent: Monday, July 02, 2007 11:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
If you don't need libpri, you could just remove it. The problem is that you
already had it installed, and it was too old for newer versions of Asterisk to
use.
- Wai Wu [EMAIL PROTECTED] wrote:
Thnx. It is working now. I though that I didn't have to that since I
am
just using RobBit.
Eagerly waiting for v1.4.x to mature a bit before getting serious about
it.
Is it ready for production yet? If that's too general, where is it in
terms
of stability compared to where 1.2.x is now. Anyone running it
successfully
in production environment and if so what sort of
or copy /usr/src/asterisk-1.4.6/configs/http.conf.sample /etc/asterisk
and edit, because make samples I believe wipes out existing configs
Russell Bryant wrote:
hugolivude wrote:
I just installed Asterisk 1.4.6. I didn't see http.conf in
/etc/asterisk. Is there a seperate install for AJAM?
I see. That's a better way of doing it. Thank you.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Parker
Sent: Monday, July 02, 2007 11:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help. Cannot
Thanks Russell...
I will apply it to my code and test it again... Will keep you updated...
Cheers,
Nitesh
Russell Bryant wrote:
Russell Bryant wrote:
Nitesh Divecha wrote:
Found some strange problem while Asterisk trying to call the AGI file.
If I pick up the call on the first
Bent Bagger wrote:
I downloaded, built and installed it without problems. However, when I
run it, it aborts after about 8 seconds with a pthread error. Where
would be the best place to ask for help - this list? the
asterisk-devel list? or some specific person?
I guess the right thing to do
hi I need a client to voip, like xlite, that can be integrated inside
jabber,
A few month ago, i saw a software that have include but I dont remember, ..
I dont want to use firefox, .. because in this way I canot use the iax or
othr protocols , only sip,
thanks in advances
On Jul 1, 2007, at 2:41 PM, Russell Bryant wrote:
bilal ghayyad wrote:
I compiled Zaptel 1.4 and Asterisk 1.4 after
downloading them using svn, but when I checked the
file zaptel.conf under etc/asterisk, I did not find
this file. Any help?
zaptel.conf is located at /etc/zaptel.conf, not
Yeah, the only time you should delete *everything* from the modules
directory,
is when upgrading between major versions, such as from 1.2 to 1.4.
what happend this situation
still i need to re-register or
just copy the g729 of 1.4 and copy the license
will this work?
ram
At 15:43 6/30/2007, Jordan Novak wrote:
Content-class: urn:content-classes:message
Content-Type: multipart/alternative;
boundary=_=_NextPart_001_01C7BB57.4F8EC993
I have three polycom 501 that are all hearing echo. The other party
sounds fine but you can hear yourself rather well.
Hi,
I'm looking for a card with 2 or more FXOs compatible
with Asterisk. Any idea where I can get it in Bombay,
India?
Thanks.
Park yourself in front of a world of choices in alternative vehicles. Visit
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
And the calls are
[EMAIL PROTECTED] wrote:
Hi,
We have quite a large setup working just fine most of the time. We have
60 outgoing lines on PRI and we never use all of these lines. But
sometimes we get the all circuits busy now message, seemingly random.
Sometimes we get it before the call even goes
Noah Miller wrote:
Eagerly waiting for v1.4.x to mature a bit before getting serious about
it.
Is it ready for production yet? If that's too general, where is it in
terms
of stability compared to where 1.2.x is now. Anyone running it
successfully
in production environment and if so what
Yes if Asterisk 1.2 has reached End-of-Life you would hope Digium uses
1.4.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Monday, July 02, 2007 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from
http://pkg-voip.buildserver.net
When misdn stuff (misdn-init start) is not started, everything is fine,
our 8 FXO (Channel 1-8) 4 FXS (21-24) are working well.
If we start the misdn stuff (one card, port 1,2,3,4 in
I was going to mention that too, even though I wasn't sure it was a
problem. I have spaces around my '=' but nowhere else in the line.
Stephen Bosch wrote:
Asif Raza wrote:
hi,
i am using Asterisk 1.4. and unable to get Voice Mail below is my config
extensions.conf
exten =
I discoved a real problem this time!
The demo doesn't work. You get a 404 as reported here:
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM)
Here's the call in one of ajamdemo.html's js functions:
astmanEngine.setURL('../rawman')
but there is no rawman
On Mon, Jul 02, 2007 at 09:27:39PM +0200, Administrator TOOTAI wrote:
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from
http://pkg-voip.buildserver.net
Does it have misdn support?
When misdn stuff (misdn-init start) is not started, everything is fine,
our 8 FXO
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different trunks to the outside
worls, so I moved some of them to a Support context.
However, dial out from this
So I've got a Voicetronix card and it looks like the kernel driver works.
Other than the 0's for ID info.
vpb: Driver Version = 4.0
vpb: major = 251
vpb: tmp [0xfc8fec00] dev-res3 [0xfc8fec00]
vpb: tmp [0xfc8c] dev-res2 [0xfc8c]
vpb: 1WS Write cycle
vpb: Manufactured 00/00/
vpb: Card
hi
Can anyone recommend a good DID provider offering numbers in Toronto ?
( 1 very stable 2 support porting numbers from Bell, primus, telus.. )
Mario
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asterisk-users mailing
I've had a good ongoing experience using http://www.unlimitel.ca. They
are responsive and reliable.
MARK.
Asterisk guy wrote:
hi
Can anyone recommend a good DID provider offering numbers in Toronto ?
( 1 very stable 2 support porting numbers from Bell, primus, telus.. )
Mario
It might help to show your Support context in outbound.conf.
MARK.
Alexander Topolanek wrote:
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different
I've had a good ongoing experience using http://www.unlimitel.ca. They
are responsive and reliable.
Ditto here - Unlimitel is small but reliable and supportive.
- Mike
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All,
It's been some time since this thread was alive but we are now seeing
some progress in this project. Which I will document.
We have ordered a T1 for the new building which we are moving (We are
getting 14 channels of the T1.) and have a Sangoma A101 card for a 3U
rack server.
The T1 will
DoH!
make checkconfig
is your friend! I had enabled = yes in manager.conf, but not
webenabled = yes. I also found that you must have an httptimeout set.
If you don't then the examples found here:
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM)
will not
On Mon, Jul 02, 2007 at 10:54:14PM +0200, Alexander Topolanek wrote:
Hi,
recently I changend a few things in the configuration of the Asterisk
1.2.17-BRIstuffed-0.3.0-PRE-1y-d of a customer. One demand was that
different groups of SIP-Phones are using different trunks to the outside
worls,
Same here. We have commercial call center clients on Unlimitel. They've
had a few outages during business hours, but Unlimitel is responsive.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: Monday, July 02, 2007 5:29
On Jul 2, 2007, at 4:31 PM, James R. Stevens wrote:
All,
It's been some time since this thread was alive but we are now seeing
some progress in this project. Which I will document.
We have ordered a T1 for the new building which we are moving (We are
getting 14 channels of the T1.) and
On Tue, Jul 03, 2007 at 12:04:08AM +0200, Administrator TOOTAI wrote:
Tzafrir Cohen wrote:
On Mon, Jul 02, 2007 at 09:27:39PM +0200, Administrator TOOTAI wrote:
We have an Ubuntu Dapper with 2.6.14 kernel, asterisk 1.2.14 debs from
http://pkg-voip.buildserver.net
Does it
To determine versions:
for zaptel, use
ztcfg -v
For asterisk, in CLI:
core show version
Moj
bilal ghayyad wrote:
By the way: How can I know the asterisk and zaptel
version extactly that I compiled them? In other words,
asterisk 1.4 and zaptel 1.4 ?
Ryan Goldberg wrote:
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
I've tried doing things like this. What got me was that SIP technology
allows for the phone not
Eric ManxPower Wieling wrote:
Maybe Digium can start using 1.4.x on THEIR production boxes like the
Digium corporate PBX and IAXTel.
Not doing so is like the city water department saying that the water is
safe to drink, but every employee of the city water department drinking
bottled
Russell Bryant wrote:
Lacy Moore - Aspendora wrote:
On 6/29/07, Ade Vickers [EMAIL PROTECTED] wrote:
What I'd like to do is have the music streaming constantly, so the on hold
caller always gets music at the current position; even if that's in the
middle or near the end of a file.
Many of
The Proposed bill S704 reads It shall be unlawful for any person within the
United States, in connection with any telecommunications service or
IP-enabled voice service, to cause any caller identification service to
transmit misleading or inaccurate caller identification information,
Please tell
Hello Dovid,
On 7/2/07, Dovid B [EMAIL PROTECTED] wrote:
Alex,
This is not correct. The reason is the calling party will be billed as
soon as you use Answer.
You want to do something like this
Exten = s,1,Dial(SIP/phone,60m)
The calling party will not be billed untill you pick up (I just
Please tell me how you can construe making a call with the the CID of a
number in your control to be Misleading or inaccurate
Sure - it goes like this - The less scrupulous among us might use a spoofed cid
to get people to do something they normally
wouldn't. Imagine a spoofed CID of your
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