Hello.
Is it possible to configure Asterisk to preserve specific SIP INVITE headers
when setting up a call?
Specifically, I have a custom SIP client that sends an additional header in
the INVITE request when originating a call. This is to request that the call
is auto-answered by the destination
A bizarre problem on this host running Asterisk. I don't actually think
this is Asterisk's problem, but I don't know who to ask, so if this is OT,
please redirect me.
CentOS 5.2 64-bit Xen domU running Asterisk 1.4.22 32-bit with a number of
SIP phones and a few SIP-PSTN gateways.
When asterisk
On Thu, Feb 05, 2009 at 02:22:12PM +0700, Asfihani wrote:
Hello,
I have an issue with Digium TDM 400 card series. When I try to make
outgoing call (PSTN call) for example, the Zap channel could not be
created and busy channel message appeared. Below is the full log :
[Feb 5 09:26:17]
Scott McNab scott.mc...@gmail.com writes:
Call-Info: sip:192.168.100.50;answer-after=0
Is it possible to configure Asterisk so that it forwards this SIP header
intact?
I know that it is possible to set up a dialplan to insert this header for
specific extensions, but I really would like to
Hi
To dial an outside line i have to dial 0. I want to have that when we dial
local numbers, that is we are in the 08 area, I don't want to have to dial 08,
how to set this up in asterisk 1.6?
Regards
/ralf
Ralf Träskman, IT
AdLibris AB,
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
// Here comes the context where user is prompted for a dest. number:
Hi!
Is it possible to define a peer where authentication is performed only
on the auth-username in the Proxy-Authorization header? Thus allowing
From username / auth user mismatch?
thanks
klaus
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On Thu, 5 Feb 2009, Ralf Träskman wrote:
Hi
To dial an outside line i have to dial 0. I want to have that when we
dial local numbers, that is we are in the 08 area, I don't want to have
to dial 08, how to set this up in asterisk 1.6?
Are your local numbers a fixed length? If so, this might
Hi
Yes i have tried to get them to dial the whole number to, but no luck. Ill try
your suggestions.
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
Sent: den 5 februari 2009 14:50
To:
my fault
regards
klaus
Klaus Darilion schrieb:
Hi!
Is it possible to define a peer where authentication is performed only
on the auth-username in the Proxy-Authorization header? Thus allowing
From username / auth user mismatch?
thanks
klaus
Hi
I have asterisk 1.6 and running queues with realtime mysql. I am trying to set
another musiconhold then default but I cant get it to work,
I have an musiconhold entry in my queue_table, but don't know what to put in
there and where to put the file.
Regards
/ralf
According to the documentation, this would also work:
exten _X.,1,Noop(Local number)
you would probably want to do it with 5 X's since most * use 3 or 4 digit
extensions.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Thursday, February 5, 2009, Ralf Träskman wrote:
To dial an outside line i have to dial 0. I want to have that when
we dial local numbers, that is we are in the 08 area, I dont want
to have to dial 08, how to set this up in asterisk 1.6?
I have this in Asterisk 1.4. My local area numbers
do you condifure the new musiconhold in the music on hold config file (or in
realtime) ?
David
2009/2/5 Ralf Träskman r...@adlibris.com
Hi
I have asterisk 1.6 and running queues with realtime mysql. I am trying to
set another musiconhold then default but I cant get it to work,
I have an
Hmm i hope i do it in realtime, how can I tell?
/ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: den 5 februari 2009 15:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
Hi,
Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a table
listing ATA/Gateways combinations.
Could anyone successfully set a Patton M-ATA to work with another one, using
Asterisk 1.4 ?
Is reinvite (canreinvite=yes) necessary or not ?
Regards
Since this information is available in debug, it is obviously there for the
taking and redistribution. Someone more versed than I will have to give you
a real answer. The Clunky/hack way to get it would be a teed log read
via AGI/AMI.
_
From: asterisk-users-boun...@lists.digium.com
hi
to add a new music on hold you need to add it to musiconhold.conf or in the
realtime table.
see the file you will know how to add a new music on hold.
and then you can make it realtime.
David
2009/2/5 Ralf Träskman r...@adlibris.com
Hmm i hope i do it in realtime, how can I tell?
/ralf
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
It was crashing frequently and seemingly randomly prior to this latest
upgrade. Not sure what version it was running prior to upgrade (it was
probably an old CVS HEAD from 2+ years go.) Anyway, currently
Thanks
I got it working now
/ralf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David fire
Sent: den 5 februari 2009 15:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] musiconhold
I've been running 1.4.21.2 on SUSE 11.0 for about 4 months. In my
experience, the fewer database interfaces you can use, the more stable it
will be.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josiah Bryan
Josiah Bryan wrote:
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
The first place I usually start is with memtest86
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is about to break loose if I can't
stop asterisk
Doug Lytle wrote:
Josiah Bryan wrote:
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
The first place I usually start is with memtest86
Here, here!
Every time I have had problems with a system (not just Asterisk)
crashing and there is nothing in
Roderick A. Anderson wrote:
Doug Lytle wrote:
Josiah Bryan wrote:
I've been using asterisk for 3+ years now, I love it, but it doesnt love
me back. :-)
The first place I usually start is with memtest86
Here, here!
Every time I have had problems with a system (not just Asterisk)
It *is* doing mysql CDR and a whole host of custom AGI scripts. AGI to
mudge the CID, AGI to handle receptionist routing/selections, AGI for
voicemail (not using builtin vm app) - all the AGI scripts do mysql
connections.
Would the CDR connection be a problem?
-josiah
Danny Nicholas wrote:
David Gibbons wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is about to break loose
Thanks for your help,
Unfortunatly neither the xx...@domain.com@domain.com nor the '
xx...@domain.com'@domain.com nor the xxx...@domain.com@domain.com worked
and when I try to do:
register = X:passw...@provider
[provider]
type=peer
host=domain.com
fromdomain=domain.com
Is there a way in the manager API to to tell it not to wait till the
first phone is answered before returning?
Jerry
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Could be. Mine works better using the CSV CDR. MYSQL isn't the stoutest
thing out there and if you're processing the kind of volume other posters
here are, it would wig out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Jerry Geis wrote:
Is there a way in the manager API to to tell it not to wait till the
first phone is answered before returning?
Jerry
I found the Async: yes option.
Jerry
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On Feb 5, 2009, at 9:32 AM, Josiah Bryan wrote:
I've ran with verbose quite high lately, but havn't left debug on.
Well,
I just opened console and turned debug on to 100 so we'll wait and see
what it shows next time it crashes. It's due for another any time
now...
If it's crashing,
Hi,
Tomorrow's guest is Mike Seto from Polycom who will be grilled on
application development for their phones. Coincidentally, there is a
contest for the best microbrowser application. Anyone can join in on
this: http://tr.im/WinPolycom
Join us in the usual places:
IRC #voip-users-conference
First posted at:
http://deancollinsblog.blogspot.com/2009/02/amazon-flexible-payment-syst
em.html
Amazon have just announced they are finally opening up their API's to
their credit card processing platform
https://payments-sandbox.amazon.com/sdui/sdui/business?sn=devfps/o
So does this
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
Well i don't see anything in the doc
please send an email to the list adding [solved] to the subject and write
the probelm and the solution.
this is to improve the list.
Thjanks
and glad to help.
David
2009/2/5 Ralf Träskman r...@adlibris.com
Thanks
I got it working now
/ralf
*From:*
2009/2/5 Olivier oza-4...@myamail.com
Hi,
Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a
table listing ATA/Gateways combinations.
Could anyone successfully set a Patton M-ATA to work with another one,
using Asterisk 1.4 ?
Is reinvite (canreinvite=yes) necessary or
2009/2/4 Mark Michelson mmichel...@digium.com
Olivier wrote:
Hi,
voip-info.org http://voip-info.org is almost silent regarding
udptl.conf except with
Depending on your fax device (such as the Linksys 3102) you may have
to edit the udptl.conf file. The error correction type that is
hello all
This is my first message to the list.
I'm using asterisk and work fine in my enterprice.
My quiestion is if i can use asterisk to authenticate my users of radius on
a service of dialup conections and whats i have installed on my server??
i have an E1 with 30 channels i want destiny 15
Josiah Bryan wrote:
David Gibbons wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is
Josiah Bryan wrote:
Roderick A. Anderson wrote:
How would I go about pinpointing / diagnosing the hardware fault? Not
sure exactly what to do with memtest86 - any pointers?
A lot of distros have memtest86 as a boot option on the CD/DVD. You
select it and let it run. It'll scan for
Hi
I've a requirement for one of my operators for an autodialler for which i plan
to deploy asterisk (I already have 3 asterisk servers on PRI running very well
! ). The scene is like : Asterisk will call a customer and play a prompt that
prompts him to press 1 if he wishes to talk to an
Doug Lytle wrote:
Josiah Bryan wrote:
Roderick A. Anderson wrote:
How would I go about pinpointing / diagnosing the hardware fault? Not
sure exactly what to do with memtest86 - any pointers?
A lot of distros have memtest86 as a boot option on the CD/DVD. You
select it and let it run.
One relevant question that hasn't been addressed is whether just the
application is crashing or the whole computer (Linux).
I would second the hardware idea, with emphasis on generic hardware, especially
RAM. I had a Suse 10 box that kept crashing and doing funny stuff. I ended up
running an
Wilton Helm wrote:
One relevant question that hasn't been addressed is whether just the
application is crashing or the whole computer (Linux).
I would second the hardware idea, with emphasis on generic hardware,
especially RAM. I had a Suse 10 box that kept crashing and doing funny
Notice that one of the prohibited items is:
# Phone Services - includes 800 or 900 phone services and audio text
services, prepaid phone cards, and prepaid phone services.
https://payments.amazon.com/sdui/sdui/about?acceptableuse
--
James Moore
ja...@restphone.com
Wow thanks for catching that - shame.
Also sorry I meant to post it to ast-bus not ast-users list.
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 Message-
From:
Here is a patch file for my change to 1.4.23.1 code
*** app_chanspy.c.origFri Dec 19 07:03:02 2008
--- app_chanspy.cThu Feb 5 09:53:32 2009
***
*** 76,81
--- 76,82
'Agent/1234'.\n
Options:\n
b - Only spy on channels involved in a
Are you locked into the 3U form factor?
We're running Asterisk on a Dell PowerEdge 1950 (1U, 2 full height PCI-E
slots [one home to an AEX-804E], 3 drive bays, redundant power).
I both the 2950 and 2970 (both are 2U, variable number of drive bays based
on the config you choose, the 2950
On Thu, Feb 5, 2009 at 7:22 AM, Geoff Lane ge...@gjctech.co.uk wrote:
The nice thing about that is that if I use MySQL I can run the
management application on another machine, and so don't need to run a
web server on the Asterisk box. However, I wonder whether the overhead
necessary to run
Hi,
Has someone met success setting a Patton M-ATA to work in T.38 ?
In my experiences here, it seems this ATA don't switch to T.38 whenever a
fax signal is heard on its FXS port.
Regards
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Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = s,2,Dial(${rgMain},${RINGTIME},t)
exten
Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten =
On Thursday, February 5, 2009, Mark Michelson wrote:
Actually, jumping to priority n + 101 is a thing of the past, and
this will only occur now if you pass the 'j' option to Dial. Dial
will just go to the next priority on a timeout now, and the
DIALSTATUS channel variable will be set to
Mark Michelson schrieb:
Actually, jumping to priority n + 101 is a thing of the past
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate of change ;-)
BTW, on a related note, I'm having some
Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most
purposes.
*gack*
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
On Thursday, February 5, 2009, Philipp Kempgen wrote:
And in addition extensions.conf is a thing of the past. ;-)
extensions.ael is cleaner and easier to maintain for most purposes.
Oh-oh ... I don't think I can keep up with the rate
I have an application where a caller leaves a voicemail message and then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature in voicemail.conf.
My script has no knowledge of the name of the
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
PrivacyManager() does nothing.
Tilghman Lesher schrieb:
On Thursday 05 February 2009 15:37:19 Geoff Lane wrote:
BTW, on a related note, I'm having some trouble with Privacy Manager
that I'd appreciate some insight with. In one priority, I'm calling
PrivacyManager(2,8). In the next priority, I've got:
Is that true? I was under the impression that .ael was still in use at your
own risk mode.
AEL certainly looks like a real programming language, but I wasn`t willing
to test it out with my dialplan last time I made serious changes.
Mike
-Original Message-
From:
Geoff Lane wrote:
On Thursday, February 5, 2009, Tilghman Lesher wrote:
The correct string is FAILED, not FAILURE.
Thanks. For info, *TFOT says:
PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to
either SUCCESS or FAILURE. If Caller ID is received on the channel,
Adam Robins schrieb:
I have an application where a caller leaves a voicemail message and then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature in voicemail.conf.
My script has no
I would replace the call to Voicemail() with your own recording and
generate a unique filename with a call to a script. Use Record().
Then, when the message is done, call another script to rename the file
to something that has a word like COMPLETE in the filename.
So, for example, assign
Another option is to have your post-processing script that watches the
new voicemail files check the output of 'lsof' to see if the asterisk
process is currently writing to any given file, and only e-mail those
that do not have open file descriptors on them.
--
Alex Balashov
Evariste Systems
I seem to be having a habit of following up my own posts lately. For
posterity, if anyone is experiencing the errors below, there is a good
chance the telco has configured inbound signalling on an RBS T1 as Pulse
Dial. I finally spoke with an engineer at the telco today and they
switched it
Adam Robins wrote:
I have an application where a caller leaves a voicemail message and then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature in voicemail.conf.
My script has no
Alex Balashov schrieb:
I would replace the call to Voicemail() with your own recording and
generate a unique filename with a call to a script. Use Record().
Then, when the message is done, call another script to rename the file
to something that has a word like COMPLETE in the filename.
Sounds scammy. Do the customers know that this autodialer will be
charging them?
j
On Thu, 5 Feb 2009, Kinjal Dixit wrote:
Sriram:
whats going on here??
unless you are developing a vas, in which case, the provider for whom you
are doing this will have to help you. each provider would
On Thu, Feb 05, 2009 at 05:04:11PM -0500, Adam Robins wrote:
I have an application where a caller leaves a voicemail message and then
I need to gpg encrypt the file before emailing it.
I wrote a perl script to do this, which is executed after a message is
left, using the externnotify feature
Hello all
I looking for answer for may problem on the web i found this message : i need
to do same in asterisk or other software in this moments i have a card tdm 410
to test an will buy an TE122B to my services
laptop with modem - PSTN line - Asterisk server - Server's broadband
internet
Josiah Bryan wrote:
snip
Problem is that its crashing for seemingly no reason at all, no errors
on the console, no logs (that I can find), nothing in /var/lib/messages
- its puzzeling! Management is screaming like banshees, calls are
dropping like flies, and all hell is about to break
App nvfaxdetect() works fine for that purpose on both Zap and mISDN.
See http://www.voip-info.org/wiki-NVFaxDetect
--
exvito
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On Feb 5, 2009, at 6:19 PM, Tzafrir Cohen wrote:
Known issue of 1.4.22 . Fixed in 1.4.23 .
If you want to fix it yourself, you can find a patch for it in our
SRPM:
http://updates.xorcom.com/astribank/elastix/repo/
Thank you. Problem solved.
Rgds,
Asfihani
When he examined the motherboards, both had capacitors around
the CPU that had visibly 'ballooned'
A good reason to look for motherboards with either Tantalum capacitors or
Organic capacitors. Its a marketing point I'm seeing these days, and as a
design engineer, I can say its worth looking
I use sipphone.com as my sip provider for testing. If I use the following
AMI packet and extensions I connect to the QueueAnswer extension, as if the
called party answered, before the called phone even rings once. This
prevents me from getting no answer or busy status returned from the Dial
Benoit wrote:
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
Well i don't see
f...@hotbox.ru wrote:
Benoit wrote:
f...@hotbox.ru a écrit :
Hi everyone!
I've set up asterisk ip-pbx to implement IVR menu and encountered such a
problem: when users dial the destinaion phone number and end it up with
# asterisk still waits until timeout in WaitExten() is reached.
Thanks for your help.
In case anyone is interested, I managed managed to get it to forward the
Call-Info SIP header using the following extension config:
exten = _X.,1,SIPAddHeader(Call-Info: ${SIP_HEADER(Call-Info)})
exten = _X.,2,Dial(SIP/${EXTEN})
Thanks again,
Scott
On Thu, Feb 5, 2009 at
Hi all,
i have now created a sourceforge project for the source made public by
mexuar - you can find it at http://sourceforge.net/projects/javaiaxphone/
Take a look at
http://lists.digium.com/pipermail/asterisk-users/2009-January/224730.html
for more information about it.
best regards,
On Thursday, February 5, 2009, Mark Michelson wrote:
I've tried it and you're correct. So it looks like the docs need a
bug report - any idea how I go about that?
Thanks again,
If you're using the 2nd edition of the book, check the preface, page xix for
contact information.
Thanks -
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