Hi guys,
I am trying to set the caller ID to 'Anonymous anonymous' if the caller is
not registered to the asterisk server. But I can't find a solution.
Any ideas?
Regards Philipp
--
Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL
für nur 16,37 Euro/mtl.!*
Ok, now for my long mail...
tir, 13 01 2009 kl. 09:05 -0700, skrev Steve Murphy:
CDR1: A - B start: e1a ans: e2 end: e4 Party: B disp:
ANSW linkedID: abc9
CDR2: A start: e1 ans: e1 end: e6 Party: A disp:
ANSW linkedID: abc9
Are start time and
--- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote:
While I don't know the OpenVOX B200P specifics, some
interface cards
need you to change physical jumpers in order to acheive
NT vs TE, mode.
Could that be the case ?
--
exvito
I've just checked the card and you were right
Hi,
I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
touchscreen friendly GUI (bigger buttons, large keypad, etc...) ?
I like Ekiga, but GUI is small and cannot be operated via touchscreen... But
maybe
Hi
I have a grandstream gxp-2000 and trying it on an asterisk 1.6.
When I call internally between extensions I can hear the other person in the
gxp2000, but when I call externally from the gxp I can't hear the person on the
other end, but he can hear me.
How do you configure the grandstream
On Wed, 14 Jan 2009, Ralf Träskman wrote:
Hi
I have a grandstream gxp-2000 and trying it on an asterisk 1.6.
When I call internally between extensions I can hear the other person in
the gxp2000, but when I call externally from the gxp I can't hear the
person on the other end, but he can
Twinkle has big buttons but it hasnt keypad to dial, the keypad is there to
send DTMF.
2009/1/14 Robert Rozman robert.roz...@comutel.si
Hi,
I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
philipp-chemn...@gmx.de writes:
I am trying to set the caller ID to 'Anonymous anonymous' if the
caller is not registered to the asterisk server. But I can't find a
solution.
Which bit is causing you trouble? Detecting that the caller isn't
registered, or setting caller ID?
The latter is
On Wed, 14 Jan 2009 09:24:05 +0100, philipp-chemn...@gmx.de wrote:
Hi guys,
I am trying to set the caller ID to 'Anonymous anonymous' if the
caller is not registered to the asterisk server. But I can't find a
solution.
put registered users in one context which dials out, and unregistered
David @ULC schrieb:
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
insecure=very this is very important
host=voiper.ipkall.com
dtmfmode=rfc2833
context=from-pstn
canreinvite=no
how will call understand that where I have to land as we DO
Hi
Yes we use voip as external.
/ralf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson
Sent: den 14 januari 2009 10:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
this is like the bible
http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf
2009/1/14 Philipp Kempgen philipp.kemp...@amooma.de
David @ULC schrieb:
If I use below code in my sip.conf ,
[123]
type=peer
qualify=no
port=5060
nat=no
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I have an Asterisk server which forwards requests to various PSTN
gateways with SIP. If the Dial() attempt is not successful I want to
differ at least these 3 options:
- called destination is busy (486): e.g.
On Wed, 14 Jan 2009, Ralf Träskman wrote:
Hi
Yes we use voip as external.
If the asterisk box is behind NAT itself, then you need to port-forward
ports 5060 and 1-2 on the firewall to the asterisk box. Then you
need to make sure that localnet= and externip= are set correctly in
Why not send the closure reason in the text message? It costs the same to
send 30 characters as it does 160. You also need to consider trunk capacity.
if you send the message out it is likely that people will react immediately
and call the number as soon as they receive the message. Depending on
Lee Wilson wrote:
--- On Wed, 14/1/09, Ex Vito ex.vitor...@gmail.com wrote:
While I don't know the OpenVOX B200P specifics, some
interface cards
need you to change physical jumpers in order to acheive
NT vs TE, mode.
Could that be the case ?
--
exvito
I've just checked the card
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP softswitch.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
But if I remember correctly I have seen patches for that somewhere.
Maybe on the bug tracker.
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock /
Hi Fadge
thanks for the comments (see inline)
asterisk wrote:
Why not send the closure reason in the text message? It costs the same to
send 30 characters as it does 160. You also need to consider trunk capacity.
We had a recent problem where the school was closed because of a burst
Philipp Kempgen schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
But if I remember correctly I have seen patches for that somewhere.
Maybe on the bug tracker.
The simple answer is that Asterisk is too high-level.
But you can change the response handlers in chan_sip.c to set various
channel variables to achieve what you want pretty easily.
Klaus Darilion wrote:
Hi!
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
I'm delighted to be able to say that as part of the agreement on my
departure from Mexuar,
the Corraleta applet source code Westhawk Ltd wrote for them has been
released under the GPL.
it is available for download at :
http://www.mexuar.com/files/corraleta_sdk.rar
Tim.
On 20 Sep 2007, at
Take a look (if it still exists) at the Asterisk B2BUA project. It has a
patch that adds direct access to SIP response codes. It takes a little
modification of the patch file to use in some of the newer asterisks
(and to strip out the one codec option that's somewhat irrelevant), but
it's a
Why are you using a text message when you could be recording a message and
sending it out? This would possibly be clearer than a read-and-callback
scenario?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Wow very cool - what is required for novices to install this application
on their websites?
Will you be making available some kind of easy install app?
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
Hi,
Am Mittwoch 14 Januar 2009 schrieb Benny Amorsen:
philipp-chemn...@gmx.de writes:
I am trying to set the caller ID to 'Anonymous anonymous' if the
caller is not registered to the asterisk server. But I can't find a
solution.
Which bit is causing you trouble? Detecting that the caller
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP softswitch.
This is IMO a stupid limitation. There are dozens of
On Wed, 14 Jan 2009, Tim Panton wrote:
I'm delighted to be able to say that as part of the agreement on my
departure from Mexuar,
the Corraleta applet source code Westhawk Ltd wrote for them has been
released under the GPL.
it is available for download at :
- Klaus Darilion klaus.mailingli...@pernau.at wrote:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP
Thank you for getting that code contributed to the community. Is there
a spec somewhere of the features supported by that applet? A version
history? Docs of the SDK it's distributed as?
On Wed, 2009-01-14 at 14:38 +, Tim Panton wrote:
I'm delighted to be able to say that as part of
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from the
mic
without being signed)
3) appropriate javascript and DHTML to implement the look and feel
Since we are all learners here, you can download the Java stuff for free
from sun, but you'd need about as much time on the Java as you spend on *.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton
Tilghman Lesher escreveu:
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.
I don't see why not. There has been no change whatsoever to that body of
On Wed, Jan 14, 2009 at 02:56:44PM -0200, Leonardo Gomes Figueira wrote:
Tilghman Lesher escreveu:
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from the
mic
without being signed)
3) appropriate javascript and DHTML to implement the
The G.729.1 wideband codec is starting to show a slight bit of
traction. There is a possibility that Asterisk could support G.729.1
- would you use it or buy it if it was available? More importantly,
does any equipment with which your systems currently exchange traffic
support G.729.1?
hi
thanks for this code is a very good contribution
is there any demo? example? or how to?
or any docs?
thanks
David
2009/1/14 Roberto Fichera ker...@tekno-soft.it
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a
John Todd wrote:
The G.729.1 wideband codec is starting to show a slight bit of
traction. There is a possibility that Asterisk could support G.729.1
- would you use it or buy it if it was available? More importantly,
does any equipment with which your systems currently exchange traffic
Danny Nicholas wrote:
Why are you using a text message when you could be recording a message and
sending it out? This would possibly be clearer than a read-and-callback
scenario?
Do you think so?
Remembering that most people, if they pick up the phone to hear a
recorded message will
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet? A version
history? Docs of the SDK it's distributed as?
All I have is the link.
I should emphasise
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from the
mic
without being signed)
3)
On Wed, 14 Jan 2009, Thomas Kenyon wrote:
Danny Nicholas wrote:
Why are you using a text message when you could be recording a message and
sending it out? This would possibly be clearer than a read-and-callback
scenario?
Do you think so?
Remembering that most people, if they pick up the
Klaus Darilion schrieb:
Philipp Kempgen schrieb:
Klaus Darilion schrieb:
Is it somehow possible to evaluate the SIP response code inside the
dialplan?
No.
Part of the reasoning is that Asterisk is meant to be a multi-
protocol PBX, not a SIP softswitch.
This is IMO a stupid
On Jan 14, 2009, at 12:27 PM, Steve Underwood wrote:
John Todd wrote:
The G.729.1 wideband codec is starting to show a slight bit of
traction. There is a possibility that Asterisk could support G.729.1
- would you use it or buy it if it was available? More importantly,
does any equipment
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets can't read from the
mic
i am set var Set(CDR(accountcode)=forkcdr-test) into agiphp
probe
$agi-exec('Set(CDR(accountcode)=5)');
$agi-exec('SetAccount','123123123');
and no work ...
how to solutions.
thanks people!
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On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2) a java code signing certificate (java applets
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote:
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet? A version
history? Docs of the SDK it's
On 14 Jan 2009, at 18:11, Matthew Rubenstein wrote:
On Wed, 2009-01-14 at 17:38 +, Tim Panton wrote:
On 14 Jan 2009, at 16:47, Matthew Rubenstein wrote:
Thank you for getting that code contributed to the community. Is
there
a spec somewhere of the features supported by that applet?
I have a single connection that seems to register ok but then becomes
unregistered immediately. This is what I see with IAX debug turned on:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 6ms SCall: 1 DCall: 0 [76.25.248.23:4569]
Tim Panton ha scritto:
On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd need:
1) a java compiler
2)
Hi,
I've been noticing a lot of these messages lately:
NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?
Is something broken? I'm running asterisk-1.4.22.1.
They seem to happen in a number of different places where a beep or
recording is played, such as when
Stéphane Van Geystelen wrote:
You know my opinion about it ;)
The G729.1 is still a 50Hz 7000kHz bandwidth. An ultra wideband codec
capabilities would be a real breakthrough.
7KHz is not ultra-wideband, it's wideband. There are already wideband
codecs out there, including G.722, G.722.1 and
Gordon Henderson wrote:
On Wed, 14 Jan 2009, Thomas Kenyon wrote:
Danny Nicholas wrote:
Why are you using a text message when you could be recording a message and
sending it out? This would possibly be clearer than a read-and-callback
scenario?
The sms will be sufficient
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS methods
are available for scripting? Something like a quickstart javascript example?
I'm great with javascript, but I havn't read thru the Java to figure out
the hooks yet - if thats whats needed, I dont mind, but I'd
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java to figure
out
the
On 14 Jan 2009, at 18:36, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 18:02, Roberto Fichera wrote:
Tim Panton ha scritto:
On 14 Jan 2009, at 17:07, Roberto Fichera wrote:
Tim Panton ha scritto:
It isn't really in a state for novices at the present
you'd
Also, I guess at this point it doesn't matter for
L1, but should I be using Point-To-Point or
Point-To-Multipoint?
Thanks
Yes, you would still need to configure mISDN correctly as
well! And
AFAIK you will need to use PTMP, as that is what the router
would expect...
--
Tim Panton wrote:
On 14 Jan 2009, at 19:53, Josiah Bryan wrote:
Tim -
Do you have any minimal docs or hints on what hooks the DHTML/JS
methods
are available for scripting? Something like a quickstart javascript
example?
I'm great with javascript, but I havn't read thru the Java to
On Wed, 14 Jan 2009, Julian Lyndon-Smith wrote:
It's about 5 seconds to send a message with a GSM terminal, so 20 minutes
for 250... Which might be OK, depending on the number of messages
required... (Although cost is another factor - for those not in the UK, it
costs to send a text message,
Nortel filed for bankruptcy today
-Karl
A/P:
http://www.google.com/hostednews/ap/article/ALeqM5gx8oAvO1SIb6Ya2KhA2d-d9SZunwD95N5HVG0
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asterisk-users mailing list
To
Hello list,
I'm using a PHP script to communicate with asterisk via AMI, and edit
configuration files. So far everything went ok, but I came up with a
little problem editing extensions.conf using 'updateconfig'.
Is it possible to edit an existing line in extensions.conf file?, e.g.
Given a
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: January 14, 2009 5:31 PM
To: Asterisk Users List
Lee Wilson wrote:
Also, I guess at this point it doesn't matter for
L1, but should I be using Point-To-Point or
Point-To-Multipoint?
Thanks
Yes, you would still need to configure mISDN correctly as
well! And
AFAIK you will need to use PTMP, as that is what the router
OCG Technical Support wrote:
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
Thanks for the quick response. maillog shows nothing if it's executed
from the System() call. Obviously maillog shows the outgoing if executed
from the terminal,
Hi All,
I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a
TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working.
Doing zap show channels etc from the Asterisk CLI results in an
error saying there's no such command.
The machine has Zaptel 1.2.9.1, which I've tried
Have you tried recompiling/installing the new zaptel source before Asterisk?
Geoff Lane wrote:
Hi All,
I'm running Asterisk 1.4.22.1 on a CentOS 5 machine fitted with a
TDM400P. When I upgraded from Asterisk 1.2.12.1, Zap stopped working.
Doing zap show channels etc from the Asterisk CLI
On Wednesday, January 14, 2009, Jose P. Espinal wrote:
Have you tried recompiling/installing the new zaptel source before
Asterisk?
Thanks for the reply.
It's the old Zaptel source that was working with Asterisk 1.2.12.1 and
so was already compiled and installed prior to upgrading Asterisk.
I think their issue is that they built their business around cheap support
in Asain countries which is a hit or miss. I know that when I pointed out an
obvious flaw that made them look stupid I got email that I had a $20.00
credit with them. I never mentioned it because I did not think it was
You can try blocking the caller ID in the dial plan. Not sure how that will
affect the CDR's. If it does not show up in there in the dial plan you can set
a variable to the caller ID then set it to be blank and on hangup update the
CDR's.
- Original Message -
From: Sriram
To:
More than just support - also core engineering.
Dovid Bender wrote:
I think their issue is that they built their business around cheap support
in Asain countries which is a hit or miss. I know that when I pointed out an
obvious flaw that made them look stupid I got email that I had a $20.00
Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend
you install Zaptel 1.4.12.1 or go to DAHDI. The first thing you need to
do is erase all the zaptel modules from the /lib/modules/kernel
version directory and do a depmod -a to make sure only the new DAHDI
or Zaptel
I use post variables. I found this on the web. Forgot where I got it from
(sorry that I can't give you credit).
?php
//Connect to the Asterisk Manager
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: username\r\n);
fputs($socket,
- Original Message -
*From:* Sriram mailto:d_r_sri...@hotmail.com
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, October 10, 2008 6:52 PM
*Subject:* [asterisk-users] Block Caller ID
Hi
Is there
I am getting this Error on my Asterisk.
How to solve it ?
ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this
SIP message, it's incomplete.
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asterisk-users
On Wednesday, January 14, 2009, Carlos Chavez wrote:
Zaptel 1.2.9.1 will not work with Asterisk 1.4.22. I would recommend
you install Zaptel 1.4.12.1 or go to DAHDI.
Thanks for the reply. Uninstalling DAHDI and switching to Zap 1.4 did
the trick. I can now make calls to and from the PSTN and
None of these examples actually create a 3-way call, which is, unless I am
mistaken, the original request. An incoming/outgoing call gets bridged to a
local channel alright, but then how do you bridge that call to yet another
call?.
I did try some alternatives and the only way I found is by
hello,
Here (http://updates.xorcom.com/astribank/bristuff/1.4/INSTALL.html) you can
read :
cd qozap
modprobe zaptel
insmod qozap.o (for kernel 2.4)
insmod qozap.ko (for kernel 2.6)
ztcfg
I thought modprobe was a replacement for insmod.
Can someone be kind enough to explain :
1. the difference
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem to be having any luck.
I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number])
Has anyone had any luck using this thing and can enlighten me on how
it's supposed to be
Well, T.38 works over IP, not TDM...
James Lamanna wrote:
Hi,
I've been trying to use the FaxGateway application to send T.38 out
over Zaptel using asterisk but I don't seem to be having any luck.
I'm executing it in the dialplan like: FaxGateway(Zap/g0/[number])
Has anyone had any luck
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
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To
Hi all,
thanks Tim and Mexuar for releasing this here...
I have already taken the source - and compiled a little java applet
which is self signed to test the whole thing.
I will put it on my site (and allow users to enter
host/user/pass/Calling Number,Calling Name,Number to dial...) for demo
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