reza naraghi wrote:
Hello
I really need to disable the call-waiting on my sip phones
I studied most of the posts on internet and did it on my asterisk but
not useful.
in fact I need a comment that I disable call-waiting but without enable
call-limit because I want to keep the waited caller
On Wed, 23 Jul 2008, reza naraghi wrote:
Hello
I really need to disable the call-waiting on my sip phones
I studied most of the posts on internet and did it on my asterisk but not
useful.
Try reading your phones manual. This is a phone function, not asterisk.
Which phone? People here might
Hi Mark,
I assure my queues.conf is full of autopause = no, in the singles and
general contexts (I'm not sure where to put it 'cause I found no docs
about it).
Moreover, this morning I checked my Asterisk with show queues and I
found another surprise:
SIP/17 with penalty 1 (paused) (Not
Hi,
i tried to compile Asterisk 1.4.21.2 on a server which i have been using with
many previous Asterisk versions,
without any problems.
But with 1.4.21.2 it failed:
--
[CC] res_adsi.c - res_adsi.o
[LD] res_adsi.o - res_adsi.so
[CC] res_agi.c - res_agi.o
On Wed, Jul 23, 2008 at 10:12:21AM +0200, Carsten Bock wrote:
Hi,
i tried to compile Asterisk 1.4.21.2 on a server which i have been using with
many previous Asterisk versions,
without any problems.
But with 1.4.21.2 it failed:
--
[CC] res_adsi.c -
On Mon, Jul 21, 2008 at 05:10:15PM +0100, Ben Thompson wrote:
[outbound-international]
exten = _900XX,1,Set(oldexten=${EXTEN})
exten = _900XX,2,Goto(international-number-length-check,s,1)
[international-number-length-check]
exten = s,1,Answer
exten = s,2,WaitExten(8)
Thanks for the hint with make NOISY_BUILD=yes:
In main/db1-ast/hash/hash_page.c
Line 654, function first_free(map)
there was an error at mask = mask 1;
hash/hash_page.c:659: error: stray '`' in program
I'm currently recompiling it from the start again, to test it.
Tzafrir Cohen schrieb:
On
Alex Balashov wrote:
It is known, as a matter of established fact, that it is possible to
disable call waiting on the eyeBeam phone. How to do it is not
something in which I can instruct you, but I gather it's a fairly
straightforward process, especially if you are autoprovisioning via the
I just re-unpacked asterisk-1.4.21.2.tar.gz and there was no '`' in the
function mentioned below.
I have no idea where it came from, i didn't edited the file before.
May be something is (terrible) wrong with the server i installed it on ... :)
= now it compiles and works
Carsten Bock schrieb:
Rob Hillis wrote:
Since when does eyeBeam have any kind of autoprovisioning? I've not
seen any reference to it in the manual or on their web site and I /have/
gone looking for it. If I've missed something, I'd be extremely
grateful if you could point it out - this is a feature I've wanted
Hello,
I'm testing Asterisk 1.6 (from SVN). In my dialplan I have:
--
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,2,Verbose(After Dial)
--
If IP denies the call I receive:
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1]
Hi,
I think i notice the problem now, but unfortunately i don't know how to fix it.
i'm using 118103 i dial 113102 i got this on asterisk server #1.
[Jul 23 18:27:48] -- Called 118102
[Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing
what i did is keep on dialing then hang up dial
I didn't say because I wanted my original email to
limit itself to facts I was sure of, but I think my SIP
problems started with 1.4.20 as well. I'm fairly sure
1.4.19 was solid... going back today.
It looks like someone at bugs.digium has found what it was,
so a fix
should be coming
Have you tried incominglimit=1 on sip.conf ??
It worked for me, no matter which softphone or ipphone / ATA I use, it
works.
You have to use it inside the configuration for every sip peer, just like
this:
[1002]
Type=friend
Host = dynamic
Port = 5060
incominglimit=1
.
.
.
De: [EMAIL
Hi to all, i'm experiencing a problem with an h323 trunk between a
Cisco Callmanager 4.2.
I'm using asterisk 1.4.21.1, openh323_v1_18_0, pwlib_v1_10_0
The problem is that sometimes (1 call every 20... but sometimes often)
the call arrives correctly on Call Manager side, and when is answered
Hello
thank u for ur attention but I did it and in fact its the same as call-limit
in newer versions.
this cmd limit ur call not disable call-waiting.
best regards
On Wed, Jul 23, 2008 at 5:02 PM, Marco Eduardo Cordeiro
[EMAIL PROTECTED] wrote:
Have you tried incominglimit=1 on sip.conf ??
Giorgio Incantalupo wrote:
Hi Mark,
I assure my queues.conf is full of autopause = no, in the singles and
general contexts (I'm not sure where to put it 'cause I found no docs
about it).
Moreover, this morning I checked my Asterisk with show queues and I
found another surprise:
David Nedved wrote:
I didn't say because I wanted my original email to
limit itself to facts I was sure of, but I think my SIP
problems started with 1.4.20 as well. I'm fairly sure
1.4.19 was solid... going back today.
It looks like someone at bugs.digium has found what it was,
so a fix
We are developing an softphone based on IAX client version 1.2 (my current
SIP softphone has many eoors), but it doesn´t have a specific function for
Conferencing (3-way calling) or to place the other party on HOLD.
I´m trying to do it through the PBX because our softphone´s lack of
functions.
Hello are you using FreePBX for your configurations? there is an option in the
extentions.conf for queues called
CWIGNORE=TRUE
try disabling it and see if it works for you .. this is the best i can help you
with .. i am using call-limit combined with busy-limit to stop the call
waiting.. i
Ok, I just tested and it works as I said before, here is the log of the
second call trying to come in:
-- Executing [EMAIL PROTECTED]:2] Dial(DGV/32, SIP/1001|20|tT) in new
stack
[Jul 23 12:05:55] ERROR[428]: chan_sip.c:3057 update_call_counter: Call to
peer '1001' rejected due to usage limit of
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote:
On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote:
My * version: 1.4.17
Please upgrade to 1.4.21.2.
Just a suggestion, Tilghman: it might have been nice to add because it
fixes your specific problem, so that we wouldn't
Hi,
On Jul/23/2008, Carles Pina i Estany wrote:
I'm testing Asterisk 1.6 (from SVN). In my dialplan I have:
--
exten = _00X.,1,Dial(SIP/[EMAIL PROTECTED],,tTwWg)
exten = _00X.,2,Verbose(After Dial)
--
Also this doesn't work either:
exten = _00X.,1,Dial(SIP/[EMAIL
Asterisk supports conferencing without using meetme. In this case you
don't have a central dial in number but a single extension can initiate
the conference call. Generally this is done the same way as with
traditional PSTN service which is that while on a call between two
parties, flash the
On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote:
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote:
On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote:
My * version: 1.4.17
Please upgrade to 1.4.21.2.
Just a suggestion, Tilghman: it might have been nice to
On Wed, Jul 23, 2008 at 10:44:12AM -0500, Tilghman Lesher wrote:
On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote:
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote:
On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote:
My * version: 1.4.17
Please upgrade
Tilghman Lesher schrieb:
On Wednesday 23 July 2008 10:15:18 Jay R. Ashworth wrote:
On Tue, Jul 22, 2008 at 06:39:28PM -0500, Tilghman Lesher wrote:
On Tuesday 22 July 2008 18:32:21 Chento Arohuanca wrote:
My * version: 1.4.17
Please upgrade to 1.4.21.2.
Just a suggestion, Tilghman: it
I´ll be upgrading my box this weekend and let you know the consequences.
I´m new at the community and it would be good for me to know what was the
problem with 1.4.17
Thanks for taking some time for me.
Daniel
On Wed, Jul 23, 2008 at 10:59 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Wed,
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
Carles Pina i Estany wrote:
Hi,
On Jul/23/2008, Carles Pina i Estany wrote:
I'm testing Asterisk 1.6 (from SVN). In my dialplan I have:
2008/7/23 MFH [EMAIL PROTECTED]:
Noah Miller wrote:
Hi Daniel -
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Anytime you need a call with more than 2 parties, you need to use some
kind of conferencing application. The default conference
Hi,
On Jul/23/2008, Anthony Francis wrote:
It is reading [EMAIL PROTECTED],,tTwWg as the device string.if you are
dialing to a sip connection called ip you would say
Dial(SIP/IP/${EXTEN},opts)
when I said IP i meant the IP value :-) not the two chars string IP.
Sorry for the confusion.
I
On Wednesday 23 July 2008 12:17:26 Steve Davies wrote:
2008/7/23 MFH [EMAIL PROTECTED]:
Noah Miller wrote:
Hi Daniel -
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Anytime you need a call with more than 2 parties, you need to use some
I have been looking for the busy-limit directive you mention but cannot
find it in any documentation for Asterisk. I can only find something
called busy-level which by its description might be what I need.
On Wed, 2008-07-16 at 15:20 +, Tariq .. wrote:
Try adding busy-limit=1 to
Hi,
Hope anyone can help me on DUNDi. I got 2 asterisk servers. configs below.
tried this on the cli:
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi lookup completed in 0 ms
*CLI dundi lookup [EMAIL PROTECTED] bypass
DUNDi lookup returned no results.
DUNDi
Would be my guess. J
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eugen Soare
Sent: Tuesday, July 22, 2008 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Recordings...
So basically,
He wants all calls recorded,
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having trouble getting my program to play a message and wait for a
keystroke.
I am able to use this code to play the file, so I know that the $msg
I'm getting close. The idea is based on the same principal as the link below.
Here's what I have done thus far:
All calls are recorded via mixmonitor. This is part of the initial dialplan
when the call comes in.
I then created an application map key sequence that is supposed to run
Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP
options packet every two to the phone to verify the remote NAT device is
allowing traffic from both sources to the phone.
Afterwards, you'll usually see this status from the servers, to verify the
phone is
Mike Diehl wrote:
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having trouble getting my program to play a message and wait for a
keystroke.
I am able to use this code to play the file,
Hi Sir,
Could it be my problem is since i'm using 2 asterisk, if an extensions
registers on asterisk#1 it will not be reachable by extensions on
asterisk#2? or it should not matter if i'm using realtime? coz this is
what i noticed:
i'm using 118103 i dial 113102 i got this on asterisk
On Tuesday 22 July 2008 02:58:38 pm Jason Lixfeld wrote:
I was looking for a Click to Dial/Web Dial solution and I found
AsteriDex. I'm looking for something I can use on the road where I
can hit an internal Click to Dial/Web Dial page from my cell, tap on a
number and have it bridge a
I am looking for a sample sip configuration of a SIP provider that runs
Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only
give me a SIP IP address to configure my asterisk box, when I call them for
support or authentication data to load on my sip.conf, they say me that I
Quoting Gustavo A Gonzalez [EMAIL PROTECTED]:
I am looking for a sample sip configuration of a SIP provider that runs
Broadsoft VoIP switch.
This is what I use:
register = 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368
[broadworks]
type=peer
host=1.2.3.5
dtmfmode=rfc2833
We have an older Meridian Norstar system and are thinking of using Asterisk
behind it
to use a SIP Voip Provider instead of our local telco.
Does anyone make an interface card that can integrate with the digital input of
the
Meridian. Not the optimal solution, but it allows for the current
On Tue, 2008-07-22 at 13:21 -0400, Jerry Geis wrote:
On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote:
/
// �[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416
handle_request_invite:
/ Call from 'devcentos5x64_to_ebox4300' to extension
Does anyone make an interface card that can integrate with
the digital input of the Meridian. Not the optimal solution,
but it allows for the current infrastructure to be retained.
By digital input do you mean a T1 interface? If so then yes several T1
interfaces are available. However I
Hi Nhadie -
Could it be my problem is since i'm using 2 asterisk, if an extensions
registers on asterisk#1 it will not be reachable by extensions on
asterisk#2? or it should not matter if i'm using realtime?
It does not matter that you're using realtime. If a phone registers
to asterisk
By digital input do you mean a T1 interface? If so then yes several T1
interfaces are available. However I think you mean is there a gateway to use
the Meridian/Norstar phones with Asterisk. If so, yes there is a company
that makes a gateway to use the Nortel p-phones with a SIP based system.
Joseph L. Casale wrote:
Well, I am not sure what is needed to interface between the two. I hoped
there was something you could use and from the sounds of it, its not worth
it. I guess the only thing I would need is a small switch in each office
then as we only have one run of cat-5e to each
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have the
feature to transfer calls (Incoming call - Answer - FLASH - Dial Number
to transfer - Answer - FLASH+4) and the call is transferred, but I have
the need to implement an internal ACD using Asterisk as the PBX, the trunks
Rob Hillis wrote:
If there were another usable softphone not tied to a specific platform
(such as Cisco CallManager) that had proper support for Plantronics CS60
USB headsets, I would have made the switch ages ago.
Does the eyeBeam have a textual local configuration file?
If so, you could
AGI can wrap calls to any dial plan applications; have you tried
calling Background() and Read() that way?
Mike Diehl wrote:
Hi all,
I'm trying to build an IVR using the Perl AGI module at
http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm
But, I'm having trouble
Hi Sir
Thanks for your reply, since i don't know how to setup DUNDi, what i did
for now is create a sip peer between the 2 servers and just use the
regserver on the realtime db.
but now with that setup i cant play the music on hold of the extension
i'm calling to, e.g i'm 118102 i called
Alex Balashov wrote:
Although, oddly enough, a lot of them can do VLAN trunking, etc.
Not odd at all as far as I'm concerned - I know a number of places that
segregate LAN traffic from VoIP traffic using multiple VLANs over the
one physical link. VLANs would be the best solution (short of
Not odd at all as far as I'm concerned - I know a number of places that
segregate LAN traffic from VoIP traffic using multiple VLANs over the
one physical link. VLANs would be the best solution (short of running
multiples cables for PC and phone) to achieve this.
I would have about 30 phones I
I resolved this problem. The key was to get the right combination of
self/callee and peer/caller. Read the instructions regarding the application
map very closely. My problem was that I was not running the StopMixMonitor
command against the proper channel. Even though mixmonitor records both
Ricardo Melendez wrote:
Hi to All, I have a PBX (MAINPBX) from a Telecomm Provider, which have
the feature to transfer calls (Incoming call - Answer - FLASH -
Dial Number to transfer - Answer - FLASH+4) and the call is
transferred, but I have the need to implement an internal ACD using
On Thu, 17 Jul 2008, Nhadie wrote:
Hi,
I'm testing using the free g723 codecs and i have successfully installed
them.
g723
g723 -
gsm 9
ulaw 9
alaw 9
g726 9
adpcm 9
slin 8
lpc1010
g72910
speex -
ilbc
Hey Guys,
New TrixBox. For some reason it works
just fine behind a WRT54GL with latest
version of stock Linksys firmware.
However, when using a GL with Tomato
firmware, can't hear the ringing or
audio from the called party.
Yes, ports 10,000 - 20,000 and
5004 - 5082 are open.
Yes, these
David,
What you sent me is almost exactly what I had, which indicated that that part
of my code was correct. So, I moved that block of code to the top of my
program and it worked. Eventually, I found a debug print() statement that I
had forgotten to take out. Once it was gone, my code
Well, I am not sure what is needed to interface between the
two. I hoped there was something you could use and from the
Joseph,
Now I'm pretty sure we are not talking about the same things. Let me see if
I have the correct picture in my head. I now think you have a Norstar in one
office and
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