I have Festival running fine on one Fedora Core 3 machine but I am
having problems getting it to work on another one.
I am using festival-1.4.2-25
I have followed the guide at
http://www.voip-info.org/wiki-Asterisk+Festival+installation and am
using the second festival command patch which is the
I forgot to add that the problem I am experiencing is that when I dial
the extension it is answered and then immediatly hung up on me. It is as
if festival is working butnot generating any sounds.
On Tue, 2005-03-22 at 15:50, Gareth Blades wrote:
I have Festival running fine on one Fedora Core 3
The most common cause for this is there being no timing source
available. Do you have the zaptel drivers correctly installed and
configured?
You could just enable 'ztdummy' and test the system using that.
On Wed, 2005-03-23 at 12:02, Bart Van Daal wrote:
Hello,
I'm running asterisk-1.0.6 on a
6436 2 (autoclean) [i810_audio]
usb-uhci 25740 0 [ztdummy]
Maybe a irrelevant question but do I need alsa or oss (alsa.conf oss.conf)
to play back these sound files?
thanks again,
Bart
-Original Message-
From: Gareth Blades [mailto:[EMAIL
On Wed, 2005-03-23 at 13:46, Eric Wieling aka ManxPower wrote:
Bart Van Daal wrote:
Thanks for your answer,
I've compiled and loaded 'ztdummy' but still no sound.
here's the relevant portion of lsmod:
ztdummy 2464 0 (unused)
wcusb 19552 0
g729 is a commercial codec and requires a license to use. You can
purchase a license for use with Asterisk. See
http://www.voip-info.org/wiki-Asterisk+G.729+licensing
g723 is also a commercial codec but Asterisk does not support it other
than in passthru mode.
I have not heard of gsmfr.
What
I have installed asterisk from the CVS source on Jan 7th and I am having
problems getting call transfers working.
features.conf contains:-
[featuremap]
blindxfer = #1 ; Blind transfer
disconnect = *0; Disconnect
automon = *1 ; One Touch Record
atxfer = *2
On Wed, 2005-01-26 at 15:50, Germn Micale wrote:
Hi,
Does someone know an ActiveX IAX softphone?
I need a free softphone to connect with Asterisk from a web page.
Regards
I use Firefly as a free IAX client and it works well.
I have also used diax which has more features (multiple line
Message -
From: Gareth Blades [EMAIL PROTECTED]
On Wed, 2005-01-26 at 15:50, Germn Micale wrote:
Hi,
Does someone know an ActiveX IAX softphone?
I need a free softphone to connect with Asterisk from a web page.
Regards
I use Firefly as a free IAX client and it works well.
I
Diax supports showing the number of new and old voicemail messages
although I have not managed to get that working yet.
There is a parameter in iax.conf which must be enabled in order to get this
functionality.
; If mailboxdetail is set to yes, the user receives
; the actual new/old
On Thu, 2005-01-27 at 13:56, Dan wrote:
Hi,
I have 'mailboxdetail=yes' in my iax.conf file.
A typical extension configuration is :-
[7000]
type=friend
regexten=7000
secret=password
host=dynamic
context=voipuk
mailbox=7000
In voicemail.conf I have entries such as :-
I am having problems getting any form of call transfer working.
I have reconfigured blind transfers to be #1 and assisted transfers to
be *2 but these are not working.
Looking at the wiki
(http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not
mention IAX so I assume I have to use the
On Tue, 2005-02-01 at 15:58, Dan wrote:
Hi,
- Original Message -
From: Gareth Blades [EMAIL PROTECTED]
...
...
I am having problems getting any form of call transfer working.
I have reconfigured blind transfers to be #1 and assisted transfers
to
be *2 but these are not working
On Tue, 2005-02-01 at 18:43, Eric Wieling wrote:
Bruno Hertz wrote:
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression that the # transfer
might need
On Tue, 2005-02-01 at 19:49, Denis Galvo - iSolve wrote:
Em Ter 01 Fev 2005 16:27, Bruno Hertz escreveu:
On Tue, 2005-02-01 at 16:27 +, Gareth Blades wrote:
Unattended transfers just does nothing. I cannot get it to do anything.
Not sure about this, but I'm under the impression
On Tue, 2005-02-01 at 20:29, Philipp von Klitzing wrote:
Hi!
Additionally to that, I enabled debugging mode on the * servers,
and could see that dtmf # arrived from gnomemeeting properly.
Plz graph your network, is it:
Gnomemeet -- * -- * -- Gnomemeet?
and insert codecs and
I have found out that the reason why my call transfers are not working
when using the IAX protocol is because Asterisk is performing a native
bridge.
If I force the user of one of the clients to use a different codec so
that Asterisk is unable to do a native transfer then it works.
How can I
When I try and use the Asterisk call transfer feature it is only
accepting 3 digits. Our extensions are 4 digits so what do I need to
change to reconfigure it?
Thanks
Gareth
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On Wed, 2005-02-02 at 14:25, Bruno Hertz wrote:
On Wed, 2005-02-02 at 09:09 -0500, Nabeel Jafferali wrote:
notransfer=yes
That prevents transfers but not bridging. As to my knowledge, there's
no way to prevent bridging.
If that is the case then it seems a serious limitation as it makes
On Wed, 2005-02-02 at 20:57, Matt Riddell wrote:
Dan Adams wrote:
Are you going to be making this one available to all. I am not sure if
Yes.
or how it is possible, but maybe you would be able to have it so that
if you right click on the contact, it has an option to iniate a call
I have a Digicom TE110p card installed in our exchange. I have compiled
and installed libpri, zaptel and recompiled and installed asterisk.
I have configured udev as I am running Fedora Core 3.
The problem that I have is that when zaptel is not running everything
works fine. However when I start
I intend to replace our Lucent Index telephone system with Asterisk and
need to buy a proper server to run it on.
I have read about the problems with the HP DL380 G4 and the TE410P
cards.
I have a TE110P and will be using a TDM400 card for the backup analogue
lines. Is there any server that you
We currently have the Grandstream GXP-2000 phones which generally work
very well except that we cannot get find a headset which works reliably
with them. Either the sound quality is poor or the other party has
difficulty in hearing us.
We therefore want to get a couple of different phones and
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten =
qualify=yes
Put in in the sip.conf file in the configuration section for the
specific phones.
On Tue, 2006-08-29 at 09:50, Tomislav Parčina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
1. you need qualify set as the wifi radio on the phone sucks big oranges
What is qualify
Make sure you have the correct version of libtiff installed.
On Thu, 2006-10-12 at 12:22, Mohammad Shokuie wrote:
Dear folks,
I have problem in fax reception. The astrisk detects the fax tone and jusmps
to the fax extension and rxfax application starts and the max machine starts
the fax
I am using the latest CVS version of * (just upgraded incase it was a
bug) and I am having problems with the directory. Whatever I enter when
dialing it the only result I get back is '6000' which is not even in the
voicemail.conf
extensions.conf contains :-
...
[voip]
exten = 6010,1,Macro(cst)
I am running the latest CVS version of Asterisk.
Calls between an IAX client and SIP phones (Grandstream SP2000 and
Sipura SPA-841) works fine and so do external call over the Internet
from the SIP desk phones.
However when I call from either the Grandstream/Sipura phones to another
one I get no
patches on mantis, but none of them worked for me. I
flipped back to stable and have had no problems since.
Anyone got any ideas?
--
Tom
On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote:
I am running the latest CVS version of Asterisk.
Calls between an IAX client and SIP phones
at 15:08, Tom Hayden wrote:
Then perhaps you have a NAT problem or some other issue.
--
Tom
On 8/10/05, Gareth Blades [EMAIL PROTECTED] wrote:
I did try installing the 1.0.9 version but I have the same problem with
that release aswell.
On Wed, 2005-08-10 at 14:14, Tom Hayden wrote
On Mon, 2005-04-11 at 15:57, Doug Millsaps wrote:
I use a headset w/out any problems, except for if my cell phone is close by
and rings. Otherwise, volume is ok and no humming. Could it be your headset?
The Labtech noise canceling headsets are the ones which seem to cause
the poblem and
Reading http://www.voip-info.org/wiki-SPA-841 I have configured Asterisk
with the following dialplan to ring using 'Simple-1' but it is not
working.
exten = 6150,1,SetVar(ALERT_INFO=Simple-1)
exten = 6150,2,Dial(SIP/6152,4,t)
exten = 6150,3,Dial(SIP/6152SIP/6148,4,t)
exten =
It works for me with the default SIP settings. I am using the latest
firmware.
I have found that you have to restart * for it to pick up on the new PIN
code however.
On Wed, 2005-04-20 at 01:58, Master Abi wrote:
if I have conf = 80,111 in meetme.conf, I dial 80# and connect to the
conference,
In particular have a look at
http://www.voip-info.org/wiki-Asterisk+x100p+echotraining
Using this guide you can set * to send an impulse down the line so that
it trains much quicker and also manually adjust the tx and rx gain to
improve things furthur.
On Thu, 2005-04-21 at 13:19, Gavin Hamill
If we get an incoming call I can edit the callerID provided to add the
leading '90' and set the name so that sales calls can be identified
according to the number called.
If however the callerID is unavailable then setting the callerID name or
number fails (it shows as unavailable on the phone).
)
exten = s,4,Hangup
exten = s,103,Goto(3)
On Thu, 2006-03-16 at 16:24, Gareth Blades wrote:
If we get an incoming call I can edit the callerID provided to add the
leading '90' and set the name so that sales calls can be identified
according to the number called.
If however the callerID
We are using the Grandstream GXP-2000 which works very well.
On Fri, 2006-03-17 at 12:11, WipeOut wrote:
Hi,
I am looking for a budget IP phone that can use preferably iLBC or GSM
codecs..
Suggestions?
___
--Bandwidth and Colocation provided
Is there any way to query the number of people in a call queue from the
dialplan?
Our freephone provider has a feature where if we busy a call they record
the voicemail and email it to us. This enables us to divert calls to
them if our incoming lines start to get full. In order to do this we
need
Cant you set the calleridname before putting the call into the queue?
On Thu, 2006-04-06 at 22:57, Shaun wrote:
I was wondering if it was possible to run a macro once the agent/member
picks up, I know I can do this with dial in the extensions.conf but wasn't
sure about the queue. Basically
I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
configured using the provisioning feature so the configuration is all
identical.
The problem I am having is that they randomly seem to stop registering
with asterisk. When they stop registering they can still make calls but
it to you.
Mark
-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED]
Sent: Monday, 10 April 2006 8:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 phones stop registering
I have about 30 GXP-2000 phones running 1.0.1.9 which have all been
On Tue, 2006-04-11 at 09:26, Tomislav Parčina wrote:
Hi group!
I have been using asterisk 1.2 for quite some time and now I need to go back
on asterisk 1.0 (because of oh323 channel driver). One thing that I can't
remember is does asterisk 1.0 support native music on hold? If it does, how
Have a look at http://www.voip-info.org/wiki/view/sipura+mass+deployment
On Tue, 2006-04-11 at 06:05, Kerry Garrison wrote:
Has anyone got any information on bulk provisioning of Linksys
SPA-941/94s? There is an overview in the admin guide but it refers to
a different provisioning guide that I
It looks to be as if you have your PRI D-channel defined as a voice
channel in your zapata.conf. For example in the UK channel 16 is the
d-channel so zapata.conf contains :-
switchtype = euroisdn
pridialplan=unknown
signalling = pri_cpe
group = 1
channel = 1-15
channel = 17-31
On Tue,
On Mon, 2006-04-10 at 21:42, Lonnie Abelbeck wrote:
Adding defaultip=10.x.x.x might solve the problem.
GXP-2000's can work without registering, using host=10.x.x.x as long as you
don't want to use BLF with the new firmware.
The new firmware is great, as long as you don't have early
Just thought I would post this as someone might find it usefull.
This is the dialplan for making outbound calls from the UK (not
internetional).
It can be set to block callerID for particular extensions. I have also
added some detection of the PRI error numbers when a call fails to give
some extra
though with the new 102x firmware branch.
I would definitely recommend it to you.
Mark
-Original Message-
From: Gareth Blades [mailto:[EMAIL PROTECTED]
Sent: Monday, 10 April 2006 8:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GXP-2000 phones stop
What version of Asterisk?
On Thu, 2006-04-13 at 12:38, Daniel Korndorfer wrote:
Hi,
i'm having problems with the MOH module. In a queue sometimes it just
stop playing, does anyone have some idea what could be wrong?
No verbose data...
Tks,
Daniel Korndorfer
Just noticed that I occasionally get these messages:-
Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq
ran 281 scheduled tasks all at once
Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq
ran 1987 scheduled tasks all at once
Apr 13 12:47:56 WARNING[11390]
See
http://bugs.digium.com/view.php?id=6457
On Tue, 2006-04-18 at 13:11, [EMAIL PROTECTED] wrote:
I'm running 1.2.3 and that seems to be the most stable version, had
problems with other versions too.
-- Original message --
From: Brian Roy [EMAIL
There is no way to do it currently on these phones. Although the web
interface supports a distinctive ring based on callerID it does not
accept wildcards.
When I contacted Grandstream about this I got the following reply :-
Unfortuantely, not with present firmware.
We will implement this
This is what I do and it works well. It uses the asterisk database so
you can define which printer the fax gets printed on and/or email
address the pdf gets mailed to.
[macro-fax]
exten = s,1,Set(FAXFILE=${UNIQUEID}.tif)
exten = s,2,Set(FAXOK=no)
exten =
Enter the 'dmesg' command. It displays a log of kernel messages etc...
and may show up a problem.
On Wed, 2006-04-19 at 03:03, [EMAIL PROTECTED] wrote:
List,
The past few days the asterisk service on my server has crashed
several times. I have had it running for months and have made no
Maybe this will help
http://www.voip-info.org/wiki-asterisk+sip+qualify
On Wed, 2006-04-19 at 14:51, Marco Mouta wrote:
I've tested maxexpirey=120 and even with this, asterisk didn't stop
the call:
Scenario: SIP user agent has left without telling to asterisk it was
leaving...
There was a
Quoting http://www.asteriskguru.com/tutorials/e1t1.html
-- configuration on SBC.
If you are being flooded (several times a second, non stop and the pri
never worked) by lines as:
Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of
Also
If you see the error Jul 14 13:55:21 NOTICE[19519]: chan_zap.c:7874
pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span
1
only occasionally, then you might have some devices in your pc (ide
cards?) taking to long when taking an intterupt.
You might want to try to put
Make sure you have the DTMF mode set to RFC.
On Fri, 2006-04-28 at 15:20, Johnny Stork wrote:
I seem to be having a problem with my GXP-2000. No matter how carefully I
type in the mailbox number and password when calling the mailbox (*98), it
keeps complaining that the password is not
If its going over a zaptel interface then you certenly can. See
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ZapBarge
On Wed, 2006-05-03 at 15:52, Olivier Saulnier wrote:
Hello,
is it possible to listen a conversation in real time, without recording it?
Best regards,
I would also recomend that you upgrade to the latest firmware 1.0.2.13
(contact grandstream) as it does fix some registeration issues and have
extra NAT/STUN features.
On Wed, 2006-05-03 at 17:15, Chris Bagnall wrote:
Greetings list,
I'm coming across an issue with some of the GXP-2000 phones
Commands are
iax2 show ...
zap show ...
sip show ...
On Wed, 2006-05-10 at 13:32, Matt wrote:
Does anyone know what would cause a 1.2.6 or 1.2.7 asterisk system to
do the following?
After inserting a zaptel card the system starts up and when I log into
the asterisk CLI I can't do a show
Have you configured the zaptel card correctly?
On Wed, 2006-05-10 at 13:32, Matt wrote:
Does anyone know what would cause a 1.2.6 or 1.2.7 asterisk system to
do the following?
After inserting a zaptel card the system starts up and when I log into
the asterisk CLI I can't do a show sip, show
It could be an old module still left behind from the previous version. I
would delete everything in /usr/lib/asterisk/modules and then reinstall
(make install) and see if it will start.
On Thu, 2006-05-11 at 14:30, Shawn Porter wrote:
Never try upgrades half-asleep and 1/4-knowledgable!
For BT in the UK I use :-
zaptel.conf
loadzone = uk
defaultzone=uk
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
zapata.conf
[trunkgroups]
[channels]
language=en
context=did
priindication = outofband
usecallerid=yes
cidsignalling=v23
usecallingpres=yes
sendcalleridafter=1
switchtype =
I just bought a couple of these units. It seems to work fine but I could
not really test it as the phones were too close together so could not
get a clear idea of the call quality.
Phoning comedian mail seemed fine and certenly acceptible considering
the gsm codec was being used.
One minor
There is some additional functionality coming in future firmware
versions. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom
On Fri, 2006-05-12 at 14:54, Forrest Beck wrote:
Asterisk 1.2.7.1 and Zaptel 1.2.5
-Original Message-
From: [EMAIL PROTECTED]
I use a X100P from www.x100p.com and it works well. It is purely a FXO
card for connecting to a PSTN line though. I have a SIP phone aswell
which is why I dont require a FXS interface.
I have an account with voiptalk.org so I can make calls over the
internet. I also have a free local rate number
Asterisk 1.2 does not save the notify status when it reloads the
configuration. The only way around this at the moment is to set a short
sip registration expiry. When the phones then re-register they will
start working again.
I believe there is a patch in trunk to save the status so this should
What firmware version are you running on the phones?
On Wed, 2006-05-24 at 11:56, asterisk wrote:
It does not happen with a reload of asterisk but when the server is
rebooted. I have set the register timeout to 1 min but this has no
effect. The BLF still does not work. The * is V 1.2.7.1. I
Why not just define a group and use :-
exten = _9X.,1,Dial(ZAP/g1/${EXTEN:1})
On Wed, 2006-05-31 at 13:08, John Joseph wrote:
Hi
I am using a 4FXO , TDM400P card
I am able to call outside , after modifiying
extensions.conf
with
exten = _9X.,1,Dial(ZAP/1/${EXTEN:1})
I had a similar problem with the GXP-2000 phones and firmware 1.0.1.9.
Upgrading to the 1.0.2 (now running 1.1) firmware fixed it.
On Fri, 2006-06-02 at 08:37, Información Capa Tres S.L. wrote:
Hello,
I have an issue with the IP Phones Grandstream BT101/102. At a random
time (more than one
I believe if you use the new native music on hold feature it always
plays the music on hold starting from the beginning.
On Tue, 2006-06-06 at 11:15, Tommaso Calosi wrote:
I want that incoming callers to hear a welcome message while the phones
ring. I know I can use Dial with the m(class)
I am running 1.1.0.13 and there are no issues which are causing a
problem for us. The speakerphone is not much use but we can live with
that.
1.0.1.9 would stop registering after a while causing incoming calls to
go straight to voicemail.
1.0.2.13 fixed this but had a bug where sometimes
Yes you can as long as you have at least the 1.0.2.13 firmware. I have
attached the template. The multi-purpose key settings are at the end.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to program the multi-purpose keys on a GXP-2000
remotely via a TFTP configuration file?
Yes you can if you are running 1.0.2.13 or later. I have the template
which I tried posting here as an attachment but it has not arrived yet.
If it does not arrive you can email me directly or contact grandstream
support.
On Fri, 2006-06-09 at 14:41, Daniel Salama wrote:
Is it possible to
I have been running the same software versions together with a digium
single port PRI card in the UK and have not experienced any problems
since the upgrade.
On Tue, 2006-06-13 at 12:00, Chris Teesdale wrote:
Hi Everyone,
This morning I upgraded Asterisk from 1.2.7.1 to 1.2.9.1 along with
G729 uses 8kbps but with the IP overhead it actually uses 30kbps so for
256k upstream you should be able to handle 8 calls but this is in ideal
conditions.
If you were to use IAX and enable trunking then you would use 30kbps for
the 1st call and 10kbps for each additional call.
See
The only issue with 1.1.0.13 which affects only certain versions of the
gxp-2000 is the display blanking issue on very early phones.
It sounds like you have a faulty phone and should return it for a
replacement.
On Wed, 2006-06-14 at 11:57, [EMAIL PROTECTED] wrote:
I have had 2 GXP-2000 for a
You need to run the java based tool from the grandstream website to
convert the template to a format the phone understands.
On Wed, 2006-06-14 at 14:05, Matthias Fechner wrote:
Hi,
i got my Grandstream GXP-2000 phone today and want to configure it
with TFTP. I downloaded the firmware
No I dont believe so. The address book is a new feature as it is very
basic in my opinion and even editing it on the phone is difficult.
I would expect a web based editing feature to be implemented at some
point and once that is done it should be possible to do a mass update of
the phones.
On
I have a couple. The audio quality is not as good as it has a noticeable
amount of hiss in the background and it also does not support message
waiting.
It does however support other codecs other than ulaw/alaw which is why
we went for it.
On Tue, 2006-06-20 at 14:51, Steve Jones wrote:
I would guess the card is actually a http://www.x100p.com/products_1.htm
and may be x100p selling it on ebay themselves.
I have one of these cards itself and it works fine. There is a bit of
echo initially but it gets cancelled our fairly quickly. Apart from
turning on echo cancellation I have
Has anyone else had problems with the Nokia E61 and E70 phones not
always answering voip calls?
We have them connected via a local access point (so no router/NAT) and
sometimes the phones dont ring when called. They are registered ok and
if you use the phone to make a voip call it works fine.
The
I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am
getting a lot of errors like this on the console :-
ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error:
Broken pipe
I have tracked it down to a perl AGI script which performs our own CDR
recording. It is
Danny Nicholas wrote:
Check out this snippet from Tilghman Lesher (one of the true Asterisk
Guru's)
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg220482.html
Thanks but that appears related to AMI not AGI.
--
Philipp von Klitzing wrote:
Hi!
Why is asterisk so slow in sending the call info via STDIn in these cases?
Is there any way this can be fixed?
Your AGI script is faulty: In at least one place you have missed to READ
the output right after you have issued a command. So go check your
Danny Nicholas wrote:
Can you post the script?
Yes private stuff is in a separate file. $mode=start works fine but
answered and completed cause the problem.
I dont know if it is a problem with teh AGI script or just the newer
asterisk reporting it as an error. It doesnt effect functionality
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, April 28, 2010 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI
script
Danny Nicholas
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is answered but during and
at the end
Danny Nicholas wrote:
Darn, that should have worked. The improvement from 1.4.22 to 1.4.23+
basically requires that every print STDOUT line be followed by a STDIN
to make util.c not choke when doing commands/setting variables. I wonder
how this rewrite would work?
sub set_variable
{
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at
all 3 stages but with different parameters on the command line to
indicate the call status. Works fine before the call is answered but
during and at the end
Steve Edwards wrote:
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via STDIN.
On Wed, 28 Apr 2010, Gareth Blades wrote
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when
Ignore me I figured it out. The dangers of copy and paste.
After looking through the code line by line I noticed the 'b' parameter
to monitor(). Fine to use before the dial command but shouldnt be used
when a call is in progress.
Gareth Blades wrote:
I have got call recording working on our
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
typo ...
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}${OFFICE2},,rtT)
Gareth Blades wrote:
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi
Can you post a sip debug
Tarek Sawah wrote:
Greetings List.
I'm facing a strange issue with one of my providers.. after sending an INVITE
request my server places the call on hold.. until the call is answered..
this is happening only with this provide although i have 3 other providers i
In my previous company we bought about 30 Grandstream GXP2000 phones.
The build and design quality of those phones were terrible (not to
mention firmware bugs).
Speakerphone and headset ports were unusable.
The external powersupply would only last a year or two before it failed.
The screen was
I am running asterisk 1.6.2.6 and have configured hints for our
extensions and have a couple of Aastra 6755i test phones. The phones
register fine but 'core show hints' shows the lines as idle even if they
are in use.
I read the wiki and see mention about needing to set call-limit in
asterisk
Richard Kenner wrote:
I read the wiki and see mention about needing to set call-limit in
asterisk 1.4 but that has been depreciated in 1.6 so what is the way it
should be done in 1.6?
I set
callcounter=yes
in sip.conf.
Thanks that works perfectly.
--
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
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