Hi all,
I am syncing call files on my secondary asterisk server but without
permission to read for asterisk. So they should be executed when I grant
the right permissions (thats when my primary asterisk server crashes or
shutsdown somehow). But asterisk only tries to read the file at the time of
and at the time of switch-over mv to outgoing directory.
Cheers
On Fri, Nov 22, 2013 at 12:36 AM, Steve Edwards
asterisk@sedwards.comwrote:
On Thu, 21 Nov 2013, Rizwan Hisham wrote:
Hi all,I am syncing call files on my secondary asterisk server but
without permission to read for asterisk. So
Hi all,
Is there any way of originating calls in future without using call files?
We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we activate
the second server but we lose the call files.
I could have a cronjob on
Hi All,
I need a softphone (PC/Mobile) which does auto redial in any case
(noanswer, answer, busy, congestion etc) after a given time interval. So
if the time interval was 5 secs, it would dial last number dialled after
every hangup (or every failure to dial).
Does anyone know such feature in a
Hi All,
I want to set the video_mode of the confbridge dynamically in the dialplan.
SO say if 5 users join the conference with follow_talker mode, it should
work like that (and it does). But if 6th user changes the video_mode to
first_marked and gets marked in the dial plan and joins the
Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.
Cheers
On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima
takehiro.dream...@gmail.com wrote:
Thank you so much.
OK, I understood that to transfer the call t is usually used, is it
right?
And I should
.
Cheers
On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Thanks a lot loan. Will try it today.
Cheers
On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias indre...@gmail.com wrote:
Updated dialplan: fix a typo when using MOH variable and now you have
truly dynamic
)
On Tue, May 10, 2011 at 9:57 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Very nice Loan. Here is the chat-room dialplan with a little tweek which
lets you set the volume up/down or mute/unmute the song.
Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song
[chat-room
/curse MOH - do you have some links to
your mp3 files? :)
BR,
Ioan (with capital i)
On Tue, May 10, 2011 at 6:59 PM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
Ooops,
here is the correct version, Missed the capital X option in meetme before
which lets you control the volume etc
, May 6, 2011 at 6:30 PM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
other
, 2011 at 8:47 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Thanks for the reply. I looked into the G option of Dial applications. No
problem with that but How do I create a ghost call?
My dial plan will look like this:
Caller A calls Caller B normally:
exten= _XXX,1,SomePreDialApps
Hi All,
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
other). I found this post which talks about creating a ghost call with the
is current) and see if your issue has been resolved.
Thanks,
--Warren Selby, dCAP
On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
Yes I have it there, here the content of the file:
i think the code is buggy,
here is a comment from the function which generated
or not.
*/
This function is used all over.
On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store
Hi list,
yesterday I converted my voicemail.conf to realtime voicemail and also
configured to store the voicemessages in a database using odbc as described
here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail and
here
google for adaptive cdr. in asterisk.
On Sun, Apr 17, 2011 at 3:58 AM, John Jolly jgjo...@gmail.com wrote:
I have a particular DID that when called will prompt the user to enter the
caller id that they want to be displayed followed by it prompting for the
phone number to dial. How would I go
not
limit your call counts. We use code and the GROUP_COUNT to limit calls. If
you use it right it is rock solid.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
--
*From*: Rizwan Hisham rizwanhas...@gmail.com
*Sent*: Monday, April 04, 2011 12:30 PM
Do this:
exten= _0522XX,1,Goto(${CONTEXT},0520${EXTEN:4:},1)
you can also use the dial command for this as well
exten= _0522XX,1,Dial(Local/0520${EXTEN:4}@${SOMECONTEXT})
replace ${CoNTEXT} and ${SOMECONTEXT} with name of your contexts which
contains 0520 numbers.
I have not tested
Hi everyone,
one of our users last night bypassed asterisk call-limit limitation. I have
no Idea how. Is it possible? Is there a bug in asterisk that can be
manipulated for this purpose?
The call-limit variable was to 2, and the user initiated 169 calls in 2
minutes each has duration at least 8
You have to use adaptive cdr for this functionality. In 1.8 the conf file
for adaptive cdr is cdr_adaptive_odbc.conf. The sample conf file should tell
you everything.
If you are using some other cdr engine then you will have to jump into the
code of asterisk to make it log the item you want,
Here is a better link for DUNDi
http://exain.wordpress.com/2010/07/03/asterisk-basics-and-load-balancing-via-dundi/
skip the part which you know already
On Wed, Mar 16, 2011 at 7:44 AM, Henrique Fernandes sf.ri...@gmail.comwrote:
[]'sf.rique
On Tue, Mar 15, 2011 at 8:36 PM, Paul Belanger
Never mind. Its in seconds :)
On Tue, Mar 15, 2011 at 6:48 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi all,
What is the unit of asterisk AMI events timestamp value?
milli/micro etc ?
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E
Hi all,
What is the unit of asterisk AMI events timestamp value?
milli/micro etc ?
--
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 6767 26
E: rizwanhas...@gmail.com
W: www.axvoice.com
--
_
--
You can try changing the priority of
'1104' = 1. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]
tp this
'1104' = 2. Goto(smvoice-mediaport-public-address,s,1) [pbx_config]
On Tue, Mar 15, 2011 at 6:59 PM, Jerry Geis ge...@pagestation.com wrote:
I am using asterisk
: Rizwan Hisham rizwanhas...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 15 Mar 2011 19:03:33
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
1.8 supports static peers along with realtime peers. I have tested.
On Mon, Feb 21, 2011 at 11:04 PM, Ricardo Carvalho
rjcarvalho.li...@gmail.com wrote:
Thanks Faisal, in fact I made a test that confirmed that in realtime
asterisk doesn’t supported static peers, like you told me.
Do you know
You can register multiple end users with only one sip account but asterisk
does not support ringing all the registered phones on single account.
Whenever a new registration comes, asterisk updates its contact info in
memory. So if the registration is coming from multiple end users (multiple
ip
If you're in the market for a custom solution for whatever reason, there's
more than a few of us who can write a custom prepaid solution. I've done
about 7 so far personally and I know there's more like me out there
Yes you are right. I am now one of them (i took the red pill :)
On Sun, Mar 6,
If you can post your extensions.conf, sip.conf and features.conf then maybe
some one can understand and help with your problem.
Thanks
On Sat, Mar 5, 2011 at 5:42 AM, Francisco Javier Cintrón Olguín
fcintr...@gmail.com wrote:
I have 2 ip phones linksys spa921 and 1 normal phone connected to a
Hi all,
I have a problem with CDRs when doing call transfers. I am using * 1.8.2.3
with cdr_odbc.
As most of you may already know, CDRs and call transfers dont go along very
well in *. I mean the developers team have done their best to bring it to an
acceptable level. But still it cannot meet the
never gave it a
second thought.
I like your idea of a gateway asterisk as well. Will try it.
Thanks
On Tue, Mar 8, 2011 at 3:27 PM, Klaus Darilion klaus.mailingli...@pernau.at
wrote:
Am 08.03.2011 11:05, schrieb Rizwan Hisham:
Hi all,
I have a problem with CDRs when doing call transfers
Anymore suggestions please.
On Tue, Mar 8, 2011 at 3:36 PM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Thanks Klaus,
Actually I got my idea from CEL. But I am more familiar with AMI, plus CEL
generates too many events for a single call. I dont want that, I already
have a library of routines
You can use threads and queues in your program to interface with AMI. Your
main thread should get all the events from * and based on some logic enqueue
it. Some other thread should be listening to the queue and in that thread
you are free to read the input whenever you want. This way you are free
You can use exten pattern matching for un allocated numbers, say exten=
_X.,1,Goto(somewhere) will match all the numbers on priority 1. But make
sure you match full extension numbers first which are allocated. Also this
extension is a security risk as well. It is recommended that you use a
filter
Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of
them in the middle of the night. And my asterisk server has no record of
these calls. The customers were getting irritated as you can imagine. I
, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.comwrote:
Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of
them in the middle of the night. And my asterisk server has no record
Carvalho.
On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham rizwanhas...@gmail.com
wrote:
Hi all,
The problem I have been experiencing since last month is that some of my
customers are getting calls with Asterisk Unknown caller id. Most of
them in the middle of the night. And my asterisk
Any suggestions on encrypting the sip and rtp. I have done some googling on
it. looks like it is not supported by most end point devices or service
providers. But still your thoughts will be appreciated on this subject.
On Mon, Feb 28, 2011 at 6:13 PM, Rizwan Hisham rizwanhas...@gmail.comwrote
Thanks Mr. Kevin.
Can anyone please also tell me which firewall is best suited for
asterisk/sip attack prevention. Is there any firewall built specially to
address sip security problems?
On Mon, Feb 28, 2011 at 6:38 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/28/2011 07:27 AM, Rizwan
Hi there everyone,
I am a bit confused these days due to some problem I am having. Its not a
technical problem. Asterisk is working fine. Most of the users are happy,
but some handful of users are getting calls in the middle of the night even
though they have enabled Anonymous Call Rejection
use the timeout option in the Dial application like so
Dial(SIP/trunk,120)
If you dont specify the timeout the default timeout used bya sterisk is
probably more than 60 seconds.
On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote:
Hi
Does anyone know how i could extend
...@firedrake.orgwrote:
On Thu, Feb 24, 2011 at 03:15:34PM +0500, Rizwan Hisham wrote:
Still last night there was a call to a customer. Plz help me figure out
the
solution for this problem.
Can you be sure that the call _is_ coming through your Asterisk server,
rather than being the result of random
try this
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
On Sat, Feb 19, 2011 at 5:00 AM, asterisk asterisk aster...@ck-lee.comwrote:
I have a sip trunk connecting to a huawei softx3000. At the moment, I can
register and dial in.
However, peer status shows not reachable
sip show
Its a pure VoIP setup. no cards.
On Thu, Feb 24, 2011 at 7:12 PM, Satish Patel satish...@hotmail.com wrote:
Do you have PRI card or FXO card?
--
Sent from my iPhone
On Feb 24, 2011, at 5:28 AM, Rizwan Hisham rizwanhas...@gmail.com wrote:
Thats what im unsure about. I think the calls
you can also set some kind of authentication on the extensions for example
ask for a pin to dialout. etc
On Thu, Feb 24, 2011 at 6:51 PM, Daniel Tryba dan...@tryba.nl wrote:
On Thu, Feb 24, 2011 at 11:38:17AM +, Roger Burton West wrote:
The relevant part of my setup is something like:
Hi List,
Can anyone please tell me how to use the phoneprov.conf to provision my
client's atas. I read the file but dont know how to actually use it.
--
Best Regards
Rizwan Qureshi
--
_
-- Bandwidth and Colocation Provided by
anymore ideas anyone please?
On Fri, Oct 15, 2010 at 8:36 PM, Josef Grand josef.gra...@gmail.com wrote:
use camailio for SIP SLB
sip load balancer
- Original Message -
*From:* Rizwan Hisham rizwanhas...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial
wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
Sent: Monday, October 18, 2010 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Hi all,
I am planning to do clustering for my company's asterisk servers. I dont
know much about it, just read some articles on the internet and learned how
to use DUNDi and some basic information about clustering.
What I need to know is:
1. can i register end user with multiple asterisk servers
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Be...
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan Hisham
*Sent:* Wednesday, October 06, 2010 10:44 AM
To: Asterisk Users Mailing List - Non-Commercial
I use the following syntax for sipura i think, and it works fine for me.
exten= s,1010,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r2)
exten= s,1020,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r3)
exten= s,1030,SipAddHeader(Alert-Info: http://127.0.0.1/Bellcore-r4)
exten=
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years. I
hav'nt used it since 2008 (version 1.4.2). Now I am trying to install
1.8.0-rc2 on centos 5.5 but getting the following errors.
app_mysql.c:33:25: error: mysql/mysql.h: No such file or directory
app_mysql.c: In
Thank you all. It is now installed.
On Wed, Oct 6, 2010 at 5:04 PM, Steve Howes steve-li...@geekinter.netwrote:
On 6 Oct 2010, at 11:35, Rizwan Hisham wrote:
Hi All,
Please refresh my memory. I am trying to install asterisk after 2 years.
I hav'nt used it since 2008 (version 1.4.2). Now
Hi All,
Please can anyone tell me the difference between 1.4, 1.6 and 1.8 asterisk
versions.
Thanks
--
Best Regards
Rizwan Qureshi
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Is there any major architectural difference between 1.4 and 1.8?
On Wed, Oct 6, 2010 at 7:17 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan
, Oct 6, 2010 at 9:58 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 6 Oct 2010, Rizwan Hisham wrote:
Is there any major architectural difference between 1.4 and 1.8?
Nope. The developer's just got tired of typing .4
Of course, the joke's on them -- 1.8 is only .4 better than 1.4
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
in advance.
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
The fax is originated from a fax machine connected to an ata which supports
t38.
On Wed, Sep 24, 2008 at 11:54 PM, C F [EMAIL PROTECTED] wrote:
On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham [EMAIL PROTECTED]
wrote:
Hi all,
Sorry to interrupt. I need some help regarding fax passthru mode
/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
of asterisk but they are not, already downloaded
and checked in asterisk 1.4.21.
How can i install these applications. Are there anyother components required
to make my asterisk a fax-passthru system.
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation
callerid=adf xyz 123
accountcode=6:0:adf
amaflags=default
disallow=all
allow=g729
allow=ulaw
allow=alaw
allow=gsm
am i doing something wrong here?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
hi all,
Is there an option of dtmf passthru mode in asterisk. If yes, how can i do
it?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register
at 6:05 PM, Alex Balashov [EMAIL PROTECTED]wrote:
What is DTMF passthru?
DTMF is regenerated by default. If the DTMF mode is inband, it's simply
part of the audio stream. If it uses named RTP events, those are
regenerated on the other call leg.
Rizwan Hisham wrote:
hi all
with a username.
Rizwan Hisham wrote:
Hi all,
I am having a problem with sip uri incoming calls. I have 2 asterisk
servers both are 1.4.2. http://1.4.2. i dial sip uri from one asterisk
server which sends the call to the other asterisk server by seeing its
domain name in the uri. Invite
there is no transfering)
I googled a little on strict and loose routing but i did not get it. maybe
someone here can help me solve this problem.
VERSIONS
asterisk 1.4.2
zaptel and libpri 1.4.0
I can send you core debug if you want it.
--
Best Regards
Rizwan Hisham
, Anthony Francis [EMAIL PROTECTED]wrote:
Rizwan Hisham wrote:
Hi all,
asterisk is giving me tough time. its been 3 days I am trying to
originate outgoing call using manager api/callfiles.
I would say remove the @TRUNK-OUT part and make sure that the context
you send the call to knows
. .$calltime.\n;
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE
have done it, and its working fine. but still expecting to receive some new
ideas.
On Wed, Aug 6, 2008 at 2:12 PM, Rizwan Hisham [EMAIL PROTECTED]wrote:
hi all,
i just finished developing some incoming call features in a macro. that
macro gets executed everytime an incoming call is received
in a dialplan.
I am using asterisk1.4.2 and asterisk addon1.4.0 package for mysql
connectivity.
Thanx in advance
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix
Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options
can solve my multiple cdr problem?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
, but i could
not find a satisfactory explanation for the this statement. Can anybody help
me understand the switch statement?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
statement?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
same is the case in 1.6, i cant set the variable still.
On Thu, May 8, 2008 at 8:43 PM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple perl script to connect with ast manager api. the
script
tries to set
Thanx a lot.that was it. will never forget to remove the new
character again. Now its working fine.
On Fri, May 9, 2008 at 4:31 PM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
same is the case in 1.6, i cant set
Can anybody help in parsing the manager events efficiently? Any ideas?
On Fri, May 9, 2008 at 5:07 PM, Gunārs Grundāns
[EMAIL PROTECTED] wrote:
On 5/8/08, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 08, 2008 at 07:39:39PM +0500, Rizwan Hisham wrote:
Hi all,
I am using a simple
not
exist. Can anybody tell me why its doing so, coz i can see on cli that the
channel exists. If i try to set the variable without stating the channel
then it sets the global variable, but i want to set the channel variable.
Anybody has a solution to this problem?
--
Best Regards
Rizwan Hisham
-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
Also it will be great if anybody can tell where i can find the explanation
of all the warnig codes and error codes of asterisk if there is any.
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Why is it doing so?
On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield [EMAIL PROTECTED]
wrote:
In article [EMAIL PROTECTED],
Rizwan Hisham [EMAIL PROTECTED] wrote:
I have been seeing a lot
asterisk
cards. if someone knows where to buy cards in pakistan, plz tell me about
it.
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
tell me how to take
out the conf files then it will also be very helpfull.
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
Hi,
Does anyone know the purpose of /n attached at the end of the dial
command below
Dial(Local/[EMAIL PROTECTED]/n Local/[EMAIL PROTECTED]/n)
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
time to
30,000 mili seconds. Can it be done in the dialplan? or should i jump into
the code?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
of the call or it just transfers the call and
then all of the responsibility of the call is handled on the other server?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
. The
From header show the name and number which i set before dialing but on cli
it shows only name:
From: salman sip:[EMAIL PROTECTED]:5238;tag=as5100f7b2
Any one knows what should i do to solve this problem?
--
Best Regards
Rizwan Hisham
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Best Regards
Rizwan Hisham
Adding fromuser option in trunk declaration in AST1 has solved all
problems though.
On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
I am having a strange problem. I am using my asterisk server AST1 to
register with another asterisk server
[13520]: chan_sip.c:8117 check_auth: username mismatch, have adf,
digest has abc*
Any solutions to this problem?
On Wed, Feb 27, 2008 at 4:36 PM, Igor A. Goncharovsky [EMAIL PROTECTED] wrote:
Rizwan Hisham wrote:
I am having a strange problem. I am using my asterisk server AST1 to
register
1 - 100 of 266 matches
Mail list logo
