My Dear,
You have used 'L03' (alphabet 'l' *EL*) in dial command instead of '103'.
Shariq
On Mon, Aug 25, 2008 at 3:26 PM, ims.asuser ims.asuser [EMAIL PROTECTED]
wrote:
Hi all,
I have a very weird problem.
I have 2 users (103 and 105). They are able to register in Asterisk, but
they
I m facing problem with TDM2400P pstn card. When someone dials, the voice
quality is crappyInstead of hearing.
Echo cancel almost works, but the callee hear what they describe as a
'background crackle/buzz' coming back when they talk.
I tried with rxgain txgain tuning but ... no effect.
Beginner Question
---
Is it necessary to have an static or fixed IP for asterisk for dialing out
on SIP.
Is there any effect on the call, if i dont have any static IP only for
outgoing calling?
Shariq
___
-- Bandwidth and
Thank u very much, Russel.
I will definitely contact with digium, then updates you.
Shariq
On Thu, Aug 28, 2008 at 5:27 PM, Russell Bryant [EMAIL PROTECTED] wrote:
Shariq Khan wrote:
I m facing problem with TDM2400P pstn card. When someone dials, the
voice quality is crappyInstead
Use winamp media player with their gsm extension.
Shariq
On Thu, Aug 28, 2008 at 10:40 PM, Gustavo A Gonzalez [EMAIL PROTECTED]
wrote:
Hi folks!
I want to play gsm agents recordings from a web interface, to do that,
someone knows some media player that launches when I click on the file
When i dial out any number through PRI it gives the following error every
time, while incoming calls works fine
I have sangoma E1 PRI card.
-- Executing Dial(SIP/2000-081b9938, Zap/g0/0501125||) in new
stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g0/0501125
PROTECTED] wrote:
Octavio Ruiz wrote:
On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman [EMAIL PROTECTED] wrote:
Octavio Ruiz wrote:
On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan [EMAIL PROTECTED]
wrote:
The output of a
CLI pri intese debug
at Asterisk CLI before make a test call would
What asterisk cli shows when you soft hangup these channels
Shariq
On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen [EMAIL PROTECTED]wrote:
V 1.4
When I do a show channels I get the following.
CLI show channels
Channel Location State Application(Data)
Use A2billing
http://www.asterisk2billing.org/
Complete solution for prepaid calling card also.
Shariq
On Fri, Dec 12, 2008 at 11:45 PM, David fire ddf...@gmail.com wrote:
prepaid solution for what?
2008/12/12 BERGANZ François franc...@acropolistelecom.net
Hello,
I am looking
Is there any way to listen SIP on multiple ports on asterisk. Is is possible
to define in sip.conf in the following way.
sip.conf
[general]
port = 5060
port = 5090
Regards,
Shariq Khan
___
-- Bandwidth and Colocation Provided by http://www.api
I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time
like
bindport = 5060,5061 OR
bindport = 5060
bindport = 5090
I want, asterisk to listen SIP on multiple ports. so that users where SIP
port 5060 blocked, can easily register to asterisk by using an alternate
port.
Shariq Khan
.
Shariq Khan
On Fri, Jan 1, 2010 at 11:41 PM, Gergo Csibra csi...@gmail.com wrote:
Friday, January 1, 2010, 7:12:54 PM, Alex wrote:
On 01/01/2010 01:06 PM, Warren Selby wrote:
Also, shouldn't the .php script be located in
/var/lib/asterisk/agi-bin?
Fact.
And on a live channel must use AGI
The problem is due to the wrong password for accessing mysql database.
You can discuss more on asterisk2billing forum
http://forum.asterisk2billing.org
http://forum.asterisk2billing.orgShariq Khan
On Tue, Apr 6, 2010 at 2:51 AM, Daniel Abreu dlab...@gmail.com wrote:
Hi guys. I am facing this
Is there any way, i can detect in asterisk that which party hanged up the
call either from A side or B.
Both parties are using SIP protocol. I am using Asterisk 1.4.27
Shariq Khan
0333-3501125
--
_
-- Bandwidth and Colocation
context=default
host=dynamic
call-limit=2
[1010]
username=1010
type=friend
secret=
mailbox=779000
context=default
host=dynamic
call-limit=2
--
Regards,
Shariq Khan
0333-3501125
--
_
-- Bandwidth and Colocation Provided by http
You mean, I need to check the DEVICE_STATUS of both (sip) users before
sending the caller into queue, otherwise skip the caller from going into
Queue by using ExecIf.
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 3:16 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Shariq
Dear Tarek,
IN_USE is other then the BUSY status, i want to skip the BUSY agent but not
IN_USE
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 4:07 PM, Tarek Sawah tareksa...@hotmail.com wrote:
Gareth
Usualy the queue has the ability to know if the agent is INUSE and skip
I also want to hear the experience of yours with Synway Cards.
--
Regards,
Shariq Khan
0333-3501125
On Mon, Sep 13, 2010 at 12:47 AM, Anita Hall anita.h...@simmortel.comwrote:
Hi
Does anyone have experience with Synway cards like SHD-240D-CT/PCI with
asterisk and SynAst driver
Dear Gareth,
DEVICE_STATE function is not available in asterisk, even DEVSTATE does not
work for me in asterisk 1.4.35. Any other method function to check the
channel status
--
Regards,
Shariq Khan
0333-3501125
On Wed, Sep 15, 2010 at 5:11 PM, Gareth Blades
list-aster...@skycomuk.comwrote
Hello Gurus,
Can i add ${HANGUPCAUSE} in CDR after the Dial command using h extension? I
want to add the Hangup reason of call in userfield of CDR.
Regards,
Shariq Khan
0333-3501125
--
_
-- Bandwidth and Colocation Provided
voice clear but P1 could not hear any voice.
My sip.conf is
[avaya]
type=peer
fromdomain=xx.xx.xx.xx
host=xx.xx.xx.xx
disallow=all
allow=ulaw
dtmfmode=rfc2833
canreinvite=yes
--
Regards,
Shariq Khan
0333-3501125
--
_
-- Bandwidth
Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??
--
Regards,
Shariq Khan
0333-3501125
--
_
-- Bandwidth
Danny,
Thanks for the support, but i need to hold the customer and play MOH after
answering the call. As you know that the signalling codes of SIP and ISDN
are almost same, that's why i was thinking that MOH can work on DAHDI as
well.
--
Regards,
Shariq Khan
0333-3501125
On Thu, Apr 7, 2011
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