Hi
I am using Asterisks as client. By console dial I can make calls.
When do "dial s" from console it wil play demo files that I can here
from headphone connected to asterisk running system(Android OS).If I
play gsm file noise is coming,but asterisk is not playing wav
files,below is the err
sory satish...my thunderbrid was not load.
Thanks for reply...
On 03/02/2011 09:59 AM, Nikhil wrote:
Any reply..
On 03/01/2011 02:50 PM, Nikhil wrote:
Hi
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Thanks
Nikhil
At least 3 replies earlier today on list.
Nikhil wrote:
Any reply..
On 03/01/2011 02:50 PM, Nikhil wrote:
Hi
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Thanks
Nikhil
--
Any reply..
On 03/01/2011 02:50 PM, Nikhil wrote:
Hi
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Thanks
Nikhil
--
_
-- Bandwidth and Colocation P
Hi all and thanks for reading.
I am experiencing a frustrating issue with asterisk where on some
calls the volume suddenly drops to inaudible o completely fades away
for a time. If you hold on long enough (20 to 30 seconds) the sound
will come back.
My asterisk server is on a public IP, and basic
On 11-02-27 09:12 PM, Stuart Longland wrote:
I've tried researching this, and so far, have struggled to find any
contemporary information on the issue, so I do apologise if asking this
irritates people who have answered this before.
I have managed to set up Asterisk 1.8 on the web server here.
On 03/02/11 01:37, Daniel Tryba wrote:
> On Mon, Feb 28, 2011 at 09:41:38PM +1000, Stuart Longland wrote:
>> Indeed, most motherboards do come with Ethernet on board. This one came
>> with one gigabit Ethernet interface. However, we needed another for a
>> connection to an ADSL router (acting in
On Tue, 1 Mar 2011 08:46:39 -0600, "Cary Fitch"
wrote:
>Wink, I think is a "start" protocol aks "wink start". It is like a flash,
>but happens as part of the predialing/dialing process.
Thanks guys.
--
_
-- Bandwidth and Col
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Tuesday, March 01, 2011 11:31 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Caller ID
_
Fro
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Tuesday, March 01, 2011 11:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Caller ID
We do not get caller ID (nam
We do not get caller ID (name) on our telco lines.
However we have a few single line extensions with consumer type handsets
that ring at odd hours with "Asterisk" before the phone is picked up, and
"Out of Area" after it is picked up.
I have read that "Asterisk" is what is reported by Aste
Hi,
here is an example:
http://www.asteriskguru.com/tutorials/mixmonitor.html
Enjoy it!
Best regards,
Fellipe
Date: Tue, 1 Mar 2011 17:06:32 +
From: salah.elharit...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] records inbound and outbound calls
thank you
thank you so much but i don't know how can i do
could you please give an example to record an external call or which file I
must to configure
Thanks a lot
2011/3/1 Danny Nicholas
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, March 01, 2011 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] records inbound and outbound calls
Hello List
i have asterisk installed in our call centre i have configured the snom
phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com
i have just one question how can i do in order to record all the calls
automatically in our server
Thanks and regards
--
___
Try this - it says it is for 1.8 but might work in 1.6
http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepika
Nijhawan
Sent: Tuesday, March 01, 2
SIP_HEADER() gives you only access to headers of the initial INVITE request
(and not, for example, the final BYE message)
How will I check sip response with this like 404 or 503?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
On Tue, 2011-03-01 at 10:08 -0600, Terry Wilson wrote:
>
> On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote:
> >
> > Seeing that 1.8.3 had been released I updated our main test server
> > to
> > that from 1.8.2.2 using the digium yum repo.
> >
> > All audio had been working fine on this server bef
On Mar 1, 2011, at 4:19 AM, Ishfaq Malik wrote:
>
> Seeing that 1.8.3 had been released I updated our main test server to
> that from 1.8.2.2 using the digium yum repo.
>
> All audio had been working fine on this server before the update but
> after the update I experienced the same as I did wit
On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote:
> I'm in the process of testing a TLS/SRTP install. My experience is improving
> with each new challenge, but this one is a great test of my 2 month
> experience with Asterisk.
> [myphones]
>
> ;exten => 6001,1,Dial(SIP/6001)
> ;exten => 6001,
On Mon, Feb 28, 2011 at 09:41:38PM +1000, Stuart Longland wrote:
> Indeed, most motherboards do come with Ethernet on board. This one came
> with one gigabit Ethernet interface. However, we needed another for a
> connection to an ADSL router (acting in bridged mode so we do the PPPoE
> directly).
-Original Message-
From: Bob Beers [mailto:bob.be...@gmail.com]
Sent: 01 March 2011 13:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Deepika Nijhawan
Subject: Re: [asterisk-users] Failover Routing
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan
wrote:
> Ya, below i
It says it for asterisk1.8. I am using asterisk1.6, can we use this function
in this version.
Is it possible for you to give example on how to use?
-Original Message-
From: Bob Beers [mailto:bob.be...@gmail.com]
Sent: 01 March 2011 13:24
To: Asterisk Users Mailing List - Non-Commercial Di
Wink, I think is a "start" protocol aks "wink start". It is like a flash,
but happens as part of the predialing/dialing process.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Tuesday, March 01, 2011
On Tue, 1 Mar 2011, Nikhil wrote:
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Asterisk chooses a file encoding based on the channel encoding. If your
channel is encoded as GSM, Asterisk will not look for a .wav of t
Do you have complied wav file support in asterisk?
--
Sent from my iPhone
On Mar 1, 2011, at 9:11 AM, "Danny Nicholas" wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Tuesday, March 01,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Tuesday, March 01, 2011 3:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] wav files are not playing asteris
http://www.carrieraccessbilling.com/telecommunications-glossary-w.asp
From: asterisk-users-boun...@lists.digium.com on behalf of Gilles
Sent: Tue 3/1/2011 7:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] [zapata.conf] What is "wink"?
Hell
On Tue, Mar 1, 2011 at 8:09 AM, Deepika Nijhawan
wrote:
> Ya, below is my routing:
> Exten => 1234,1,Dial(SIP/abc)
> Exten => 1234,n,Dial(SIP/xyz)
>
> If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE} variable.
> For this I don't want it to try SIP/xyz.
> But overall, if we get SIP
Ya, below is my routing:
Exten => 1234,1,Dial(SIP/abc)
Exten => 1234,n,Dial(SIP/xyz)
If 1234 is unallocated on SIP/abc it returns 1 in ${HANGUPCAUSE} variable.
For this I don't want it to try SIP/xyz.
But overall, if we get SIP 4xx reason then call should hangup like it sends
back 404 not found
I think he meant the opposite he is sending calls to a sip trunk and would like
to know when to failover and send calls to a different sip trunk
I haven't really looked at this but maybe check the header of the packet for
which response your getting
Also are you sure you are getting the hangup
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,
If I use dialstatus variable, it doesn't give exact reasons for failure like
for unallocated numbers it sends Congestion. Whereas, for unallocated number
I don't want to go to failover routing. But need to go to failover routing
for other congestion reasons.
So, is there any way to check
Hello
I couldn't find information about what "wink" is in zapata.conf:
www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters
Does someone know what it is, and how it differs from flash?
Thank you.
--
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-- B
Thanks all. I appreciate your support a lot. Your suggestions helped make a
game plan finally. Otherwise i was just shooting in the dark before. Anyways
here is what I am going to do now.
install a better firewall, possibly the one mentioned by mr j stapleton and
tighten its security settings. use
Hello list,
I'm having a problem with a BRI card, a B200E (PCI-e) card on an Intel
server with a riser that I'm guessing is what's making the problem.
The output of lspci -vvv is:
07:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Colo
Hello Phil,
Yealink is amazing product. The factory makes some excellent pieces of SIP
phones, especially its a value for money at their current selling prices.
Make sure you buy it from a local vendor, helpful for warranty issues.
Regards
Mike
On Fri, Feb 25, 2011 at 10:34 PM, --[ UxBoD ]--
On Mon, 2011-02-28 at 13:40 +, Ishfaq Malik wrote:
> I've just installed 1.8.3-rc3 on a test server as we really needed that
> deadlock involving REFER fix on our server but now I'm having an odd
> issue with one way audio with a specific type of call.
>
> If I do extension to extension calls
Hello
I'd like to know what my options are to bridge two channels after
calling each through Dial().
I know about MeetMe, Conference, and Konference. Are there other
options available just to bridge two calls?
www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
www.voip-info.org/wiki/view/Asterisk+c
Try insecure=port,invite
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-us
You don't need to put quotes "" around AGI name.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oguzhan Kayhan
Sent: Tuesday, March 01, 2011 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
S
Update,
My first question solved already.
There was an error on my agi script.
But second problem still valid.
On Tuesday, March 01, 2011 11:04:50 am Oguzhan Kayhan wrote:
> Hello,
> I want to make an agi script to match incoming DIDs with usernames.
>
> I tried to do such entry in incoming tr
Hi
I try to play a wav file in asterisk ,but its accepting only gsm
files.Do u know where I need to change to make it works.
Thanks
Nikhil
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Aster
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI("did.php")
exten = 3130XXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:201
On Mon, 28 Feb 2011 12:53:36 +0100, Gilles
wrote:
>Flash() now works :-)
>
>However, after putting call #1 on hold, Asterisk is unable to dial the
>second number:
>== extensions.conf
>[from_fxo]
>exten => s,1,Wait(2)
>exten => s,n,Set(GLOBAL(CID)=${CALLERID(num)})
>exten => s,n,Hangup(
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