mention the complete scnario and your sip.conf.
Regards,
Faisal
(sent from phone)
Rafael Visser rafael_vis...@hotmail.com wrote:
Hi Gurus..
I use asterisk for just for ivr.
My issue is that when the switch changes it's host name from MSSASU1.MYDOMAIN
to MSSSASU1.MYDOMAiN.COM or MSSSASU1
hi,
you can simply avoid this by using local ring r option in dial command.
azterisk pass local ring voice to caller and will not bridge b leg audio until
b leg is answered.iin
Regards,
Faisal Hanif
(sent from phone)
Steve Davies davies...@gmail.com wrote:
Hi SIP Gurus,
I've tried to find
You can create trunk/route specific dial command parameters.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Friday, August 24, 2012 8:40 PM
To: Asterisk Users Mailing
The easiest way for you to use MySQL-Relay or MySQL-Proxy with ODBC.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Owais Ahmad
Sent: Tuesday, June 05, 2012 7:54 PM
To: Asterisk Users
If I understand correct you need to increase qualify value.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Tuesday, May 22, 2012 5:02 PM
To: Asterisk Users Mailing List
Why don't you use FQDN in phone instead of IP of server and configure DNS
Server to failover resolve to next IP while set SIP reg expiry same as DNS
TTL.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
I have tried EyeBeam and it worked fine with x members audio conference
however it need resources (Processing + RAM) per additional line.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
In hardware I used some snom phones up to six lines. You can check on
http://www.snom.com/ for appropriate model.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, November
Hi Everyone,
I am in search of a reliable open source tool for the real time monitoring of
ASR ACD, so any help or suggestions will be highly appreciated.
Regards,
Faisal Rehman--
_
-- Bandwidth and Colocation Provided
U can also use VICIDIAL for it
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal
Shriyan
Sent: Saturday, August 20, 2011 12:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
+musiconhold.conf) so it
will get all mog class info from DB in realtime.
2-Before dialing a call create a moh class in db by hitting a query and
associate your target voice.mp3 files with that class.
3-Dial the call and associate that moh class using parameter.
Regards,
Faisal Hanif
You can have all this plus a lot more. What you need is configurations and
dialplan code.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neo haux
Sent: Thursday, August 11, 2011 6:12 AM
To: asterisk-users@lists.digium.com
Subject:
If you take a bit deep analyses on SIP packet you will be able to understand
the issue,
Iptables filter on layer-3 while SIP is on layer-7. It is easily possible to
generate a SIP packet with different source-ip than physical interface.
You can also simulate it if you set
Dundi just give you location of extensions. For ring you should have capable
dialplan and peering.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Wednesday, August 03, 2011 1:06 PM
To: Asterisk Users Mailing List -
switch.
Regards,
Faisal Hanif
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Did you tried to execute Set(CALLERID(num)=you-required-callerid)?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman
Sent: Friday, July 29, 2011 1:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
.
If anyone need it contact me direct at email imfa...@gmail.com I will send
the software as attachment.
Regards,
Faisal Hanif
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Yep. Look the dtails of option of Dial command and features.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinod
Dharashive
Sent: Friday, July 29, 2011 8:51 PM
To: asterisk-users@lists.digium.com
I have tried asterisk on windows XP using Cygwin and it worked fine.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Antonio Modesto
Sent: Thursday, July 28, 2011 1:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
If it is just matter of billing you can pass billing related info in
additional SIP headers on single trunk.
If you must need multiple trunk you can add multiple IPs of different subnet
class to both interfaces and configure asterisk to listen of all IPs. Then
use one trunk per IP Subnet
asterisk. You can compile asterisk as portable and copy compiled asterisk to
multiple locations/directories (as many instances you need). Each copy will
have its own configuration files where you can play as you like.
Regards,
Faisal Hanif
Did u tried by disabling relaxdtmf?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Friday, July 08, 2011 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Problem in
I think yes. Check queuetimout variable.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deka, Rajib IN
MAA SL
Sent: Friday, July 08, 2011 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] dialout time
Use Filter command in dia-plan to get numeric only string,
Set(MYNEWCLI=${FILTER(0123456789,${CALLERID(number)})
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday
As asterisk is an B2BUA you can handle 503 at asterisk and hang caller end
using the response code compatible with eyebeam as
Hangup(16)
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
You can't use WaitExten to receive two digits. Use Read() command.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith
Sent: Wednesday, July 06, 2011 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Community can help you better if you provide some details about you scenario
and requirement.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Wednesday, July 06, 2011 5:03 PM
To:
Hi,
As per my experience YATE is the best option for H323=SIP Proxy.
Regards,
Faisal
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Thursday, July 07, 2011 2:48 AM
To: asterisk-users
The problem you are reporting is not related to realm but can be context or
domain.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Tuesday, July 05, 2011 11:59 AM
To: Asterisk Users Mailing
as time on each
trunk can be monitored via any queue monitoring tool. !!
or better use queue_log in realtime DB
As per my view this is most easy and optimized approach while keeping all
possible data in queue logs. Hope this will helpful for you.
Regards,
Faisal Hanif
-Original Message
If the problem always related to some specific module then try clean
recompiling asterisk if it is with random modules then check you system RAM.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina
Berretta
Sent: Wednesday, July
You have to provide channel ID to command like channel request hangup
SIP/12316156-sad4d46a5.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Wednesday, July 06, 2011 9:50 AM
To: Asterisk Users Mailing List -
Hi,
I don't think there is a way for it inside asterisk but you achieve it by
adding static route in Linux routing table and make interface having that IP
as default route for the interested IPs traffic.
Regards,
Faisal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users
When you make a call asterisk always create a channel named as below,
CheannelType/PeerName-uniquecode
Like
SIP/jon-312abf
So here jon is the peer name. This can help you to identify a peer as long
as A-Leg is active.
Regards,
Faisal
-Original Message-
From: asterisk-users-boun
Have you tried SIP session timer values in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject:
Have you installed sample configuration files package?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paolo De Michele
Sent: Monday, June 27, 2011 4:26 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] not find files
Call file are not suitable for you as asterisk process these files in serial
mode (single threaded) and in case of large number of files processing of
last file can be that much delayed that some portion of message may be
already played or the 1st phone may be hanged.
-Original Message-
If you can explain a bit more what exactly you need?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, June 27, 2011 9:16 AM
To: asterisk-users@lists.digium.com
Subject: Re:
It depend on Hypervisor. if it is full virtualization then it will not be
more than a part sharing from system resources depends on VM configuration
and processing load.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Barot
Asterisk-SNMP could be an option for u.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu
Sent: Friday, June 24, 2011 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor
Fail2ban
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, June 16, 2011 11:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to secure our Asterisk server
Try by reversing the line number of permit deny
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Thursday, March 10, 2011 6:39 PM
To: Asterisk Users Mailing List - Non-Commercial
, you have permit=172.16.16.0/24 whereas suggestion was
permit=0.0.0.0/0.0.0.0
On 3/10/2011 1:48 AM, RR wrote:
On Thu, Mar 10, 2011 at 2:12 AM, Faisal Hanif fai...@vopium.com wrote:
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use
Asterisk doesn't have all features of SBC like relay and forward request on
packet level but all depends on your scenario what you need.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, March 10, 2011
It just have ACL concept. You can add permitted IPs List to any peer then
only from that IPs user can register. If you want to permit all you can add
0.0.0.0 to ACL
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of RR
Sent: Thursday,
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] [1.8] Unable to Register: Registration denied
because of contact ACL
On Thu, Mar 10, 2011 at 12:20 AM, Faisal Hanif fai...@vopium.com wrote:
It just have ACL concept. You can add permitted IPs List to any peer
AMI is single threaded link so waiting on it will bring things to hang mode
but FastAGI dialplan is multithread. Better to manage all info by AMI in a
local hash or array and use sleep/waiting on AGI till required info
populated to hash/array by AMI.
-Original Message-
From:
When you compile asterisk you can select multiple language files by using
make menuselect additionally you find lot of free sources on internet for
language files. Simply create a folder with language short-code in sounds
and then set channel's language variable to that short-code.
-Original
-users] [1.4] Reading phone number the French way?
On Tue, 8 Mar 2011 17:31:26 +0500, Faisal Hanif fai...@vopium.com
wrote:
When you compile asterisk you can select multiple language files by
using make menuselect additionally you find lot of free sources on
internet for language files. Simply create
This settings are for ISDN configurations I think.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Monday, March 07, 2011 6:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
If you dialout call without answering and allow all codec for both peers
then codec negotiation will be direct between endpoints and asterisk will
only do media pass-through.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
http://www.danielaliaman.com/blog/files/AsteriskSNMPtutorial.pdf
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Sunday, March 06, 2011 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Well a solution for you to put original context name in variable and then
use that variable in goto statement on h.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Friday, March 04, 2011
1-Check signaling type on gateway PSTN ports
2-Set RTP timeout in SIP trunk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Friday, March 04, 2011 7:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
AstPP jbilling
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Saturday, March 05, 2011 10:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
You can find lots by googling but none can give realtime stats as it depends
on network. Packet drop, retransmit, codec type will make lot of vibrations
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, March
I don't remember exact name but there are two authorities which provide
real-time portability information online but you need subscription.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
www.numberingplans.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Piotr Górski
Sent: Thursday, March 03, 2011 12:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Testing from where
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
If your PRI provider permit you to associate any ANI to any Circuit-ID you
can do this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, February 24, 2011 12:17 PM
To: Asterisk Users Mailing List -
Well you simple use dtmfmode=info in peer configuration of Snome phone.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, February 18, 2011 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
The difference you will feel when using callback files or AMI.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles
Sent: Friday, February 18, 2011 1:31 PM
To: asterisk-users@lists.digium.com
Subject:
This is not Digium's customer support address but free public emailing list
for asterisk user's contributed by community volunteers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jan Zieher
Sent: Friday, February 18, 2011 2:19 PM
Did you checked if you extension.ael doesn't have syntax error?
Did you upgraded anything after last compile?
Or
Try a clean recompile
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Borin
Sent: Friday, February
The only specific you need to do in extensions.lua is create a table to put
your extensions in like,
Extension{
}
Else all will be LUA code and all asterisk applications can be called as
app.application_name.
Regards,
Faisal Hanif
From: asterisk-users-boun
: ast_compile_ael2
On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif fai...@vopium.com wrote:
Did you checked if you extension.ael doesn't have syntax error?
I think there is no error. I loaded the standard ael first (provided by
asterisk) then my test config, got the same result.
Did you
If you are using asterisk 1.8.x you don't need to type \ for spaces you can
write simple query and use spaces as normal it will work fine.
Faisal
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felipe
Figueiredo
Sent: Friday
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To:
It is in client but not in asterisk sip channel
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to diable echo
Did you executed Answer() before it?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] function Echo() doesn't work
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] function Echo() doesn't work
Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS
Stick). Just only no echo on SIP. Any suggestion?
2011/2/16 Faisal Hanif fai...@vopium.com
Did you executed Answer() before
-, ) in new stack
== Spawn extension (von-voip-provider, 1174614, 2) exited non-zero on
'SIP/sipgate-account-'
here is the log. It is as same as I got from CAPI and Datacard. I just didn't
hear the echo from SIP connection.
2011/2/16 Faisal Hanif fai...@vopium.com
Check
in Queue?
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] function Echo() doesn't work
I tried to set allow=all in sip.conf. But it still doesn't work.
2011/2/16 Faisal Hanif fai...@vopium.com
I faced same issue for sipgate but got it resolved by allowing all
You can do it using callback files or AMI.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu
Sent: Wednesday, February 16, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play one audio file to the
seconds.
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo
Carvalho
Sent: Wednesday, February 16, 2011 9:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] trunk not working
: Faisal Hanif
Subject: Re: [asterisk-users] trunk not working if I register a phone at the
same IP as the trunk peer's IP
Isn't this a limitation that can be surpassed with some configuration that
I'm lacking in my sip.conf or extensions.conf of my asterisk?
Ricardo.
On Wed
EAGI could be your target application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, February 16, 2011 11:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pipe audio
You can simply use Portable LinuxLive USB Creator 2.6 or grub4dos. And
make your USB bootable by any Linux Live ISO.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan
Sent: Wednesday, February 16, 2011 11:24 PM
To: Asterisk Users
In case of asterisk you simply can't accept registration from an IP which
you have mentioned as static host for IP authentication.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:37 PM
You need to use relay request in your SBC instead of forward.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pezhman Lali
Sent: Tuesday, February 15, 2011 5:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Check if dtmfmode is properly set on SIP trunk ask with your carrier which
dmtfmode they support.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk
asterisk
Sent: Wednesday, February 16, 2011 5:39 AM
To: Asterisk Users Mailing
You may need to share your LUA code and the extension your call is need to
execute.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Pires
Sent: Wednesday, February 16, 2011 3:29 AM
To: Asterisk Users
List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:
Check if dtmfmode
-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not detected, time out
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I check further?
On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif fai...@vopium.com wrote:
Check if dtmfmode is properly
. Can pluged to asterisk PBX
machine and used as FXO device.
Regards,
Faisal Hanif
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan
Sent: Wednesday, February 16, 2011 10:49 AM
To: asterisk-users@lists.digium.com
Subject
Well I think you need major changes as application in android run in sandbox
instead of direct Linux APIs. Till now no news on it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil
Sent: Monday, February 14, 2011 6:46 PM
To:
Better to report a BUG to cisco.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Miller
Sent: Monday, February 14, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Cisco 7960 asterisk 1.8.22
You may need to provide some more scenario detail
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, February 14, 2011 7:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] issue with
Go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
and use proper parameters to dial command to pass early media.
-Original Message-
From: Benoit Panizzon
Sent: Thursday, February 10, 2011 4:08 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Early audio SIP
Well. I suggest to use DB function instead of modifying asterisk source. You
can add one additional column and write and after-insert trigger in your cdrs
table which convert dattime to your required format and update the value of
added column.
From: Rodrigo Lang
Sent: Thursday, February 10,
The settings you are asking varies in different countries and providers. You
need to contact you provider for it.
From: Roi Stork
Sent: Thursday, February 10, 2011 9:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] zaptel/dahdi settings for singtel E1
There are some flags in general settings of dialplan which enable/disable
modify this behaviors of dialplan. Have a look on sample extensions.conf for
general tab settings. I will see if I can have time today to tell you exact
parameter name.
From: Dovid Bender
Sent: Thursday, February 10,
But if you are getting calls all the way on VoIP then you can have calls in HD
audio using HD audio codec on all locations (Server and Client). In that case
you either need use some available 3rd party solution which uses packet
capturing to trace the calls and record call using packet capture
Yes. The technology need to be used on LAN switches is port mirroring or
line tapping
-Original Message-
From: Sherwood McGowan sherwood.mcgo...@gmail.com
Sent: Tuesday, February 8, 2011 7:34am
To: Asterisk Users Mailing List - Non-Commercial Discussion
Why don't you use single callfile and set CLI and other perameters in dial-plan
as unique as you need?
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
${HANGUPCAUSE} value is available on h extension.
-Original Message-
From: Shariq Khan shariqrazak...@gmail.com
Sent: Tuesday, February 8, 2011 8:30am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: [asterisk-users] ${HANGUPCAUSE}
Just verified I faced the same issue once and got it reolved by adding /n like
Channel: [mailto:Local/0036701234567@CustomCallOut-1/n]
Local/0036701234567@CustomCallOut-1/n in you case.
-Original Message-
From: Tamás Dajka tda...@gmail.com
Sent: Tuesday, February 8, 2011 8:49am
To:
()
exten = h,1.NoP()
Regards,
Faisal
-Original Message-
From: Feng Xu felto...@yahoo.com
Sent: Monday, February 7, 2011 11:55pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] standalone NOTIFY message handling
to dialplan and
totally in controll.
Regards,
Faisal
scussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Can a duration limit be specified in spool call
file?
Bruce,
All in all, I don't think it's that hostile, it just goes through
cycles...maybe a good number of us may indeed
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