mention the complete scnario and your sip.conf.
Regards,
Faisal
(sent from phone)
Rafael Visser 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
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
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 wrote:
>Hi SIP Gurus,
>
>I've tried to fin
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.digiu
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
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
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 Provide
onfig+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
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: [aster
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: [as
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 external-ip=some
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 -
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
Subject:
n 1.2.
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 u
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: [a
BX or VoIP 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
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
Sub
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 class.
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
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 configuration
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
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 Of
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
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
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: asterisk-users@list
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
Subjec
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 - Non-Co
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 0
ed per trunk as well 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
-
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
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
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-
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: [asterisk-users]
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-
Fr
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 i
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: [asterisk-u
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 Ast
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
Sent
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 Discussi
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 3:
, 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 wrote:
You can add following line to your peers configuration
permit=0.0.0.0/0.0.0.0
It will allow to use that peer's account
AM
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 wrote:
It just have ACL concept. You can add permitted IPs List to any peer then
only
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, Mar
[asterisk-users] [1.4] Reading phone number the French way?
On Tue, 8 Mar 2011 17:31:26 +0500, "Faisal Hanif"
wrote:
>When you compile asterisk you can select multiple language files by
>using "make menuselect" additionally you find lot of free sources on
>inter
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.
-Origina
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: asterisk-use
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: [asterisk-user
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: [ast
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] O
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] He
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'
Subje
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 4:38
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 0
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 numb
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
Sent:
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
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 - Non-C
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: F
: ast_compile_ael2
On Fri, Feb 18, 2011 at 3:29 PM, Faisal Hanif 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 upgraded any
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
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, Feb
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
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: [asteris
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
S
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
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 s
on
Cc: 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.
me is 120 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 no
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 c
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
I faced same issue for sipgate but got it resolved by allowing all codec i
on 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 wrote:
If you use
Echo("SIP/sipgate-account-", "") 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 SI
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
Did you executed Answer() before it?
From: ast
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
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 can
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: asterisk-user
. 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
-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 wrote:
Check if dtmfmode is properly set on SIP trunk
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 wrote:
Check if dtmfmode is properly set on
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 Maili
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 L
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: Re
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
To
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 some
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 ringlist
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: aster
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, 2
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 s
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, 2
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
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"
Sent: Tuesday, February 8, 2011 8:49am
To: "Asterisk User
${HANGUPCAUSE} value is available on h extension.
-Original Message-
From: "Shariq Khan"
Sent: Tuesday, February 8, 2011 8:30am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] ${HANGUPCAUSE} in CDR
Hello Gurus,
Can i add ${HANGUPCAUSE} in CD
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"
Sent: Tuesday, February 8, 2011 7:45am
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call files error
Hi All,
I'm having some
Yes. The technology need to be used on LAN switches is "port mirroring" or
"line tapping"
-Original Message-
From: "Sherwood McGowan"
Sent: Tuesday, February 8, 2011 7:34am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: Re: [asterisk-users] Call Recording aud
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
UAs.
Regards,
Faisal
P peers calling each other:
A (g722, alaw) calls B (alaw,ulaw) via asterisk.
My setup:
allow=g722,alaw
preferred_codec_only=no
Note that when B calls A, codec alaw is used on both ends, fine, but it does
not seem to work the reverse way (A is using g722, B is using alaw
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