Hi,
I just realize in these few days there are many calls that already hangup
but not detected by Asterisk.
Those calls occupy PSTN lines and need to be manually terminated through
Flash Operation Panel or phycally disconnect the PSTN lines.
This never happen before but as long as I can remember,
Hi all,
In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)
After i filled in the mac-addresses of the BT-adapter and the one from
my phone, i see it is recognized, got connected, and
Hello,
I experience the same problem, and I would really appreciate if someone
could give us a hint on that.
Hoggins!
Le 17/09/2012 19:22, Mehdi Rahimi a écrit :
Hi all,
I need to handle a problem from AGI please guide me
in extensions_custom.conf :
exten = s,1,Answer
exten =
On Tuesday 18 September 2012, Satria Anamarta wrote:
Hi,
I just realize in these few days there are many calls that already hangup
but not detected by Asterisk.
Those calls occupy PSTN lines and need to be manually terminated through
Flash Operation Panel or phycally disconnect the PSTN
Hi,
Just following this thread for few days, I've some basic troubleshooting
questions for you.
1- What do you mean by calling from landline? How is your Landline /mobile
reaching your asterisk box ? is there a Hardware card ! or a VoIP provider.
2- Enable SIP traces and keep an eye on the
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
Mehdi
On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 18 September 2012, Satria
ِDear Sammy,
Thank you for your following ,
1- Land line i mean telco company which is calling to my server , i
use FXO VOIP CARD (ATCOM 4 port) and test on a gateway too.
2-please explain me more about Enable SIP traces and keep an eye on
the originating BYE request
Regards,
Mehdi
On Tue, Sep
In article caehsoweantztyoebdobjchoeszhfk_z9sigaujsij15xx-u...@mail.gmail.com,
Mehdi Rahimi mrm.ci...@gmail.com wrote:
Hi all,
I need to handle a problem from AGI please guide me
in extensions_custom.conf :
exten = s,1,Answer
exten = s,n,AGI(hang.php)
exten = s,n,Hangup
in
Hi,
So basically the FXO cards configurations need to be tweaked i.e
hanguponpolarityinverse=yes etc.
Since this is a Hangup request initiated by the SIP client, Asterisk then
atleast it should close all the media streams and channel should get
deleted.
Keeping an eye on BYE : *CLI sip set debug
Hello
In indications.com are the tones for several countries
On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote:
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
On Tuesday 18 September 2012, Mehdi Rahimi wrote:
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
Mehdi
Unfortunately I am not familiar with the Iranian telephone system.
In article cajujwtig7yzk4+kb3c6sdu6zhb_+vwsg-oy0pibw0maeeed...@mail.gmail.com,
SamyGo govoi...@gmail.com wrote:
So basically the FXO cards configurations need to be tweaked i.e
hanguponpolarityinverse=yes etc.
Since this is a Hangup request initiated by the SIP client, Asterisk then
atleast
Hi Tony,
Thank you for your attention , and appreciate your contribution .
You are right we can not do anything till the caller hangup BUT how
can we prevent to hearing DTMF when someone else is trying on another
extension ?
to clearance :
someone calls (from landlines os mobile , no difference)
On Tuesday 18 September 2012, Mehdi Rahimi wrote:
Hi Tony,
Thank you for your attention , and appreciate your contribution .
You are right we can not do anything till the caller hangup BUT how
can we prevent to hearing DTMF when someone else is trying on another
extension ?
to clearance :
The latest version of res_speech_lumenvox.so doesn't seem to work and
nobody seems to know when a version that works will be available. It
looks to me like this is some sort of timeout issue. Does anybody
have a workaround to allow this to be used? (I know about UniMRCP,
but find it quite
You could go back to a version that it works in and apply patches to it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, September 18, 2012 9:48 AM
To:
Hi all.
I compiled the module chan_sccp, now its possible deploy trunk SCCP with
Callmanager? Anyone?
Regards--
_
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New to Asterisk? Join us for a live
Hi Hans,
On 18/09/12 08:04, Hans Witvliet wrote:
Hi all,
In one of my other project i had a look at chan_mobile.
I build 1.8.15.1 with the apropiate module. (in my distro asterisk is
build without chan_mobile ;-)
After i filled in the mac-addresses of the BT-adapter and the one from
my phone,
Hi all,
I would like to know, is there a way to trigger Asterisk after data
inserted into mysql DB? Like here what I'm trying to do, when the new data
inserted into MySQL DB, it sends the request to Asterisk along with the new
data (that is inserted in DB) for making outbound call i.e. Realtime.
On 9/18/2012 3:41 PM, Ahmed Munir wrote:
Hi all,
I would like to know, is there a way to trigger Asterisk after data
inserted into mysql DB? Like here what I'm trying to do, when the new
data inserted into MySQL DB, it sends the request to Asterisk along with
the new data (that is inserted in
On Tue, 2012-09-18 at 17:43 +0100, Sebastian Arcus wrote:
Hi Hans,
The following page has some useful info:
http://www.voip-info.org/wiki/view/chan_mobile
Sebastian
Indeed. Didn't realise it was so picky.
just bought a couple of bt-adapters.
Will try again tomorrow and feed the
- Original Message -
From: upendra uppi...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, September 18, 2012 12:30:04 AM
Subject: Re: [asterisk-users] Asterisk Test Suite error
Hi Matthew ,
i have enabled the
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