Hi Richard,
i update a new version of asterisk to 1.8.9.1 and checked the issue are
still same and my call
getting answer while it is in ringing.
here is brief details for finding root cause.
Dahdi -Version: 2.4.1.2 Echo Canceller: OSLEC[channels]
File : chan_dahdi.conf
context=from-pstn
FXO ports are considered Answered as soon as dialing completes. This is the
way analog FXO ports work. Use PRI or SIP if you need correct Answer
supervision.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Hi,
I've read here and there how Asterisk could send SMS but I didn't find
much about how to receive SMS and forward them to an email box.
1. First of all, I don't think my telco would let me receive any SMS
my landline.
2. Maybe I could find providers selling this service for a monthly fee;
I have Avaya IPOffice 403 talking to my Asterisk 1.8.x with virtually no
issues using OOH323.
I am having some minor issues with the name portion of the caller ID
sent to Avaya. That may be relalted to a way FreePBX created the dial
plan. Maybe not. Never had time to systematically look into
Le 16/02/2012 16:55, Olivier a écrit :
Hi,
I've read here and there how Asterisk could send SMS but I didn't find
much about how to receive SMS and forward them to an email box.
1. First of all, I don't think my telco would let me receive any SMS
my landline.
Why? If I assume well you're in
2012/2/16, Administrator TOOTAI ad...@tootai.net:
Le 16/02/2012 16:55, Olivier a écrit :
Hi,
I've read here and there how Asterisk could send SMS but I didn't find
much about how to receive SMS and forward them to an email box.
1. First of all, I don't think my telco would let me receive
https://reviewboard.asterisk.org/r/1599/
I so wish that this patch would be backported to the 1.8 branch! I am
considering switching to trunk just for this alone.
I know it's a stretch but, given the popularity of running Fail2Ban
alongside Asterisk, could it not fall under the pretense of
A definite me too from my side. Always wondered why it wasn't like that.
It would do wonders to help us fix our own problems instead of filling in
bugs or posting here ;-) (hint hint)
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
On 16-02-12 20:18, Luke Hamburg wrote:
https://reviewboard.asterisk.org/r/1599/
I so wish that this patch would be backported to the 1.8 branch! I am
considering switching to trunk just for this alone.
I know it's a stretch but, given the popularity of running Fail2Ban
alongside Asterisk,
- Original Message -
From: Patrick Lists asterisk-l...@puzzled.xs4all.nl
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 16, 2012 6:32:13 PM
Subject: Re: [asterisk-users] High verbose set at console effects the
Hi,
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they can give the right greeting message when answering.
E.g. I have
Fair enough.
Giving up on the backport to 1.8 or 10 for now, I had a thought for a
kludge.
How about a shell script (scheduled with cron) that checks for any 'active'
consoles -- any connected consoles where there has been user input within
the last X minutes. If it finds none, then set the
Kevin,
You might have luck changing the callerid number so its not empty, that might
override the Anonymous label.
exten = 12345678,1,GotoIf,$[${LEN(${CALLERID(num)})} != 0]?3
exten = 12345678,2,Set(CALLERID(num)=0)
exten = 12345678,3, Your code starts here
Good luck!
Mark
On Feb 16,
On Fri, Feb 17, 2012 at 12:56:15PM +1030, Kevin Shanahan wrote:
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they
Hello,
Thanks for taking out tome for my query. Yes I do have an actual problem.
I've a monitoring tool to record the VoIP QoS (Asterisk servers port
mirrored to it). My end points(soft-phones) are sending RTCP connection
strings to asterisk, and Asterisk then forwards their call to their
Hi Eric,
but in this case dialing is not completed ring is still going on, so it
should not answered
thanks
Dhaval
On Thu, Feb 16, 2012 at 7:52 PM, Eric Wieling ewiel...@nyigc.com wrote:
FXO ports are considered Answered as soon as dialing completes. This is
the way analog FXO ports work.
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