Hi
Its surprising that no one is responded. Does this mean, that nobody
has ever used FastAGI and AsyncAGI?
Does that also mean, that FastAGI AsyncAGI should not be used?
I am using Asterisk 1.8.xx
Thanks Regards,
Amit Patkar
On 10/8/2012 11:26 AM, Amit Patkar | ATPL wrote:
Hi
Hi Matthew,
you are right...it seems that extensions.conf behaviour has been changed
from asterisk 1.4.
Thank you.
Giorgio Incantalupo
On 10/03/2012 05:40 PM, Matthew Jordan wrote:
- Original Message -
From: gincantalupogincantal...@fgasoftware.com
To: Asterisk Users Mailing
Hi dear asterisk gurus,
I'm trying to use Lua dialplans ; I'd like to include from extensions.lua some
regular (non-lua) dialplans, and it seems to fail (silently).
I tested it including, in my lua dialplan
1- a context from the same lua dialplan OK
2- a context from extensions.conf dialplan
Sip trunk?
2012/10/10 Deepesh D deep.d2...@gmail.com
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which registers as a
Tomorrow evening I'll be at a customer site installing 2 Digum cards - a
4 port analog and 2 port T1. I'd appreciate any tips, resources and
links that you have that might help if we run into trouble. It will, of
course, be fairly late at night and relatively high pressure to get it
working,
On 10/10/2012 11:34 AM, Mitch Claborn wrote:
Tomorrow evening I'll be at a customer site installing 2 Digum cards -
a 4 port analog and 2 port T1. I'd appreciate any tips, resources and
links that you have that might help if we run into trouble. It will,
of course, be fairly late at night
I've been running an Asterisk server (1.6.2.17.2) for over a year
without any major issues. All of a sudden people are unable to login
to their voicemail because Asterisk is seeing DTMF twice for each
digit the caller pushes. We've noticed the problem only consistently
happens to callers from
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have two asterisk servers S1 and S2.
There is a third asterisk server C1 which
Hi,
Not exactly a solution, but I'm sure you must've taken pcap traces of a few
such sample calls. See in their RTPs that you are receiving repeatedly same
RTPs which will tell you that any DTMF packet is coming in twice by the
source or not !
just one such simple pcap will help you identify at
I am a complete novice at T1's, etc. What else besides framing and
coding do I need to ask about?
Mitch
On 10/10/2012 10:41 AM, Jose P. Espinal wrote:
From my own experience, get sure that the Telco actually gives you the
*correct* information about the T1 (framing, coding, etc.).
I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?
On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote:
Hi,
Not exactly a solution, but I'm sure you must've taken pcap traces of a few
such sample
On Wed, 10 Oct 2012, Mitch Claborn wrote:
I am a complete novice at T1's, etc. What else besides framing and
coding do I need to ask about?
Lots.
How are the Ts supposed to be configured?
Will you have access to a 'turnup tech' who will sit on the phone with you
and can make changes?
On Wed, 10 Oct 2012, Mitch Claborn wrote:
I am a complete novice at T1's, etc. What else besides framing and coding
do I need to ask about?
I have a strong preference for ISDN PRI Ts.
--
Thanks in advance,
-
Steve
On Wednesday 10 October 2012, Mitch Claborn wrote:
I am a complete novice at T1's, etc. What else besides framing and
coding do I need to ask about?
Ask how numbers come through on incoming calls: in international format
(with IDD and STD codes, but without the leading double-zero and zero
On Wed, 10 Oct 2012, A J Stiles wrote:
Usually, channels 1-15 and 17-31 are B-channels and 16 is the D-channel;
but again, check this with the telco. If your box can't find a
D-channel, it won't work at all.
Or, on this side of the pond, 1-23 B, 24 D is common.
--
Thanks in advance,
On Wed, 10 Oct 2012, Mitch Claborn wrote:
I am a complete novice at T1's...
On Wed, 10 Oct 2012, Mitch Claborn wrote:
It will, of course, be fairly late at night and relatively high pressure
to get it working...
Good luck. Bring caffeinated beverages :)
--
Thanks in advance,
Usually, channels 1-15 and 17-31 are B-channels and 16 is the D-channel;We don't use E1s here in the USA.I just finished installing a PRI line, and being a complete novice at it myself, this is what I wish someone had told me:- the dahdi program dahdi_genconf creates 2 files 1)
On Wed, 10 Oct 2012, C. Savinovich wrote:
- Not written anywhere, but the way chan_dahdi.conf is read, there are
no [] separators, so any parameters in the file apply to the first
channel statement it finds.
(I'm sure its written somewhere, I just can't be bothered to look right
now.)
Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN
number.
We can get ring and a connected call but no audio
SIP = ASTERISK = MOTIF
Is there any specific configurations for getting audio to work?
--
_
--
Robert wrote:
Hola,
Please in the future don't cross post as you have done to both the
developer list and users list. If it's not related to development of
Asterisk the users list is where it should stay.
Just installed 11 and trying to get MOTIF / XMPP working to E164/PSTN
number.
We can
(I'm sure its written somewhere, I just can't be bothered to look right now.) If you come across it, please let me know because other than getting a hint somewhere after 2 hours of googling, I would not have known.And aren't the characteristics 'cumulative' ? Can't tell you. You seem to hint that
Once an option is set in the chan_dahdi.conf file it applies to every channel
= line listed after the setting, until the option is changed. This is all you
really know.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
After comparing packet captures of good and bad calls. It looks like
the double digit is coming from rfc2833 and dtmf inband. It looks
like the inband tone is splitting the rfc2833 in two? Is there some
way to resolve this???
On Wed, Oct 10, 2012 at 12:28 PM, Vik Killa vipki...@gmail.com wrote:
My apologies I will clarify the situation.
We set up Motif per Digium's new WIKI on Google Voice for Asterisk 11.
It completed dialing / ring and answer BUT NO AUDIO.. No errors on the
console.
We upgrade to SVN pull of Asterisk 11 and now Motif gives new errors (ICE).
I gave up as there is
Robert wrote:
My apologiesŠ I will clarify the situation.
We set up Motif per Digium's new WIKI on Google Voice for Asterisk 11.
It completed dialing / ring and answer BUT NO AUDIO.. No errors on the
console.
I've experienced this once or twice and narrowed it down to the Google
Voice
There is actually only a single T1. When we ordered the card, customer
thought there were two, but found out later there is only 1.
Mitch
On 10/10/2012 11:50 AM, Steve Edwards wrote:
What is the relationship between the 2 Ts? NFAS? I've pissed away many
an hour trying to (remotely)
Hi,
Perhaps can someone tell me if i had the wrong expectancies
If one sip-clinet only supports GSM-codec, and another only supports
g711-U, they still can call each other and asterisk does the transcoding
Correct?
If i try to do the same with an AV-call, (one only h264, the other only
Hans Witvliet wrote:
Hi,
Hola,
Perhaps can someone tell me if i had the wrong expectancies
If one sip-clinet only supports GSM-codec, and another only supports
g711-U, they still can call each other and asterisk does the transcoding
Correct?
If codec_ulaw and codec_gsm are both
Is this happening for all callers, or just iPhone callers?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vik Killa
Sent: Wednesday, October 10, 2012 11:29 AM
To: Asterisk Users Mailing List -
Hi,
Are there any thoughts about how cpu-expensive motif is?
Does it only translate SIP -- jingle (during call-setup)
if so, impact will probably neglectible.
or does asterisk remains constantly in between the data-stream?
In that case, it might be something to pay serious attention to, when
Hans Witvliet wrote:
Hi,
Hola,
Are there any thoughts about how cpu-expensive motif is?
No more expensive than SIP I'd expect.
Does it only translate SIP-- jingle (during call-setup)
if so, impact will probably neglectible.
It's not a straight SIP--Jingle signaling translation layer.
I've tested asterisk 1.8.17.0 and I'm still getting the repeated error message
on the command line:
iax2-provision.c:266 iax_provision_version: ast_db_get failed to retrieve
iax/provisioning/cach
--
Joseph
--
_
-- Bandwidth
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