hi Bharat,
why you are giving same answer as mine over and over ? please read
posts carefully.
On Wed, Mar 20, 2013 at 6:48 AM, Bharat Lalcheta
bharatlalch...@gmail.comwrote:
Did u changed rtp.conf ?
port is showing 39408. Asterisk definetly drop rtp packet for this port if
not updated in
Hi
Does anyone know what the different return codes from AGI script
execution mean? I'm getting a lot of
AGI Script script-name completed, returning 4
I'm using asterisk 1.8.7.0
Thanks in advance
--
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44
On Wednesday 20 March 2013, Ishfaq Malik wrote:
Hi
Does anyone know what the different return codes from AGI script
execution mean? I'm getting a lot of
AGI Script script-name completed, returning 4
I'm using asterisk 1.8.7.0
Thanks in advance
You need to check the exit statements
On Wed, 2013-03-20 at 11:07 +, A J Stiles wrote:
On Wednesday 20 March 2013, Ishfaq Malik wrote:
Hi
Does anyone know what the different return codes from AGI script
execution mean? I'm getting a lot of
AGI Script script-name completed, returning 4
I'm using asterisk 1.8.7.0
On 20/03/2013, at 10:15 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
I wrote the script! I've not put exit status values in it either.
I've been having a look the source for res_agi_c (as a non c coder) and
there is a variable called returnstatus which is instantiated with the
value
On Wed, 2013-03-20 at 22:32 +1100, Andrew Yager wrote:
On 20/03/2013, at 10:15 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
I wrote the script! I've not put exit status values in it either.
I've been having a look the source for res_agi_c (as a non c coder) and
there is a variable
Hi Ishfaq,
On 20/03/2013, at 10:46 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Wed, 2013-03-20 at 22:32 +1100, Andrew Yager wrote:
Hi Andrew
Thanks for the advice, I will look into it (I'm using php)
The script executes successfully over 99% of the time, it is run very
very
On Wed, 2013-03-20 at 22:52 +1100, Andrew Yager wrote:
Hi Ishfaq,
On 20/03/2013, at 10:46 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Wed, 2013-03-20 at 22:32 +1100, Andrew Yager wrote:
Hi Andrew
Thanks for the advice, I will look into it (I'm using php)
The script executes
Hi ishfaq,
if you want just loging some info into db you can do in dialplan without
any AGI.
i am doing billing on the fly in dialplan and mysql for every single user
without AGI and enhanced call capacity almost double.
let me know you need some examples.
On Wed, Mar 20, 2013 at 12:56 PM, Ishfaq
On Wednesday 20 March 2013, Ishfaq Malik wrote:
I have an even simpler fix for this particular script. This one isn't
really a true AGI script, all it's doing is taking the arguments
presented to it and logging them in a db table.
I'm going to try the system command instead, should have done
Hello;
I am facing a problem to let Cisco IP Phone 7942G register on Asterisk. The
firmware has been downloaded from the TFTP successfully and currently I am
running this load SIP42.9-3-1SR2-1S*
I feel that there is a problem in the SEPMAC.cnf.xml but really I do not know
which one to be used
I just wanted to send out some information that will hopefully help
others. I don't know, maybe I'm the only one that's been having
problems with this. I've been pulling my hair out for a while
wondering why Google would not send my incoming calls to my Asterisk
box. The calls would just roll
Great info, thanks for sharing! I was resigned to the fact that the same
account couldn't be logged into Gtalk simultaneously. (Never had issues
with being logged into Gmail, though.)
On Wed, Mar 20, 2013 at 6:57 AM, Chris Gentle gent...@gmail.com wrote:
I just wanted to send out some
Hello everyone,
I wonder if there's a product that I can install on my debian-based server
to extract CDRs (it'd be better if Excel's downloads are available), also
it would be desirable if I can access additional table to update rows (e.g.
sip for realtime)
Please let me know what you know.
Hello;
The phones are registering now. I found a SEPMAC.cnf.xml file and I used sip
firmware version 8.3 and I configured nat=no at sip.conf and nat to be false in
xml file.
But I am facing a time problem, I am in Kuwait country and the time that is
appearing at the Phones screen is delayed
That change did not fix the problem. Below is another trace from a
failed call this morning. 172.16.0.71 is the client, 172.16.0.245 is
the Asterisk server. All the RTP packets after the SIP are from server
to client.
Any further ideas are appreciated. (If I don't get this fixed this
On Wed, Mar 20, 2013 at 7:11 PM, Asghar Mohammad asghar...@gmail.comwrote:
hi,
problem seem to client end i am going to install SFLPhone i will let you
know when finish, have you check firewall on clients pc?
--
_
--
There is no firewall on the client.
I've compared the SIP messages between a successful call and a failed
call, and I can see no difference except for things like port numbers
and call IDs.
It only fails occasionally, not on every call.
Mitch
On 03/20/2013 01:16 PM, Asghar Mohammad wrote:
On Wed, 20 Mar 2013, bilal ghayyad wrote:
I am facing a problem to let Cisco IP Phone 7942G...
How I can access the Phone via ssh or http to be able to see the logs
and understand what is happening?
I have a 7960 and the only 'way in' is telnet. I think the default
password is 'cisco'.
client phone not sending rtp at all there is nothing to do with sip
invites. some firewall blocking rtp packets or softphone is missconfigured.
On Wed, Mar 20, 2013 at 7:25 PM, Mitch Claborn mitch...@claborn.net wrote:
There is no firewall on the client.
I've compared the SIP messages between
Mitch Claborn wrote:
Where is a good place to find documentation on the various fields in the
INVITE SIP message and the response? I'd like to dig into them and try
to understand them more completely.
For the SIP headers:
http://en.wikipedia.org/wiki/Session_Initiation_Protocol
Z
--
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To
hi,
sflphone work fine installed and tested on debian with nat and without nat.
please check setting in preferences my sflphone use alsa device. you should
check with alsamixer maybe sometime mic get muted or you agent mute the mic.
also check out what advice Mitch.
NB. you can test with IAX also.
Thank you for that most excellent post. I had guessed at most of the
SDP fields and meaning.
I have wireshark traces from the client and the RTP packets are not in
the trace, which I think means that the client software is simply not
producing them. I have opened a ticket with SFL phone
I wonder if anyone in the list find useful to write stdexten in AEL
instead of in extensions.conf syntax. I do, but probably I'm just
alone.
On Wed, Oct 31, 2012 at 11:44 AM, Octavio Ruiz tac...@tacvbo.net wrote:
Almost two years ago, a change between how AEL code is built into
Asterisk
Ii
-Original Message-
From: Octavio Ruiz tac...@tacvbo.net
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 20 Mar 2013 16:21:02
To: Asterisk Users Mailing List - Non-Commercial
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