Hi All,
I want packets [request/response] capture for ISUP packets , i have E1 line
terminated to my digium card
i just want a packets flow between my machine and teleco side, is any tool
or utility [command] availabele for
observation this packets and data.
any help appericiated
Thanks
Dhaval
Hallo Barry,
extensions_additional.conf is supposed to be edited by FreePBX.
Gopal, on using extensions_custom, the SIP phones work however the details
are not captured in the reporting mechanism of FreePBX, which is what I need
most.
Richard Zulu
Twitter
www.twitter.com/richardzulu
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the
we too had enough issues with ribbit support and moved to tringme for
web based phone.
On Wed, Aug 10, 2011 at 3:29 PM, Dean Collins d...@cognation.net wrote:
Any thoughts on why they did this?
-
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:
On Thu, 2011-08-11 at 08:50 +0300, Richard Zulu wrote:
Hallo,
I have a production asterisk server running on Ubuntu however all my
configs where done using the CLI.
I would like to implement a reporting element into the server so I can
know the number of calls made, for what duration,
We have a client that has sporadic problems with the *8 pickup facility.
The server they are using is 1.8.5 and they are using Snom phones.
Every now and then when they try to do a pickup from another phone they
get a forbidden message on the phone and I can see the following in the
logs.
[Aug
Hi all,
Continuing a previous post regarding background music during a call,
there is a strange miserable problem which I am unable to understand.
It works fine on some systems as I have tested it with sip and with
dahdi + pri signalling with digium hardware on one of my production
server, but
Anyone?
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We have difficulty setting up the incoming termination for our
clients. Both
for start you could disable guest access in sip.conf, I guess you do not
need it
On 2011.08.11 14:29, Jim Boykin wrote:
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM,
Ah, now this is interesting as one of our clients had the same problem the
other day; in our case when they performed the *8 they got an extension
unavailable from a completely different dialplan! This was on Asterisk 1.6
though with Snom phones.
--
Thanks, Phil
- Original Message -
On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote:
Ah, now this is interesting as one of our clients had the same problem the
other day; in our case when they performed the *8 they got an extension
unavailable from a completely different dialplan! This was on Asterisk 1.6
though with
Hi,
In my office I have 1000 ext, each users has it's own DID number.
What I would like is that each user can get a fax using his own number.
I'm fighting with this since some time too.
My experience is, that faxing over IAXModem is not reliable.
So I'm going with T38Modem, my provider also
Thorolf Godawa wrote:
My experience is, that faxing over IAXModem is not reliable.
My experience is just the opposite, but I'm not faxing over IP. PRI,
over slinear.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
in my case i use snom 320 and 370
i flow this link and i can do the pichup with any issue
http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Pickup
Regards.
2011/8/11 Ishfaq Malik i...@pack-net.co.uk
On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote:
Ah, now this is interesting as
2011/8/11 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk
On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote:
Ah, now this is interesting as one of our clients had the same
problem the other day; in our case when they performed the *8 they
got an extension
On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote:
Hi All,
I want packets [request/response] capture for ISUP packets , i have E1 line
terminated to my digium card
i just want a packets flow between my machine and teleco side, is any tool
or utility [command] availabele for
On Thu, Aug 11, 2011 at 12:43:43PM -0500, Russ Meyerriecks wrote:
On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote:
Hi All,
I want packets [request/response] capture for ISUP packets , i have E1 line
terminated to my digium card
i just want a packets flow between my
Paul Hayes p...@provu.co.uk writes:
If you time the *8 just right so it is being handled during the end of
the Dial then I got:
[Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data
is NULL
[Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data
is NULL
Does
Hello list,
I presently use the 1.4 releases because I enjoy sleeping
at night. I understand that 1.4 reaches end-of-life in a little over 8
months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I
also know (as best as I can) that no genie is going to make
On Thu, Aug 11, 2011 at 3:04 PM, Danny Nicholas da...@debsinc.com wrote:
Hello list,
I presently use the 1.4 releases because I enjoy sleeping
at night. I understand that 1.4 reaches end-of-life in a little over 8
months
Does anybody ever update the software on the Panasonic phone system they
had installed 30 years ago? Maybe if it ain't broke don't fix it.
Hello list,
I presently use the 1.4 releases because I enjoy
sleeping at night. I understand that 1.4 reaches end-of-life in a
On 11-08-08 09:50 AM, Miguel Molina wrote:
El 08/08/11 11:46, J Gao escribió:
On 11-08-06 10:06 AM, Miguel Molina wrote:
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0
and under 1.6.2.16.1 and I get the same error:
[Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok
[Aug 11 06:50:20] VERBOSE[3023] tcptls.c: == Problem setting up ssl
connection:
Dear all
may be it isn't related .
but I want to shared my VOIP experiences in my new weblog.
http://blog.lopl.net
Help me to improve it by your comments and ideas.
Best
--
Pezhman Lali
--
_
-- Bandwidth and Colocation Provided
Hi Paul,
Maybe you can give some help here:
I'm trying to compile and build the debian source file
of asterisk_1.8.5.0.orig.tar.gz
and asterisk_1.8.5.0-1digium1~natty.debian.tar.gz.
Howerver every time I'm trying to compile it, using ./configure of
dpkg-buildpackage -rfakeroot -us -uc I get
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