Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk
crashed while executing meetme kick all CLI command from manager interface.
The link says the issue has been closed however I am not able to identify in
which release of asterisk this issue has been fixed.
HI
Have you tried:
sendrpid = pai ; Use the P-Asserted-Identity header
; to send the identity of the remote party
in the sip.conf?
Regards
Ish
On 16 February 2014 20:29, Nick Cameo sym...@gmail.com wrote:
Hello Markus,
Thank you so much
On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:
Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk
crashed while executing “meetme kick all” CLI command from manager
interface. The link says the issue has been closed however I am not able
to identify in which
On Friday 14 Feb 2014, Carlos Chavez wrote:
[stuiff omitted]
Does anyone know of a dialer for Asterisk that can
take several phone numbers for the same contact and if any of those
answers it will not try the other numbers?
You can do that in your dialplan, without any additional software!
Hello Ishfaq,
I just tried it and it did create a P-Asserted header however it
contains the extension
of the asterisk peer not what was passed by our switch. From the
previous example:
P-Asserted-Identity: 222 sip:222@192.168.2.10 (which is asterisk
peer extension and not)
P-Asserted-Identity:
On 15/02/14 20:05, Jerry Geis wrote:
I have a confbridge in asterisk 11.
I am using an AGI to bring people in the conf automatically.
I want to speak a pre-recorded wave file message into the conf to
all users.
how might I do that?
Thanks,
Jerry
You could initiate a call which would
On Mon, Feb 17, 2014 at 4:29 AM, Nick Cameo sym...@gmail.com wrote:
Hello Ishfaq,
I just tried it and it did create a P-Asserted header however it
contains the extension
of the asterisk peer not what was passed by our switch. From the
previous example:
P-Asserted-Identity: 222
Hi,
Anyone know if it is possible with Asterisk 11.7 using a Tigase 5.1.5 server
to stop receiving notice about old and new messages waiting? Looking at
xmpp.conf there does not seems to be a setting to disable voicemail
notification. Only the Device State is useful for us for now and
Asterisk is a B2BUA -- think of it as two calls, one inbound call from your
switch to Asterisk and one for outbound call from Asterisk to the destination.
Using SIPAddHeader or similar is the proper way to copy headers from the
inbound call to the outbound call in Asterisk.
-Original
Hey Guys, I really appreciate this and I apologize for asking however,
we do not have any way to test in advance outside of our live
environment. Can someone kindly provide a working extension rule that
will retain the following P-Asserted info that is existent from the
inbound-leg to the
How about: SipAddHeader(${SIP_HEADER(P-Asserted-Identity)})
Might have some issues with the ; character being see as start of comment.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo
Sent: Monday,
On Sun, 16 Feb 2014 21:21:30 -0500
Nick Cameo sym...@gmail.com wrote:
Our environment is a register free setup, and our phones are set as
host=dynamic.
The problem we are experiencing is for inbound calls:
Name/username HostDyn Forcerport ACL Port
Status
Shiza Sounds about right but is it true? Anyone else?
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Hi all,
I'm trying to build a fax relay mechanism where faxes come in and get
relayed out to their final destination. I'm using the h extension to store
various results from both legs. This data is being saved correctly for the
first (receiving) leg. The second leg isn't calling the h extension
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