Yes, I am pretty sure that if a Polycom unit is set DND and you initiate a
multicast page from another Polycom handset on a page or PTT channel that the
DND handset is subscribed to (like the emergency channel), then you will hear
audio on that handset.
BUT Polycom handsets cannot be configured
Tim,
I THINK but I'm not sure that you can do this with the Polycom multicast
page function. Have you attempted this yet?
Thanks
david
On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson wrote:
> Greetings-
>
> As many of your are Polycom "experienced", I was hoping some kind soul
> could provide dir
On Wednesday, September 17, 2014 04:35:14 PM Russ Meyerriecks wrote:
> Patch for this has been committed to master here:
> http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=b9a8000bbd1
> b6120f22627c105a2c2194dcc793d
>
> I expect to release a v2.10.1 for this soon.
> Thanks for the
Patch for this has been committed to master here:
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=b9a8000bbd1b6120f22627c105a2c2194dcc793d
I expect to release a v2.10.1 for this soon.
Thanks for the report.
--
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis
On Wednesday 17 Sep 2014, Anurag Rana wrote:
> Thanks, That worked. :)
>
> Anurag Rana
> http://newbie42.blogspot.in/
Good; it's always nice to hear that someone has got something working!
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address
Thanks, That worked. :)
Anurag Rana
http://newbie42.blogspot.in/
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On Wednesday 17 Sep 2014, Anurag Rana wrote:
> Oh, Sorry My mistake, I misspelled it in mail.
> It is already ${DIALEDPEERNUMBER}, still returning null.
>
> Anurag Rana
> http://newbie42.blogspot.in/
Hmm. I've looked a bit further. According to the documentation,
${DIALEDPEERNUMBER} is set by
Oh, Sorry My mistake, I misspelled it in mail.
It is already ${DIALEDPEERNUMBER}, still returning null.
Anurag Rana
http://newbie42.blogspot.in/
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On Wednesday 17 Sep 2014, Anurag Rana wrote:
> in dialplan:
> exten=>h,n,NoOp(${DIALLEDPEERNUMBER)
>
> variable ${DIALLEDPEERNUMBER} is returning null.
>
> Suggestions please?
>
> Thanks
>
> Anurag Rana
> http://newbie42.blogspot.in/
Asterisk has it mis-spelled as "DIALEDPEERNUMBER" (sic).
Try
Call file syntax:
Channel: SIP/
MaxRetries: 2
Context: demo1
Extension: s
Priority: 1
WaitTime: 30
RetryTime:
60
in dialplan:
exten=>h,n,NoOp(${DIALLEDPEERNUMBER)
variable ${DIALLEDPEERNUMBER} is returning null.
Suggestions please?
Thanks
Anurag Rana
http://newbie42.blogspot.in/
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Hi,
I am initiating a call using call files. In 'h' extension I am trying to
collect the value of ANSWEREDTIME variable but it is returning null.
While It works fine when call is not generated using call files instead is
generated from softphone.
any idea what might be wrong?
thanks
Anurag Ran
On Wednesday 17 Sep 2014, Rizwan H Qureshi wrote:
> Hi All,
> I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying
> to use for kind of a call intercept between two GSM users. Call comes
> through one SIM and goes out through another Sim with our Asterisk in
> between to log
Hi All,
I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying
to use for kind of a call intercept between two GSM users. Call comes
through one SIM and goes out through another Sim with our Asterisk in
between to log the call. This works fine but we need the original callerid
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