Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
got Polycom to issue a hotfix firmware version. I'll be happy to share it
with you offlist, just email me.
Officially Polycom will fix the issue in 5.3 in a few months..
Thanks
David
On Mon, Mar 9,
On Monday 09 Mar 2015, janani m wrote:
The Error Which I face I have attached.
I need a clarification of Why I face this error and how to overcome this.
Anybody know Please help..
That's a very common error and what it means is, the AGI script
/var/lib/asterisk/agi-bin/googletts.agi
Hi Guys,
We are getting a strange issue on certain polycom phones, sometimes when a
call comes in it just flashes to show there is a call but does not play
any sound.
This problem is very intermittent and happens to maybe 2 out of 10 calls.
Has any else experienced this before?
--
I'll add that it appears to happen when you have users in a ring group or
call queue and BLF is being used in some capacity..
dw
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We
On Monday 09 Mar 2015, janani m wrote:
The Error Which I face I have attached.
Please do not attach pictures. Please cut and past text.
On Mon, 9 Mar 2015, A J Stiles wrote:
That's a very common error and what it means is, the AGI script
/var/lib/asterisk/agi-bin/googletts.agi either has
Joshua Colp wrote:
Have you configured any transports? PJSIP does not create any by
default, you have to explicitly configure them. Without them no traffic
can come in or go out. You can also remove the explicit transport from
the endpoint.
Yes I have two transports
[transport-udp]
Chirag Desai wrote:
snip
Here's my PJSIP conf:
[kamailio]
type=endpoint
transport=transport-udp
context=from_kamailio
disallow=all
allow=alaw
allow=g722
allow=ulaw
aors=kamailio
direct_media=no
rtp_symmetric=no
force_rport=no
rewrite_contact=no
[kamailio]
type=identify
endpoint=kamailio
Hi,
I want to have Kamailio in front of one or more Asterisk boxes.
I don't think it is necessary for Kamailio and Asterisk to register with
one another. I'd like for PJSIP to recognise Kamailio by its IP address.
I have two boxes, both have public IP addresses, they also have private IP
On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov d...@belkam.com wrote:
Hello!
Just installed asterisk 13.2.0 and see many such messages in log, I see them
in console during calls, really something like this:
-- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6,
SIP/6166@asterisk) in
Chirag Desai wrote:
I've tried explicitly setting the IP in bind and leaving it as above.
Nothing seems to come into asterisk. Although, as mentioned I can see the
SIP messages when I ngrep 5061.
I got it working, I can see the sip traffic in the CLI now.
I was trying to match on the IP of
The Error Which I face I have attached.
I need a clarification of Why I face this error and how to overcome this.
Anybody know Please help..
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Only slightly asterisk related I suppose, but hoping someone has
attempted this...
I have an old installation with a bunch of IP501s, and one died. I
replaced it with an IP450, and the user sorely misses his DND button. I
hated those DND buttons anyway, as I couldn't control them
I'd love to get server-side DND working with an on-screen notification as well,
for my own customers - I haven't sat down to nut this one out though.
In the meantime, they can change the phone-side DND status through the menu's.
It's the same behaviour as the handsets which have a hard DND
On Mon, Mar 9, 2015 at 9:40 AM, David Wessell da...@ringfree.biz wrote:
Welcome to our hell.
We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally
got Polycom to issue a hotfix firmware version. I'll be happy to share it
with you offlist, just email me.
Officially
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