Fair enough.
Giving up on the backport to 1.8 or 10 for now, I had a thought for a
kludge.
How about a shell script (scheduled with cron) that checks for any 'active'
consoles -- any connected consoles where there has been user input within
the last X minutes. If it finds none, then set the verbo
https://reviewboard.asterisk.org/r/1599/
I so wish that this patch would be backported to the 1.8 branch! I am
considering switching to trunk just for this alone.
I know it's a stretch but, given the popularity of running Fail2Ban
alongside Asterisk, could it not fall under the pretense of 'secur
Try TAPIRex
http://www.tapirex.com/en/
It's not free, but I've been using it with Asterisk + Outlook 2010
successfully. Users can also click on the screenpop and it will open up the
contact in Outlook. Pretty handy. You will need to make dialplan
modifications to send out the call info to the u
e: [asterisk-users] Blind transfers being cancelled by asterisk &
hanging up on remote caller
On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg wrote:
Doug:
for what it's worth I am having the exact same nightmare. Not sure exactly
when it started but I believe it was a change in 1.8.8.1
Carlos-
Sorry if this is too much of a digression but this piqued my interest as
I've been pretty happy with Polycom in my limited experience (haven't used
the SPAs much, just Yealink & Polycom, and an occasional Snom here and
there). If the config files were not the issue for you, then what _wer
Doug:
for what it's worth I am having the exact same nightmare. Not sure exactly
when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am
running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers
are broken. All legs of the call are dropped when the xfer is
[asterisk-users] How to use menuselect.makeopts?
On 10/19/2011 03:08 PM, Jason Parker wrote:
> On 10/18/2011 09:52 PM, Luke Hamburg wrote:
>> I think this might actually be a bug.
>> https://issues.asterisk.org/jira/browse/ASTERISK-18137
> It is indeed a bug, but it's
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Lists
Sent: Tuesday, October 18, 2011 9:23 PM
To: asterisk-us
Have you tried adding 'qualify=no' in the peer definition?
Are all the natted phones using port 5060 as their SIP port? I don't know
how many "a bunch" is but if it's not too many you might try having each
phone bind to a different port for its SIP signaling, sometimes that is
helpful with stric
Interesting. I just signed up with Gafachi (haven't even tested the service
yet) but I planned to make use of their T38 support since they are listed at
voip-info as being one of the ITSP's that _do_ support T38. Have you tried
contacting Gafachi with these results about their broken implementati
Right -- I've been wrestling with this problem for over a year now.
Tinkering with modules.conf load order, "noload= / preload=" etc I even
modified the internal "priority" of the timers in the C code of the timing
modules so that after a reload the pthread timer would still be at a higher
prio tha
Danny Nicholas wrote:
>> 2. Don't know if moving to 10.x would help you, but since that is still
considered beta, that's probably not an option anyhow.
Yup, not really an option for me. I actually use this system daily and
don't want to muck around with 10.0 just yet.
>> 3. My understanding is t
Hi all-
This is my first post to this list so please don't flog me if this is not
the appropriate place to post this. I've had an issue for over a year
affecting MOH + DAHDI timers, I reported it at:
https://issues.asterisk.org/jira/browse/ASTERISK-17474
Basically the MOH channel goes dead a
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