Hi All.
Iam testing asterisk-1.2.9.1 linked with one pabx Siemens HiPath 4000 in protocol QSIG.
I noticed that when the occupation is linearly orderly increasing, or either initiating for the first circuit, the call is completed successfully. But when the occupation is disordered and initiating
I just downloaded, compiled and installed Asterisk 1.2.9.1. I did this
specifically
to get the Dial M(x^y) feature so that I could implement call completion
confirmation over IAX2 channels (not available in 1.0.7). The problem is that
the
call is always completed--even without the required
Does anyone know how
to successfully register a Motorola VT1005? What firmware should be
used?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Does someone know where I can get the last sip version? My Polycom
reseller doesn't have it and I need to enable the buddy for 14 contacts.
Thanks in advance.
Lucas Alvarez
Douglas Garstang wrote:
I've never seen that problem, and I've only ever used 1.2+ with
Polycom and buddies.
Ohoh... Kevin, what version of SIP software are you running?
One of my Polycom phones just rebooted itself for no apparent reason.
-Original Message-
From: Kevin Smith [mailto:[EMAIL PROTECTED]
Sent: Friday, June 30, 2006 1:07 PM
To: Asterisk Users Mailing List - Non-Commercial
In that case, it is likely your reseller is not a Polycom certified VoIP
reseller. Contact me off-list and I'll help you.
Nabeel Jafferali
www.voipdepot.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Lucas Alvarez
Sent: June 30, 2006 6:08 PM
I have quite the same strange behaviour in Dial(...M(x^y) )
I use it to play various annoucements and operates differents operations
to agents that answers and when the dial timeout reachs to its end the
macro is hangup although the DIALSTATUS is set to ANSWER
I'll post some logs and
yeah this post is old and there have been dozens of replies, but here's
some feedback for the list, now that i have some.
we're using a sangoma a102 card (no hw ec) with 2 pris from sbc.
asterisk 1.2.7.1, zaptel 1.2.6 (much testing previously with 1.2.5). we
first used:
KB1 (not
M.Hockings wrote:
One weakness is the incoming PSTN line, what is the best way to
protect that beyond the device at the premises entry ?
The device at the entry, assuming it was even installed correctly, is
there to protect the PSTN Central Office, NOT your equipment.
There are many sites
No log entries yet that might show whats happening and you are correct, I
cant run under strace as it would hit performance quite bad.
:( I will continue to look into the logs and hope something will show up so
I can post further.. If anybody else experiencing this can come up with some
log
I want to set some variables for each phone. For that I use setvar in
Real-time. At the beginning of each context should be this include
statement before all other include statements.
How can I rewrite the dial plan, so that after the include var-key other
include statements are still used?
anyone have information on how the call back features work with asterisk? I means the dial plan or what so ever. thanks
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
Hi,IhaveAudioCodesMP-124device.Itworksfine,butIwanttofixoneproblem,butIdon'tknowhow!InAudioCodesI'vecreatedtwolinesfortests:52040668 and 52040669.Here is my sip.conf fragment sniffedtraffic:
sip.conf[52040668]type=friendaccountcode=1videosupport=yescontext=bandymassecret=12345host=dynamic
I upgraded from 1.6.2 to 1.6.6. After which, the problems started to
happen. While it isn't a good thing, at least I'm not crazy and someone
else is having the problem as well. ;).
I also turned on the logger on asterisk with full debug information.
Sure it's crazy, but maybe if one of the
101 - 114 of 114 matches
Mail list logo
