2015-03-27 10:52 GMT-06:00 Carlos Rojas crt.ro...@gmail.com:
I Ricky
I have worked with this gateway few years ago, it's good product, they have
gateways with PRI connectors and SIP.
The quality is good, and it woks good with asterisk or regular PBXs.
Hi carlos , thank for your advice, I
,
__TIMESTR=20150327-183519) in new stack
-- Executing [s@sub-record-check:20] Set(SIP/300-0192,
__FROMEXTEN=300) in new stack
-- Executing [s@sub-record-check:21] Set(SIP/300-0192,
__CALLFILENAME=out-0176XX-300-20150327-183519-1427481319.470) in new
stack
-- Executing [s@sub
I Ricky
I have worked with this gateway few years ago, it's good product, they have
gateways with PRI connectors and SIP.
The quality is good, and it woks good with asterisk or regular PBXs.
On Thu, Mar 26, 2015 at 11:16 PM, ricky gutierrez xserverli...@gmail.com
wrote:
Hi, I know there are
please no body has som with aastra can help me in this issue
2015-03-26 11:02 GMT+00:00 Salaheddine Elharit salah.elharit...@gmail.com:
hello list
i need your help please regarding an issue with snom300 and aastra6731i
using asterisk
11.13.0 asterisk
snom 300 8.7.3.25
astra 6731i
You would need to give more information really.
Your sip.conf file listing the entries for the phones especially which
codecs are permitted.
A copy of the 'asterisk -rvvv' console output when you make the call.
On 27/03/15 17:05, Salaheddine Elharit wrote:
please no body has som with aastra
All;
I have a problem that Ive been working on for a while now, but Im
stuck and cant see what the solution is. I have an Asterisk 1.11 server on
a public IP address and have two phones registered from behind a NAT. I can
send a page to/from each phone without a problem. My problem is that
On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
You have to consider whether you really want anonymous calls, or you
just want to enable SIP calls from trusted companies/partners. The
latter means setting up routes to these companies and (ideally)
registration between peers.
This is
That is your issue.
You can enable a 'sip debug' and make the call again and get a trace of
the SIP message asterisk is sending to the phone.
We can take a look here to see if anything looks wrong.
If you could post a trace from a phone that can call that destination it
might be easier to
Hi,
What is your experience: if you plan to make a test SIP call to check voice
quality overa connection what would be the best call duration? The point is
that we should have a call long enough to be able to catch/hear impairments
that the connection may have.
This is partly a matter of
James,
I'm a systems and telecom professional with experience going back more than
thirty years, to the days of teletype, current loop, POTS (2600hz signalling
anyone?) and echo
cancellation via analog level control and hybrid balance.
Your read of the intent of the VOIP/SIP design correctly.
On 27/3/15 8:03 pm, James B. Byrne wrote:
One only accepts VOIP calls from known correspondents. I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.
Because on the whole most people don't *want* to receive calls from
random
To answer your first question, what you refer to as the PSTN is also quite
dangerous. There is a lot of fraud going on over analog lines - usually
hackers try to find an outside line by calling in to a PBX and trying lots of
digits. or, in some cases fooling a naive user to forward them to an
Hi James,
Fortunately, your theory about common run for dollars is false with many
contra-examples. :)
jh
-- Původní zpráva --
Od: Bruce Ferrell bferr...@baywinds.org
Komu: asterisk-users@lists.digium.com
Datum: 28. 3. 2015 0:17:54
Předmět: Re: [asterisk-users] Anonymous
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