I’m looking at setting up a VoIP GSM gateway to connect to my
asterisk box. What experience have people on this list have with
GSM gateway hardware. I have been looking at the 2N voiceblue
products.
We are using the voiceblue that supports a maximum of 4 x sims (and
are using all four
In my relentless quest for knowledge, I pose this question: who's got
the biggest dialplans, and how big are these monsters?
Business System with 120 users:
-= 332 extensions (1412 priorities) in 45 contexts. =-
___
--Bandwidth and Colocation
I have had a problem with a few Snom 320's on several sites locking up
after a few days. I am running application ver 6.2.2 with the latest
jffs2 ver and tried the latest 5.x ver with similar results. Is
this also
experienced with other Snom users?
not sure if this will help you
. A little bit of work to change the
dial plans around and removes the dependancy from the VoIP phone. But
still it would be nice if it could would from the VoIP phone.
On 27/06/2006, at 11:19 PM, Steve Davies wrote:
On 6/26/06, Peter J Dean [EMAIL PROTECTED] wrote:
I have a issue trying
I have a issue trying to understand why Asterisk-PBX, when a SNOM
(320 or 360) successfully redirects/diverts a call when it is a local
extension, but fails when you enter external number.
Both the local extension dial and external extension dial are within
the same context [from-sip] and
direct issues with DTMF, I just hope that v1.4
will be released sooner rather than later.
On 09/06/2006, at 5:11 AM, Martin Joseph wrote:
On Jun 8, 2006, at 3:13 AM, Peter J Dean wrote:
I have an issue with DTMF. DTMF is being partly recognised by some
external IVR systems (banks, billing
I have an issue with DTMF. DTMF is being partly recognised by some
external IVR systems (banks, billing, etc), other IVR systems have
intermittent issues. Call our VSP directly and using their IVR system
without issue, and our internal IVR works just fine. Currently i have
all voip devices
Ummm, not sure if I am missing anything, but I have never experienced
this before, where the asterisk release didn't compile.
Downloaded all the newest releases from Digium, I compile everything
first before installing to minimise downtime of the phone system.
( cd zaptel ; make
Installing the zap and lib before compiling asterisk worked, will
change my installation processes to meet the expectations of the new
version.
Thanks for everyone that replied.
On 31/05/2006, at 12:19 PM, trixter aka Bret McDanel wrote:
On Wed, 2006-05-31 at 12:09 +1000, Peter J Dean
Ummm, when you do a show channels does it show that Asterisk still
has an open and active channel with them? If so, you could try Set
(TIMEOUT(absolute)=60) in your extension context before paging the
phones, we had an issue whereby the paging channels didn't close
after the pager (calling
You need to have an established and open channel before the audio can
be played.
exten = x,1,Answer
exten = x,n,Playback(audio,noanswer)
exten = x,n,BackGround(out)
exten = x,n,Hangup
On 24/04/2006, at 10:25 PM, FaberK wrote:
Hi Folks,
using this:
exten =
,
but if I Answer, I'll be billed, isn't it?
What I need is to play an announce of the service cost, so that if
the guest do not want to go ahead for the cost, can hungup without
pay.
I'll try your solution.
Thanks.
2006/4/25, Peter J Dean [EMAIL PROTECTED]: You need to
have an established
I had an incident, whereby the caller didn't either hang-up their SIP
phone properly or the disconnect/hang-up information didn't properly
find their way back to the Asterisk-PBX and it left the company phone
system in intercom mode with about 90 phones overnight (624mins, CPU
utilisation
Try, http://www.snom.com/wiki/index.php/Asterisk_1.2_Firmware_R4
On 17/04/2006, at 4:29 PM, Ronald Wiplinger wrote:
I tried to find this in asterisk wiki, but each link I found was
broken.
How can I use my Snom 190 or 360 softphone as Intercom ?
bye
Ronald Wiplinger
I have to this some questions:
1. I have not seen n(tryiax01) construction before. Can you
explain it, please and how you give this to the macro?
I know only exten = s,4,Goto(s-${DIALSTATUS},1)
tryiax01 is a reference label (which can be more meaningful for
other's such as
We do it slightly different, rather than multiple macros, we do it
within a single macro.
;
;
;
[macro-outbound-calling]
exten = s,1,NoOp(Debug: Outbound Call from ${CALLERID})
;
exten = s,n(tryiax01),NoOP(Debug [${CONTEXT}]: Trying 1st IAX2
Service)
exten =
The Dial line that I am using, which is called from within a Macro;
exten = s,n(dialextension),Dial(SIP/${ARG1:-3},15,tT)
Where ARG1 is a 3-digit extension number.
On 10/04/2006, at 9:24 PM, Eric ManxPower Wieling wrote:
Peter J Dean wrote:
I have an issue with trying to ensure
I am not sure how this works in other countries, but in Australia -
according to Telstra - the caller id is managed by the terminating
device which is the telephone. If the caller id has been flagged as a
silent number then generally the caller id is not display, but rather
display with an
Why not use the busy command, in combination with the groupcheck
commands - refer to http://www.voip-info.org/wiki/index.php?
page=Asterisk+cmd+Busy
On 09/04/2006, at 5:01 PM, Miles Scruggs wrote:
C F wrote:
use groups, check the commands/functions group and checkgroup.
I guess I can
I have an issue with trying to ensure that when dialling an extension
that it continues to ring up to the timeout value. But what I am
finding is that the timeout is all over the place. Sometimes half the
timeout value and other times within a few seconds of the timeout value.
I am running
We have a SNOM360 (ext 226) configured for redirection (away on annual leave) to another SNOM360 (ext 225), being tested from a SNOM320 (ext 227) which appears on the surface to be an easy adjustment.Was receiving the following message, "Got SIP response 302 "Moved Temporarily" back from"of
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