-administered DDI ranges.
Peter
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, maybe a combination,
to restart the dialplan with your variable set? (Might need a _ or two
on the variable name to get it to survive)
Peter
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Peter Bowyer
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asterisk
Caller*ID data on the channel.
[Description]
Gets or sets Caller*ID data on the channel. The allowable datatypes
are all, name, num, ANI, DNID, RDNIS.
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Peter Bowyer
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it says loop detected. can some one tell me what is wrong.
Your dialplan.
(Since you didn't get around to posting any configuration or log
information, that's about as close as anyone's going to get to your
problem).
Peter
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Peter Bowyer
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Not very good at answering followups to your ads, are you, Sam?
On 01/12/06, Peter Bowyer [EMAIL PROTECTED] wrote:
On 30/11/06, Sam Tam [EMAIL PROTECTED] wrote:
We do have @cough VoIP GSM Gateway for sell as well @ cough
Try to search on ebay for gsm voip gateway and you will see some
Have you looked at his website, www.netenable.co.uk ? Looks like he pays bills
the same way as he answers followups ;-)
g
-Original Message-
From: [EMAIL PROTECTED] on behalf of Peter Bowyer
Sent: Sun 03-Dec-06 8:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
On Monday, November 27, 2006 10:23 AM Louis-David Mitterrand wrote:
Hello,
With the following setup:
- asterisk 1.2.13,
- zaptel 1.2.10
- bristuff 0.3.0-PRE-1v
- quadbri card,
Have you tried using bristuff 1v with the qozap driver of 1s? All qozap
versions after 1s had serious
.
If you must plug it here, please be honest about what it is and what it's not.
Peter
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=.asf
But how to get mplayer and asterisk to work together?
My setup is trixbox 1.2.3
Hoping someone know how to put asterisk and mplayer to work.
Regards
Peter Vedstesen
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On 01/12/06, yusuf [EMAIL PROTECTED] wrote:
Hi,
I realise this might be an insane noob question, but I'm on a huge brain
freeze, and I'm trying to
decide this:
Is Asterisk a SIP Gateway or SIP proxy?
http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy
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Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a
semicolon as separator.
I was thinking the 'CUT' function would be perfect for this.
But the
_The functions:_
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Have you tried to put a '\' in front of the ';': Set(sep='\;')?
Peter
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Hi,
we just upgraded from 1.2.10 to 1.2.13 and now encounter strange problems with
our snom phones (FW 6.2.3 to 6.5.1). Upon phone boot everything works fine.
Phone registers and asterisk is happy. Soon afterwards the registration is lost
however. Sometimes after a few minutes the phone
On Wed, 2006-11-15 at 01:47 -0800, Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Howard
Sent: 14 November 2006 20:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: Problem found Re
On Tue, 2006-11-14 at 02:10 -0800, Steve Langstaff wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Peter Howard
Sent: 14 November 2006 00:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Problem found Re
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
any logs/errors when you do a verbose 6 and a sip debug ?
I've got a log from a call under asterisk 1.4.0-beta3 attached. The
behaviour was the same; the call connected and audio worked, but no
video.
On 11/13/06, Peter Howard [EMAIL
On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
any logs/errors when you do a verbose 6 and a sip debug ?
I've got a log from a call under asterisk 1.4.0-beta3 attached. The
behaviour was the same; the call connected and audio
-time_Transport_Protocol might help.
Peter
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On Tue, 2006-11-14 at 09:41 +1100, Peter Howard wrote:
On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
any logs/errors when you do a verbose 6 and a sip debug ?
I've got a log from a call under asterisk 1.4.0-beta3
On Tue, 2006-11-14 at 09:28 +1100, Peter Howard wrote:
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote:
any logs/errors when you do a verbose 6 and a sip debug ?
I've got a log from a call under asterisk 1.4.0-beta3 attached. The
behaviour was the same; the call connected and audio
the allow line has spent a lot of time with
restricted codecs to see if that makes a difference.
I can provide the full sip.conf, extensions.conf, and debug output if
anyone wants to see them.
Any suggestions as to where things are falling down?
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URSYS
13 Burwood Rd,
Burwood
Oops,
Asterisk version is 1.2.12 (on Ubuntu)
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine
On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote:
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote:
Greetings all,
I'm playing with asterisk and two Polycom VSX300 videoconferencing
units. And I'm having zero luck getting video working over SIP.
The two units register fine
in
offer: data 49218 RTP/AVP 100
The rest of the output seems to be normal. I can regenerate it, but
right now I've put 1.4-beta3 on to see if that improves things (so far
it hasn't, but I've tried one run)
On 11/13/06, Peter Howard [EMAIL PROTECTED] wrote:
On Mon, 2006-11-13 at 00:57
It''s a USB Sound card / keypad / display, not a phone. It contols a
softphone on the PC it's plugged into - they say it works with XLite -
the SIP setup will be done in Xlite, not the 'phone'.
Peter
On 05/11/06, Thufir [EMAIL PROTECTED] wrote:
I'm looking at the http://support.a-link.com
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
to Asterisk.
Thanks
Peter
This e-mail and any attachment is for authorised use by the intended
recipient(s) only. It may contain proprietary material, confidential
information and/or be subject to legal privilege. It should not be copied,
disclosed to, retained or used by, any other party. If you
of staying
with a supportable version of software, especially if it's open
source. If there's a security-related bug found in your version, will
it get patched, or will you have a forced upgrade several versions
ahead on your hands in a hurry?
Peter
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Email: [EMAIL PROTECTED
Yusuf,
I am using this card and it works very well for me. To use it you need
to download a driver addition and recompile Zaptel. Not a big problem
really. In all other aspects it works like the Digium card.
Peter
Yusuf wrote:
Hi,
http://www.openvox.com.cn/products_detail.php?genre_id
Sure thing, count me in
Paul Hales wrote:
We are currently writing a reception console for Asterisk - if anyone is
interested in beta testing it, feel free to ask.
Paul Hales
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If your phones reinvite then it doesn't matter if Asterisk
supports G722, only that both endpoints support the codec.
Peter
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jessee J
HolmesSent: Saturday, 14 October 2006 4:09 AMTo: Asterisk
Users Mailing List - Non
If your phones are using DHCP rather than static IPs it might be worth
checking the DHCP server logs. If the phone cannot renew its lease or if it
receives a new IP address, it will reboot.
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Garey
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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. =-
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perhaps the rest of your dialplan is expecting the call to come in
with a destination which matches your DID - in which case, put the DID
number there instead of the 101.
Peter
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On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.
And that new-fangled electricity will never catch on - lets stick with
gas-lamps...
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Peter Bowyer
Email: [EMAIL PROTECTED
On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
On Mon, 2006-10-09 at 08:59 +0100, Peter Bowyer wrote:
On 09/10/06, Joseph [EMAIL PROTECTED] wrote:
I just upgraded to Asterisk 1.2.12 from 1.0.1 and it seems to me
Asterisk 1.2 is not ready for PRIME TIME.
And that new-fangled electricity
On 09/10/06, Daniel Cyt [EMAIL PROTECTED] wrote:
Hi Peter,
Thank you for your answer.
I did:
register = DID:[EMAIL PROTECTED]/DID
exten = DID,1,...
Now when I call the DID number It doesnt reach the Asterisk.
sip show registry shows me the line is registered but when I dial out from
my
Brian,
Take a look at www.intertex.se I believe they have what you are looking for.
Peter
Brian Candler wrote:
I'm looking for a way to set up a VOIP network in branch offices where one
or more phones have lifeline capability, i.e. can place calls if the IP
network or VOIP service dies
On 06/10/06, ram [EMAIL PROTECTED] wrote:
Hi
can some one clarify
does the aterisks act like a SER
http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+not-proxy
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see where the problem is.
I tested Asterisk 1.2.5 and current SVN 1.2.
Thanks in advance
Regards
Christian Peter
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Sorry to reply to myself,
if I dial out with ISDN it works. I don't have a different SIP account
to test dial in SIP_PROVIDER_1 and dial out with SIP_PROVIDER_2.
Am Donnerstag, den 05.10.2006, 11:14 +0200 schrieb Christian Peter:
Hi list,
I hope somebody already had this kind of problem
Nick,
I use one and it works just fine for me with 2 FXO and 2 FXS at the
moment. I would say it is a great board to have and experiment with and
as you say not too big or too small.
Peter
Nick Ellson wrote:
I know it's not a digium product, but the 12 port A1200P card with a
single FXO
Hi,
scenario:
Call comes in via ISDN BRI on Asterisk A. Callerid (set by zapata) is let's say
0151123456789. In the incoming context I prepend a 0 to that callerid. My snom
correctly displays 00151123456789. The call is also forwarted to Asterisk B.
On the incoming context of Asterisk B I
Sorry for replying to my own post: I just switch the connection from Asterisk A
to Asterisk B from SIP to IAX without changing anything else (dialplans on both
system are the same). Now the correct callerID is logged. The behaviour changed
from 1.2.9 to 1.2.10 I suppose since this worked
Ok... I got it. Someone changed the CDRs to reflect CALLERID(ANI) instead of
CALLERID(number) in 1.2.10. According to the release notes this was taken back
in 1.2.12. I do not know why this was not done for IAX as well so it would have
been consistent at least but well...
I am either going
On Monday, September 04, 2006 3:22 PM Ronald Wiplinger wrote:
What's happen to you guys?
Nothing. Why?
I am not yelling, just asking.
Maybe in a bit stressed out kind of way.
It is sure not a dialplan question!
Without having all necessary information that is hard to say. Maybe one
On Tuesday, September 05, 2006 2:06 AM Ronald Wiplinger wrote:
In my opinion Asterisk remembers all numbers and therefore it does
not wait for the 4, since it found a match. This is in VoIP (in my
If both phones enter the dialplan the same way and one phone does work then it
should not be a
all that for free. Enjoy!
Peter
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Hi,
can anybody comment on patton inalp voice gateways and Asterisk? How good is
there echo cancellation? How good is the interop with Asterisk? I am especially
looking for reports on 4630 and 45xx series with BRI.
Thanks a lot in advance!
Kind regards,
JP
On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote:
try that way. However, I have doubts as well. If you are right, than
why snom phone does not have this problem? Would not here also the
first match count?
Because the transfer button on the SNOM is using a totally different
seconds? A minute if
you're a slow typist...
Yes, you can do this. #include is a literal text include, as the last
poster said.
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on Tuesday.
Peter: No need to be an ass about it, pal. Not all of us are as adept
at this as you are.
You've still not got it. #include is a general text include - can be
used anywhere. Well, perhaps it has to be at the start of a line.
Contexts, not even the [general] section which isn't actually
I'm monitoring my tftp servers' logs and my Cisco 7960 test phone
won't download dialplan.xml to the phone. I know this from the logs
and from the behavior of the phone. I see it downloading other files
like the ring tone file, etc.
Is there something that needs to be set in the cnf files to
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
of the
column. I've tried turning the column into a text and Asterisk copes
with that but still truncates it somewhere. Is this possible / is there
a constant I can change somewhere? :)
Thanks,
Peter.
This message has been comprehensively scanned for viruses,
please visit http://www.avg.power.net.uk
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On Thursday, August 10, 2006 10:43 AM Christophorus Laube wrote:
Hi list,
how can I realize explicit call transfer? I want to transfer a call
which I answered to another phone and it the other one answers I want
to hang up so that my resources are freed. Is that possible with
Zaptel or
If developers only ever get feedback from other developers, how will
they ever produce something that the market needs? Shouln't you also
listen to your customers?
And surely someone who uses Plesk already is ideally placed to give an
opinion on whether it's suitable?
Peter
On 08/08/06
Switch your echo canceller to MG2. Look for zconfig.h in zaptel and
recompile. Also experiment with echocanel=256 and adjust rxgain/txgain.
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On Wednesday, August 02, 2006 8:35 PM Vadim Berezniker wrote:
No idea, but DIALEDPEERNAME should contain the same value as
BRIDGEPEER. Try that.
Already did. Both are empty.
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Hi,
an all of my installations (1.2.9.1 bristuffed) the parameters BRIDGEPEER and
DIALEDPEERNAME are empty after a successful dial command. Can someone please
try to confirm this? I am not sure what I could have done to the implementation
to cause this so it might very well be a bug.
Thanks,
On Wednesday, August 02, 2006 6:49 AM Eric ManxPower Wieling wrote:
Zap/10-43 would indicate that this is the 43rd call (call waiting) on
channel 10. Obviously this would have to be removed to do it the way
you want.
Obviously. :-)
Or we find another solution for the problem/challange...
On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote:
DIALEDPEERNUMBER contains the exact peer spec for the peer that
picked up. You can use that.
Consider yourself my hero of the day! That looks VERY promising. It does not
show the technology so
On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote:
when you park a call (asterisk feature defautl keys: #700 ...) at
your isdn phone and you forgot to catch the call on another phone,
the phone from where you parked the call, should ring after 45
seconds (default)
does this work for you?
On Friday, July 28, 2006 3:12 PM Kai Ober wrote:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I would love to see the called
On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:
I am trying to have thier PC run thru the port on the phone and the
phone give prioroty to itself and the rest to the PC. When my client
does a big download the phone call gets real bad. The docs from SNOM
on TOS (or DIFFSERV) is poor and I
On Monday, July 31, 2006 9:22 AM Marcus Carlson wrote:
So, my question is; How can I make SIP/ext1 call continual all the
time?
Use a local channel. Have a look at voip-wiki cmd Dial. There should be an
example there. Basically you create a local channel in your dialplan that first
waits
On Monday, July 31, 2006 3:12 PM Kai Ober wrote:
(How do you get to the dial command, can you send the extension for
this?)
the idea is to to use $EXTEN.call a macro with $EXETN as an argument
...
The problem is this:
exten = 43,1,Dial(SIP/phone_1Zap/g1/43)
I need to find out who
On Montag, 31. Juli 2006 6:27 William Piper wrote:
Doesn't the dstchannel in the CDR's show this already?
Does not work for zapata BRI/PRI combinations. If I call Zap/g1/43 e.g. and
group 1 is span 1 and 2, dstchannel will be something like Zap/1-1 or Zap/2-1.
Imagine
How about up.oneTouchVoiceMail=1 in your sip.cfg
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein
Sent: Sunday, 30 July 2006 8:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Polycom 1.6.7 Firmware Messages Button
On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:
Does anyone know how to set up QoS on the SNOM 360 ? Thanks.
What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a
Snom 360 that will manage things for you. AFAIK all you can do is tell the
phone (or * or whatever)
Hi,
if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who
exactly picked up the call? In the cdrs dstchannel I can see the channel but
not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1
which does not help me unfortunatly.
Any ideas?
Kind
Hi,
we are having some trouble with CDR records. Example:
Case 1: Customer 12345 calls extension 10. Extension 20 takes the call
using Pickup (e.g. *810). I now have two CDRs:
1: 12345 - 10
2: 20 - *810
I could live with the second CDR but the first gives the impression as
if 12345 was talking
.
Peter
On 27/07/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I opened bug #0007490 the other day. The issue was that when you do a 'sip
debug' on the Asterisk console, there was no way to have this output go
_only_ to the messages file. Someone with the id of 'russell' in his
infinite wisdom
Hi Issac,
If I recall correctly, out of band DTMF didn't seem to work for us on
our Vega 50 (atleast not when using the Vega with Asterisk). We had to
tell Asterisk to use dtmfmode=inband in our sip.conf. It didn't seem
like we had to change any settings on the Vega, because it was sending
both
for us.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
Doyle
Sent: Tuesday, July 25, 2006 12:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] vegastream 50 FXO DTMF Problem
Hi Issac,
If I recall
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On Montag, 17. Juli 2006 6:40 ted jones wrote:
I have been trying to read up and understand Asterisk. I have a
small office of 25 people growing to 50 and have a dedicated DSL for
Asterisk
What kind of DSL? Synchronous, Async? What speed?
and another DSL for computer use and was wondering
On Freitag, 14. Juli 2006 10:13 Adrià Vidal wrote:
Someone using these phone Snom 300 with his own headset ?
We used to but the quality was horrifying. Since we changed to Plantronics
Noise Cancelling headsets everything is wounderful.
We got horrible static noise on them?
Maybe the
to set a
callerid
for this dial() command, without changing the original channel's
callerid.
/snip
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrea
Spadaccini
Sent: 14 July 2006 14:35
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
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support say? I presume they were your first port of
call, since they're the people prividing you with service
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On Friday, June 23, 2006 4:08 PM Steven wrote:
Exchange changes
http://www.microsoft.com/exchange/techinfo/tips/mailtip01.asp
Looks promising and helps a bit. Still no use of precedence bulk etc. though.
Very poor detection of lit mails.
___
of signalling.
Perhaps it would be easier to have it configured as either ETSI ISDN
(CPE) or QSIG, and then configure your digium as per the
samples/wiki/etc... (as either QSIG or ISDN NET)
Cheers,
Peter
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On 05/07/06, Kai Fürstenberg [EMAIL PROTECTED] wrote:
Just dial the international number completely (e.g. for Germany 0049etc.)
In your extension above a number beginning with 011 is being dialed.
That is not an international number.
Where were you assuming the OP was dialling from?
--
Peter
Dear List!
I'm looking for a way to display the current status of call forwarding with
PHP in a webpage.
Does anyone has an idea how to do this?
Can I get this info with a command line batch?
something like asterisk -r -x commandtodisplaycfwdstatus
Thanks to all!
peter
, packet delay)
* something else?
What is a solution? I think the jitterbuffer in 1.2.6 is broken, yes?
Beckman
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. 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
Yes - every message I've sent to the list in the past few weeks is now
arriving back here. I'd ignore it, it's harmless...
Peter
On 27/06/06, Mike Fedyk [EMAIL PROTECTED] wrote:
Is anyone else getting messages from the lists.digium.com mail server
with errors about a mail loop?
I've been
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
Hi,
Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I
am currently using Firmware 5.5 without serious problems but wanted to make
sure 6.X will work as well (including subscription etc.)
Kind regards,
JP
smime.p7s
Description: S/MIME cryptographic signature
On Thursday, June 22, 2006 8:13 PM Anthony Rodgers wrote:
We use MS Exchange too and, as far as I am aware, it is cognizant of
mailing list headers and doesn't send OOO notices to mailing list
postings. The only mailing list from which I receive my own OOO
notices is one that doesn't have the
The Polycom 501's or 601's are the way to go
On 6/23/06, shadowym [EMAIL PROTECTED] wrote:
I love my Aastra 9133i with v1.4 firmware.Pretty much everything justworks with Asterisk right out of the box and it has all the features I need.
-Original Message- From: Jonathan k. Creasy
,Playback(tt-weasels)
You have it backwards. The callerid to match goes after the extension,
not before.
--
Peter Bowyer
Email: [EMAIL PROTECTED]
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d: This flag trumps the 'H' flag and intercepts any dtmf while waiting for the call to be answered and returns that value on the spot. This allows you to dial a 1-digit exit extension while waiting for the call to be answered - see also
On 6/22/06, John Klimek [EMAIL PROTECTED] wrote:
Anybody
---
Peter Beckman Internet Guy
[EMAIL PROTECTED] http://www.purplecow.com
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