.
Dave Cotton
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On 10/03/11 12:55, Gilles wrote:
On Tue, 08 Mar 2011 13:22:18 +0100, Gillescodecompl...@free.fr
wrote:
I need to write a script which prompts the callee to type a number,
and then read it back to them as confirmation:
Apparently, the right way to read a phone number back to the user is
not to
On 08/12/10 17:53, Danny Nicholas wrote:
Thanks
Just my .02, but since you’re going to (quite possibly) have a long(ish)
timeout if internet connection or SIP provider is down, I would have an
AGI run in front of my dial that did a ping to verify internet and sip
provider
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden.
Dave Cotton
On 22/10/10 11:05, Hans Witvliet wrote:
On Fri, 2010-10-22 at 09:20 +0200, Dave Cotton wrote:
On 21/10/10 22:04, Hans Witvliet wrote:
For suse there is a precompiled version on the OBS (vitsoft)
Package search on the OBS shows nothing for 1.8.0 at all.
Perhaps you know where it is hidden
Just done a clean install of rc5 on a totally new machine and found the
following:-
/etc/init.d/asterisk start
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
Dave Cotton
On 21/10/10 17:05, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
What OS are you running
On 21/10/10 16:40, Dave Cotton wrote:
More interesting is that after make samples I have no iax2 available.
Adding more info :-
[Oct 21 17:14:04] WARNING[17255]: loader.c:387 load_dynamic_module:
Error loading module 'res_crypto':
/usr/lib/asterisk/modules/res_crypto.so: cannot open shared
On 21/10/10 17:19, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Adding more info :-
Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely
are missing libssl-dev (openssl) on our box.
Yes and ./configure and make
On 21/10/10 19:26, Paul Belanger wrote:
On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Yes and ./configure and make menuselect did not signal it. :(
Did the patch at-least work for you?
I'd already edited the init file so I didn't use it..
Dave Cotton
On 31/07/10 16:06, bruce bruce wrote:
Now that I check again, I see that DNSMasq for DHCP and DNSMasq for DNS
is NOT enabled. Which one should I enable and also can you please detail
what DNSMasq really does?
Look at
http://www.dd-wrt.com/wiki/index.php/DNSMasq_as_DHCP_server
DC
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On 30/07/10 16:15, bruce bruce wrote:
Adria,
How can I build a dns cache into my lan? I am using a Linksys 48 port
POE switch and running a micro DD-WRT firmware on a linksys router.
DD-WRT supports DNSMasq which would do exactly what you need.
DC
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On 05/02/10 09:15, Randy R wrote:
On Fri, Feb 5, 2010 at 8:41 AM, Olle E. Johansson o...@edvina.net wrote:
What I have seen on my asterisk box when I had a up/down adsl line was
that the asterisk box couldn't do dns resolution and would hang( well no
other internal calls could be made, seemed
On 05/02/10 14:21, Nikhil Nair wrote:
Hi again,
OK, I've now installed a local caching nameserver, but don't see any
change at all.
IN detail, what I did:
- Installed Debian packages resolvconf and dnsmasq (resolvconf just takes
care of dynamic nameserver allocations in
On 05/02/10 16:01, Jeff LaCoursiere wrote:
On Fri, 5 Feb 2010, Vinícius Fontes wrote:
I solved similar issues by setting srvlookup=no, having bind running
locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf.
Your local bind is what solved the problem. The srvlookup=no
On 19/11/09 15:37, Asterisk Development Team wrote:
The Asterisk Development Team is pleased to announce the release of Asterisk
1.4.27, 1.6.0.18, and 1.6.1.10. These releases are available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk/
The site page still
]: Leaving directory `/usr/src/linux-2.6.31.5-0.1-obj/x86_64/default'
make: *** [modules] Error 2
But actually I only want dahdi dummy at the moment where could I modify
the Makefile to just do this?
Dave Cotton
smime.p7s
Description: S/MIME Cryptographic Signature
is asterisk does not get any further than
*CLI == Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
I look farther tomorrow.
Dave Cotton
smime.p7s
Description: S/MIME Cryptographic Signature
hadi motamedi wrote:
Dear All
Can you please do me favor and let me know if there is an facility in
Asterisk server that can be used to have remote access to the server ?
Please be informed that we have installed commissioned our Asterisk
server at remote site with DECT telephony service
reza adinata wrote:
Hi guys,
I am trying to compile zaptel, using debian 4r5. However what I get in
zaptel 1.2.27 after make is below :
You do not appear to have the sources for the 2.6.18-6-486 kernel
installed (under ).
make: *** [modules] Error 1
tried to change the source with
Paul Chambers wrote:
I'd recommend dnsmasq. I've been running it for a few years, and it
works very well for me. Besides DNS, it optionally supports DHCP
(integrated with DNS) and TFTP. A basic (i.e. normal :) configuration is
easy to set up, though there's plenty of depth if you need to
Andres wrote:
Phil Knighton wrote:
Hello all
What I'm looking for is some plain speaking advice on ISDN.
Currently using 4 analog lines connecting via a four port TDM400P FXO
card. We need to physically move our installations, and on requesting
the analog lines be moved - our
Hans Witvliet wrote:
There's not much that can stand lightning (not just a direct hit), so
you cant't blame the sipura box for that.
Even when it was build, using a Faraday-cage with double insulation with
optocouplers, the amount of energy picked up by a 3 km line is beyond
commercial
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them at our respective homes. Rock solid stable.
No issues
Joseph wrote:
On 07/11/08 18:37, Dave Cotton wrote:
SIP wrote:
Joseph wrote:
I need another Sipura 3K and the replacement I think is Linksys SPA3102.
Any input on how reliable is it?
We have a few dozen subscribers using them at any given point in time. I
and my wife even use them
Simon wrote:
Hi There,
Has anyone managed to get 2 AVM ISDN Fritzcard's working in with a 2.6
kernel system?
Yes, with Suse 10.2/10.3 and chan_misdn.
/usr/sbin/misdn-init config wrote this:-
#
# Configuration file for your misdn hardware
#
# Usage: /usr/sbin/misdn-init
Matt Watson wrote:
On June 30, 2008 08:44:44 pm Simon wrote:
Hi There,
I am looking to build an Asterisk server with dual AVM Fritz!PCI cards
linked to 2 BRI in New Zealand. Just wondering if anyone has done
this, and if you have any ideas about the best disto choice for this
task?
Let
Brent Davidson wrote:
Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.25 and 1.4.10. These releases contain many bug fixes as
well as performance enhancements.
A couple of the more major changes include: modifications to
them.
Where is that? I don't seem to get that option. What I want is an announced
call transfer to another SIP device.
I've always used # which, in my features, conf is configured for
attended transfer and ## which is configured as blind.
Dave Cotton
to pay.
Dave Cotton
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On Friday 25 January 2008 05:25:57 Lyle Giese wrote:
You need to do a 'make' before the 'make install'.
make install will do all that is necessary to install a program including
making any files necessary.
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On Tuesday 11 December 2007 06:35:24 dave cantera wrote:
!DOCTYPE html PUBLIC -//W3C//DTD HTML 4.01 Transitional//EN
html
Please send to the list in text and not HTML
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installed mISDN. I was up and running in minutes and now all my
ISDN PBXs are running chan_misdn with either Digium , Junghanns, AVM Fritz or
Bewan cards.
Dave Cotton
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asterisk
://www.talkshoe.com/talkshoe/web/tscmd/tc/22622
Good luck with the book, Stephan!
I'll echo that, it's already solved one problem for me regarding pattern
matching. Great work.
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interesting was that you can even do that when your ISP is Orange a
division of France Telecom the company that will loose the telephony
business. It's not too well advertised though.
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while he’s still hearing the ringing tone.
How can I investigate those 2 problems in order to find what’s
happening ?
Contact the Trixbox mailing lists?
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On Tue, 2007-05-15 at 19:57 +0300, Diego Iastrubni wrote:
On Tuesday 15 May 2007 19:11, Dave Cotton wrote:
Contact the Trixbox mailing lists?
Why is that? You think some fancy-shmancy GUI will fix this? The problem is
obviously in the zaptel area. But hey... this is asterisk-users...
/me
On Tue, 2007-05-15 at 19:16 +0200, Dave Cotton wrote:
Perhaps the fancy-shmancy GUI is hiding the configs.
Al Bochter has just told me off list that Trixbox is Asterisk
But according to their site
trixbox is a complete application platform. When you install trixbox you
have a powerful
spoken up, perhaps I am
off base. Please help clear up what I am missing.
Should be clearer now.
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in *, or is it likely to be
sipura-specific?
No. The phone in question didn't answer the call so it's a missed call
as far as it's concerned.
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this.
http://www.voip.com.sg/voip_products/voip_ip_phone_provisioning_tool.html
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)
And that's where it started to go wrong.
(S0:[EMAIL PROTECTED]:5060)
will do what you want.
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it has a pass
through for the phone power supply.
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look like this
#
#
# 480i
#
#
sip screen name: Linux Autrement
sip line1 auth name: 2001
sip line1 password: password
sip line1 user name: 2001
sip line1 display name: Dave Cotton
sip line1 screen name
that the main problem with tapes was that the bits fell off,
how many people actually test the restore capability before they
actually need it?
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asterisk-users
this
purpose? Is there a provider(voip prefer) who offer a special account which
is able to handle multiple calls simultaneously?
Thanks in advance.
I love it, a question like this from a _hotmail_ address.
Of course he could have a legit reason, but that email address.
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to make 50 simultaneous calls. What difference does the length
and frequency make.
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; done
if you substitute aastra for polycom?
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aastra for polycom
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about chan_bluetooth you are talking
about chan_cellphone.
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of pickup is
already in place.
Need much more information.
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phone when the called phone answers.
I have just tried calling my mobile using the fixed line and saw this
Call on SIP/2001-081f7e58 left from hold so did have two sound.
The fixed line is connected via an SPA3000 so SIP is there also.
Thoughts anyone.
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On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote:
Hello All
This month we would like to offer our GSM Gateway range for less to
clear up some spaces.
etc
Perhaps, you could explain what is NON COMMERCIAL about your post.
I would not buy anything from a spammer.
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the best results in a productivity
enviroment?
Aastra 9133i and 480i
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and send the info back to the phone.
Any chance of sharing the config line, my wife sees missed calls and
starts panicking.
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On Thu, 2007-01-18 at 20:13 +0200, Cosmin Prund wrote:
How about the Digium Wildcard B410P card? It seems to be Digium, it
has hardware echo cancel and I can buy this in Romania. Is this card any
good?
Well I hope so, I'm installing my first one next week.
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.
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frequencies)
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it with Firefox, try again.
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I'm running chan_capi on a number of systems in France, France Telecom
offer the possibility of having the caller's name, but say we must
configure for EuroISDN+. Google doesn't show much and the best I could
see was in Dutch.
Any Europeans solved this one?
Rgds
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day I saw a setting Global in the
dropdown for the FXO since then no one has complained about echo on any
of the SPA3000 units I've got installed. Give it a try it might or might
not help.
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On Thu, 2006-10-26 at 13:08 -0700, Martin Joseph wrote:
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
Since they are incorporated in a single product which is doing the
configuration, consistency where possible would be good...
That product is designed to link the two
'wilder' is now TOO LAGGED (2055 ms)!
How is this a problem with IAX? The notice is telling you that the delay
on the lan/wan is such that the call would be awful.
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different, so setting qualify=3000 will
ping peer every 3s,
quite inconsistent, imho
So are you saying that in your world two different things, created by
totally different people, must have the same configuration settings.
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networking should work at that time or not at all, not
somewhere in between.
I've seen this type of nonsense with newer versions of Suse. I've never
bothered to find out why, just changed the priority manually to make
sure it does what _I_ want.
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On Thu, 2006-10-12 at 14:28 -0700, Tom Lynn wrote:
Dave,
Are you in the US?
No, France.
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the registration, then when an internal phone tries to
re-register it can't, in the end all the phones go out of service one by
one according to their re-registration period.
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On Thu, 2006-10-12 at 12:00 -0400, Jay R. Ashworth wrote:
On Thu, Oct 12, 2006 at 04:53:51PM +0200, Dave Cotton wrote:
It's more likely directly linked with how asterisk deals with
registrations to external SIP/IAX servers it appears to sit there for
ever trying to do the registration
because all calls via the sip providers are outgoing. But it does not
alter the fact that if I do register with a provider and the ADSL is
down, * blocks.
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or issues with these?
Well, I have had echo issues. Then I find out the echo cancellation on
PSTN line is switched off by default. I switched on, and no echo any
more :)
I have had echo with the SPA3000 but I switched to Global impedance on
the FXO and since then clear as a bell.
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) causing the internals to go to No service. So when someone came
in in the morning ...
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+33 (0)4 90 23 30 81
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...
Oh, that's why I'm not getting any phone calls.
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9133i or 480i
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Autrement
sip line1 auth name: 2001
sip line1 password: password
sip line1 user name: 2001
sip line1 display name: Dave Cotton
sip line1 screen name: Dave Cotton
sip line1 vmail: *10#
sip line2 auth name: 2001
sip line2 password: password
sip line2 user name: 2001
sip line2 display name: Dave Cotton
On Sat, 2006-09-30 at 09:35 +0200, Dave Cotton wrote:
and 00085D183552.cfg (not uppercase) contains
Whoops note
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On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote:
I’m unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Confirmed here, I only noticed because of this message.
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Dave Cotton [EMAIL PROTECTED
, the 9133i is solid and professional
looking and works very well with *. My experience with support is A1.
Message waiting is well signalled as is no service.
The switch and POE save a lot of cabling.
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?
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On Tue, 2006-09-12 at 09:00 -0600, [EMAIL PROTECTED] wrote:
I've ever post this question many times on asterisk
users without success ?
As I and many others have probably noted.
I found then neatly filed in junk mail.
Perhaps you're getting your just deserts.
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On Sun, 2006-09-10 at 09:07 -0500, Michael Graves wrote:
The POE idea explains a lot. I was wondering how one forces a reboot
on the phones in order to direct uptake of new settings.
In the case of Aastra phones (and others, I only use Aastras) use
sip_notify.
about that patch, nice to see the manufacturer's people on
the list. Gives confidence in their product.
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On Thu, 2006-08-17 at 14:22 +0600, Dualcall.com wrote:
2000+ lines:D
Film Script?
No, SCO lines.
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.
Please actually read the GPL.
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set up the Sipura as show in the document I sent you
privately 2 weeks ago? Because it all works on my systems.
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:-) 192.168.1.1 is my Asterisk server, and the ATA is at
192.168.1.113.
Yes, just set
PSTN CID For VoIP CID:
to guess what:- YES.
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On Wed, 2006-07-19 at 19:04 +0800, Sam Tam wrote:
You will need an asterisk server + X100P + GSM Gateway say from
cyber-telecom.net
Not forgetting that the above person IS cyber-telecom.net.
Therefore his advice is not impartial.
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as you think it does?
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! with chan_capi until
someone pointed me in the direction of the codec setting in capi.conf
;ulaw=yes;set this, if you live in u-law world instead of a-law
I thought I am in the u-law world but evidently I am not.
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you can support the protocol on a connection
tracking/NATing firewall.
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count 6 of them) all
the time to make and receive POTS calls and have not had any trouble at
all.
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On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote:
What brand/model phones are you using.
Aastra all models
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than to say it does happen elsewhere.
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at the often totally stupid
posts on his part.
C'est la vie.
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quality is good across the range.
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sub directory ?
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On Thu, 2006-06-15 at 11:33 +0300, Khaled Chehab wrote:
I am using [EMAIL PROTECTED] version 2.6 how can I update the existing to
be 2.8.
Any ideas
Ask on the AAH lists?
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Dave Cotton [EMAIL PROTECTED]
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On Thu, 2006-06-15 at 10:16 -0500, Aaron Daniel wrote:
Dunno if anyone else has seen this yet:
And that is perhaps why the current version of Asterisk is 1.2.9.1.
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Dave Cotton [EMAIL PROTECTED]
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/sounds'
make: *** [datafiles] Error 2
but the file is definitely there.
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Dave Cotton [EMAIL PROTECTED]
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On Thu, 2006-06-08 at 09:26 +0200, Dave Cotton wrote:
I'm getting this error when compiling:-
make[1]: Entering directory `/usr/src/asterisk.svn/sounds'
--09:22:12--
http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz
= `asterisk-core
On Thu, 2006-06-08 at 16:28 +0200, [EMAIL PROTECTED] wrote:
Hello,
I have to dial prefix 9 for non local numbers however
when i missed calls i Can't redial this number
because of 9 is not append .
I use polycom phones .
What Can i do ?
RTFM?
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Dave Cotton [EMAIL PROTECTED
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