A T carrier cable is not the same as an ethernet cable. A T carrier cable uses
a real metal shielded RJ-45 and loosely twisted pair wire. With most modern T
carrier equipment, you can use a CAT-5 ethernet cable instead of a real T
carrier cable. A T-carrier crossover cable does not have the
should be fine. As far as the shielding goes, I use UTP cables and
Connectors all the time and some of my X-connects run over 100 feet.
Paul Mahler - [EMAIL PROTECTED]
www.signate.com
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk
We still have a seat open in the London
Introduction to Asterisk class.
TKS
Paul
Paul Mahler
[EMAIL PROTECTED]
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of McQuiggan, Mark xt46480
Sent: Sunday, March 05, 2006 12:20
PM
To: asterisk-users
There are still seats open in our March 21st to 23rd Introduction to
Asterisk and VoIP telephony course. More information is available at
www.signate.com.
Paul Mahler
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing
Based on our benchmarking, I am VERY
skeptical of this number. Im guessing that you dont really have
RTP streams going through the NIC.
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Joash Herbrink
Sent: Wednesday, February 01, 2006
12:23 AM
To: Asterisk Users
Have you verified that you are actually
sending sound over the RTP streams?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith
Sent: Friday, January 27, 2006
11:13 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re:
Signate sells a single server that can get
you to the call volumes you need.
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vic
Sent: Saturday, January 28, 2006
7:16 PM
To:
asterisk-users@lists.digium.com
We still have a seat open in our Asterisk training course next week in
London. You can find more information at our Web site, www.signate.com
I'm going to be teaching the class.
Paul
___
--Bandwidth and Colocation provided by Easynews.com --
to customers, or is it more sysadmin oriented?
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Mahler
Sent: Thursday, January 05, 2006 9:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk
I would be MUCH more tempted to use an IAXy or SIP adaptor and a cordless
phone. It will be less expensive and it will likely work better.
Paul
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joash Herbrink
Sent: Monday, January 09,
The second edition of my book VoIP Telephony with Asterisk is now in
print and available. You can find out more about it at our web site
http://www.signate.com/products.php
This book is written for beginners. It will make it easier for you to get
started. The second edition is reorganized and
The second edition of my Asterisk book VoIP Telephony with Asterisk is now
in print. It's reorganized and expanded.
TKS
Paul Mahler
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
___
--Bandwidth and Colocation provided by Easynews.com
We sell a complete Web facing interface called sigMan that works with
realtime.
http://www.signate.com
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe
Sent: Friday, December 23, 2005 10:40 AM
To: [EMAIL PROTECTED]; Asterisk Users
-Users] Avaya 4612 IP phones with Asterisk?
I have gotten 4620's to work ( convert to sip ) It works ok... at best. I
have a 4612 at work I will try tomorrow.
good luck with yours .
Dave
_
From: [EMAIL PROTECTED] on behalf of Paul Mahler
Sent: Tue 11/8/2005 2:06 PM
To: 'Asterisk
Has anyone been able to make these phones work with *? If you have, what
does it take?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk
Why do you want to use a SIP provider instead of a PSTN connection?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier Taylor
Sent: Wednesday, November 02, 2005 4:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE :
We have a turn-key solution available that does exactly what you are asking
for. You can reach someone for more information at 415.442.4010.
TKS
Paul
[EMAIL PROTECTED]
trixter aka Bret McDanel wrote:
I am tasked with evaluating ready made solutions for a voip provider.
Does anyone have
:
http://lists.digium.com/mailman/listinfo/asterisk-users
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
___
--Bandwidth and Colocation sponsored
I am looking for someone who knows how to configure cisco routers to work with
*.
You can contact me at
Paul Mahler
[EMAIL PROTECTED]
Thanks!
Paul
Paul Mahler
www.signate.com
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk
we used sipp, the opeh source benchmarking software sponsored by HP. We can
send you our benchmark, if you like.
We did run into a problem, though. The benchmark suite core dumps on us at
about 5100 simultaneous SIP streams.
Regards,
Paul
Paul Mahler
www.signate.com
?
Thanks,
Adam
Paul Mahler
www.signate.com
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
to Paul Mahler at [EMAIL PROTECTED]
Paul Mahler
www.signate.com
___
Asterisk-Dev mailing list
Asterisk-Dev@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman
=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
group=1
callgroup=1
pickupgroup=1
immediate=no
context=main
callerid=Your Name 555-1212
channel = 1
Hope this helps.
Paul Mahler
/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.322 / Virus Database: 267.4.1 - Release Date: 6/2/2005
Paul Mahler
Slash-dot is pointing to this article on Asterisk and Pingtel.
http://www.theregister.co.uk/2005/05/22/pingtel_voip/
Paul
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman
.
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-Virus.
Version: 7.0.308 / Virus Database: 266.9.13 - Release Date: 4/16/2005
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update
There's a long chapter in my book about re-programming the 7960 from skinny to
SIP that might help you out. Figuring it out was non-trivial. You can get the
book at Amazon.
TKS,
Paul Mahler
I can't get the 7960 to reconfigure and work. I am a newbie to voip. I went
through the list and read
/ Virus Database: 266.8.6 - Release Date: 3/30/2005
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
with Asterisk Book and CD Set is $89.95 and Signate's
Asterisk Installation 2005 CD is $49.95. They are available at Amazon, Signate
or Ebay.
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
.
Version: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
: 7.0.308 / Virus Database: 266.8.1 - Release Date: 3/23/2005
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
You can also use * and linux to partialize the T1. You better plan on
spending a lot of time on making it work. You have to install the Linux
packages to split the line. NON trival. Works great,
though.
Paul
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
On Mon, 2005-03-21 at 21:16 -0700, Tim
Are you at run level 3? X can cause this if you are at run level 5.
Paul
Paul Mahler
[EMAIL PROTECTED]
www.signate.com
Hello
We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work
with our call agent.
Unfortunately **VERY** frequently, asterisk stops responding
I haven't used their 24 port gateway, only the four port gateway, but they have
been excellent.
http://www.mediatrix.com/products_devices.php?prodid=3
Paul
Paul Mahler
www.signate.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
s Introduction to
Asterisk and dCAP Certification Review prepares test takers for the
certification exam.
A Signate training class schedule is posted at
http://www.signate.com/training.php.
Certification is given under license from Digium.
=
Paul Mahler
www.signate.com
I have a new HP IpaQ 6315. I run SJPhone on it with a bluetooth headset. Works
great!
Paul
paul mahler
www.signate.com
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Friday, March 04, 2005 11:23 AM
We supply an * server that can support as many users as you want, 5,000 is a
small system.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED
HI,
1. Make sure you are running asterisk with the command
asterisk
With no arguments.
2. Make sure you are booting to run level 3 so that X-windows isn't running.
Paul
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of TELUX
Sent: Thursday,
Are you using oh323 ?
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jorge Alayon
Sent
The linksys BEFSR81 does QoS.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew Boehm
Are they just sending dnis? Do you have feature group D?
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
we run Asterisk on RedHat 9 with no problems. Works
great!
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
CA94107-1901Asterisk Services and
Training
From: [EMAIL PROTECTED
Beginners.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Sys. Concept Inc.
Sent: Saturday
Asterisk should run well with any Linux distribution.
Mepis, www.mepis.org, is pre-configured for *
and might make your installation faster andeasier.
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco
to get * running. You can get Linux installed and * running VERY
quickly if you start with Mepis.
Hope this helps,
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message
The expansion module is NOT supported with SIP.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Under what circumstances? If the first T1 is down, for example?
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third
I have a recent version installed. I am having problems with hangup
detection on my zap channels.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From
for. You should spend your money on getting a copy of each of the
two books that are now available and learn *. Then it will be clear to you
that you don't really want what you are asking for.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
should just be happy that Asterisk will do what you
want, even if SIP won't.
If you really, really want to do this, up the bounty to about $50,000 and
get the SIP specification changed.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
It's not what SIP does with SER, it's what SER does with SIP.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
, for example the IAX
jitterbuffer setting.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
Does asterisk provide quality of service(QoS)? If it does,
how do I use it? The reason why I ask
certainly an issue when making the decision to move off the
PSTN. Is the performance of your VoIP system going to be comparable to the
performance of your PSTN system? Sounds like a reasonble question to me.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San
as the address
of record, then the second client comes in with the same registration and
becomes the address of record?
Andy, I'm in your hands.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
Not from me. I think the more books the better. I'm looking forward to
getting my copy.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED
Das is aber schöen!
Paul von Wachter Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jo
Sent:
. Much less expensive, too.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ken D'Ambrosio
, add two additional extensions to the
7960. The admin can tell who is being called by the extension that rings.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message
I ordered a copy, but they said it's six weeks or so 'till delivery.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto
Ok, you asked for it, so here it is. ;-)
Fabulous! Works great! Love Firefly! Magnificant job!
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From
the stable branch.
TKS
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
CA94107-1901Asterisk Services and
Training
signate small logo.gif
/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED])
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Telnet to the phone and look at the sip debug trail. Probably a wrong ip
address somewhere.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL
You could also install ngrep and watch the traffic go by on port 5060.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
username=514
secret=password
context=inside
callerid=Paul Mahler 4154424024
qualify=1000
host=dynamic ; This host is not on the same IP addr every time
canreinvite=no
[EMAIL PROTECTED] ; Activate the message waiting light for
waiting messages
;defaultip=192.168.0.102
If you use the mepis debian release, it's a piece of cake to install *. It
takes about 15 minutes to install Mepis and *.
www.mepis.org
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
I'm having this problem too.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED
does anyone have an example they would please share for turning on stutter
dialtone for a zaptel channel when there is a message waiting?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Please count me in for testing!
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
; spans one and two connect to the Adit 600 channel bank
signalling=fxo_ks
group=1
context=inside
channel = 1-48
callgroup=1
pickupgroup=1
callerid=Paul Mahler 100
context=inside
[EMAIL PROTECTED]
channel = 1
I have the following entry in zapata.conf, but I don't get stutter dialtone
when there is a message waiting. Suggestions? Please?
callgroup=1
pickupgroup=1
callerid=Paul mahler 100
context=inside
mailbox=100
channel = 1
Thanks,
Paul
___
Asterisk
I have had good experiences with Adit. Their customer service and
documentation are excellent.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message-
From: [EMAIL PROTECTED
I copied voicemail files to a replacement system. When vm tries to play the
file * throws an error messages:
Unexpected header size 16
unable to open fd on /
How can I copy the VM to the new machine?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.signate.com/
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http
It's avaialble at:
http://www.carrieraccess.com/support/products/index.cfm/fuseaction/default_p
rod/cat_id/21.htm
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message-
From: [EMAIL
Title: Message
Does the Cisco softphone work with SIP? The factsheet
only talks about SKINNY.
Paul
Mahler [EMAIL PROTECTED]
Signate, LLC665 Third
StreetSuite 100San Francisco,
CA94107-1901Asterisk Services and
Training
Excellent answer. Thank you very much.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andreas Frackowiak
Sent: Saturday, May 15, 2004 1:32 AM
To: [EMAIL
they
are calling from.
Thanks Guys
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.signate.com/
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
___
Asterisk
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.signate.com/
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
Cylogistics sells a sayson phone that's very nice.
http://cylogistics.comtelligence.net
http://www.sayson.com/product/analog_phone.htm
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original
Using the sunc from the T1 line made my problems go away.
Thanks Andrew!!!
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
The Adit channel bank we are using, and XO communications who provisioned
the T1 are both showing a LOT of framing errors on our system.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message
you need to combine both sides of the conversation. This
should be covered in the archives.
Paul
Paul
Mahler [EMAIL PROTECTED]
Signate, LLCPO Box
60430Palo Alto, CA94306VoIP Systems, Training
Consulting
From: [EMAIL PROTECTED
Well, for me it was a problem with the T1 line. XO fixed the line and the
ticking sound is gone!
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems Training Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
i'm getting a tick every second or so on all my calls. All channels are zap
channels.
Does anyone know how to fix this?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
http://www.signate.com/
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training
I guess vocera doesn't have any RF engineers to tell them they can't do it.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
I'm using the stable
branch. Is voicemail or voicemail2 deprecated?
TKS
Paul
[EMAIL PROTECTED]
' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
== Spawn extension (macro-zapdial, s, 3) exited non-zero on 'Zap/49-1' in
macro 'zapdial'
== Spawn extension (main, 100, 1) exited non-zero on 'Zap/49-1'
-- Hungup 'Zap/49-1'
Paul Mahler
I got it! Nothing like posting to the mailing list when you're going to look
stupid to help you find the answer yourself!
The answer is to use waitforring(1)!
Thanks!
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training Consulting
in sip.conf
[general]
port = 5060 ; The TCP/IP port for SIP communiations
bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses
on server.
context=other ; Default for incoming calls
disallow=all
allow=ulaw
allow=gsm
in extensions.conf
[general]
Is anyone successfully using call queues and call groups? If so do you have
an example configuration?
The wicki and mailing list information I found is pretty old.
Thanks!
Paul
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL
You need a timing source for conferencing or music on hold. Voice mail works
fine without a timer. If there is no Zaptel card installed, you will have to
find timing from a USB driver, or recompile the real time clock.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo
Paul Mahler
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks!
Paul Mahler
[EMAIL PROTECTED]
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk
unpowered switch.
It's not the phone,
the problem moves from phone to phone. If it's happening to a phone and I
restart everything, when everything is back up a different phone will have the
problem.
has anyone seen
this?
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
Where and when is the rollout meeting? I'd love to attend.
Thanks!
Paul
Paul Mahler
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven M. Sokol
Sent: Monday, March 29, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk
, real technical support you should be willing to
pay for it, or in this case part of it.
Paul Mahler
mailto:[EMAIL PROTECTED]
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Saturday, March 27, 2004 7:37 PM
To: [EMAIL PROTECTED]
Subject: RE
1 - 100 of 149 matches
Mail list logo
