Thanks a lot Peter, you are 100% right, we were using G729. Changed
the codec to G711 and the %&/$!! confrnc button appeared. ;-)
Thanks again for your help,
Cristian
On 4/27/06, Peter Johnson <[EMAIL PROTECTED]> wrote:
> Are you using G.729? You cannot conference G.729 calls locally on this
> ph
Jay Milk wrote:
Could you kindly let us know what numbers those survey-calls will be
coming from, so we can all add them to our blacklists? Thanks!
*snickers*
--
JP Carballo
http://www.netfone2x.com
Bringing the world closer.
It might look like I'm doing nothing, but at the cellular level
Ronald Wiplinger wrote:
I tried now the examples in the wiki, but they do not fit!!!
If I use in configure Require Pins Yes then everyone needs a pin code!
If I use in configure Require Pins NO then calling in people will
just need to know a valid card number!!!
How can I overcome this?
Mo
Ronald Wiplinger wrote:
I want to use something like:
What is your card number:
Enter your pin:
Enter your destination phone number: phone number>
Is there a code snip available for that?
Not that I know of.
Just juggle the way the routines are called.
Everything you need (or most of
Joseph, I'm getting exactly the same issue as you. Periodically, I get
CHANUNAVAIL back from the PRI span and then we fail over to VoIP. There is
plenty capacity spare on the span.
If you get any further with figuring this out, please let me know. I'll do
likewise.
Cheers,
Mark
-Original Me
Benchev wrote:
astcc is such a neat and stable piece that I would hardly dare to
to mess with it.
You make it sound like it's anathema to modify it :)
My idea was 1) you need a PIN=YES because otherwise pins are
not generated;
You mean to say PINS are not required, right? Because they are
Dovid Bender wrote:
A while back some one posted some code that he used
that took out the flag in astcc that kept track if
there was a call in progress for that pin or not. Dont
know if it wil work for real time though.
Dovid
I don't know if you were pertaining to what I posted in the messag
OK, assuming the usbaudio sees the conference phone and can work it,
how would you write an extension to ring that?
on Thursday 04/27/2006 Steve Feinstein([EMAIL PROTECTED]) wrote
> It's a standard USB audio device. While I haven't tried it, I'm pretty
> sure the Linux USB audio driver will pr
Hi,
I was wondering if the Dev community could help me out with a few
answers to questions I am having a hard time getting info on.
Firstly, is there a document describing HINTING and how it works so I
can program a tool to monitor it?
Also, I have noticed with my Grandstream phone. They r
8<
Thanks for the pointer Nathan. I slapped something together quick 'n
dirty but this is not working. The problem is that the call file is
only
generated when the originating caller stays on the phone until the
remote caller hangs up. Suggestions how to fix this much appreciated.
>8
We
Time to report back. We took out the daughter board (no more hardware echo
canceling) on the TE411P, and the problem is gone. Guess we have to RMA the
card.
From: [EMAIL PROTECTED] on behalf of Wai Wu
Sent: Thu 4/27/2006 3:30 PM
To: Asterisk Users Mailing List -
Time to report back on this. We took out the daughter board from the TE411P as
per Digium's request, and the problem went away. We might just had a bad card.
One question thought, does the hardware echo cancellation work much better than
software?
From: [EMAIL
Please read my report on my interaction with digium under this subject. It
might be helpful for future reference.
From: [EMAIL PROTECTED] on behalf of Andrew Kohlsmith
Sent: Thu 4/27/2006 9:20 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] P
On Thursday 27 April 2006 14:18, Eric "ManxPower" Wieling wrote:
> >>> span=1,0,0,esf,b8zs
> >>> span=2,0,0,esf,b8zs
> Digium cards only support 1 timing source per card.
Setting the timing to '0' tells the card NOT to try to recover clock from that
span. Not what you want to do with telcos,
On Thursday 27 April 2006 12:18, Wai Wu wrote:
> My TE411p does not seem to like to have two PRIs from different telcos
> (span 1 and span 2). I can get one working, but not the other. However,
> if I use vpmsupport=0 when loading the wct4xxp module, they both work.
> But here is the problem, vpmsu
Andreas Sikkema wrote:
I believe this is called camp on. Found some examples on voip-info.org
but they assume that you do not hangup the originating phone. Anyone
have an idea how to implement this feature as described above?
When I worked at Philips there were two variants:
- camp on busy
- ca
On Thursday 27 April 2006 18:35, William M Conlon wrote:
> sip show peers showed no ip address for my 501. Rebooted the phone
> and it's ok now.
Without packet traces (just log UDP/5060 or you'll capture the entire audio
stream too) you'll never know.
I'm wondering if Asterisk or the IP501 "forg
I have a group of local users. They want to participate on a conference
call to a PSTN line in USA.
To connect they need to enter a code there as well.
I want to use a local conference room, where LAN users and local users
can call in. The conference room should be connected to the conference
I would like to set-up a "info system".
A. weather
I found the code to get the weather report and with festival I can
create the audio file.
However, I have different location of my useres, which are easy to
distinguish with the users starting phone number. E.g. users in region A
would start t
HDLC Abort errors are usually caused lost or corrupted data from the T-1.
If you configured the span as PRI in Asterisk and it's not a PRI, I can
imagine you might get this message.
Also, if there is a problem with the line itself, I can imagine you
might get this message.
However, in my ex
I was wondering if someone could help me with this, I’ve
searched high and low to find more info but with no success. When I want to transfer
a call from another ringing sip phone to my sip handset I dial *8. This works
but the caller ID shows up as *8 on my handset. What I want to do is be
Hmm sounds good.. Thanks for the input!
Terrelle
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Luki
Sent: Thursday, April 27, 2006 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Looking for input on wh
Bob McDowell wrote:
My Asterisk server believes that a Digium TE405P and sound are
incompatible. Basically, no matter what else I do to the machine it
terms of hardware, if the TE405P is installed, none of the
playback/background/etc commands work. MOH works fine.
Bob,
Can you be more spe
I would be interested in trying that our for our
outbound call centre, if you can get it working.
Kris Cote -
XS!te Networks Inc.Wholesale and Retail Voip in Canada and the
USA
- Original Message -
From:
TV JOE
To: asterisk-users@lists.dig
Terrelle,
I've implemented a similar setup about a year ago. Here are couple
observations worth sharing. YMMV, but these are my experiences:
1) A small LAN (~40 devices: PC, printers, phones) does not need QOS.
Even when a workstation floods it with 100 Mbps traffic there is no
quality problems o
I've got two Polycom 501s in an Asterisk 1.2.6/zaptel 1.2.5 with
Digium connected to POTS. My legacy key system is still running in
parallel.
Today my 501 didn't ring on an inbound call, nor could I call out.
sip show peers showed no ip address for my 501. Rebooted the phone
and it's ok n
It's a standard USB audio device. While I haven't tried it, I'm pretty
sure the Linux USB audio driver will probably see it.
-Steve
John covici wrote:
Any way to use this on a Linux box so I could use this with asterisk?
I have a windows box on the same network, but how would I get asterisk
t
Looks like a timing problem - zaptel.conf and zapata.conf, please.
A.
On Apr 25, 2006, at 3:05 AM, Nico Giefing wrote:
Hello,
I get an Error every minute on the second card of two installed TE410P
Cards in our System.
The error is:
PRI got event. HDLC Abort (6) on Primary D-channel of
I doubt we will need a 1:1 ratio, but I did want the possibly to have all 12
lines maxed out, and not run into a situation where someone gets a busy
signal or "all lines are busy" when that 13th call comes in.
I would love to go ip phones, but I fear the "folks who's putting out the
money" will ch
Are you using G.729? You cannot conference G.729 calls locally on this
phone. IIRC on some firmware releases, even if you are not using G729, but
G729 is in the phones codec list, the conf facility is disabled. Try
configuring the phone without the G729 codec option.
Peter
-Original Message--
amna saleem wrote:
Hi!
I have been using ASterisk 1.0.3 on Red hat Linux 9.0 for a long time
now on my Home PC.
I want to shift to a PC having SATA hard disk .Can I install Redhat
9.0 on SATA hard disk ??some people are telling me that I have to go
for Linux Enterprise 4.0.I don`t want to l
Before someone blasts you for not posting this to the biz list, I need
to ask -
This is NOT for US use, correct?
GSM wrote:
Selling analog GSM gateways at 119.00 USD. Polarity reverse for "call
answer" and "call end". Compatible with all Digium/Asterisk analog FXO
cards. Suitable for low-traf
Don't discount the 24-port analog cards to quickly. Both digium and
sangamo sells them with or without hardware EC. I know the sangoma one
works fine and am just about ready to swap to the digium TDM2400 for
equivalent testing. Both will handle 24 ports of fxo or fxs; mix or match.
Kerry Garr
What would the difference be with using IP phones or ATA's in that case? You
are still talking about a network device. Even with 50-60 stations installs
you are highly unlikely to run into a need for QoS internally.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL
I changed the DHCP overrides to 1, restarted the phone, now it showing the
right day but it is 4 hours and 29 minutes fast. Arrrgh.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rich Adamson
> Sent: Thursday, April 27, 2006 1:24 PM
> T
Kerry Garrison a écrit :
You will kick yourself up and down the block for not using IP phones in the
end. What are you going to spend? $65 for an ata and $25 for a phone? Spend
the extra money and get SNOM or Linksys phones. You then need to figure out
how to get 12 analog lines into Asterisk. U
Kerry Garrison wrote:> I am ready to pull my hair out. I cannot seem to get the Polycoms to > read the time properly. Regardless of the server they are pointed to our > the offset, i am getting the correct time, but 24 hours ahead. So for > today it is showing Friday April 28 but with the correct t
You will kick yourself up and down the block for not using IP phones in the
end. What are you going to spend? $65 for an ata and $25 for a phone? Spend
the extra money and get SNOM or Linksys phones. You then need to figure out
how to get 12 analog lines into Asterisk. Using 3 TDM400's is not reall
> >> Just to give you an idea
> >> I would suggest you to make two .agi files:
> >> astcc.agi and astcc-disa.agi
> >> In astcc.agi you'd leave everithing as it is, and enable
> >> PIN =YES through the astcc-admin.cgi.
> >> Thus you could dial without interogation:
> >> exten => _1NXXNXX,1,Dead
I don't have a polycomm manual handy, but I think I'd change the
overrideDHCP parameter to "1" and test. You are apparently in the PST
timezone?
If that doesn't do it, my next step would be to use ethereal to capture
one of the ntp request/response pkts and analyze the content. If that
looks
If you use a Sangoma T1 card, (A10x) card you can send both voice and
data down the same T1 and have the Sangoma card split it for you.
If you are talking about non-hobby usage, stay away from FXO adapters
and go with a T1.. You'll be much happier in the long run. For a
fractional T1, don't worry
I have a wired problem, i can recive call but i can't make any call.
ATT say that is my problema because the call is operator mode (???)
The log is
MFC/R2 Chan 1: Call control(1)
MFC/R2 Chan 1: Make call
MFC/R2 Chan 1: Making a new call with CRN 32772
MFC/R2 Chan 1: 0001
->
[1/
Are you passing the Offset through the DHCP server as well? On a linux
DHCP server this would be:
option
time-offset
-18000; # Eastern Standard Time
option
ntp-servers
192.168.x.x
The fact that the date is wrong, but the time is correct, seems a l
Here is the sip.cfg file
Using the same IP for the server on a Linksys SPA-941 everything is correct,
using this configuration on the 501 shows 24 hours ahead.
-Kerry
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Rich Adamson
> Sen
> > But then the call file just keeps sitting in the
> > /var/spool/asterisk/outgoing directory and it seems that * is doing
> > nothing with it?? Only after 10-30 seconds sometimes even much longer
> > the call file is picked up.
Check if the system times are in sync; if you copy a file with sam
Hey guys!
I'm the past week and a half, I have really learned a lot from the mailing
list and the wiki's posted online.
Now I have a question regarding different ways I can setup my asterisk
server for a small business with 12 extensions in the office. Cost is a
great concern, so I know cheap anal
Hello, i am newly in asterisk 1.2.2 and I am use the FXO card. I have the same problems about the zap channels (analog) that never detect when a person answers the call.
How can I do?, please help me.
Thanks,
Yenfry
___
--Bandwidth and Colocation
Yes. It is always the same pri regardless port on the TE411P. If I disable the
hardware echo canceling with vpmsupport=0, they will all work. I called Digium
tech support the last hour. They ssh into my machine and can't find anything
wong with the configuration, and he suggest to remove the dau
Title: Messaggio
Hi there Tommaso,
You've probably already tried this, but a reset and
reboot often fixes certain flakey behaviours on our Snom 320s. We've had
to revert back to the 4.5 firmware because of some issues with the 5.0 and
later, mostly that additional calls come in when our re
I haven't received any messages after 3/27/06 and I have tried to resub
twice without any success?
I miss the flood of messages, and I have other stupid questions to ask
;~)
Marty
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk
On Apr 27, 2006, at 1:18 PM, Eric "ManxPower" Wieling wrote:
Steven Totaro wrote:
Try setting the timing to zero on both spans?
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
Digium cards only support 1 timing source per card.
That's irrelevant. His current configuration means that the secon
I have 3 PRIs from different providers, and the fourth port is to a
legacy Meridian system...
Is it always the same PRI that does not work regardless of which port
is used on your Digium card?
I don't know if it's significant, but the order of my zaptel.conf is
different than yours:
span=1,1,0,esf
Your Dial command must have both T and t in it to be able to transfer
both incoming and outgoing calls.
in features.conf, you can change # for blind transfer to ## -- this lets
you use # in banks and voicemails and other auto attendants.
Moj
Matt wrote:
Hi I'm a little confused here...
Kerry Garrison wrote:
I am ready to pull my hair out. I cannot seem to get the Polycoms to
read the time properly. Regardless of the server they are pointed to our
the offset, i am getting the correct time, but 24 hours ahead. So for
today it is showing Friday April 28 but with the correct time
I'm building an app that will do the following:
1. Force the caller to record their name.
2. Dial the party to call.
3. Play a short menu:
1 = Accept Call
2 = Decline Call, go to VM if available
3 = Accept Call forever, never ask again
4 = Decline Call
I am having a problem with createlink not wanting to be disabled in my agents.conf file. No matter what when an agent picks up the phone, it appends the filename. Is there something other than 'createlink=no' that I should be adding to my
agents.conf to prevent this?Thanks,Kyle Sexton
__
Steven Totaro wrote:
Try setting the timing to zero on both spans?
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
Digium cards only support 1 timing source per card.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
__
Steven Totaro wrote:
Open the console with verbose turned up. Make a test call and see where
it is hanging. That will isolate the problem.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo
Sent: Thursday, April
With so many vendors offering so many different bundles, packages and
various PBX systems, can anyone suggest either/or a well know, reliable
SOHO sized VoIP/PBX, based on Asterisk, as a software-only or
appliance bundle?
Basically I am looking for something economical (under $500.00), which
I cou
Hello,
Aheeva offers a complete Call Center solution including IVR.
I read in the wiki-info about a large scale deployment in Portugal (40 E1, I guess) made by an american company.
I am programming a php-ivr library, but I think it takes me some time.
DanielOn 4/27/06, Dovid Bender <[EMAIL PROT
Title: Messaggio
I have a preoblem
with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2
firmware. All have same settings in the advanced panel. On 2 phones when I
press the hold or transfer key nothing happens and * does not start the
musiconhold. In the The hold
I just tried it. Same problem, one of the two spans is not working. If I
load wct4xxp with vpmsupport=0, then both spans working.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Totaro
Sent: Thursday, April 27, 2006 1:23 PM
To: Asterisk Users Maili
Haven't see this posted yet but keep in mind the polycom does offsets in
seconds not in hours... I spent three days figuring that out...
tcpIpApp.sntp.gmtOffset="-18000" is the same as GMT -5
Sean
Aaron Daniel wrote:
Whoops... meant dhcp... Keep in mind that if you're using windows' dns
se
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Waldo Rubinstein
>
> Make sure dtmf-mode is set to rfc2833 in both sip.conf as
> well as in the GXP-2000.
OK, thanks, but problem is more general: often phone doesn't send even a
packet to the Ast
I have several Polycoms (301, 501, 601) all working fine -- some with
sip ver 1.5.2, others recently upgrade to 1.6.5. The only problem
I've had was incorrect time for the first few minutes when the phones
boot-up, but that's been fixed by Polycom in newer versions.
The section of sip.cfg in 1.
Thanks - rfc2833 setting for DTMF in the in the phone GFUI did the trick.
- Original Message -
From: "Waldo Rubinstein" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, April 27, 2006 10:50 AM
Subject: Re: [Asterisk-Users] GrandStream GX
I've got dns on the brain... pretend I said DHCP anywhere that I really
said DNS...
On Thu, 27 Apr 2006, Aaron Daniel wrote:
Whoops... meant dhcp... Keep in mind that if you're using windows' dns
server, it doesn't allow negative offsets, but the linux one does. That was
a pain for us as wel
Selling analog GSM gateways at 119.00 USD. Polarity reverse
for "call answer" and "call end". Compatible with all Digium/Asterisk analog FXO
cards. Suitable for low-traffic call termination, inbound calls from cellular
network or for phone bill optimization applications.
Main Functions
Grandstream setting needs to be the 3rd radio button (cannot remember
the label)
Asterisk needs to be the rfc2833 This is the standard setup for
Grandstream phones to work with Asterisk.
--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database
Open the console with verbose turned up. Make a test call and see where
it is hanging. That will isolate the problem.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo
> Sent: Thursday, April 27, 2006 11:16
Try setting the timing to zero on both spans?
> > span=1,0,0,esf,b8zs
> > span=2,0,0,esf,b8zs
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of C F
> Sent: Thursday, April 27, 2006 12:29 PM
> To: Asterisk Users Mailing List - N
" Repeat these definitions for each provider then add the corresponding
lineX_ parameters as needed."
Hi Steve
Can you please clarify what you mean by LineX_ parameters.
Thanks
Dan
On 26/04/06, Steve Blair <[EMAIL PROTECTED]> wrote:
>
>
> Joe Greco wrote:
>
> >>Hello all,
> >>
> >>
Steve,
Im new to the asterisk mailing list and new to the asterisk
scene in general so excuse me if this is a stupid question but, I found a post
in the mailing list archive where you stated that you have multiple Quintum
Tenor’s configured with Asterisk. Im having some trouble getting
Whoops... meant dhcp... Keep in mind that if you're using windows' dns
server, it doesn't allow negative offsets, but the linux one does. That
was a pain for us as well.
On Thu, 27 Apr 2006, Aaron Daniel wrote:
What dns server are you running?
On Thu, 27 Apr 2006, Kerry Garrison wrote:
I
DNS is Windows 2003
Using the NTP server from CentOS 4.3
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Aaron Daniel
> Sent: Thursday, April 27, 2006 9:36 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asteris
Polycom 501
Firmware: 1.6.2.0041
Bootrom: 3.1.0.0269
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Matt Florell
> Sent: Thursday, April 27, 2006 9:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-
Hi,
I'm trying to receive faxes with asterisk. Everything works fine except
the tif to pdf conversion. Even though the tif file is okay, the pdf
always turns out to be empty (blank)..
I read that this might be caused by incompatible libtiff and that I
should install another version.
But when try
Make sure dtmf-mode is set to rfc2833 in both sip.conf as well as in
the GXP-2000.
- Waldo
On Apr 27, 2006, at 12:28 PM, dataman wrote:
We are having trouble getting the GrandStream GXP-2000 (1.0.2.13)
to work with the Asterisk (1.2.6) voice mail prompts. We access
voice mail but extens
On 10:28, Thu 27 Apr 06, dataman wrote:
> We are having trouble getting the GrandStream GXP-2000 (1.0.2.13) to work
> with the Asterisk (1.2.6) voice mail prompts. We access voice mail but
> extension and password dial tones are not accepted. An the voice mail times
> out. Dial tones work fin
Kerry Garrison wrote:
I
am ready to pull my hair out. I cannot seem to get the Polycoms to read
the time properly. Regardless of the server they are pointed to our the
offset, i am getting the correct time, but 24 hours ahead. So for today
it is showing Friday April 28 but with the cor
What dns server are you running?
On Thu, 27 Apr 2006, Kerry Garrison wrote:
I am ready to pull my hair out. I cannot seem to get the Polycoms to read
the time properly. Regardless of the server they are pointed to our the
offset, i am getting the correct time, but 24 hours ahead. So for today i
Hello Kerry, you has some NTP server installed in its system? Which distribution uses? I wait to have helped.
GreetingsJosué
2006/4/27, Kerry Garrison <[EMAIL PROTECTED]>:
I am ready to pull my hair out. I cannot seem to get the Polycoms to read the time properly. Regardless of the server they ar
What Polycom phone model?
What firmware version?
What bootROM version?
Older versions of Polycom phones only worked with SNTP time servers not NTP.
MATT---
On 4/27/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
>
> I am ready to pull my hair out. I cannot seem to get the Polycoms to read
> the
We are having trouble getting the GrandStream GXP-2000 (1.0.2.13) to work
with the Asterisk (1.2.6) voice mail prompts. We access voice mail but
extension and password dial tones are not accepted. An the voice mail times
out. Dial tones work fine with voice menus and dialing but not voice mai
You should really take this up with Digium support, and don't forget
to share your experience.
On 4/27/06, Wai Wu <[EMAIL PROTECTED]> wrote:
>
> My TE411p does not seem to like to have two PRIs from different telcos
> (span 1 and span 2). I can get one working, but not the other. However,
> if I u
On Apr 27, 2006, at 11:18 AM, Wai Wu wrote:
My TE411p does not seem to like to have two PRIs from different telcos
(span 1 and span 2). I can get one working, but not the other. However,
if I use vpmsupport=0 when loading the wct4xxp module, they both work.
But here is the problem, vpmsupport=0
Benchev wrote:
Hi Ronald,
Small mistake, see bellow:
Benchev
Just to give you an idea
I would suggest you to make two .agi files:
astcc.agi and astcc-disa.agi
In astcc.agi you'd leave everithing as it is, and enable
PIN =YES through the astcc-admin.cgi.
Thus you could dial without interogati
Hi Benjamin,
How do you setup early media in asterisk ?
Harry
--- Benjamin Lawetz <[EMAIL PROTECTED]> a écrit :
> Hello all,
>
> I've been playing around with early audio, and I'm
> able to get some things
> working
>
> We have PSTN calls coming in to asterisk in SIP from
> a Cisco AS5300. If
I am ready to pull
my hair out. I cannot seem to get the Polycoms to read the time properly.
Regardless of the server they are pointed to our the offset, i am getting the
correct time, but 24 hours ahead. So for today it is showing Friday April 28 but
with the correct time. Any clues?
Kerry
My TE411p does not seem to like to have two PRIs from different telcos
(span 1 and span 2). I can get one working, but not the other. However,
if I use vpmsupport=0 when loading the wct4xxp module, they both work.
But here is the problem, vpmsupport=0 disables the on board echo
cancellation. Any
> I actually tried that before but it didnt seem to work. I tried once again
> and still nothing rings, whether I set the destination to a single
> extension, or a ring group. But the suggestion from another user below did
> work, but wont go to voicemail yet when its not answered.
>
>
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> [from-ps
On Thu, 27 Apr 2006, Klaus Darilion wrote:
> Hi Armin!
>
> Armin Schindler wrote:
> > I'm not aware of such a cable to buy. Normaly, when you create a NT-side
> > the connection is not made with just one cable (like I did because both
> > device are just 10cm away from each other). In most cases y
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Hi,
Is it possible to get the callergroup or pickupgroup of a phone in the
dialplan? So I can make decisions depending on the caller/pickupgroup.
chris...
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Could you kindly let us know what numbers those survey-calls will be
coming from, so we can all add them to our blacklists? Thanks!
TV JOE wrote:
I write perl applications for a living and have developed code to
talk to all kinds of hardware. What I'd like to do is pull a list of
phone numb
Hi Eric,
this is my zapata.conf (zap/1 is a FXS but not used during tests):
;-
; Channel: zap/2 [in] - Telecom (lasciare libera)
;-
language = us
musiconhold = default
signalling = fxs_ks
channel => 2
I actually tried
that before but it didnt seem to work. I tried once again and still nothing
rings, whether I set the destination to a single extension, or a ring group. But
the suggestion from another user below did work, but wont go to voicemail yet
when its not answered.
[from-pstn]
in
Joe Pukepail wrote:
I have a question, we have some locations were I'm just planning on
putting in a PRI, management also wants analog lines incase the PRI is
down and someone calls 911. Is there a way to use asterisk to seize a
phone line from the fax machine?
I don't want to have to have
Hi Armin!
Armin Schindler wrote:
I'm not aware of such a cable to buy. Normaly, when you create a NT-side the
connection is not made with just one cable (like I did because both device
are just 10cm away from each other). In most cases you have an ISDN bus
cabled in the rooms where the necessa
I have a Dell PE SC420 (a no-no with a TE110P) connected to a Mitel SC-200. The Mitel gets Slip and Frame errors that cause the T1 card in the Mitel to go offline and this causes a service interruption. Could the SC-420/TE110P be causing these errors? I know it is listed on the incompatibility list
On 4/27/06, Rich Adamson <[EMAIL PROTECTED]> wrote:
Joe Pukepail wrote:> I have a question, we have some locations were I'm just planning on> putting in a PRI, management also wants analog lines incase the PRI is
> down and someone calls 911. Is there a way to use asterisk to seize a> phone line
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