Hi,
We are the American distributors for these phones. The link is below.
http://ipphone.eezeephone.com
AT723 is discontinued. A 2 Port ATA and a 4 Port ATA is on the cards very
soon.
Seshu Kanuri
Netweb Group, Inc.
Ph:1-732-387-4133
[EMAIL PROTECTED]
www.netwebgroup.com
This e-mail message
On Wed, 2004-09-22 at 17:13, duncan hall wrote:
Hi,
I am currently trying to find a replacement for a dinosaur PBX and want
to replace it with a VoIP solution.
We have just moved our lines over to an Optus Multiline from a Telstra
ISDN Onramp 30 service with 100 lines.
My question
Sounds like you'll need a TE410p (Austel approved) or an E100p (non Austel
approved). Which provide 4 or 1 E1/T1 interfaces respectively. Depending
on your number of internal extensions and need for call queues etc one
server running Asterisk could handle everything. We currently have an
On late august, there was a thread about
setting up some meetme conferences to
be able to follow Astricon remotely.
This indeed could be nice for those
that can't attend for various reason.
And of course is a demonstration of
Asterisk capabilities... :)
(Astricon without a remote conference
for
What is the type/model of the Adtran box?
I was under the impression the Optus Multinet network (of which
MultiLine is a product of) used 2 boxes onsite. One an SHDSL NTU and
the other a voice router. That is, unless things have changed since I
left.
The service has definately been completely
hi
if someone cares to add the solution to the bug tracker, others may
find it...
bkw's suggestion about enabling SIP_USERS worked perfectly
thanks
roy
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Matteo Brancaleoni wrote:
so, bring on and demostrate to the world
what asterisk can do!
I have packed all the necessary gear to stream the Developers Meeting on
Friday. I am looking for people with Big Pipe(tm) to get crazy and link
multiple Asterisk MeetMe's together. Lets see how much Pipe
Hi
I'm setting up a SIP gateway, serving quite a few potential users, and
I wonder if I should purchase a Opteron or Xeon based system. Xeon has
it's HT, but is it worth it? Has anyone tested Opteron on asterisk?
Does it work well?
There'll be no transcoding in this system - G.711A all way
Perhaps OT, but how I can apply this patch to an existing intalled * ?
Bye
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Helo all,
Some times meeting my asterisk with the message: The previous reload command
didn't finish yet. Therefore, it loses the communication with the
telephones.
What necessary to make to decide this problem?
Thank you
JMoura
___
Asterisk-Users
Hi,
I needed to create config files for downloading to Grandstream devices and made a
little script for it. It seems to work fine for the HT486.
The script needs a config file (cfg.in) which is in this format:
P2 = blah
P10 = hrm
...etc...
The configfile may contain comments (starting with '#')
On 05:36 Wed 22 Sep , Jeremy McNamara wrote:
Matteo Brancaleoni wrote:
so, bring on and demostrate to the world
what asterisk can do!
I have packed all the necessary gear to stream the Developers Meeting on
Friday. I am looking for people with Big Pipe(tm) to get crazy and link
hi,
i've got big problems with the Echo Cancellation of Asterisk. Entering zap show
channel 1 (while telephoning / making a call) gets me this line:
Echo Cancellation: 128 taps, currently OFF
And there's a heavy echo noticeable.
When I'am called, Echo Cancellation is enabled, and there's no
Hi,
I have the following config, which I can elaborate on if
necessary:
TDM400P REV E/F
X100P (X101P)
PII-450
Linux version 2.6.8-gentoo-r3
gcc version 3.3.4 20040623
Asterisk CVS-HEAD-09/05/04-09:28:57
Last night I had called into this system
Hi!
When I call a colleague of mine from my Cisco (via Asterisk), they get
on their display:
From Evert
asterisk
How do I remove/change the 'asterisk' part?
Regards,
Evert
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On Wed, 22 Sep 2004 14:06:51 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
Hi!
When I call a colleague of mine from my Cisco (via Asterisk), they get
on their display:
From Evert
asterisk
How do I remove/change the 'asterisk' part?
Regards,
Evert
You need to set a valid caller
Hello,
On Mon, 20 Sep 2004 16:04:39 +0200 (CEST), [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Yeah that works perfectly. Now trying to merge that with the
configurable transfer button patch :)
Should Supervised call transfer not already be possible without
patching asterisk?
Its
I am trying to set up VOIP link between India and US using *. I talked to 2 small
carrier and they both suggested me to go with a very expansive ($15,000 a month)
dedicated point to point solution which i can not effort. They said that this is the
only way to guarantee the quality of service.
/me too
this morning was all okie, now I can't connect.
I have an asterisk server ready for replicate the conference
here in .it, as soon as the link will be up with someone,
I'll post the IAX2 url
Matteo.
--
Matteo Brancaleoni
System Administrator
I moved the phone to the same subnet as the * server and I got a bit further
as you indicate is the way it needs to be for now. It's giving me a #3
registration error.
Could still use a couple of pointers on the uniden*.txt files as to what
they really need in there. I still have something
On 14:48 Wed 22 Sep , Matteo Brancaleoni wrote:
/me too
this morning was all okie, now I can't connect.
I have an asterisk server ready for replicate the conference
here in .it, as soon as the link will be up with someone,
I'll post the IAX2 url
How can I set it up to replicate,so I
Good fill in local time of day
I'm looking for a piece of hardware that we can place in two offices
that have decent bandwidth, but are in two different US states.
There are phone systems on both sides, that have extra CO analog line
ports that I'd like to connect through. One side has an IVR,
What type of phone systems do you have in either office? I have done
different applications for my customers in the past that wanted the same
type of service. Basically if you are just using co ports this would be a
tie line service. There may be a better solution though.
-Original
Hi Mike,
Several companies including MCI can set up a frame or Internet VPN the cost
is a little more than internet connection but I believe they will guarantee
QOS.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ofer Dagan
Sent: Wednesday, September
Henry Devito wrote:
What type of phone systems do you have in either office? I have done
different applications for my customers in the past that wanted the same
type of service. Basically if you are just using co ports this would be a
tie line service. There may be a better solution though.
1) it could be the x100p. Have you tried merely disconnecting the phone
line and plugging it back in?
2) it could be the phone line connected to the x100p. A red alarm is an
indicator of the co talk battery is missing on the line jack.
Lyle
- Original Message -
From: Mark C. Thomas
Just wanted to say Thanks to the Asterisk community -- all links are
bookmarked now!
Jeb Campbell
[EMAIL PROTECTED]
On Sep 21, 2004, at 4:54 PM, John Hill wrote:
Her is the 7905-12 page
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
-Original Message-
From: [EMAIL PROTECTED]
I'm presently using meetme extensively on my server. I have a rather
strange question. I'm using it with one person in talk-only mode and
everybody else in monitor mode. I'm running an athlon xp 2800 with
1gb of ram. I can handle 40 users. Does anybody know of any
adjustments that could
I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It
installed pretty easily and has worked great so I went to order some more
of these units today.
When I logged into Tech Data this morning, the PAP2-NA was now marked as
discontinued and no longer available and only the
(careful of line wrap...)
http://www-1.ibm.com/press/PressServletForm.wss?MenuChoice=pressreleasesTemplateName=ShowPressReleaseTemplateSelectString=t1.docunid=7293TableName=DataheadApplicationClassSESSIONKEY=anyWindowTitle=Press+ReleaseSTATUS=publish
Now THIS would be a nice adddition to *...
Hi,
Has anybody had any problems getting digium hardware lately?
Regards
Greg Cirino
___
Cirelle Enterprises Inc.
603-425-2221
www.cirelle.com Website Design
www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster
www.cedata.com Web, FTP, Email Hosting
Hi All,
I am look for recommendations for a good SIP phone, specifically with a good speaker phone. I have tried the SNOM 100 and the speaker phone quality is quite poor. Can any one share there experiences with this.
Much Appreciated,
Phil
___
I have an Asterisk system running on T1 PRI trunks using a TE405P. It
seems to be running ok, but one thing puzzles me.
Every so often I get a raft of messages like this:
-- B-channel 0/1 successfully restarted on span 1
-- B-channel 0/2 successfully restarted on span 1
...
-- B-channel 0/22
Hey all,
Wondering if this is possible.. Incoming call is
answered through X100P, then an extension is dialed
using the same X100P card. Basically I want to dial
in, enter 9 + phone# and have it do a flash then
have it dial *08 the same phone number + # on the
same PSTN line to have it transfer
Lyle Giese wrote:
Could still use a couple of pointers on the uniden*.txt files as to what
they really need in there. I still have something wrong in there.
I'll send you my config files in a separate email.
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[EMAIL
On Wed, 22 Sep 2004 08:21:50 -0500, Henry Devito [EMAIL PROTECTED] wrote:
Hi Mike,
Several companies including MCI can set up a frame or Internet VPN the cost
is a little more than internet connection but I believe they will guarantee
QOS.
depends where you are in india. If you are in
Every so often I get a raft of messages like this:
-- B-channel 0/1 successfully restarted on span 1
-- B-channel 0/2 successfully restarted on span 1
As far as I know, this is expected behavior. There is code to reset inactive B-channels periodically. I think the default is once an hour. I
Private Leased Line between India and US are expensive :-(
If you took a decent internet connection in India, around 1Mbps, you
will be paying $1500 to $2000/month there. That is still a lot cheaper
than private leased line. If your call center is in Mumbai (Bombay) the
delays are not bad.
One
On Wed, 2004-09-22 at 10:18, Tony Mountifield wrote:
I have an Asterisk system running on T1 PRI trunks using a TE405P. It
seems to be running ok, but one thing puzzles me.
Every so often I get a raft of messages like this:
-- B-channel 0/1 successfully restarted on span 1
-- B-channel
Cisco 7940 :)
- Original Message -
From: Phil Siegrist [EMAIL PROTECTED]
Date: Wed, 22 Sep 2004 10:15:57 -0400
Subject: [Asterisk-Users] SIP Phone
To: [EMAIL PROTECTED]
Hi All,
I am look for recommendations for a good SIP phone, specifically with
a good speaker phone. I have tried
I haven't tried disconnecting the phone line, I'll try that
next time.
If it was a co problem, I wouldn't think reloading the
wcfxo module would have fixed it - or would it?
Thanks for the info...
--- Lyle Giese [EMAIL PROTECTED] wrote:
1) it could be the x100p. Have you tried merely
Do you have a price range?
I use Polycom IP500s and the speaker phone is awesome. It picks up
speakers in the room very well at 5-6 feet.
Polycom has always made an exceptional speaker phone even on plain ole
phones.
Their implementation on the IP phones is excellent so they are my
preference.
I
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Mark C. Thomas
Subject: Re: [Asterisk-Users] Red Alarm on X100P
I haven't tried disconnecting the phone line, I'll try that
next time.
If it was a co problem, I wouldn't think
On Wed, 22 Sep 2004 16:40:04 +0200, Michael Bielicki [EMAIL PROTECTED] wrote:
Cisco 7940 :)
I'll concur with that.
The Cisco 7940 and 7960 phones have great speakerphones :)
As for ones to stay away from - the Grandstream BT-100 series. The
sound is fine on the local end, but is very low for
Anyone know where we could get a cheap free maybe would be nice sip
phone... We've been playing with an Innomedia MGCP and SIP adapters and
failing - so thinking that testing with a real phone might be good..
Robert A. Huddleston, KF4BYY
IT Support Analyst
Cavalier Telephone LLC.
(Cell)
On Wed, 22 Sep 2004 07:56:48 -0700, Wiley E. Siler [EMAIL PROTECTED] wrote:
Do you have a price range?
I don't know about pricing in the US, so I'll skip this (I buy mine in
Australia).
I use Polycom IP500s and the speaker phone is awesome. It picks up
speakers in the room very well at 5-6
Matteo Brancaleoni wrote:
/me too
this morning was all okie, now I can't connect.
I am in the conference without any trouble. There is nobody in it, so
it is quiet.
Jeremy McNamara
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[EMAIL PROTECTED]
Yes, I have placed two orders for TDM04B cards and I ran in to issues
both times.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Cirelle Enterprises
Sent: Wednesday, September 22, 2004 10:14 AM
To: [EMAIL PROTECTED]
Subject:
3) It could be the motherboard. We're on the cheap here and used
available components to make our server...worked fine with two x100's,
then the boss wanted another line. Once I got the damn thing to accept
them all on different IRQs, two would red alarm nearly every night at
random times.
The free phones I have heard of are soft phones...
X-Lite is excellent...
Cheap phones to test with, Grandstream is cheap but like the man said,
you get what you pay for.
eBay is a good source for cheap phones to test with but cheap is
relative
I consider cheap as sub $100.
You can pick
Is there an IAXtel 1-700 number by chance that can get me in to the
conference?
On Wed, 2004-09-22 at 05:36 -0400, Jeremy McNamara wrote:
Matteo Brancaleoni wrote:
so, bring on and demostrate to the world
what asterisk can do!
I have packed all the necessary gear to stream the
The problem is some calls from the PSTN have hidden caller id so if you want to change
it to something else then modify chan_sip.c
#define CALLERID_UNKNOWNAsterisk
I've changed mine to:
#define CALLERID_UNKNOWNUnknown
-Original Message-
From: Shaun Ewing
--- Brent Franks [EMAIL PROTECTED] wrote:
Hi Mark,
We had a similar issue about a year ago. We eventually
figured out that
our Promise Array (ATA RAID) Controller cards were
causing the X100P
card to go into RedAlarm. It would come out on its own
sometimes,
others not. What other
Mike Benoit wrote:
Is there an IAXtel 1-700 number by chance that can get me in to the
conference?
Then to truly experience Asterisk's power, join the same conference...
...Using SIP: [EMAIL PROTECTED]
I can't make the sip address work either, at least not with the fwd
communicator beta.
If
Hi all,
I have a person trying to sell me Cisco 7910 IP Phones. Does anyone know if SIP is supported on these
phones? I have CCO login also so if they
do support SIP does anyone know where I could download the software?
Thanks in advance.
The 7910 does not support SIP. It is SCCP only.
-Shaun
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
Date: Wed, 22 Sep 2004 10:44:02 -0500
Subject: [Asterisk-Users] Cisco IP phone
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Hi all,
Title: Message
Hi
all,
I have looked
through the wiki guides and also Siemens user guides but they haven't proven
useful. Nor has the normally trusty googling. Also have upgraded to
the latest Optipoint 400 Standard SIPfirmware.
Having read a few
previous threads on the Optipoint it
On Tue, 21 Sep 2004 23:50:57 -0400, Kenton Powell [EMAIL PROTECTED] wrote:
I am trying to setup the Asterisk server
and softphone on the same machine with the hope of extending extensions
to other computers or phones after I test the configuration.
OK. Now, it is not impossible to run both the
We are using 2 X-Lite Soft Clients, and 1 Avaya 4602 IP
phone, the Asterisk server has an analog port card in it. The Avaya can
support up to 2 call appearances. When I call the Avaya from an external phone
or another SIP client, it works fine, I then hang up. The second time also.
But
There seems to be a bug in app_queue that prevents
calls from reaching agents. If a call is directed to an agent, and that agent
transfers the call using the transfer facility on Cisco phones (SIP firmware
7.2) then show agents shows the call still at that same agent. This prevents
calls
When I logged into Tech Data this morning, the PAP2-NA was
now marked as discontinued and no longer available and only
the PAP2 version was available which is the Vonage branded version. :(
I saw someone on the list say that they heard from Cisco that
these units were not due out until
Shite, I ordered some a few days ago from TD and they have my order on hold.
Gary
I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It
installed pretty easily and has worked great so I went to order some more
of these units today.
When I logged into Tech Data this morning,
Does anyone have * working with full MS SQL
support (CDR, VM)?
--
Richard Cook
[EMAIL PROTECTED]
Tel: 705-497-9320- ext
2010
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
In article [EMAIL PROTECTED],
Tony Mountifield [EMAIL PROTECTED] wrote:
I have an Asterisk system running on T1 PRI trunks using a TE405P. It
seems to be running ok, but one thing puzzles me.
Every so often I get a raft of messages like this:
-- B-channel 0/1 successfully restarted on span
This is about big business. No ILEC is going to just sit idle and watch
billions in revenue go out the window. It will be interesting to see if port
blocking ever becomes an issue. Did I buy my Internet service with out
without restrictions? H. Cisco sells to Telco's and Cable guys. Vonage
has
This really chaps my hide. The situation as it's been explained to me
is: Apparently, too many *consumers* were accidentally buying the
PAP2-NA (unlocked) version and then complaining/returning them to
Linksys b/c they didn't understand that they need a service provider to
be able to
Hello all,
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for
this card.
M CAPI2.0 support
[*] Verbose reason code reporting (Kernel size +=7K)
[*] CAPI2.0 Middleware support (EXPERIMENTAL)
M CAPI2.0 /dev/capi support
[*] CAPI2.0 filesystem support
M CAPI2.0
This really chaps my hide. The situation as it's been explained to me
is: Apparently, too many *consumers* were accidentally buying the
PAP2-NA (unlocked) version and then complaining/returning them to
Linksys b/c they didn't understand that they need a service provider to
be able to place and
Brandon Patterson (peering) wrote:
This is about big business. No ILEC is going to just sit idle and watch
billions in revenue go out the window. It will be interesting to see if port
blocking ever becomes an issue. Did I buy my Internet service with out
without restrictions? H. Cisco sells to
Hello everyone. I am trying to do a cvs update. I do the make update; make
upgrade and this is the error that I am getting.
make[1]: Entering directory `/usr/src/asterisk-cvs/asterisk/channels'
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
Ryan Wilkins
On Wed, 22 Sep 2004, Brandon Patterson (peering) wrote:
This is about big
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
Hello all,
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for this card.
M CAPI2.0 support
[*] Verbose reason code reporting (Kernel size +=7K)
[*] CAPI2.0 Middleware support (EXPERIMENTAL)
Ok, an italian link to nufone astricon conf room
is up running.
Connect it to:
IAX2/[EMAIL PROTECTED]/meetme
OR
IAX2/[EMAIL PROTECTED]/meetmeq
The first one is to listen speak.
The second one is to listen only, use that
if you wanna listen, perhaps with a speakerphone,
in order to not send
On Wed, Sep 22, 2004 at 07:15:19PM +0200, [EMAIL PROTECTED] wrote:
Hello all,
Sorry for my first mail which answers the 2nd part:(
I'm trying to setup a AVM C2 card.
I have read the kernel requirements for this card.
M CAPI2.0 support
[*] Verbose reason code reporting (Kernel size
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
Thats the first thing I'm going to try when we get our units. I'll
IAX2/[EMAIL PROTECTED]
you can connect gsm/g726/alaw or ilbc
server is in NL
On Wed, 22 Sep 2004 19:56:28 +0200, Brancaleoni Matteo
[EMAIL PROTECTED] wrote:
Ok, an italian link to nufone astricon conf room
is up running.
Connect it to:
IAX2/[EMAIL PROTECTED]/meetme
OR
Hi All,
I am testing out Asterisk with IAX between 2 machines on local
IP addresses and I want one machine to act as an IAX gateway with the
other connecting to it. Anyone know of or can supply me an example of
how to do this?
Cheers,
Dee
___
Anyone confirmed a stocking vendor we can purchase these from?
Gary
Ryan Wilkins wrote:
This begs the question, again, that someone else posted originally.. what
about loading SPA-2000 or PAP2-NA firmware in the PAP2? If it's the same
hardware, there shouldn't be any reason not to try it.
Try app_conference. In this configuration, you should be able to
handle 200++ users without problems. It's ideal for this kind of
thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
-SteveK
On Sep 21, 2004, at 11:13 PM, Darren Wiebe wrote:
I'm presently using meetme extensively on
We are supposed to have some of these on the way. Barring some unforeseen
cicumstance we should have them early next week. E-Mail me off-line if you
want me to set one aside for you.
Michael Crown
Managing Partner
The VoIP Connection
-Original Message-
From: Gary Carr [mailto:[EMAIL
--On Wednesday, September 22, 2004 14:06 -0400 Steve Kann
[EMAIL PROTECTED] wrote:
Try app_conference. In this configuration, you should be able to handle
200++ users without problems. It's ideal for this kind of thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
Is there a link/WIKI
steve how stable is that ?
On Wed, 22 Sep 2004 14:06:29 -0400, Steve Kann [EMAIL PROTECTED] wrote:
Try app_conference. In this configuration, you should be able to
handle 200++ users without problems. It's ideal for this kind of
thing.
(it's located in iaxclient CVS at iaxclient.sf.net).
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no [EMAIL PROTECTED] for ME :-(
John Millican
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Stay away from the 7910 if your going SIP. It will not support it.
Lethol
On Thu, 23 Sep 2004 01:45:23 +1000, Shaun Ewing [EMAIL PROTECTED] wrote:
The 7910 does not support SIP. It is SCCP only.
-Shaun
- Original Message -
From: Henry Devito [EMAIL PROTECTED]
Date: Wed,
Folks!
Our Phones are cheap and they are selling well. We have no complaints so
far. These phones are made by ATCOM, 2nd largest maker of VOIP gear in
China. We are ATCOM's US distributors.
We want to beat grandstream both at features and price.
I can sell these industry standard PA1688 Chip
You have some contact info for Bottom Line Tech? We are a ISP/CLEC and want
to order some of these.
Gary
Eric,
I was told by Bottom Line Tech that Linksys told them to pull all units
and
stop all shipments unless there customer could prove they were and ISP,
which i am not so i can not, so no
There is an asterisk-biz list for this type of post.
Asterisk-users is the non-commercial forum.
Thanks!
Scott Stingel
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
SeshKanuri wrote:
We want to beat grandstream both at features and price.
I can sell these industry standard PA1688 Chip enabled phones with IAX2, yes
I said IAX2 (along with SIP, H323 and MGCP and a few more such protocols
already enabled) at bulk rates to anyone interested in them.
So does that
Here is the contact info For Bottom Line Tech
Bottom Line Telecommunications
www.shopblt.com
457 Route 164
Preston, CT 06365-8111
(860) 886-1011 / (561) 791-3308
John
I would love to have contact info for Bottom Line Tech also.
Then we do not have to go with all the trouble getting to them.
- Original Message -
From: Gary Carr [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Wednesday, September
Well just did a search on bottom line and they do not have the PAP2-NA
listed anymore. They may still have them in stock if you call them though.
Sorry
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Millican
Sent: Wednesday, September 22, 2004
Can I contact you off-list?
Please provide email address.
Yiannis Costopoulos.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of SeshKanuri
Sent: 22 September 2004 22:41
To: Asterisk Users Mailing List - Non-Commercial Discussion; 'Shaun
Ewing'
Shawn Kelley [EMAIL PROTECTED] writes:
I am preparing to setup a system using Cisco 7940 and 7960's I have the
7.1 SIP firmware on them.
One issue I have run into is how to silence the ringer if a call comes
in and you don't want to take it.
Many phones have a DND button. I know the 79XX has
This is the guy that i talked to and he seemed helpfull
David Durel
Bottom Line Telecommunications
http://www.shopblt.com/
[EMAIL PROTECTED]
Voice / FAX: (860) 886-1011
Monday - Thursday, 9:00 - 6:00 Eastern Time
as i said before i searched the site and the PAP2-NA is no longer listed.
May be a
Michael Bielicki wrote:
IAX2/[EMAIL PROTECTED]
you can connect gsm/g726/alaw or ilbc
server is in NL
Ok, an italian link to nufone astricon conf room
is up running.
Connect it to:
IAX2/[EMAIL PROTECTED]/meetme
OR
IAX2/[EMAIL PROTECTED]/meetmeq
The first one is to listen speak.
The second one
On Wed, 22 Sep 2004 10:13:58 -0500, Bob Klepfer [EMAIL PROTECTED] wrote:
3) It could be the motherboard. We're on the cheap here and used
available components to make our server...worked fine with two x100's,
then the boss wanted another line. Once I got the damn thing to accept
them all
Does anyone know which physical interrupt line out of the four on the PCI backplane
the TE405P uses? Or is it somehow configurable by hardware or software?
I'm trying to diagnose a problem where the card generates no interrupts in one
system, but is fine in another system. These systems are
On Saturday 18 September 2004 06:21 pm, Lyle Giese wrote:
Perfectly normal. On analog lines, the caller id is set between the 1st
and 2nd rings. So Asterisk has to wait for the caller id and depending on
the speed of the computer that hosts Asterisk, 13 seconds is exactly right.
A normal
On Sunday 19 September 2004 10:28 pm, C Wegrzyn wrote:
I ran the LiveCD version of Asterisk on my hardware and it worked. I am
trying to run it natively on a 2.6 kernel (Gentoo distro), but it keeps
getting a seg fault using the sample configuration files. Does Asterisk
not work with the
Since upgrading to
7.2, I've noticed a random problem where I dial a number and hear all the
correct tones in the handset, but the display won't show all the numbers I
dialed. So you sit there waiting for the dialplan to kick the call off
(b/c you heard the proper amount of tones played and
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