You need to either have a zap channel available or zap_dummy in place to
get this working... SIP setup only requires a zap channel for meetme...
K. Callis
On Thu, 2003-09-25 at 00:43, C. Johnson wrote:
Ok.. I got * and SIP working internally now .. still wrestling with
connecting two remote
] On Behalf Of
Kim C. Callis
Sent: Thursday, September 25, 2003 2:46 AM
To: Asterisk User Mailing List
Subject: Re: [Asterisk-Users] Meetme question
You need to either have a zap channel available or zap_dummy
in place to get this working... SIP setup only requires a zap
channel
I have been trying to send and receive calls through sipphone.com. Can
someone give me an example of sip.conf and extensions.conf to support
that?
Thanks
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The interface is quite nice... It would be nice to see it integrated
with the web configuration interface.
On Thu, 2003-09-25 at 19:20, Jamie Carl wrote:
Ok, wow. Didn't expect as many responses as I got.
Didn't think this would spark so much interest.
Anywayz. Some of the questions
On Wed, 2003-09-24 at 10:41, Mike Hjorleifsson wrote:
Has anyone successfully run asterisk with a VIA processor ?
I have tried unsucessfully, do I have to run make with any specific switches
?
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Yesterday, I started to experiment with Cisco to Cisco SIP calls using
the g729 codec. According to the documentation, both the ATA-186 and
7960 are able to make use of the g729.
From an earlier e-mail, I made a change to the configuration of the ATA,
changing the values:
LBRCodec:3
RxCodec: 3
I have come to realize that I don't have to have a g729a license in
order to make use of an ATA-186 or 7460 connecting to another 7460. I
just need to allow the codec in sip.conf.
Now what ramification does that have when I dial out over one of my
analog line (connected to * by a channelbank and
Has anyone put together a template if you will can be used to generate
feature codes for Asterisk. Even if some of the feature codes are
dialplan specific, if there was a template, it could be changed
accordingly. Besides the feature codes, there could also be something
for voicemail, etc.
Just a
their
technology, and would rather look at someone on the screen at a whopping
3 frames per a second, as opposed to talking in the meetroom... Go
figure!
Kim C. Callis
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to purchase SIP
phone or have to have the telephone that goes with the system, along
with some other telephone device on the desk.
Any answers, and I will sumerize...
Thanks,
Kim C. Callis
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http
I was reading on www.vovida.org/applications/downloads/G729A/ (home of
VOCAL) pages, and that there is a free license use for non-commercial
for G.729A. Is that usable under Asterisk or strictly a Vovida offering?
Kim C. Callis
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Asterisk-Users
was babbling about the wonders of Bayonne. The only thing that
was successful in that meeting about VOIP solutions was tabling that
discussion until a future (as in way, way in the future) date.
Just a thought!
Kim C. Callis
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I was finally playing around with the various databases,
blacklist and cidname. The format
for these are family name, key and value. Would that mean that the key
is the name to enter, and the value the number? For instance:
database put
blacklist Kim Callis 3235551212? And if that is in
I noticed in the source for app_meetme.c,
that there is an arg to allow someone to be an admin.
What exactly does that mean? What extra privileges do they gain? And
furthermore, how is one made an admin in a room?
Are there any other codecs that
can be used with the 7960 and the ATA-186? I have been using the default gsm codec and wanted to see if I could make use of
something a little less bandwidth intensive
Kim Callis
:[EMAIL PROTECTED] On Behalf Of Kim C. Callis
Sent: Tuesday, July 22, 2003 11:56 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Codecs
for use with Cisco 7960 and ATA-186
Are there any other codecs that can be used with the 7960
and the ATA-186? I have been using the default gsm codec and wanted
7960 and
ATA-186
Hi,
For local connection to Asterisk (LAN), G.711 is the best option.
BR,
Dan
- Original Message -
From: Kim C. Callis [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 23, 2003 10:13 AM
Subject: RE: [Asterisk-Users] Codecs for use with Cisco 7960
I was thinking of adding QoS to my
Linux based router. I thought I would add all my IP phones and my * box into a VLAN,
and then would do a QoS setup for that particular
VLAN. Has anyone did any QoS
setups for better performance? Has it made any change to the performance?
Kim C. Callis
Alright, I am basically cheap, and I have a cellular plan
which allows for free incoming calls (Nextel). I was wondering if there was any
way to do sort of a dialback trick in the extensions.conf I call into the system from my cell
phone (maybe via DISA), I dial an internal extension, and
with parking. As for transfers, I can do a flash
and get dial tone, but then what are my options at that point for transferring?
Any help would be greatly appreciated!
Kim C. Callis
, considering that I dont have anything
attached to the FXS.
So would that mean that the configuration is incorrect? Maybe
I have the channels incorrectly listed? The one thing that I cant seem
to do is to get any dialtone off the FXS card, at
least seemingly
Kim C. Callis
Has anyone attempted to use Efax
as a virtual fax? I know that * will answer and route to a fax device, but the
question is, would * answer the fax tone and then call my e-fax phone number
and then route the fax to that? And if so, how does one go about doing that?
Thanks,
Kim C
Steve,
What exactly would be classified as a digital ZAP device?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven Critchfield
Sent: Sunday, July 06, 2003 8:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Accurate
as well.
This is my view of how it would work, but if anyone has
another way to provide accountability that would work as well
Kim C. Callis
Actually, in many of the mailing lists that I have subscribed to, this
one is one of the friendliest. Nevertheless, I have been remiss in
reading your postings, so if you repost for me, I will do my best to
answer.
Kim C. Callis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
there is to create a database for billing and create a relationship that
will extract from the CDR database.
Kim C. Callis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Angelo
Sampietro
Sent: Wednesday, July 02, 2003 7:06 AM
To: Scott Stingel
Cc: [EMAIL
sits behind a NATd firewall, any ideas what might
cause the de-registration?
Kim C. Callis
If there is a way possible, would someone tell me how I can
setup a dial by name feature under vmail2?
Thanks,
Kim Callis
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