it into a fractional PRI and run the fa0/0 as the WAN or added
another T1 port and used that as the WAN.
Mark
--
Mark Willis
Star One Telecom
Office: 1-800-889-7001
Cell: 210 880 5097
http://staronetel.com
--
_
-- Bandwidth
by Fedex to the US.
Thanks for your interest and if you have any questions, email me at
m...@staronetel.com or call me directly on 210 880 5097.
Mark Willis
--
Mark Willis
Star One Telecom
Office: 1-800-889-7001
Cell: 210 880 5097
http://staronetel.com
On 2011-02-13 15:21, Gilles wrote:
noload = codec_speex.c
Try noload = codec_speex.so
Mark
--
Mark Willis
Star One Telecom
Office: 1-800-889-7001
Cell: 210 880 5097
http://staronetel.com
--
_
-- Bandwidth and Colocation
On 2011-02-13 15:49, Gilles wrote:
On Sun, 13 Feb 2011 15:32:03 -0600, Mark Willis
marksli...@markwillis.net wrote:
Try noload = codec_speex.so
That dit it. However, I'm puzzled by the fact that the default
filenames in modules.conf all ended with .c instead of .so:
===
/etc
Is it possible to do SIP-Asterisk-TDM in a single step with FFA? Or
does FFA always use TIFF files?
I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102
ATA's at the fax machines and send faxes directly over a PRI.
Mark
--
Mark Willis
Star One Telecom
Office: 1-800-889-7001
On 2010-02-06 21:42, C F wrote:
For a case like this I would go with overhead paging.
On Fri, Feb 5, 2010 at 4:50 PM, Mark Willismarksli...@markwillis.net wrote:
Has anyone done any large scale intercom deployments with Asterisk? I've
been asked about building a system to one-way page
would be taking a single input stream
and exploding it out to 500 endpoints.
- There are 500 near-simultaneous INVITEs being sent. Can the SIP
channel handle that?
Any suggestions or war stories are appreciated.
Mark Willis
Cartama Consulting LLC
210 698 5097
I thought of that too, but the phones will be spread over a large number
of rooms in several buildings, so that won't be too much of an issue.
Mark Willis
On 2010-02-05 15:55, jon pounder wrote:
Mark Willis wrote:
This could potentially create a very weird audio situation where the
delay
Thanks everyone, I'll look at multicast. The customer prefers Snom
phones, luckily.
Mark
On 2010-02-05 16:32, Philipp von Klitzing wrote:
Hi!
Has anyone done any large scale intercom deployments with Asterisk?
I've been asked about building a system to one-way page 500 phones
On 2010-02-05 16:20, Jeff LaCoursiere wrote:
On Fri, 5 Feb 2010, Mark Willis wrote:
- My limited math capabilities suggest 41 Mbps of RTP traffic, which
seems like a lot, plus asterisk would be taking a single input stream
and exploding it out to 500 endpoints.
How did you get
Noah Miller wrote:
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
- Noah
So asterisk can't do SLA with Polycom phones?
mark
___
Noah Miller wrote:
I don't believe that Polycom's version of SLA does anything with
Asterisk. You have to use asterisk's SLA implementation
(http://www.asterisk.org/node/48342).
So asterisk can't do SLA with Polycom phones?
Asterisk can do SLA with Polycom, just not using
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
Mark Willis
___
-- Bandwidth
Has anyone done SLA with Polycom phones? I've got a large project coming
up where the customer is keen on SLA for trunks and extensions. Trunks
will be on a PRI.
We may do this with Cisco phones if they work better.
You really want to do SLA with all 23 lines of the PRI? That's a
Francisco Seratti wrote:
Hi pals, im trying to save some money
in
cellphones calls, so i bought a GSM gateway and a Sipura SPA3000
gateway.
The GSM gw is currently working, and now im trying to configure the
SPA, but every call i send, i get a 503 service unavailable.
It does that
Has anyone done any installations using the Xorcom TS-1 and multiple
T1's? I'm looking for a reliable box to put in a closet and route calls
between T1's. No voicemail, IVR, etc. Any other suggestions?
Mark Willis
Cartama
___
--Bandwidth
Adam Robins wrote:
This is definitely something that changed in the 1.07 to 1.24
upgrade. We have a pair of identical 1.07 servers connected via the
same network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes a
difference.
Suppose I had a SIP FXO that sent the 200 OK immediately after dialling,
and no way to do call progress. But I need to not send answer
supervision to my carriers until the call is really answered. Short of
detecting hello, hello the best way to handle it seems to be to delay
the answer
I tried to add money to my account last week, but the web page didn't
allow me to do so. If they have no money it's because they don't seem to
have a way for their customers to give them any.
Mark
Carlos Chavez wrote:
I have been using Sixtel from the beginning of the year and service
To answer my own question... the solution is to have both ends run the
same version.
Mark
Mark Willis wrote:
Two asterisk servers, one running a recent HEAD, the other 1.0.9. I
have both ends set up with trunk=yes, notransfer=yes, type=friend. I
notice that the trunking works from HEAD
Two asterisk servers, one running a recent HEAD, the other 1.0.9. I have
both ends set up with trunk=yes, notransfer=yes, type=friend. I notice
that the trunking works from HEAD to 1.0.9 only (the direction in which
calls are originated). I know this by bandwidth usage and by iax2 trunk
debug.
I think you're looking at this the wrong way. Take a look at automon in
features.conf. Play the for-quality-purposes disclaimer/misleader on
all incoming calls to these extensions and use the w option on Dial().
What you've done below won't record an existing call.
Mark
Swapna Gupta wrote:
You can do it in voicemail.conf like this:
100 = 1234,Mark,[EMAIL PROTECTED],,tz=central24
Mark
Max Clark wrote:
Hi all,
I am curious if it is possible to have multiple timezones registered
on an Asterisk server for Voicemail (i.e. so that PST users get PST
time, and EST users get EST
I have some lines outside the USA that I can't get to for up-close
testing. The issue is that when I make an outbound call on these lines,
I don't get answer indication until 20 seconds into the call. Asterisk
always thinks the line is answered, even if it's still ringing or busy.
On incoming
Need some help understanding codec preferences:
I have 2 asterisk servers.
Server 1 sends calls to the PSTN and has allow=g729 allow=gsm and
allow=ulaw in iax.conf
Server 2 receives calls and routes them to server 1. It has the same
allow lines.
We receive calls from a phone co and route them
Jacob Cazzell wrote:
Looking at alternative VoIP providers and I found Teliax. One of the
features listed on their pay-as-you-go plan is unlimited
incoming/outgoing connections.
I am working on setting up a conference calling system for some of our
traveling salepeople to call into for their
That works for me too, just not obvious. Pity Conference doesn't work the same
way.
And 1.0.5.18 fixed the Message button, thanks to those who suggested the
location.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Goddard
Sent: Friday,
I never could get attended transfer to work with the BT-100 on 1.0.5.16. Where
did you get 1.0.5.18? It's not anywhere obvious on Grandstream's web site.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Thursday, December 09, 2004
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