I have been searching for an alternative network provider, but I'm told
that they would all take the same route from the US into Canada, as
there is simply no major backbone running into NB east of Toronto.
I would have to agree with what I am hearing here - I would look at
alternative network
Can anyone recommend me ATA device that REALLY has T.38 built in.
While I have not tested it myself (one just arrive for me try out), I
have been told that the Mediatrix products have a working T38
implementation. Of course my suggestion would be check with the
provider tho you plan to use the
Hey Paul,
What kind of problem are you having? You need to post more information if you expect help.
The two ports on a 2002 will register and work independently (as if
they were 2 devices). You should have no problem calling between them
(or ringing them both etc). Check your dialplan and see
I have a UK Livevoip DID that is down, and has been for several days.
I'm looking to replace my London DID, low usage but need at least 2
channels and a local London number.
Please email me off list if you can provide this.
J
On 6/26/05, Darren Wiebe [EMAIL PROTECTED] wrote:
Is there
We have been having serious quality problems using the westcoast
server - been using the East coast server with increased success but
seeing some issues related to going cross continent.
Voipjet, you listening?
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Asterisk-Users mailing list
I ran into a similar problem with a CVS from around then... 04/18/05
and after getting advice from the list moved to 02/10/05 and have been
running stable since.
Cheers,
J
On 6/6/05, Umair Bari [EMAIL PROTECTED] wrote:
Dear All,
I am using Asterisk CVS-HEAD-05/05/05-15:46:09, and found it
Colin,
I'm not sure this helps the problem, if you want to try DIALs the
caller is still left hanging during each 30 or 40 second period.
As for timing the call, be careful of voicemail on busy as I think
you'll find that most cell phone voicemail will also answer if the
line is
Hello everyone,
I'm been toying with the idea of allowing my users to use meetme but
have had some service quality issues (which I know are being
addressed) but am concerned about making work for myself for something
I can outsource...
Junction Networks (http://www.junctionnetworks.com) seems
Signate offers an interesting product they call 'webcall', which
basically contacts a client at a number they provide then connects
that person to a sales staff. Some potential for abuse but a nice idea
for support etc.
I know that it is possible to do (obviously) and well documented but
has
It is actually a different animal because you're not using a softphone
etc at all, give it a try on the site to see what I mean.
http://signate.com/callme.php
It actually calls you on a pstn number the proceeds to connect you to
a staff member. This is why I mentioned the potential for abuse. It
Here is a working sample that I use for the same thing on my home
box... note that I use AreskiCC so that I can easily and nicely track
usage..
The SetAccount is used so that AreskiCC doesn't ask for the calling
card number and directly prompts me to dial but if anyone else calls
in it asks for a
Thanks, lots of insight and an improved yet more complex solution...
My initial thoughts were if I wanted to use it for a calling card type
environment was to simply dump the user into the calling card AGI
after the first leg of the call came up and let the AGI do what is
good at. This removes
UPDATE
As for my problem - at the suggestion Luki I backtracked to a previous CVS.
As I had some problems with a src sent to me I grabbed a CVS from the
same day (CVS-D2005.02.10.05.00.00) and haven't had a problem since
performing the change; nearly 24 hours.
I would love to provide more info
Hey Everyone,
I've been running a version of the CVS without issue until late last
week when suddenly Asterisk would randomly hit 99% CPU and stop
registering my DIDs.
If I stop Asterisk with a 'stop now' and restart Asterisk all is
well... for a bit.
So far I have deducted the following.
thanks for the help... I knew I missed some info...
Music on hold.. I am not using any form of it.
As for AGIs... I do have AreskiCC installed but it is used for only
some calls. I discounted it as being the culprit as the problem seems
to occur even when no one is connected and for sure when
Can you post your dialplan?
If you are using the extension 's' change it to the actual DID number.
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Sorry, I misread the second message. I have the following for SIP
based config that works fine... hope that helps.
register = uid:[EMAIL PROTECTED]
[livevoip]
type=user
secret=mysecret
deny=0.0.0.0/0.0.0.0
permit=217.160.244.186/255.255.255.0
context=default
notransfer=yes
disallow=all
Hello Mark,
I have been working on a similar plan but am still looking for
reasonable/tested hardware - can you tell me what devices you are
using?
Thanks,
Jonathon
On Apr 7, 2005 7:01 AM, Mark Dutton [EMAIL PROTECTED] wrote:
Hi there
I have a SIP ATA with a fax machine attached
With another provider I was having similar problems where numbers
would almost always be wrong. To fix this issue I had the provider set
dtmf to 'info' and everything has worked perfectly since.
I don't know if this is possible with Livevoip or IAX (I use SIP) but
I would suggest trying it.
On
On the ringback notion, I solved my ring back issue by moving my
connection to SIP where it seems to function fine.
On Thu, 31 Mar 2005 20:57:52 -0500, Moody [EMAIL PROTECTED] wrote:
Same here,
I have a 1-888 number via SIP and DTMF works fine.
I do not have a DTMF type specified
/notice.cgi?NoticeID=1471
Quote from above link: profiting from the unauthorized distribution
of our copyrighted works
It's all a little much for me...
On Thu, 31 Mar 2005 15:12:51 -0500, Lee Azzarello
[EMAIL PROTECTED] wrote:
On Sun, 2005-03-27 at 21:19 -0500, Moody wrote:
I don't want to move
Same here,
I have a 1-888 number via SIP and DTMF works fine.
I do not have a DTMF type specified but assume it is defaulting
rfc2833 (anyone confirm?)
Hope that helps,
J
On Thu, 31 Mar 2005 17:51:42 -0500, MF Hulber [EMAIL PROTECTED] wrote:
I don't have any difficulty with DTMF with
Hey Chris,
What type of phone are you using for testing? I found a big difference
when I switched from a cheap testset to a better phone. The only
problems I get with voipjet is when people talk over each other - but
I'm not sure how to fix that but everything else has been very good.
J
Looks interesting,
From the FAQ it looks like a 'metered' plugin for CDRs is coming but
not available yet. Is this out of date or am I missing something?
Of course you could just do the translation yourself from what I read...
On Wed, 30 Mar 2005 12:24:23 +0200, Klaus Darilion
[EMAIL
Not to feed the fire - but I would be interested to hear about open
source projects that arn't free. It's pretty hard to charge for
something you legally have to provide the source for...
Some try with subscriptions, but citing Sveasoft I'd suggest that
model is both unacceptable (IMHO) and
Although I'm not that familiar with it, I have heard good things about...
http://www.bsdmall.com/saadpcico.html
Don't know about hardware QOS on it tho... I'm assuming just shaping
via the host machine.
J
On Mon, 28 Mar 2005 08:26:38 -0500, steve szmidt [EMAIL PROTECTED] wrote:
On Sunday 27
I'm not sure, but even using CVS I'm having a hell of a time getting it going.
Can anyone tell me if the wiki has the correct information? I'm
getting the well documented but unsolved (publicly from what I can
tell) No such context 'macro-screen^... error.
It is listed in Mantis but I can't
I don't want to move this thread towards a discussion of Sveasoft, but
I would ask anyone considering this option to make sure they do some
reading about Sveasoft and their version of opensource before
sending them a check.
IMHO, Do the community a favor and check out OpenWRT (see earlier
posting
If you are just looking to start using them, voipjet provides a basic
sample config, including your login details, should be right next to
CDRs on the site menu. Itshould just be a cut and paste unless you
actually are trying to do something different...
J
On Sat, 26 Mar 2005 23:19:17 -0500, Jon
I've managed to get it to compile using patched .c but Asterisk 1.0.6
doesn't see to want to start, hanging at the following message.
I've googled and can't find anyone else with a similar problem, but
I'm guessing its a database error as thats what the other similar
messages seem to be a bout
Sounds very interesting, would providors be willing to insert pricing
or would you need to enter all the data?
I would suggest a set of rules like pricewatch.com uses to keep people honest.
Keep us informed,
Cheers,
Jonathon
On Thu, 17 Feb 2005 10:29:54 +, Alistair Cunningham
[EMAIL
I've been trying to resolve some quality issues and I was hoping
someone might be able to provide some insight.
To give you an idea the calls are coming in via a SIP DID and sent out
via an IAX2 connection. Latency to both the SIP equipment and IAX
equipment are around 80ms with 0 packet loss
Sounds like maybe the wrong DTMF setting ?
On Fri, 11 Feb 2005 10:57:38 -0500, Philip Siegrist [EMAIL PROTECTED] wrote:
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.
On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED]
I have been qualityhaving problems although I don't have all the
details at this point - I currently suspect mine are due to Latency
issues as both ends are colo boxes which don't seem to drop packets in
testing (ping).
Latency right now seems to be in the 80ms range on each end of the server.
Very interesting,
I had disabled that previsouly on my SPA but if I am the midpoint
between an inbound SIP connection and outbound IAX connection any
silence detection is not on my end it would be with the company
providing the DIDs, correct?
Or is IAX using it and I'm not aware.
Cheers,
On
Sounds like maybe you don't have either Postgres installed or PHP
confirgured to use it.
If you use RPMs, check for something in the php-pgsql family (%yum
install php-pgsql)
As a warning, you will also need to enable PHP globals in your php config.
Hope that helps,
J
On Tue, 8 Feb 2005
I would suggest the most important concern is that you set your
extensions high enough that people calling from the outside have say
options 1,2,3 and the extensions don't interupt those otherwise you
get people dialing the options by accident which can be annoying if
one of those options is
) to 90.0.4.76:5060
Sip read:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP
90.0.4.57:5060;branch=z9hG4bK3483fca1;received=90.0.4.57
From:
sip:[EMAIL PROTECTED];tag=as6c243000
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
Content-Length: 0
MARK G. MOODY
Network Architecture
--
| Chris Moody
| | | Silicon Hotrod - Linux Architects
:|::|: | Network Security Engineer
:|||: :|||: | [EMAIL PROTECTED]
.:|||:..:|||:. |~-~-~
CCNP - CCNA - CCDA| I route, therefore you
anyone successfully use the Cisco 12SP+ phone with asterisk?
They are VoIP phones, and SHOULD work...but wanted to know if anyone had
tried these sucka's out. Any caveats?
Cheers,
~Chris
--
| Chris Moody
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