Yes I should really upgrade, just have to make sure that asterisk-java
will work properly with 1.8
/H
Den 2013-01-02 22:25 skrev Danny Nicholas da...@debsinc.com:
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I
On Friday 04 January 2013, Selva M wrote:
Hi,
I tried the option and got following message.
PBX1*CLI
-- Starting simple switch on 'DAHDI/1-1'
== Starting DAHDI/1-1 at from-pstn,s,1 failed so falling back to exten
's' == Starting DAHDI/1-1 at from-pstn,s,1 still failed so falling
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I
Did you set externip and localnet in your sip conf ?
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New to Asterisk? Join us for a live introductory webinar every Thurs:
hello,
any news about WebM/VP8 support in asterisk?
some bounty where can i contribute?
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---
Marek Cervenka
===
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-- Bandwidth and Colocation
Hi
sip show peer 21342
gives me peer 21342's parameters. I am interested in the MaxCallBR line i.e.
MaxCallBR: 384 kbps
What exactly does this mean?
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Its maximum call Bit rate available for that peer. Default is 384 kbps.
Your call for that peer allowed max bit rate or bandwidth of 384 kpbs only
Regards,
Bharat Lalcheta
On Fri, Jan 4, 2013 at 7:09 PM, XBrian bobo...@yahoo.co.uk wrote:
Hi
sip show peer 21342
gives me peer 21342's
I have a Polycom IP6000 conference phone, along with a lot of Polycom IP550
units. I've been updating all the 650s to Polycom's from 3.2.3 to the 4.0.3
software release, by hodling 468*and having them pull the update.
It's been fine with the 650s, but the IP6000 (held 68* for that one)
Having an issue with receiving faxes, but when I pass through the fax.
Currently, I receive the fax with Digium's Fax for Asterisk, store it and
the initiate an outbound call to our fax server. (XMedius Fax). This
works, but we would prefer to have Asterisk simply route the call directly
to
Its so obvious now that you've made it clear
Thanks
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Hello,
I want to use an Huawei stick model K3765 which support voice with
asterisk. I'm begginer with this kind of interaction from asterisk
with external devices.
Can someone guide me what should i configure to use this device?
Thank you for support,
Regards,
Jonson.
---
www.Mobile-Wi.Fi
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Justin,
I haven't seen it on that model, but I did have a case awhile back where
it happened to me with a different conference phone. Pretty much the same
symptoms you had. Even more fun it was remote so I couldn't get my hands
on it.
I tracked mine down to being an incorrect firmware for
From: Carlos Alvarez car...@televolve.com
It may be too late for this, but in working with another RBOC who
didn't want to deal with Asterisk, I just asked what they do
support, and modified the headers sent by Asterisk to claim that it
was one of the devices on that list. Done.
Like
- Original Message -
From: Matthew J. Roth mr...@imminc.com
Your email documents the same experience we had years ago. It was
strange reading it and I was shocked that nothing has changed in that
much time. Asterisk will work with Verizon's IP trunking product,
but
they're trying
Trust me, Verizon doesn't really provide support.What they will do is tell
you something different (often conflicting stuff) when you send in a ticket.
One time they tell us the From must be in e.164 format, other times they say it
does not.We asked for an updated Interop guide weeks
On Fri, Jan 4, 2013 at 11:18 AM, Eric Wieling ewiel...@nyigc.com wrote:
Trust me, Verizon doesn't really provide support.What they will do is
tell you something different (often conflicting stuff) when you send in a
ticket.One time they tell us the From must be in e.164 format, other
Hi Danny,
Can you please elaborate on how in the dialplan we can set faxdetect on and
off?
We currently have it set on in sip.conf.
Thanks.
On 3 January 2013 17:21, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Carlos Alvarez wrote:
Sounds like the same huge effort it takes to work with Qwest/
Centurylink, and in the long run we found it simply isn't worth it.
The few benefits of working with an RBOC are countered by the many
drawbacks of working with an RBOC.
Also we recently acquired a half
- Original Message -
From: Carlos Alvarez car...@televolve.com
Sounds like the same huge effort it takes to work with
Qwest/Centurylink, and in the long run we found it simply isn't
worth it. The few benefits of working with an RBOC are countered by
the many drawbacks of working
- Original Message -
From: Matthew J. Roth mr...@imminc.com
At least Verizon maintains a consistent customer experience. ; )
Overall, we've found the service to be reliable and stable, but when
there are problems or changes needed you're dealing with Verizon and
the
Don't think you actually can, per se. What you can do is set a variable and
redirect to the line that has this defined or undefined.
Let's say that you have 4 lines; SIP/1001 and DAHDI/1 have faxdetect=yes
defined in sip.conf and chan_dahdi.conf. SIP/1002 and DAHDI/2 have
faxdetect=no in the
On Thu, Jan 3, 2013 at 9:39 PM, David Cunningham
dcunning...@voisonics.comwrote:
We have all calls going to an AGI, which decides where the number will get
routed to, and if fax detection should be enabled for this call. The choice
should only apply to the current call.
What criteria would
I believe Asterisk 11 is the first version which allows you to enable and
disable faxdetect on the fly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Friday, January 04, 2013 2:42 PM
To:
On Fri, 4 Jan 2013, Danny Nicholas wrote:
Simple Perl AGI to set dialplan variable:
print STDOUT SET VARIABLE USEFAX \ON\ \n;
print STDOUT SET VARIABLE USEFAX \OFF\ \n;
You need to read the response from each request to comply with the AGI
protocol.
--
Thanks in advance,
Hi all,
I'm attempting to install dahdi-linux-complete-2.6.1+2.6.1 on an OpenSUSE
system running kernel 3.7.1-17
The build is failing like this:
ct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tone_detection.c oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.c
Looks like this issue:
https://issues.asterisk.org/jira/browse/DAHTOOL-60
--
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com www.asterisk.org
--
Hi all,
We want to use res_calender_ews to close users extensions if there are busy in
there exchange calenders.
It is possible to use shared calenders ? Ie, I have a resource user that have
access to the users calenders, so I dont need to maintain every users
password/username in asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Löfqvist
Sent: Friday, January 04, 2013 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calender and EWS with shared calenders
Hi
4 jan 2013 kl. 23:59 skrev Danny Nicholas
da...@debsinc.commailto:da...@debsinc.com:
From:
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus Löfqvist
Sent: Friday, January 04, 2013 4:46
Hi there,
I support a large number of enterprise users who contractually must connect to
our support center via a 4G VOIP connection.
I simply want to be able to auto detect all poor quality calls in realtme (as
they are being made), play a message and drop the call - without user
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