Hi,
I was facing the very same issue and created a ticket...
https://issues.asterisk.org/jira/browse/ASTERISK-20221
best regards,
Sven
2012/8/13 James Mortensen james.morten...@a-cti.com
Andrew Latham lathama at gmail.com writes:
On Mon, Aug 13, 2012 at 2:58 PM, James Mortensen
On 14-08-12 08:29, Gopalakrishnan N wrote:
If I change autoload=no then asterisk is starting, but post to that
loading modules even chan_sip.so asterisk hangs. Its strange, only in
OpenSuse I am facing this. In CentOS, Ubuntu its working fine, same
Asterisk version with same hardware.
Please
All,
I ran into a case today where using the Console/DSP that the SIP
channel had been
hung up a long time ago. I tried to call back into the Console/Dsp and I
got busy.
There was no active channel any longer. Some how it did not get the hangup.
I am running 1.4.43
How can I make sure my
Hi,
I would like to know, anyone who worked in Email to Fax scenario? If so
please share the idea for implementing it.
As on other hand I configured Asterisk for inbound Fax which is working
good i.e. later forward the fax via email but don't know how can I
implement for outbound fax in this
8 aug 2012 kl. 14:07 skrev Kevin P. Fleming:
On 08/08/2012 06:30 AM, Kannan wrote:
Where can I get a complete set of RFCs and other specifications
supported by Asterisk?
To my knowledge there is no such list. In addition, Asterisk (like many other
pieces of software) does not claim
Hi,
We've got a client with a Polycom IP331 on Virgin Media behind a VMDG280 Router.
Asterisk is in a data centre with a public IP.
According to the link below, there seems to be a fault with the router that
blocks all SIP but the client doesn't want to change it.
please check it. might be it will help
http://ictfax.org/content/installation-guide
On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
I would like to know, anyone who worked in Email to Fax scenario? If so
please share the idea for implementing it.
As on other
please read CHANNEL variable. it will help you in this case...
On Tue, Aug 7, 2012 at 4:01 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
I discover that I have to place the wave files in the
/var/lib/asterisk/sounds/custom/
So, can I understand that the only solution I have is to
mailsvb mailsvb at gmail.com writes:
Hi,
I was facing the very same issue and created a ticket...
https://issues.asterisk.org/jira/browse/ASTERISK-20221
best regards,
Sven2012/8/13 James Mortensen james.mortensen at a-cti.com
Andrew Latham lathama at gmail.com writes:
On
On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen
james.morten...@a-cti.com wrote:
mailsvb mailsvb at gmail.com writes:
Hi,
I was facing the very same issue and created a ticket...
https://issues.asterisk.org/jira/browse/ASTERISK-20221
best regards,
Sven2012/8/13 James Mortensen
hello all,
I have call queue management system where all call comes in, put in the
queue while the caller speak with the online support team / teacher.
However, my major concern is those under MOH (in the queue) will not be
able to listen to the teacher until their turns and this is required.
my
On Tue, 2012-08-14 at 18:43 +0100, Goke M Aruna wrote:
hello all,
I have call queue management system where all call comes in, put in
the queue while the caller speak with the online support team /
teacher.
However, my major concern is those under MOH (in the queue) will not
be able to
Hi Gang,
Hopefully somebody out there has a doh for this one. My
dialplan announces the date and time using SayUnixTime. When I run this:
exten = 36225,1,Set(ABA=9)
exten = 36225,n,Background(telbank/${ABA}/${CHANNEL(language)}/thetimeis)
exten =
Hi,
I'm looking for SIP client that supports T.38 Fax other than zoiper.
Please advise at earliest.
--
Regards,
Ahmed Munir Chohan
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New to Asterisk?
I'm looking for any pros/cons of running an Asterisk based PBX over a
metro ethernet pipe. The system will have about 40 handsets and 6
DIDs.
Kind Regards,
Chris
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hello Carlos,
Thanks. I will try that and give back later.
Regards
On Tue, Aug 14, 2012 at 6:49 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Tue, 2012-08-14 at 18:43 +0100, Goke M Aruna wrote:
hello all,
I have call queue management system where all call comes in, put in
the
On Tue, Aug 14, 2012 at 12:44 PM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
I'm looking for any pros/cons of running an Asterisk based PBX over a
metro ethernet pipe. The system will have about 40 handsets and 6
DIDs.
As opposed to what, a regular internet connection? A metro
On Tue, Aug 14, 2012 at 4:44 PM, Chris Nighswonger
cnighswon...@foundations.edu wrote:
I'm looking for any pros/cons of running an Asterisk based PBX over a
metro ethernet pipe. The system will have about 40 handsets and 6
DIDs.
How oversold is the MetroE service? Can you get any latency /
Andrew Latham lathama at gmail.com writes:
On Tue, Aug 14, 2012 at 1:20 PM, James Mortensen
james.mortensen at a-cti.com wrote:
mailsvb mailsvb at gmail.com writes:
Hi,
I was facing the very same issue and created a ticket...
Hi James,
after applying the patch, I got the 400 bad request message as well...
This seems to be related to the sipml5 client (same issue with sip-js)
generating a wrong request. Take a look at the contact header in the
REGISTER message.
I was not able to fix the js code to generate the correct
Hi list,
is there an easy and proper way to revoke certificat created by the
script ast_tls_cert? I always get a name error on the CN/ O/ subject ...
--
Daniel
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-- Bandwidth and Colocation Provided by
mailsvb mailsvb at gmail.com writes:
Hi James,
after applying the patch, I got the 400 bad request message as well...
This seems to be related to the sipml5 client (same issue with sip-js)
generating a wrong request. Take a look at the contact header in the REGISTER
message.
I was
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