Thanks for all of your suggestions - I shall try both!
Glen
On 10/9/2011 14:42, Jean-Denis Girard wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Le 09/10/2011 03:40, Silverthorne Wystead a écrit :
I have a Grandstream GXP2000 and I would like to use tftp or some other
utility to
Hey Elliot;
Would you mind posting your dialplan for your Google Voice config? I am
having a hell of a time getting it to do *anything*.
Perhaps I am just fat-fingering.
Would you mind? Thanks in advance.
Glen
On 6/13/2011 19:02, Elliot Murdock wrote:
Hello,
I am using 1.8.4.2 and while
Hey;
I figured I would ask here as I seem to get better results.
I am using the voip.ms http://voip.ms/ VoIP service. I have no problem
configuring my
Asterisk server 1.8x to dial out with my Softphone.
HOWEVER, for some reason, I cannot get inbound. All that I hear is a
busy signal.
I know
Hello Folks;
Perhaps I am chasing my tail here.
Before I go any further, is this compatible/supported in Asterisk 1.6x?
If so, I would be willing to post any manager.conf or http.conf snippets
needed.
When I attempt to open the Asterisk Web GUI, I get a 'page not found'.
I am sure this is
Works well - however, I see you included the API access.
Are there more parameters that we can pass to get more information?
Example, when we go to the web site, it gives you the
City/State/Province/Postcode and carrier.
G
On 5/29/2011 07:47, Michael R. Wally wrote:
FreeCNAM.org is
Hello Folks;
I appreciate all of the help so far - thanks.
Another question: I am using MixMonitor() to record calls and I would
like to include the called number/extension in the filename:
In my dialplan, I am able to save the file with the caller id in the
filename. However, what I am a
Dan et al;
Okay - I have declared DYNAMIC_FEATURES=MixMonApp in the [global]
section of my extensions.conf
I dial into my trunk, the softphone rings, I answer and I press '*1' - I
hear the tones, but I see no indication in the Asterisk CLI and I see no
.wav file being created.
I must
Hi Dan et al;
I had actually done a sip reload, dialplan reload, module reload
res_features.so and logger reload.
However, upon seeing your email, I restarted the Asterisk server
completely to see if I had missed anything. I still see the same behaviour.
I am at a loss.
Glen
On 4/10/2011
Hey!
I did a little bit of digging - and I solved my issue!
Apparently, in my extensions.conf, I specified the wrong variable.
I had DYNAMIC_FEATURES=callrec (which is the name of my macro)
I changed it to DYNAMIC_FEATURES=MixMonApp, which is what is it aliased
to in the features.conf.
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
Dan et al;
This looks like a perfect solution.
However, I have one issue. If I initiate the macro manually (put it in
the proper context/dialplan) it works. I see the *.wav file being
created and growing in the /var/spool/asterisk/monitor directory.
If I try to implement it adding the
Hello Everyone;
I am looking for a solution to record calls that come into our Asterisk
server. I am hoping for something that is easy to use - however, if I
have to modify it to make it easier to use, I do not mind.
Does anyone know of any opensource or otherwise solutions out there that
I
Forgive my ignorance on this as I am still fairly new to Asterisk.
I have noticed lately that there have been several attempts to hack our
Asterisk server. I see multiple attempts to log in with a particular
extension from the same IP address, perhaps hundreds of times per
second. It causes
Let me explain:
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication.
The calling party, from the SIP trunk hears nothing - the extension
rings on the Asterisk server (you can see it in the CLI and hear it at
the
Raza wrote:
Hi
Try Nat=yes in general settings
On 06-Nov-2010 9:57 PM, Silver Thorne zora...@gmail.com
mailto:zora...@gmail.com wrote:
Let me explain:
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication
Hey Everyone;
I inherited an Asterisk box where the dialplan is a real mess. ( I would
actually be embarrassed to post some of the stuff!)
So, here is what I need to do - and again, I am looking for fishing nets
and places to cast them - if I don't figure it out, I will never
freakin' learn!
extension not found.
So, when I call the 33173793697 number, the above entry is what I see in
the log.
Glen
On 11/1/2010 17:32, Steve Edwards wrote:
On Mon, 1 Nov 2010, Silver Thorne wrote:
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839
Hey;
I never thought of that.
It is causing an issue for me. One SIP UA works fine - ring, forward,
etc. While the other does not.
I am a little clueless here - where would I start with this?
Thanks
Glen
On 11/1/2010 19:15, Philipp von Klitzing wrote:
Hi!
[Nov 1 19:55:49]
Hello Folks;
Again, excuse my cluelessness.
I have an Asterisk server in the US - and I want to connect it to one in
Europe.
Here is my scenario:
1. call a phone number, my Asterisk box in the US answers
2. perhaps a 'please wait' voice message
3. it dials an extension on the other
Hey;
Anyone see this before:
[Nov 1 19:55:49] WARNING[30497] chan_sip.c: username mismatch, have
6839, digest has 3169
G
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