Hello all,
Can asterisk be used to provide PTT feature? Anyone using or tested it?
Thanks:
-AEE
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I just implemented Asterisk as Voicemail server on our GSM network. Also MWI
notifications are working fine for me.
There are still some glitches but overall it's working for cellular network.
What type of cell switch you have? Do your GSM cell switch is also VOIP
gateway?
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Hello,
I am in bit confusion as I am not able to find point of trouble.
All extensions are configured same way. All are registered, have same
context, voicemail context.
There are around 112 extensions. So I am giving example of 2 extensions. one
going to voicemail fine and other not.
===
On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria wrote:
> Poster is having problem when he disallows anonymous sip peers. Do you know
> at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet
> seen the dialplan for FreePBX.
>
It's very simple to find the actually issue, if th
On 09/12/10 07:06, Zeeshan Zakaria wrote:
>
> "I think this may be because ..."
>
>
> So you think, don't know. Maybe you knew if you knew the FreePBX
> code, or bothered to look into it.
>
For God's sake, stick a sock in it. Others are attempting to help. You
are not.
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On Sep 10, 2010, at 10:07 PM, bruce bruce wrote:
> Hi Everyone,
>
> I have a provider whose DID used to come into the box just fine but recently
> stopped working. Nothing has been changed on our end.
>
> Here is what I get when doing "sip set debug peer PROVIDER":
>
> Sending to 123.123.123.
[snip]
customers, who all connect from behind their home nat gateways of all
kinds. I still don't know why that fixed it.
Sorry you took it so harshly Zeeshan, but the only posts that stick out
to me from you are the ones where you are bashing people for posting
Poster is having problem when he disallows anonymous sip peers. Do you know
at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet
seen the dialplan for FreePBX.
When there is enough detail in the post and I am aware of the problem, I
always try to help. I don't believe in ma
> --
> www.ilovetovoip.com
>
> > On 2010-09-11 7:22 PM, "Paul Belanger"
> > wrote:
> >
> >
> >
> > On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere
> > wrote:
> > >> Sending to 123.123.12...
> >
> > > Either you changed the peer parameters or they did...
> > >
> >
> > If he is not receivi
Actually it is a very easy to understand and fix issue, but looking at the
code taking care of anonymous sip calls is the key. Those who post third
party GUI related issues should at least post the underlying asterisk config
or code here, so the asterisk part of the problem can be fixed.
Zeeshan A
On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere wrote:
>> Sending to 123.123.123.123 : 5060 (no NAT)
>>
> Either you changed the peer parameters or they did...
>
If he is not receiving any response, it is most likely a routing issue.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul
So you are sure it has NOTHING to do with extensions.conf. This clearly
shows your absolute ignorance about what poster is asking and how FreePBX
works. Had the problem code been posted, this problem would already have
been solved by now.
And sorry if you think this is policing. You can think what
>
> "I think this may be because ..."
>
So you think, don't know. Maybe you knew if you knew the FreePBX code, or
bothered to look into it.
>
> j
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digita
On 09/09/10 17:59, Steve Davies wrote:
> On 9 September 2010 17:52, Antonio Berrios
> wrote:
>
>> Steve Davies wrote:
>>
>>> Hi,
>>>
>>> I am using 1.6.2.11, and I need to be able to include the name of the
>>> channel that answered a call in the call-recording filename.
>>>
>>> At a gu
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
> This is not elastix or FreePBX forum and asking non-asterisk related
> questions here is misusing this mailing list. Allow anonymous sip is
> not an asterisk feature. Look in the code in extensions.conf what it
> is programmed to do and y
On Fri, 2010-09-10 at 23:07 -0400, bruce bruce wrote:
> Hi Everyone,
>
>
> I have a provider whose DID used to come into the box just fine but
> recently stopped working. Nothing has been changed on our end.
>
>
> Here is what I get when doing "sip set debug peer PROVIDER":
>
>
> Sending to 1
Mr. John,
This is not about policing and this is asterisk-user mailing list. Poster is
a FreePBX user. I am very well aware of Asterisk IS involved, but the fact
is this is not a FreePBX mailing list. If the poster examines the problem
code from extensions.conf, or post it here, it'll made him and
On Fri, Sep 10, 2010 at 11:07 PM, bruce bruce wrote:
> I have a provider whose DID used to come into the box just fine but recently
> stopped working. Nothing has been changed on our end.
>
Have you considered contacting your provider? I would think that is
your first step.
--
Paul Belanger | d
Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list.
Since the poster may not be sure this isn't an Asterisk problem, and
Asterisk IS involved, your position is unreasonable.
Self appointed list police a
Allow anonymous SIP and enable debug then check if calls coming from
same IP which you have configured in peer?
Regards,
Faisal Hanif//
On 9/11/2010 8:07 AM, bruce bruce wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working. Nothi
Hi,
An intelligent way is to maintain numbers list in any Database (could be
SQlite if you don't want to use proper DB engine) then use ODBC-Function
if the number is there and decide routing.
2nd option is to use Perl AGI with DBI::CSV and manage numbers list in a
CSV file.
Regards,
Fai
2010/9/10 Hose
>
> Does anyone have a suggestion on how to handle this? For example, if I
> have a list of numbers that I want to go out a certain sip channel and
> another that I want to go out the dahdi device, is there a way to do
> this? None of the numbers will fit into a pattern, so just
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous sip is not an
asterisk feature. Look in the code in extensions.conf what it is programmed
to do and you'll figure out why it is happening. Or maybe post the code and
as
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