Michael Jerris wrote:
> because your not running limit at all when you are doing an originate
> directly. You can use loopback to originate through a dialplan
> extension.
>
> Mike
>
> On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote:
>
>
>> Hi
>> I set the limit to 1 on the extension like that
Do you have anything on that extension?
On Jul 22, 2009, at 7:21 PM, Luis F Urrea wrote:
I don't know if this may be related but in voicemail.conf.xml by
default the two params that follow are defined:
And pressing 9 during the greeting does not send me to the operator.
I am on trunk re
you are using a channel created with a script and you did not set
js
session.autoHangup(0)
lua
session:autoHangup(0)
so when the * makes the call transfer the script kills the channel.
On Wed, Jul 22, 2009 at 3:13 PM, Lars Zeb wrote:
> Brian,
>
>
>
> When calling into FreeSWITCH and pressin
I don't know if this may be related but in voicemail.conf.xml by default the
two params that follow are defined:
And pressing 9 during the greeting does not send me to the operator.
I am on trunk rev 14123M
On Wed, Jul 22, 2009 at 2:13 PM, Lars Zeb wrote:
> Brian,
>
>
>
> When calling into
Brian,
When calling into FreeSWITCH and pressing * during the greeting, the call
immediately hangs up.
It used to ask for the mailbox number to retrieve messages. It no longer
works. I don't know if my dialplan is causing the error or something in
FreeSWITCH has changed.
Any ideas?
0. Probably, gonna have to go get them on the phone.1. No, but the information I need to process is stored in separate databases by gateway. There is no single table that has all of the DIDs and which gateway they belong to. I could create/maintain that table, but if I can determine the gateway b
FYI,
Brian West called to my attention that one of our community members, John
Wehle, has been very good at submitting useful bug reports, in many cases
with patches. His style of reporting is worthy of imitation, so I've added a
few links to the JIRA section of the Reporting Bugs wiki page:
http
because your not running limit at all when you are doing an originate
directly. You can use loopback to originate through a dialplan
extension.
Mike
On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote:
> Hi
> I set the limit to 1 on the extension like that
>
>
>
> When I am trying to make a call
you could hang on the event socket and catch the conference events, then
play the sounds via the "conference" api commands
-Ray
Rudolf Denert wrote:
> Hallo everybody!
>
> I would like to play soundfiles in a existing conference.
>
> The procedure is this:
>
> Someone calls the number of the con
On Jul 22, 2009, at 8:04 AM, Rupa Schomaker wrote:
On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller
wrote:
0) Rupa, you are absolutely right, I forgot that. ports was never
an issue because previous gateways all REGISTERed. I will have to
swap my ports around as bandwidth is not flexible
On Wed, Jul 22, 2009 at 5:11 AM, Pete Mueller wrote:
> 0) Rupa, you are absolutely right, I forgot that. ports was never an issue
> because previous gateways all REGISTERed. I will have to swap my ports
> around as bandwidth is not flexible.
>
You can't tell bandwidth.com to use port 5080?
>
>
Hi
I set the limit to 1 on the extension like that
When I am trying to make a call the that destination i transfered to
limit_exceeded dialplan, just like I want
The problem is, that when I am trying to make a call using "originate" I
am not getting the limitation.
Why is that?
__
The far end you're calling is sending a 302 can you check the sip
traffic please.
/b
On Jul 22, 2009, at 12:45 AM, Łukasz Zwierko wrote:
> I'm not sure how this exactly works, but I suppose that it is a single
> leg call, which upon answer would be attached to the conference (?)
> somehow. But
On Jul 22, 2009, at 5:11 AM, Pete Mueller wrote:
0) Rupa, you are absolutely right, I forgot that. ports was never
an issue because previous gateways all REGISTERed. I will have to
swap my ports around as bandwidth is not flexible.
What do you mean here?
1) I thought of this, but I hav
Hallo everybody!
I would like to play soundfiles in a existing conference.
The procedure is this:
Someone calls the number of the conference. Then this person types the pin in
to his phone. The next step is that he has to say the name for example "John".
This file is saved in a special folder.
0) Rupa, you are absolutely right, I forgot that. ports was never an issue because previous gateways all REGISTERed. I will have to swap my ports around as bandwidth is not flexible.1) I thought of this, but I have hundreds of DID, (around 600 at the moment) and maintaining that mapping in the dia
On Tue, Jul 21, 2009 at 11:35 PM, Pete Mueller wrote:
> My goal is:
> 0) figure out why the bandwidth gateway is being processed as "internal"
> (this is more of a security thing)
>
they are probably terminating traffic on port 5060 rather than 5080. 5060
is internal, 5080 is external.
>
> 1)
Szymon Olko wrote:
> Eli Hayun pisze:
>
>> Raymond Chandler wrote:
>>
>>> Eli Hayun wrote:
>>>
>>>
Is there is a way to initiate a call without making any dial manually?
>>> i think the api command "originate" is what you're looking for
>>>
>>>
Eli Hayun pisze:
> Raymond Chandler wrote:
>> Eli Hayun wrote:
>>
>>> Is there is a way to initiate a call without making any dial manually?
>>>
>>>
>> i think the api command "originate" is what you're looking for
>>
>> -Ray
>>
>> ___
>>
>
Raymond Chandler wrote:
> Eli Hayun wrote:
>
>> Is there is a way to initiate a call without making any dial manually?
>>
>>
> i think the api command "originate" is what you're looking for
>
> -Ray
>
> ___
>
Thanks, I figure that out, but n
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