Re: [asterisk-users] Looking for the carrier that owns a particular DID

[Frank Vanoni]

On Thu, 2017-11-02 at 11:33 -0400, Tech Support wrote:

> How do I find out which carrier owns the DID in question?

Try here:

https://www.twilio.com/lookup



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Re: [asterisk-users] Return without Gosub: stack is empty

[Richard Mudgett]

On Thu, Nov 2, 2017 at 1:25 PM, Dmitriy Serov  wrote:

> [sub-out-do-dial]
> exten => s,1,NoOp(Dial)
>  same => n,NoOp(FirstChannel: ${CHANNEL})
>  same => n,Dial(,60,gF)
>  same => n,NoOp(SecondChannel: ${CHANNEL})
>  same => n,Return()
>
> [some]
> exten => s,1,GoSub(sub-out-do-dial,s,1)
>
> In case of the destination channel hangs up in log i see:
> ERROR[26570][C-00e7] app_stack.c: Return without Gosub: stack is empty
>

I think you mean if the calling channel hangs up you get the error message.


>
> Yes. SecondChannel != FirstChannel.
> Is it a wrong way to use Dial(g)+Gosub? Or any Dial in sub?
>

The first channel (calling party) knows about being in the subroutine.  It
was the
channel executing dialplan to get into the subroutine.  Use of the g option
with this
dialplan is OK as the calling channel knows where the subroutine was called
from.

The second channel (called party) knows nothing about being in a subroutine
so it
cannot return from one.  It has not executed any dialplan before.  The F
option simply
tells it to start executing dialplan at the priority after the dial
application.  This dialplan
needs to be able to distinguish between the two channels and act
accordingly.  Using
the F() option with a dialplan location is the simplest way to distinguish
between the
two channels.

Richard
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[asterisk-users] Return without Gosub: stack is empty

[Dmitriy Serov]

[sub-out-do-dial]
exten => s,1,NoOp(Dial)
 same => n,NoOp(FirstChannel: ${CHANNEL})
 same => n,Dial(,60,gF)
 same => n,NoOp(SecondChannel: ${CHANNEL})
 same => n,Return()

[some]
exten => s,1,GoSub(sub-out-do-dial,s,1)

In case of the destination channel hangs up in log i see:
ERROR[26570][C-00e7] app_stack.c: Return without Gosub: stack is empty

Yes. SecondChannel != FirstChannel.
Is it a wrong way to use Dial(g)+Gosub? Or any Dial in sub?

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Re: [asterisk-users] Looking for the carrier that owns a particular DID

[Antony Stone]

On Thursday 02 November 2017 at 16:33:04, Tech Support wrote:

> I have a customer who is looking for a particular DID. (I dialed it and
> it is not in service). I searched through my preferred upstream provider's
> list but I came up empty. I wrote them, and this is their reply.
> 
> "We currently do not have that specific number in stock as this number is
> owned by another carrier that we do not have a business relationship with."

This suggests that they either:

 - identified who owned it, and hence established that they had no business 
relationship with the owner, or

 - identified that it wasn't owned by any of their business partners, in which 
case they may genuinely not know who does own it.

Either way, it's probably worth asking them, in case they have no objection to 
telling you, but simply didn't provide the information as it wasn't an answer 
to your question.

> So my question is this. How do I find out which carrier owns the DID in
> question?

Failing the above, I would start with the relevant country's telecoms 
regulator.


Antony.

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Perl is executable line noise.

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[asterisk-users] Looking for the carrier that owns a particular DID

[Tech Support]

All;

I have a customer who is looking for a particular DID. (I dialed it and
it is not in service). I searched through my preferred upstream provider's
list but I came up empty. I wrote them, and this is their reply.

 

"We currently do not have that specific number in stock as this number is
owned by another carrier that we do not have a business relationship with."

 

So my question is this. How do I find out which carrier owns the DID in
question?

Thanks;

John V. 

 

Tech Support

Tech Support

VoIP Business Solutions

240-215-3479 x325

  supp...@voipbusiness.us

 

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Re: [asterisk-users] pjsip insecure=port,invite

[Dmitry Melekhov]

Thank you very much!

It works.
Although there is one strange thing here-
there is no ringback tone on cisco's side when pjsip is use,
but this is another story :-)



02.11.2017 14:17, Joshua Colp пишет:

On Thu, Nov 2, 2017, at 04:50 AM, Dmitry Melekhov wrote:

Hello!

Looks like faq, but...

Could you , please, point me on how to convert this


[cisco]
type=friend
host=192.168.22.253
insecure=port,invite


to pjsip?

The equivalent is this:

[mytrunk]
type=identify
endpoint=mytrunk
match=203.0.113.1

 From the page you linked. That says "Match incoming traffic from
203.0.113.1 and use endpoint mytrunk for it".

You also need an endpoint defined like:

[mytrunk]
type=endpoint
context=from-external
disallow=all
allow=ulaw




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Re: [asterisk-users] pjsip insecure=port,invite

[Joshua Colp]

On Thu, Nov 2, 2017, at 04:50 AM, Dmitry Melekhov wrote:
> Hello!
> 
> Looks like faq, but...
> 
> Could you , please, point me on how to convert this
> 
> 
> [cisco]
> type=friend
> host=192.168.22.253
> insecure=port,invite
> 
> 
> to pjsip?

The equivalent is this:

[mytrunk]
type=identify
endpoint=mytrunk
match=203.0.113.1

>From the page you linked. That says "Match incoming traffic from
203.0.113.1 and use endpoint mytrunk for it".

You also need an endpoint defined like:

[mytrunk]
type=endpoint
context=from-external
disallow=all
allow=ulaw

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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] ASR Suggestions for small dictionnary (<1000 entries) lookup in France/french

[Bertrand LUPART - Linkeo.com]

Hello Olivier,


> I'm in the early stages of designing an Emergency calling service IVR 
> application.
> 
> The IVR application asks simple one or two questions like "which is the 
> postal code of the area you are currently calling from ?" "Is the correct ?". 
> The expected values are a 5-digits number like 
> "twenty-five-thousand-two-hundreds-twelve" or "twenty-five-two-twelve" 
> variations or "yes or no".
> 
> From received answers, this application would route incoming call to an 
> appropriate Public Safety Answering Point.


Have you considered ways for determining caller's position on the backend or 
use integrated GPS of his cellphone?

For example : 
https://www.telecom.gouv.fr/normau/


My guess user input in case of an emergency call should remain optional.


Greetings,

-- 
Bertrand LUPART


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[asterisk-users] pjsip insecure=port,invite

[Dmitry Melekhov]

Hello!

Looks like faq, but...

Could you , please, point me on how to convert this


[cisco]
type=friend
host=192.168.22.253
insecure=port,invite


to pjsip?


as you can see another side is very old cisco router, so I can't change 
anything there.



I don't see any examples here

https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-ExampleSIPTrunkConfiguration


on insecure=port,invite  psjip  equivalent.


Thank you!



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