Re: [asterisk-users] GSM Gateway behind SIP ATA?

[Belal]

Dear sir,
 
what about receiving call from a GSM gateway. I didn't see the caller ID?. 
is it happen to you? and what is the solution,Please.? 

thanks,
Belal


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Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

[A J Stiles]

On Monday 25 Jan 2016, waqas.mehmood90 wrote:
> I am working on asterisk ivr .i am facing problrm in crontab.when i run
> example it give bash 5:command not found then i check and found that no
> crontab for root user kindly guide me please

Hello, is that the vet?  One of my animals is poorly.  What should I do?

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Re: [asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 So once a transport is pulled from the transports table in realtime during 
asterisk startup it can't get any updates?
 Can a new transport be added to the table and the associated endpoints be 
updated to use the new transport, or are transport types only read at 
startup across the board?
  
 Thanks

Bryant
  


 From: "Joshua Colp" 
Sent: Tuesday, January 26, 2016 8:10 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant Zimmerman wrote:
> Joshua
> Since there is no automated way currently built in to update the
> external signaling and media address information.
> Does the realtime pjsip support having the transport contexts section
> being pulled from a database table?
> I was thinking a cron script updating the table and forcing a reload
> each time an IP address changed might a workable solution.

No, once loaded the transports can not be changed.

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Re: [asterisk-users] PJSIP Stun/ICE

[Joshua Colp]

Bryant Zimmerman wrote:

Joshua
So once a transport is pulled from the transports table in realtime
during asterisk startup it can't get any updates?
Can a new transport be added to the table and the associated endpoints
be updated to use the new transport, or are transport types only read at
startup across the board?


Transports can only be loaded at startup. This stems from PJSIP not 
being dynamic with transports (it doesn't like its environment changed 
to that degree while in use). I'm afraid if your IP changes you'd have 
to restart Asterisk when you are using PJSIP.


--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] PJSIP - Realtime - Transports module?

[Bryant Zimmerman]

Does anyone know which module the type=transport loads under.
 I am trying to set up transports to load from a realtime table. I added 
the following under [res_pjsip] and it does not poll the associated 
database.
  
 [res_pjsip]
transport=realtime,vap002_ps_transports
  
 We also set the associated values in extconfig.conf as well. My best guess 
is that transports are loaded under a different module's context. Anyone 
have an idea?
  
 Thanks

Bryant

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Re: [asterisk-users] PJSIP Stun/ICE

[Daniel Heckl]

Bryant,

I have the same problem with dynamic public IPs and PJSIP. What is your idea to 
solve the problem?

My suggestion would be to write a script that monitors the change, 
pjsip.transports.conf updated and Asterisk restarts?

Daniel

> Am 26.01.2016 um 14:21 schrieb Joshua Colp :
> 
> Bryant Zimmerman wrote:
>> Joshua
>> So once a transport is pulled from the transports table in realtime
>> during asterisk startup it can't get any updates?
>> Can a new transport be added to the table and the associated endpoints
>> be updated to use the new transport, or are transport types only read at
>> startup across the board?
> 
> Transports can only be loaded at startup. This stems from PJSIP not being 
> dynamic with transports (it doesn't like its environment changed to that 
> degree while in use). I'm afraid if your IP changes you'd have to restart 
> Asterisk when you are using PJSIP.
> 
> -- 
> Joshua Colp
> 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] PJSIP Stun/ICE

[Bryant Zimmerman]

Daniel
  
 Thank you for your response. I was considering this as well. I have a 
script that monitors the IP Address now. I was hoping to use the real-time 
transports table now that alembic creates. I am trying to figure out which 
pjsip module is responsible for the transports contexts as I need to now 
configure it in the sorcery.conf file. I thought it would be under the 
[res_pjsip] context, but it is not even trying to pull from my transports 
table when it is there.  I am hoping someone will know what module it is in 
so I can move my configuration under the correct context.
  
 Thanks

Bryant
  


 From: "Daniel Heckl" 
Sent: Tuesday, January 26, 2016 10:15 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant,

I have the same problem with dynamic public IPs and PJSIP. What is your 
idea to solve the problem?

My suggestion would be to write a script that monitors the change, 
pjsip.transports.conf updated and Asterisk restarts?

Daniel

> Am 26.01.2016 um 14:21 schrieb Joshua Colp :
>
> Bryant Zimmerman wrote:
>> Joshua
>> So once a transport is pulled from the transports table in realtime
>> during asterisk startup it can't get any updates?
>> Can a new transport be added to the table and the associated endpoints
>> be updated to use the new transport, or are transport types only read 
at
>> startup across the board?
>
> Transports can only be loaded at startup. This stems from PJSIP not being 
dynamic with transports (it doesn't like its environment changed to that 
degree while in use). I'm afraid if your IP changes you'd have to restart 
Asterisk when you are using PJSIP.
>
> --
> Joshua Colp
> 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] PJSIP Stun/ICE

[Daniel Heckl]

Bryant,

that sounds interesting. I am searching for a script which monitors and updates 
the ip address. Does this your script? Can you share your script with us?

Thanks
Daniel

> Am 26.01.2016 um 16:39 schrieb Bryant Zimmerman :
> 
> Daniel
>  
> Thank you for your response. I was considering this as well. I have a script 
> that monitors the IP Address now. I was hoping to use the real-time 
> transports table now that alembic creates. I am trying to figure out which 
> pjsip module is responsible for the transports contexts as I need to now 
> configure it in the sorcery.conf file. I thought it would be under the 
> [res_pjsip] context, but it is not even trying to pull from my transports 
> table when it is there.  I am hoping someone will know what module it is in 
> so I can move my configuration under the correct context.
>  
> Thanks
> 
> Bryant
>  
> From: "Daniel Heckl" 
> Sent: Tuesday, January 26, 2016 10:15 AM
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Subject: Re: [asterisk-users] PJSIP Stun/ICE
>  
> Bryant,
> 
> I have the same problem with dynamic public IPs and PJSIP. What is your idea 
> to solve the problem?
> 
> My suggestion would be to write a script that monitors the change, 
> pjsip.transports.conf updated and Asterisk restarts?
> 
> Daniel
> 
> > Am 26.01.2016 um 14:21 schrieb Joshua Colp :
> >
> > Bryant Zimmerman wrote:
> >> Joshua
> >> So once a transport is pulled from the transports table in realtime
> >> during asterisk startup it can't get any updates?
> >> Can a new transport be added to the table and the associated endpoints
> >> be updated to use the new transport, or are transport types only read at
> >> startup across the board?
> >
> > Transports can only be loaded at startup. This stems from PJSIP not being 
> > dynamic with transports (it doesn't like its environment changed to that 
> > degree while in use). I'm afraid if your IP changes you'd have to restart 
> > Asterisk when you are using PJSIP.
> >
> > --
> > Joshua Colp
> > 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] GSM Gateway behind SIP ATA?

[John Kiniston]

Have you turned on sip debugging?

Do you see the caller ID in the invite from your Gateway to your PBX?

On Tue, Jan 26, 2016 at 2:07 AM, Belal 
wrote:

> Dear sir,
>
> what about receiving call from a GSM gateway. I didn't see the caller ID?.
> is it happen to you? and what is the solution,Please.?
>
> thanks,
> Belal
>
>
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Re: [asterisk-users] PJSIP Stun/ICE

[James Cloos]

> "JC" == Joshua Colp  writes:

JC> This stems from PJSIP not being dynamic with transports (it
JC> doesn't like its environment changed to that degree while
JC> in use). I'm afraid if your IP changes you'd have to restart
JC> Asterisk when you are using PJSIP.

Wow.

I say this having voted for pjsip over the listed alternatives back when
the plan to depricate chan_sip was first floated:

That should have excluded pj from the options.  Which of course means
there were no reasonable options.

Can ari get around that bug? 

Lack of full support for traversing nat makes pjsip worthless for a
large number of users.  And the whole point of realtime is to have all
of the rt config fully dymanic.

If ari cannot avoid that limitation, chan_sip should get full ongoing
maintainance until pjsip is fixed.

-JimC
-- 
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Re: [asterisk-users] PJSIP Stun/ICE

[Joshua Colp]

James Cloos wrote:

"JC" == Joshua Colp  writes:


JC>  This stems from PJSIP not being dynamic with transports (it
JC>  doesn't like its environment changed to that degree while
JC>  in use). I'm afraid if your IP changes you'd have to restart
JC>  Asterisk when you are using PJSIP.

Wow.

I say this having voted for pjsip over the listed alternatives back when
the plan to depricate chan_sip was first floated:

That should have excluded pj from the options.  Which of course means
there were no reasonable options.


PJSIP doesn't like changing existing transports, the NAT functionality 
is provided by the Asterisk implementation and can't be reloaded as a 
side effect due to the heavy handed restriction. With work it could be 
changed to allow the non low level things to be changed. What you can't 
do with PJSIP is create a UDP transport, reload, and have it removed. 
Once it's there it is there unless you restart.




Can ari get around that bug?


ARI is a REST interface to Asterisk, it doesn't have anything to do with 
this.




Lack of full support for traversing nat makes pjsip worthless for a
large number of users.  And the whole point of realtime is to have all
of the rt config fully dymanic.


I disagree that it makes it worthless for a large number of users. It's 
only within the last few days that a few people have run into this 
particular issue where they have a public IP address that is changing a 
lot and PJSIP does not support changing it without a restart. If it were 
a huge sweeping issue we'd be seeing it more often. If it continues to 
show up a community member or us (heck maybe even myself in my spare 
time) may look into implementing it.




If ari cannot avoid that limitation, chan_sip should get full ongoing
maintainance until pjsip is fixed.


The support level for chan_sip has already been changed and was 
announced long ago. Patches will continue to be accepted for it and 
community members can support it. We (Digium) are putting our effort 
towards PJSIP.


--
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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] PJSIP Stun/ICE

[Francis Mendoza]

Hi JC,

I have the same case as you are my server has static public IP assigned and
my client has public dynamic IP address in order to connect them without
issue what I did was to setup openvpn in my other side that has
public static IP and then the client server asterisk will connect into it
and they will communicate with the VPN local IP adresses that I assigned.
hope this 'workaround' helps

~Cheers

On Wednesday, 27 January 2016, Joshua Colp  wrote:

> James Cloos wrote:
>
>> "JC" == Joshua Colp  writes:
>>>
>>
>> JC>  This stems from PJSIP not being dynamic with transports (it
>> JC>  doesn't like its environment changed to that degree while
>> JC>  in use). I'm afraid if your IP changes you'd have to restart
>> JC>  Asterisk when you are using PJSIP.
>>
>> Wow.
>>
>> I say this having voted for pjsip over the listed alternatives back when
>> the plan to depricate chan_sip was first floated:
>>
>> That should have excluded pj from the options.  Which of course means
>> there were no reasonable options.
>>
>
> PJSIP doesn't like changing existing transports, the NAT functionality is
> provided by the Asterisk implementation and can't be reloaded as a side
> effect due to the heavy handed restriction. With work it could be changed
> to allow the non low level things to be changed. What you can't do with
> PJSIP is create a UDP transport, reload, and have it removed. Once it's
> there it is there unless you restart.
>
>
>> Can ari get around that bug?
>>
>
> ARI is a REST interface to Asterisk, it doesn't have anything to do with
> this.
>
>
>> Lack of full support for traversing nat makes pjsip worthless for a
>> large number of users.  And the whole point of realtime is to have all
>> of the rt config fully dymanic.
>>
>
> I disagree that it makes it worthless for a large number of users. It's
> only within the last few days that a few people have run into this
> particular issue where they have a public IP address that is changing a lot
> and PJSIP does not support changing it without a restart. If it were a huge
> sweeping issue we'd be seeing it more often. If it continues to show up a
> community member or us (heck maybe even myself in my spare time) may look
> into implementing it.
>
>
>> If ari cannot avoid that limitation, chan_sip should get full ongoing
>> maintainance until pjsip is fixed.
>>
>
> The support level for chan_sip has already been changed and was announced
> long ago. Patches will continue to be accepted for it and community members
> can support it. We (Digium) are putting our effort towards PJSIP.
>
> --
> Joshua Colp
> 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] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 I look forward to improvements as time goes on with PJSIP.
 I have been trying all day to get the Transport objects to pull from a 
real-time table. The documentation says it is possible, but does not show 
any examples. I am hoping to have the Transports pulled from the table at 
asterisk startup and then add additional as necessary. Using reloads to 
make the new Transports available. I understand the limitation of not being 
able to change existing and can live with that for now.   
  
 Do you know if there is anything special I have to do in the sorcery.conf 
to make the Transports pull from the real-time side of things. All my other 
tables are working.
  
 I disagree with the user that things PJSIP is worthless. There are some 
issues to work out long term, and documentation will get better over time 
as more of us work with it and contribute back.  Thanks for all you have 
assisted with around PJSIP.
  
 Bryant 


 From: "Joshua Colp" 
Sent: Tuesday, January 26, 2016 8:40 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE   
James Cloos wrote:
>> "JC" == Joshua Colp writes:
>
> JC> This stems from PJSIP not being dynamic with transports (it
> JC> doesn't like its environment changed to that degree while
> JC> in use). I'm afraid if your IP changes you'd have to restart
> JC> Asterisk when you are using PJSIP.
>
> Wow.
>
> I say this having voted for pjsip over the listed alternatives back when
> the plan to depricate chan_sip was first floated:
>
> That should have excluded pj from the options. Which of course means
> there were no reasonable options.

PJSIP doesn't like changing existing transports, the NAT functionality
is provided by the Asterisk implementation and can't be reloaded as a
side effect due to the heavy handed restriction. With work it could be
changed to allow the non low level things to be changed. What you can't
do with PJSIP is create a UDP transport, reload, and have it removed.
Once it's there it is there unless you restart.

>
> Can ari get around that bug?

ARI is a REST interface to Asterisk, it doesn't have anything to do with
this.

>
> Lack of full support for traversing nat makes pjsip worthless for a
> large number of users. And the whole point of realtime is to have all
> of the rt config fully dymanic.

I disagree that it makes it worthless for a large number of users. It's
only within the last few days that a few people have run into this
particular issue where they have a public IP address that is changing a
lot and PJSIP does not support changing it without a restart. If it were
a huge sweeping issue we'd be seeing it more often. If it continues to
show up a community member or us (heck maybe even myself in my spare
time) may look into implementing it.

>
> If ari cannot avoid that limitation, chan_sip should get full ongoing
> maintainance until pjsip is fixed.

The support level for chan_sip has already been changed and was
announced long ago. Patches will continue to be accepted for it and
community members can support it. We (Digium) are putting our effort
towards PJSIP.

--
Joshua Colp
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] PJSIP Stun/ICE

[Joshua Colp]

Bryant Zimmerman wrote:

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is
running the PJSIP Stack
It is registering to another asterisk 13 server that is on a Static IP
outside of the firewall at a different location (also on the PJSIP Stack).
How do we implement STUN/ICE on the server behind the dynamic Address.
It does not appear to be registering properly without knowing the NAT
pubic address. When I manually add external_media_address and
external_signaling_address to the pjsipconfig registration seams to
work, but knowing that the IP could change really means I need some kind
of STUN/ICE similar to what we ran with chan_sip.
I can find limited documentation on this, and what I have found does not
show how to set a stun server to make the ice_support field work on an
endpoint.
Can anyone advise where I could find an answer to this.
Thanks in advance for any ideas you can offer.
Bryant


The res_pjsip module does not currently support an auto-updating 
mechanism for the external signaling and media address information.


--
Joshua Colp
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] PJSIP Stun/ICE

[Joshua Colp]

Bryant Zimmerman wrote:

Joshua
Since there is no automated way currently built in to update the
external signaling and media address information.
Does the realtime pjsip support having the transport contexts section
being pulled from a database table?
I was thinking a cron script updating the table and forcing a reload
each time an IP address changed might a workable solution.


No, once loaded the transports can not be changed.

--
Joshua Colp
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] PJSIP Stun/ICE

[Bryant Zimmerman]

Joshua
  
 Since there is no automated way currently built in to update the external 
signaling and media address information.
 Does the realtime pjsip support having the transport contexts section 
being pulled from a database table?
 I was thinking a cron script updating the table and forcing a reload each 
time an IP address changed might a workable solution.
  
 Thanks
 Bryant
  


 From: "Joshua Colp" 
Sent: Tuesday, January 26, 2016 7:39 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] PJSIP Stun/ICE   
Bryant Zimmerman wrote:
> I have an asterisk 13 server behind NAT on a dynamic IP Address. It is
> running the PJSIP Stack
> It is registering to another asterisk 13 server that is on a Static IP
> outside of the firewall at a different location (also on the PJSIP 
Stack).
> How do we implement STUN/ICE on the server behind the dynamic Address.
> It does not appear to be registering properly without knowing the NAT
> pubic address. When I manually add external_media_address and
> external_signaling_address to the pjsipconfig registration seams to
> work, but knowing that the IP could change really means I need some kind
> of STUN/ICE similar to what we ran with chan_sip.
> I can find limited documentation on this, and what I have found does not
> show how to set a stun server to make the ice_support field work on an
> endpoint.
> Can anyone advise where I could find an answer to this.
> Thanks in advance for any ideas you can offer.
> Bryant

The res_pjsip module does not currently support an auto-updating
mechanism for the external signaling and media address information.

--
Joshua Colp
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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[asterisk-users] PJSIP Stun/ICE

[Bryant Zimmerman]

I have an asterisk 13 server behind NAT on a dynamic IP Address. It is 
running the PJSIP Stack
 It is registering to another asterisk 13 server that is on a Static IP 
outside of the firewall at a different location (also on the PJSIP Stack).
  
 How do we implement STUN/ICE on the server behind the dynamic Address. It 
does not appear to be registering properly without knowing the NAT pubic 
address.  When I manually add external_media_address and 
external_signaling_address to the pjsipconfig registration seams to work, 
but knowing that the IP could change really means I need some kind of 
STUN/ICE similar to what we ran with chan_sip. 
 I can find limited documentation on this, and what I have found does not 
show how to set a stun server to make the ice_support field work on an 
endpoint.
  
 Can anyone advise where I could find an answer to this.
  
 Thanks in advance for any ideas you can offer.
  
 Bryant

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   http://lists.digium.com/mailman/listinfo/asterisk-users