SIP trunk provider lab

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SIP trunk provider lab

home-2
Hi all!
 
can anybody please point my in the right direction?
 
I'm trying to setup a lab for a possible [several pstn gw's]->[sip trunk DID] provider.
 
what I mean is basicly, incomming calls from [pstn gataways] should be dynamicaly routed based on registrations.
 
e.g.
 
 
if we have a SIP subscriber registered with a user name companyA which has 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly route all incomming calls to the "location" of the regisration.
 
Is this possbile? I mean is this possible to achive dynamicaly? I already can do it staticly, script based. If so please point modules that I should use.
 
Thank you.
 
 

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Re: SIP trunk provider lab

Iñaki Baz Castillo
El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
> if we have a SIP subscriber registered with a user name companyA which has
> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
> route all incomming calls to the "location" of the regisration.
>
> Is this possbile? I mean is this possible to achive dynamicaly?

Of course, use ENUM or dbaliases.


--
Iñaki Baz Castillo

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Re: SIP trunk provider lab

home-2
Hi,

thank you for your response!
correct me if I'm wrong, but aren't we going to loose dest phone number if
we use dbaliases?
Also please provide hints on ENUM, how do I use it for this purpose?

Thank you.

Elnour


----- Original Message -----
From: "Iñaki Baz Castillo" <[hidden email]>
To: <[hidden email]>
Sent: Saturday, March 14, 2009 3:59 PM
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab


> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>> if we have a SIP subscriber registered with a user name companyA which
>> has
>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>> route all incomming calls to the "location" of the regisration.
>>
>> Is this possbile? I mean is this possible to achive dynamicaly?
>
> Of course, use ENUM or dbaliases.
>
>
> --
> Iñaki Baz Castillo
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> !DSPAM:49bb9c46608611888415445!
>
>


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Re: SIP trunk provider lab

Iñaki Baz Castillo
El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
> Hi,
>
> thank you for your response!
> correct me if I'm wrong, but aren't we going to loose dest phone number if
> we use dbaliases?

Yes, the RURI will become the Contact of the regietered SIP account associated
to these phone numbers. But you can store the original RURI in an AVP, append
it to the request in "P-Called-Party-ID" (RFC 3455)...
Also, the called number remains in the To header (except in case it was
diverted previously, but it is not possible if you receive calls from a
gateway).


> Also please provide hints on ENUM, how do I use it for this purpose?

ENUM associates phone number is E.164 format to SIP accounts. OpenSIPS has a
function to perform a ENUM query (take a look to "enum" module in OpenSIPS).

--
Iñaki Baz Castillo

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Re: SIP trunk provider lab

home-2
Thank you.

I'll post here how it went.


----- Original Message -----
From: "Iñaki Baz Castillo" <[hidden email]>
To: <[hidden email]>
Sent: Saturday, March 14, 2009 4:37 PM
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab


> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>> Hi,
>>
>> thank you for your response!
>> correct me if I'm wrong, but aren't we going to loose dest phone number
>> if
>> we use dbaliases?
>
> Yes, the RURI will become the Contact of the regietered SIP account
> associated
> to these phone numbers. But you can store the original RURI in an AVP,
> append
> it to the request in "P-Called-Party-ID" (RFC 3455)...
> Also, the called number remains in the To header (except in case it was
> diverted previously, but it is not possible if you receive calls from a
> gateway).
>
>
>> Also please provide hints on ENUM, how do I use it for this purpose?
>
> ENUM associates phone number is E.164 format to SIP accounts. OpenSIPS has
> a
> function to perform a ENUM query (take a look to "enum" module in
> OpenSIPS).
>
> --
> Iñaki Baz Castillo
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> !DSPAM:49bba51f608618071319683!
>
>


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Re: SIP trunk provider lab

home-2
In reply to this post by Iñaki Baz Castillo
Hi Iñaki,

This might be off topic, but by any chance, do you know a way to make Cisco
Call Manager Express use information form the 'To' header to route the
calls..
And as a side none, in order to use 'P-Called-Party-ID' in CCME we have to
have Cisco Unified Border Element license.

Thank you.

Elnour

----- Original Message -----
From: "Iñaki Baz Castillo" <[hidden email]>
To: <[hidden email]>
Sent: Saturday, March 14, 2009 4:37 PM
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab


> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>> Hi,
>>
>> thank you for your response!
>> correct me if I'm wrong, but aren't we going to loose dest phone number
>> if
>> we use dbaliases?
>
> Yes, the RURI will become the Contact of the regietered SIP account
> associated
> to these phone numbers. But you can store the original RURI in an AVP,
> append
> it to the request in "P-Called-Party-ID" (RFC 3455)...
> Also, the called number remains in the To header (except in case it was
> diverted previously, but it is not possible if you receive calls from a
> gateway).
>
>
>> Also please provide hints on ENUM, how do I use it for this purpose?
>
> ENUM associates phone number is E.164 format to SIP accounts. OpenSIPS has
> a
> function to perform a ENUM query (take a look to "enum" module in
> OpenSIPS).
>
> --
> Iñaki Baz Castillo
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> !DSPAM:49bba51f608618071319683!
>
>


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Re: SIP trunk provider lab

Iñaki Baz Castillo
El Domingo, 15 de Marzo de 2009, [hidden email] escribió:
> Hi Iñaki,
>
> This might be off topic, but by any chance, do you know a way to make Cisco
> Call Manager Express use information form the 'To' header to route the
> calls..
> And as a side none, in order to use 'P-Called-Party-ID' in CCME we have to
> have Cisco Unified Border Element license.

No idea sorry.

--
Iñaki Baz Castillo

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Re: SIP trunk provider lab

Jesus Rodriguez
In reply to this post by home-2
Hello,


> Hi Iñaki,
>
> This might be off topic, but by any chance, do you know a way to  
> make Cisco
> Call Manager Express use information form the 'To' header to route the
> calls..
> And as a side none, in order to use 'P-Called-Party-ID' in CCME we  
> have to
> have Cisco Unified Border Element license.


About the To: header, i know is possible to do it with some TCL  
scripting, but never tried myself:

http://www.cisco.com/en/US/docs/ios/12_3t/12_3t2/feature/guide/gt_tcl.html

Saludos
JesusR.



> ----- Original Message -----
> From: "Iñaki Baz Castillo" <[hidden email]>
> To: <[hidden email]>
> Sent: Saturday, March 14, 2009 4:37 PM
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
>
>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>> Hi,
>>>
>>> thank you for your response!
>>> correct me if I'm wrong, but aren't we going to loose dest phone  
>>> number
>>> if
>>> we use dbaliases?
>>
>> Yes, the RURI will become the Contact of the regietered SIP account
>> associated
>> to these phone numbers. But you can store the original RURI in an  
>> AVP,
>> append
>> it to the request in "P-Called-Party-ID" (RFC 3455)...
>> Also, the called number remains in the To header (except in case it  
>> was
>> diverted previously, but it is not possible if you receive calls  
>> from a
>> gateway).
>>
>>
>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>
>> ENUM associates phone number is E.164 format to SIP accounts.  
>> OpenSIPS has
>> a
>> function to perform a ENUM query (take a look to "enum" module in
>> OpenSIPS).
>>
>> --
>> Iñaki Baz Castillo
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> !DSPAM:49bba51f608618071319683!
>>
>>
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users





Saludos
JesusR.

------------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
[hidden email]
http://www.voztele.com
Tel. 902360305
-------------------------------------





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Re: SIP trunk provider lab

home-2
Hi,

Thank you. I'll look into it.

Regards,
Elnour

----- Original Message -----
From: "Jesus Rodriguez" <[hidden email]>
To: <[hidden email]>
Cc: <[hidden email]>
Sent: Sunday, March 15, 2009 11:45 PM
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab


> Hello,
>
>
>> Hi Iñaki,
>>
>> This might be off topic, but by any chance, do you know a way to  make
>> Cisco
>> Call Manager Express use information form the 'To' header to route the
>> calls..
>> And as a side none, in order to use 'P-Called-Party-ID' in CCME we  have
>> to
>> have Cisco Unified Border Element license.
>
>
> About the To: header, i know is possible to do it with some TCL
> scripting, but never tried myself:
>
> http://www.cisco.com/en/US/docs/ios/12_3t/12_3t2/feature/guide/gt_tcl.html
>
> Saludos
> JesusR.
>
>
>
>> ----- Original Message -----
>> From: "Iñaki Baz Castillo" <[hidden email]>
>> To: <[hidden email]>
>> Sent: Saturday, March 14, 2009 4:37 PM
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>>
>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>> Hi,
>>>>
>>>> thank you for your response!
>>>> correct me if I'm wrong, but aren't we going to loose dest phone
>>>> number
>>>> if
>>>> we use dbaliases?
>>>
>>> Yes, the RURI will become the Contact of the regietered SIP account
>>> associated
>>> to these phone numbers. But you can store the original RURI in an  AVP,
>>> append
>>> it to the request in "P-Called-Party-ID" (RFC 3455)...
>>> Also, the called number remains in the To header (except in case it  was
>>> diverted previously, but it is not possible if you receive calls  from a
>>> gateway).
>>>
>>>
>>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>>
>>> ENUM associates phone number is E.164 format to SIP accounts.  OpenSIPS
>>> has
>>> a
>>> function to perform a ENUM query (take a look to "enum" module in
>>> OpenSIPS).
>>>
>>> --
>>> Iñaki Baz Castillo
>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
> Saludos
> JesusR.
>
> ------------------------------------
> Jesus Rodriguez
> VozTelecom Sistemas, S.L.
> [hidden email]
> http://www.voztele.com
> Tel. 902360305
> -------------------------------------
>
>
>
>
>
> !DSPAM:49bd5b83608618629088827!
>
>


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Re: SIP trunk provider lab

Bogdan-Andrei Iancu
In reply to this post by home-2
Hi Elnor,

[hidden email] wrote:
> Hi,
>
> thank you for your response!
> correct me if I'm wrong, but aren't we going to loose dest phone number if
> we use dbaliases?
>  
Not necessary - depends of how you define the alias. For ex, you can do:
    DID@server -> DID@ trunk

preserve the username part and change only the domain to point to the trunk.


Also, if you have blocks of DIDs which are easy to detect based on
regexp, you may consider using the dialplan module:
       http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html

See example 1.4.1.2 -
http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187

Regards,
Bogdan

> Also please provide hints on ENUM, how do I use it for this purpose?
>
> Thank you.
>
> Elnour
>
>
> ----- Original Message -----
> From: "Iñaki Baz Castillo" <[hidden email]>
> To: <[hidden email]>
> Sent: Saturday, March 14, 2009 3:59 PM
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
>
>  
>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>    
>>> if we have a SIP subscriber registered with a user name companyA which
>>> has
>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>> route all incomming calls to the "location" of the regisration.
>>>
>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>      
>> Of course, use ENUM or dbaliases.
>>
>>
>> --
>> Iñaki Baz Castillo
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> !DSPAM:49bb9c46608611888415445!
>>
>>
>>    
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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Re: SIP trunk provider lab

Robert Borz
Hi Bogdan,

could you be more specific here, please?

I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:

[hidden email]
[hidden email]
[hidden email] -- forward to ---> [hidden email]
...
[hidden email]

Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.

How can I achieve this behaviour? Thanks a lot!


Regards,
Robert


-----Original Message-----
From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, March 16, 2009 9:14 AM
To: [hidden email]
Cc: [hidden email]
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab

Hi Elnor,

[hidden email] wrote:
> Hi,
>
> thank you for your response!
> correct me if I'm wrong, but aren't we going to loose dest phone number if
> we use dbaliases?
>  
Not necessary - depends of how you define the alias. For ex, you can do:
    DID@server -> DID@ trunk

preserve the username part and change only the domain to point to the trunk.


Also, if you have blocks of DIDs which are easy to detect based on
regexp, you may consider using the dialplan module:
       http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html

See example 1.4.1.2 -
http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187

Regards,
Bogdan

> Also please provide hints on ENUM, how do I use it for this purpose?
>
> Thank you.
>
> Elnour
>
>
> ----- Original Message -----
> From: "Iñaki Baz Castillo" <[hidden email]>
> To: <[hidden email]>
> Sent: Saturday, March 14, 2009 3:59 PM
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
>
>  
>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>    
>>> if we have a SIP subscriber registered with a user name companyA which
>>> has
>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>> route all incomming calls to the "location" of the regisration.
>>>
>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>      
>> Of course, use ENUM or dbaliases.
>>
>>
>> --
>> Iñaki Baz Castillo
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> !DSPAM:49bb9c46608611888415445!
>>
>>
>>    
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


_______________________________________________
Users mailing list
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Re: SIP trunk provider lab

Bogdan-Andrei Iancu
Hi Robert,

you mean, at the end, you want to sent it to [hidden email] phone, but
with [hidden email] in RURI, right ?

Regards,
Bogdan

Robert Borz wrote:

> Hi Bogdan,
>
> could you be more specific here, please?
>
> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>
> [hidden email]
> [hidden email]
> [hidden email] -- forward to ---> [hidden email]
> ...
> [hidden email]
>
> Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.
>
> How can I achieve this behaviour? Thanks a lot!
>
>
> Regards,
> Robert
>
>
> -----Original Message-----
> From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
> Sent: Monday, March 16, 2009 9:14 AM
> To: [hidden email]
> Cc: [hidden email]
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> Hi Elnor,
>
> [hidden email] wrote:
>  
>> Hi,
>>
>> thank you for your response!
>> correct me if I'm wrong, but aren't we going to loose dest phone number if
>> we use dbaliases?
>>  
>>    
> Not necessary - depends of how you define the alias. For ex, you can do:
>     DID@server -> DID@ trunk
>
> preserve the username part and change only the domain to point to the trunk.
>
>
> Also, if you have blocks of DIDs which are easy to detect based on
> regexp, you may consider using the dialplan module:
>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>
> See example 1.4.1.2 -
> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>
> Regards,
> Bogdan
>
>  
>> Also please provide hints on ENUM, how do I use it for this purpose?
>>
>> Thank you.
>>
>> Elnour
>>
>>
>> ----- Original Message -----
>> From: "Iñaki Baz Castillo" <[hidden email]>
>> To: <[hidden email]>
>> Sent: Saturday, March 14, 2009 3:59 PM
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>>
>>  
>>    
>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>    
>>>      
>>>> if we have a SIP subscriber registered with a user name companyA which
>>>> has
>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>> route all incomming calls to the "location" of the regisration.
>>>>
>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>      
>>>>        
>>> Of course, use ENUM or dbaliases.
>>>
>>>
>>> --
>>> Iñaki Baz Castillo
>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> !DSPAM:49bb9c46608611888415445!
>>>
>>>
>>>    
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>  
>>    
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>  


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Re: SIP trunk provider lab

Robert Borz
Hi Bogdan,

yes, that's exactly what I want... :-)


Regards,
Robert.

-----Original Message-----
From: [hidden email] [mailto:[hidden email]]
Sent: Tuesday, April 21, 2009 11:38 AM
To: [hidden email]
Cc: [hidden email]; [hidden email]
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab

Hi Robert,

you mean, at the end, you want to sent it to [hidden email] phone, but
with [hidden email] in RURI, right ?

Regards,
Bogdan

Robert Borz wrote:

> Hi Bogdan,
>
> could you be more specific here, please?
>
> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>
> [hidden email]
> [hidden email]
> [hidden email] -- forward to ---> [hidden email]
> ...
> [hidden email]
>
> Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.
>
> How can I achieve this behaviour? Thanks a lot!
>
>
> Regards,
> Robert
>
>
> -----Original Message-----
> From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
> Sent: Monday, March 16, 2009 9:14 AM
> To: [hidden email]
> Cc: [hidden email]
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> Hi Elnor,
>
> [hidden email] wrote:
>  
>> Hi,
>>
>> thank you for your response!
>> correct me if I'm wrong, but aren't we going to loose dest phone number if
>> we use dbaliases?
>>  
>>    
> Not necessary - depends of how you define the alias. For ex, you can do:
>     DID@server -> DID@ trunk
>
> preserve the username part and change only the domain to point to the trunk.
>
>
> Also, if you have blocks of DIDs which are easy to detect based on
> regexp, you may consider using the dialplan module:
>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>
> See example 1.4.1.2 -
> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>
> Regards,
> Bogdan
>
>  
>> Also please provide hints on ENUM, how do I use it for this purpose?
>>
>> Thank you.
>>
>> Elnour
>>
>>
>> ----- Original Message -----
>> From: "Iñaki Baz Castillo" <[hidden email]>
>> To: <[hidden email]>
>> Sent: Saturday, March 14, 2009 3:59 PM
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>>
>>  
>>    
>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>    
>>>      
>>>> if we have a SIP subscriber registered with a user name companyA which
>>>> has
>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>> route all incomming calls to the "location" of the regisration.
>>>>
>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>      
>>>>        
>>> Of course, use ENUM or dbaliases.
>>>
>>>
>>> --
>>> Iñaki Baz Castillo
>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> !DSPAM:49bb9c46608611888415445!
>>>
>>>
>>>    
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>  
>>    
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>  


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: SIP trunk provider lab

Bogdan-Andrei Iancu
What you can do is:

1) before applying aliases, save the RURI into an AVP
2) do alias ->userXY, do whatever other routing stuff, including
lookup("location");
3) just before sending out the request, do: if $du (destination URI) is
empty, copy the current RURI into $du ($du = $ru); (id $du already set,
skip that step). After that, copy the stored AVP into RURI and send it out.

more or less you will the desitnation URI to point to userXY and put in
RURI the alias stuff..

Regards
Bogdan

Robert Borz wrote:

> Hi Bogdan,
>
> yes, that's exactly what I want... :-)
>
>
> Regards,
> Robert.
>
> -----Original Message-----
> From: [hidden email] [mailto:[hidden email]]
> Sent: Tuesday, April 21, 2009 11:38 AM
> To: [hidden email]
> Cc: [hidden email]; [hidden email]
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> Hi Robert,
>
> you mean, at the end, you want to sent it to [hidden email] phone, but
> with [hidden email] in RURI, right ?
>
> Regards,
> Bogdan
>
> Robert Borz wrote:
>  
>> Hi Bogdan,
>>
>> could you be more specific here, please?
>>
>> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>>
>> [hidden email]
>> [hidden email]
>> [hidden email] -- forward to ---> [hidden email]
>> ...
>> [hidden email]
>>
>> Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.
>>
>> How can I achieve this behaviour? Thanks a lot!
>>
>>
>> Regards,
>> Robert
>>
>>
>> -----Original Message-----
>> From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
>> Sent: Monday, March 16, 2009 9:14 AM
>> To: [hidden email]
>> Cc: [hidden email]
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>> Hi Elnor,
>>
>> [hidden email] wrote:
>>  
>>    
>>> Hi,
>>>
>>> thank you for your response!
>>> correct me if I'm wrong, but aren't we going to loose dest phone number if
>>> we use dbaliases?
>>>  
>>>    
>>>      
>> Not necessary - depends of how you define the alias. For ex, you can do:
>>     DID@server -> DID@ trunk
>>
>> preserve the username part and change only the domain to point to the trunk.
>>
>>
>> Also, if you have blocks of DIDs which are easy to detect based on
>> regexp, you may consider using the dialplan module:
>>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>>
>> See example 1.4.1.2 -
>> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>>
>> Regards,
>> Bogdan
>>
>>  
>>    
>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>>
>>> Thank you.
>>>
>>> Elnour
>>>
>>>
>>> ----- Original Message -----
>>> From: "Iñaki Baz Castillo" <[hidden email]>
>>> To: <[hidden email]>
>>> Sent: Saturday, March 14, 2009 3:59 PM
>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>
>>>
>>>  
>>>    
>>>      
>>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>>    
>>>>      
>>>>        
>>>>> if we have a SIP subscriber registered with a user name companyA which
>>>>> has
>>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>>> route all incomming calls to the "location" of the regisration.
>>>>>
>>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>>      
>>>>>        
>>>>>          
>>>> Of course, use ENUM or dbaliases.
>>>>
>>>>
>>>> --
>>>> Iñaki Baz Castillo
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> [hidden email]
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> !DSPAM:49bb9c46608611888415445!
>>>>
>>>>
>>>>    
>>>>      
>>>>        
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>  
>>>    
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>  
>>    
>
>
>  


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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|

Re: SIP trunk provider lab

Robert Borz
Hi Bogdan,

I tried it and it works great! Thanks again for your help. :-)

But one thing... how can I achieve this the other way around?
[hidden email] should be able to place calls, which should look as originated from one of its "alias" accounts.

I think it must be as easy as your workaround... but I'm currently a little lost. :-/ Just not my day...


Regards,
Robert

-----Original Message-----
From: [hidden email] [mailto:[hidden email]]
Sent: Tuesday, April 21, 2009 11:51 AM
To: [hidden email]
Cc: [hidden email]; [hidden email]
Subject: Re: [OpenSIPS-Users] SIP trunk provider lab

What you can do is:

1) before applying aliases, save the RURI into an AVP
2) do alias ->userXY, do whatever other routing stuff, including
lookup("location");
3) just before sending out the request, do: if $du (destination URI) is
empty, copy the current RURI into $du ($du = $ru); (id $du already set,
skip that step). After that, copy the stored AVP into RURI and send it out.

more or less you will the desitnation URI to point to userXY and put in
RURI the alias stuff..

Regards
Bogdan

Robert Borz wrote:

> Hi Bogdan,
>
> yes, that's exactly what I want... :-)
>
>
> Regards,
> Robert.
>
> -----Original Message-----
> From: [hidden email] [mailto:[hidden email]]
> Sent: Tuesday, April 21, 2009 11:38 AM
> To: [hidden email]
> Cc: [hidden email]; [hidden email]
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> Hi Robert,
>
> you mean, at the end, you want to sent it to [hidden email] phone, but
> with [hidden email] in RURI, right ?
>
> Regards,
> Bogdan
>
> Robert Borz wrote:
>  
>> Hi Bogdan,
>>
>> could you be more specific here, please?
>>
>> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>>
>> [hidden email]
>> [hidden email]
>> [hidden email] -- forward to ---> [hidden email]
>> ...
>> [hidden email]
>>
>> Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.
>>
>> How can I achieve this behaviour? Thanks a lot!
>>
>>
>> Regards,
>> Robert
>>
>>
>> -----Original Message-----
>> From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
>> Sent: Monday, March 16, 2009 9:14 AM
>> To: [hidden email]
>> Cc: [hidden email]
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>> Hi Elnor,
>>
>> [hidden email] wrote:
>>  
>>    
>>> Hi,
>>>
>>> thank you for your response!
>>> correct me if I'm wrong, but aren't we going to loose dest phone number if
>>> we use dbaliases?
>>>  
>>>    
>>>      
>> Not necessary - depends of how you define the alias. For ex, you can do:
>>     DID@server -> DID@ trunk
>>
>> preserve the username part and change only the domain to point to the trunk.
>>
>>
>> Also, if you have blocks of DIDs which are easy to detect based on
>> regexp, you may consider using the dialplan module:
>>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>>
>> See example 1.4.1.2 -
>> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>>
>> Regards,
>> Bogdan
>>
>>  
>>    
>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>>
>>> Thank you.
>>>
>>> Elnour
>>>
>>>
>>> ----- Original Message -----
>>> From: "Iñaki Baz Castillo" <[hidden email]>
>>> To: <[hidden email]>
>>> Sent: Saturday, March 14, 2009 3:59 PM
>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>
>>>
>>>  
>>>    
>>>      
>>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>>    
>>>>      
>>>>        
>>>>> if we have a SIP subscriber registered with a user name companyA which
>>>>> has
>>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>>> route all incomming calls to the "location" of the regisration.
>>>>>
>>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>>      
>>>>>        
>>>>>          
>>>> Of course, use ENUM or dbaliases.
>>>>
>>>>
>>>> --
>>>> Iñaki Baz Castillo
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> [hidden email]
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> !DSPAM:49bb9c46608611888415445!
>>>>
>>>>
>>>>    
>>>>      
>>>>        
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>  
>>>    
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>  
>>    
>
>
>  


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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|

Re: SIP trunk provider lab

Bogdan-Andrei Iancu
Hi Robert,

Robert Borz wrote:
> Hi Bogdan,
>
> I tried it and it works great! Thanks again for your help. :-)
>  
glad it worked.
> But one thing... how can I achieve this the other way around?
> [hidden email] should be able to place calls, which should look as originated from one of its "alias" accounts.
>
> I think it must be as easy as your workaround... but I'm currently a little lost. :-/ Just not my day...
>  
First you need somehow to determine what alias you want to use as new
identity. After that, use uac_replace_from() to change the From hdr (if
you have SIP destinations) or try using PAI/RPID hdr if the destination
is a GW.

Regards,
Bogdan

>
> Regards,
> Robert
>
> -----Original Message-----
> From: [hidden email] [mailto:[hidden email]]
> Sent: Tuesday, April 21, 2009 11:51 AM
> To: [hidden email]
> Cc: [hidden email]; [hidden email]
> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>
> What you can do is:
>
> 1) before applying aliases, save the RURI into an AVP
> 2) do alias ->userXY, do whatever other routing stuff, including
> lookup("location");
> 3) just before sending out the request, do: if $du (destination URI) is
> empty, copy the current RURI into $du ($du = $ru); (id $du already set,
> skip that step). After that, copy the stored AVP into RURI and send it out.
>
> more or less you will the desitnation URI to point to userXY and put in
> RURI the alias stuff..
>
> Regards
> Bogdan
>
> Robert Borz wrote:
>  
>> Hi Bogdan,
>>
>> yes, that's exactly what I want... :-)
>>
>>
>> Regards,
>> Robert.
>>
>> -----Original Message-----
>> From: [hidden email] [mailto:[hidden email]]
>> Sent: Tuesday, April 21, 2009 11:38 AM
>> To: [hidden email]
>> Cc: [hidden email]; [hidden email]
>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>
>> Hi Robert,
>>
>> you mean, at the end, you want to sent it to [hidden email] phone, but
>> with [hidden email] in RURI, right ?
>>
>> Regards,
>> Bogdan
>>
>> Robert Borz wrote:
>>  
>>    
>>> Hi Bogdan,
>>>
>>> could you be more specific here, please?
>>>
>>> I want to setup a similar configuration. I want to forward let's say 300 numbers (sip accounts) to a single SIP account without loosing the R-URI, because the box receiving the forwarded calls should be able to distinct which number was dialed. What do I have to insert in the alias table to do this:
>>>
>>> [hidden email]
>>> [hidden email]
>>> [hidden email] -- forward to ---> [hidden email]
>>> ...
>>> [hidden email]
>>>
>>> Another thing I'm currently completely lost in is how to handle the outgoing part. [hidden email] should be able to place outgoing calls with the originator set to one of the [hidden email] ... [hidden email] addresses.
>>>
>>> How can I achieve this behaviour? Thanks a lot!
>>>
>>>
>>> Regards,
>>> Robert
>>>
>>>
>>> -----Original Message-----
>>> From: [hidden email] [mailto:[hidden email]] On Behalf Of Bogdan-Andrei Iancu
>>> Sent: Monday, March 16, 2009 9:14 AM
>>> To: [hidden email]
>>> Cc: [hidden email]
>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>
>>> Hi Elnor,
>>>
>>> [hidden email] wrote:
>>>  
>>>    
>>>      
>>>> Hi,
>>>>
>>>> thank you for your response!
>>>> correct me if I'm wrong, but aren't we going to loose dest phone number if
>>>> we use dbaliases?
>>>>  
>>>>    
>>>>      
>>>>        
>>> Not necessary - depends of how you define the alias. For ex, you can do:
>>>     DID@server -> DID@ trunk
>>>
>>> preserve the username part and change only the domain to point to the trunk.
>>>
>>>
>>> Also, if you have blocks of DIDs which are easy to detect based on
>>> regexp, you may consider using the dialplan module:
>>>        http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html
>>>
>>> See example 1.4.1.2 -
>>> http://www.opensips.org/html/docs/modules/1.4.x/dialplan.html#id227187
>>>
>>> Regards,
>>> Bogdan
>>>
>>>  
>>>    
>>>      
>>>> Also please provide hints on ENUM, how do I use it for this purpose?
>>>>
>>>> Thank you.
>>>>
>>>> Elnour
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: "Iñaki Baz Castillo" <[hidden email]>
>>>> To: <[hidden email]>
>>>> Sent: Saturday, March 14, 2009 3:59 PM
>>>> Subject: Re: [OpenSIPS-Users] SIP trunk provider lab
>>>>
>>>>
>>>>  
>>>>    
>>>>      
>>>>        
>>>>> El Sábado, 14 de Marzo de 2009, [hidden email] escribió:
>>>>>    
>>>>>      
>>>>>        
>>>>>          
>>>>>> if we have a SIP subscriber registered with a user name companyA which
>>>>>> has
>>>>>> 8 DID phone lines from PSTN say 5551001-08, our setup should dynamicaly
>>>>>> route all incomming calls to the "location" of the regisration.
>>>>>>
>>>>>> Is this possbile? I mean is this possible to achive dynamicaly?
>>>>>>      
>>>>>>        
>>>>>>          
>>>>>>            
>>>>> Of course, use ENUM or dbaliases.
>>>>>
>>>>>
>>>>> --
>>>>> Iñaki Baz Castillo
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> [hidden email]
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> !DSPAM:49bb9c46608611888415445!
>>>>>
>>>>>
>>>>>    
>>>>>      
>>>>>        
>>>>>          
>>>> _______________________________________________
>>>> Users mailing list
>>>> [hidden email]
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>  
>>>>    
>>>>      
>>>>        
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>  
>>>    
>>>      
>>  
>>    
>
>
>  


_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users