[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
2 Apr
2008
2 Apr
'08
4:29 p.m.
Hi, maybe this question is a bit off-topic so I'm sorry for that.
My question is about SIP providers using OpenSer that associate PSTN numbers
to their local clients (SIP accounts):
Usually the client must register to OpenSer in order to receive calls. Then it
will appear in location table with "Contact=sip:clientX@IP".
Suppose clientX has two PSTN numbers associated in a ENUM entry:
+34999000111
+34999000222
When anyone in PSTN world calls to +34999000222 the call will arrive to the
OpenSer from a gateway in an INVITE like:
INVITE sip:+34999000222@gateway SIP/2.0
To: <sip:+34999000222@gateway>
The OpenSer will do the lookup in location table and finally send this INVITE
to the clientX:
INVITE sip:clientX@IP_clientX SIP/2.0
To: <sip:+34999000222@gateway>
The info about the called PSTN number is just available in "To" header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
Thanks for any comment. Regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
2 Apr
2 Apr
4:49 p.m.
New subject: [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
Iñaki Baz Castillo writes:
When anyone in PSTN world calls to +34999000222 the
call will arrive to the
OpenSer from a gateway in an INVITE like:
INVITE sip:+34999000222@gateway SIP/2.0
To: <sip:+34999000222@gateway>
why gateway in the host part? isn't host more likely the ip or hostname
of your proxy?
The info about the called PSTN number is just
available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
your proxy could add other headers to the invite, such as diversion or
history-info, if your sip ua knows to look for them.
-- juha
5:15 p.m.
El Wednesday 02 April 2008 14:49:33 Juha Heinanen escribió:
Yes, that would be better than examining the "To" header, since "To"
could be
modified previously (forwarding and so...).
So in conclusion, it seems that there is not a extended way of handling it,
rigth?
Thanks a lot.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Iñaki Baz Castillo writes:
Yes, sorry, it was a failure of copy&paste ;)
When anyone in PSTN world calls to +34999000222
the call will arrive to
the OpenSer from a gateway in an INVITE like:
INVITE sip:+34999000222@gateway SIP/2.0
To: <sip:+34999000222@gateway>
why gateway in the host part? isn't host more likely the ip or hostname
of your proxy? The info about
the called PSTN number is just available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
your proxy could add other headers to the invite, such as diversion or
history-info, if your sip ua knows to look for them. 
5:35 p.m.
Hola Iñaki,
El 02/04/2008, a las 16:29, Iñaki Baz Castillo escribió:
Hi, maybe this question is a bit off-topic so I'm
sorry for that.
My question is about SIP providers using OpenSer that associate PSTN
numbers
to their local clients (SIP accounts):
Usually the client must register to OpenSer in order to receive
calls. Then it
will appear in location table with "Contact=sip:clientX@IP".
Suppose clientX has two PSTN numbers associated in a ENUM entry:
+34999000111
+34999000222
When anyone in PSTN world calls to +34999000222 the call will arrive
to the
OpenSer from a gateway in an INVITE like:
INVITE sip:+34999000222@gateway SIP/2.0
To: <sip:+34999000222@gateway>
The OpenSer will do the lookup in location table and finally send
this INVITE
to the clientX:
INVITE sip:clientX@IP_clientX SIP/2.0
To: <sip:+34999000222@gateway>
The info about the called PSTN number is just available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
You can also add multiple aliases to the same user. All INVITEs to
different aliases will be sent to the resolved user.
Saludos
JesusR.
------------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------------
5:40 p.m.
El Wednesday 02 April 2008 15:35:40 Jesus Rodriguez escribió:
Yes, but after the "lookup" the RURI username will be definitively the
username the client sent in the REGISTER's "Contact", so the INVITE will not
change, will it?
Thanks..
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
The info about
the called PSTN number is just available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
You can also add multiple aliases to the same user. All INVITEs to
different aliases will be sent to the resolved user.
5:48 p.m.
Hola Iñaki,
El 02/04/2008, a las 17:40, Iñaki Baz Castillo escribió:
El Wednesday 02 April 2008 15:35:40 Jesus Rodriguez
escribió:
Yes, but after the "lookup" the RURI username will be definitively the
username the client sent in the REGISTER's "Contact", so the INVITE
will not
change, will it?
The RURI will be changed by the registered Contact value. The To:
header is not modified.
Saludos
JesusR.
------------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------------
The info
about the called PSTN number is just available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
You can also add multiple aliases to the same user. All INVITEs to
different aliases will be sent to the resolved user.
6:01 p.m.
El Wednesday 02 April 2008 15:48:10 Jesus Rodriguez escribió:
Hola Iñaki,
El 02/04/2008, a las 17:40, Iñaki Baz Castillo escribió:
Exactly, but a client will have (probably) just one sip account registered, so
just one entry will appear in "location" table for him AoR.
Imagine a user "clientX@sip_proxy" using an Asterisk and registering with:
REGISTER sip:sip_proxy SIP/2.0
To: <sip:clientX@sip_proxy>
Contact: <sip:s@ip_asterisk>
It will appear in location table with "Contact=sip:clientX@IP".
Suppose clientX has two PSTN numbers associated in a ENUM entry:
+34999000111
+34999000222
When it receives a call from PSTN to +34999000222 the INVITE arriving to
Asterisk will be:
INVITE sip:s@ip_asterisk SIP/2.0
To: <sip:+34999000222@sip_proxy>
So the info about the real PSTN number dialed by the call originator just
remains in the "To" header. So the client (Asterisk) must read the
"To"
header in order to have a behaviour different for +34999000111 and
+34999000222 (or adding in OpenSer other custom header containing also dialed
number 999000222 as Juha suggests).
I can't understand how using aliases in OpenSer can help here. ¿?
Thanks a lot and regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
El Wednesday 02 April 2008 15:35:40 Jesus
Rodriguez escribió:
Yes, but after the "lookup" the RURI username will be definitively the
username the client sent in the REGISTER's "Contact", so the INVITE
will not
change, will it?
The RURI will be changed by the registered Contact value. The To:
header is not modified. The info
about the called PSTN number is just available in "To"
header, so a
way to get different behaviour for each associated PSTN number is
matching "To" URI.
Is common to do it? which other alternatives are there?
You can also add multiple aliases to the same user. All INVITEs to
different aliases will be sent to the resolved user.
6:21 p.m.
Hola Iñaki,
El 02/04/2008, a las 18:01, Iñaki Baz Castillo escribió:
El Wednesday 02 April 2008 15:48:10 Jesus Rodriguez
escribió:
Yes, in theses cases you have to relay on To: header value. The
problem is that not all "multiport" devices are "multi account" as
Asterisk is and you need to use aliases to send all different DIDs to
the same SIP account... some of these "multiport" devices can read the
To: value and send the call to the right place even registering only
one account.
Saludos
JesusR.
------------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------------
Hola Iñaki,
El 02/04/2008, a las 17:40, Iñaki Baz Castillo escribió:
Exactly, but a client will have (probably) just one sip account
registered, so
just one entry will appear in "location" table for him AoR.
Imagine a user "clientX@sip_proxy" using an Asterisk and registering
with:
REGISTER sip:sip_proxy SIP/2.0
To: <sip:clientX@sip_proxy>
Contact: <sip:s@ip_asterisk>
It will appear in location table with "Contact=sip:clientX@IP".
Suppose clientX has two PSTN numbers associated in a ENUM entry:
+34999000111
+34999000222
When it receives a call from PSTN to +34999000222 the INVITE
arriving to
Asterisk will be:
INVITE sip:s@ip_asterisk SIP/2.0
To: <sip:+34999000222@sip_proxy>
So the info about the real PSTN number dialed by the call
originator just
remains in the "To" header. So the client (Asterisk) must read the
"To"
header in order to have a behaviour different for +34999000111 and
+34999000222 (or adding in OpenSer other custom header containing
also dialed
number 999000222 as Juha suggests).
I can't understand how using aliases in OpenSer can help here. ¿? El Wednesday 02 April 2008 15:35:40 Jesus
Rodriguez escribió:
The RURI will be changed by the registered Contact value. The To:
header is not modified. > The info about the called PSTN number is just
available in "To"
> header, so a
> way to get different behaviour for each associated PSTN number is
> matching "To" URI.
> Is common to do it? which other alternatives are there?
You can also add multiple aliases to the same user. All INVITEs to
different aliases will be sent to the resolved user.
Yes, but after the "lookup" the RURI username will be definitively
the
username the client sent in the REGISTER's "Contact", so the INVITE
will not
change, will it?
6:30 p.m.
El Wednesday 02 April 2008 16:21:59 Jesus Rodriguez escribió:
Yes, in theses cases you have to relay on To: header
value. The
problem is that not all "multiport" devices are "multi account" as
Asterisk is and you need to use aliases to send all different DIDs to
the same SIP account... some of these "multiport" devices can read the
To: value and send the call to the right place even registering only
one account.
Ok, thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
24 Feb
24 Feb
9:56 a.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
Hi Iñaki,
did you find a way to solve this problem. I have the same problem with
devices that can't read the To: value.
For customers with 10/100 assigned number I routed the entire root number to
the customer's IP (loosing auth...), but for users with only one
supplementary number it is not practicable.
Has someone a suggestion for this problem?
Thanks to all.
Regards.
Antonio.
-----Original Message-----
From: users-bounces(a)lists.openser.org
[mailto:users-bounces@lists.openser.org]
On Behalf Of Iñaki Baz Castillo
Sent: Wednesday, April 02, 2008 6:30 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] [OT] How to handle different DID's in
incoming
calls for a registered client?
El Wednesday 02 April 2008 16:21:59 Jesus Rodriguez escribió:
Yes, in theses cases you have to relay on To:
header value. The
problem is that not all "multiport" devices are "multi account" as
Asterisk is and you need to use aliases to send all different DIDs to
the same SIP account... some of these "multiport" devices can read the
To: value and send the call to the right place even registering only
one account.
Ok, thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users 
10:17 a.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
25 Feb
25 Feb
11:35 a.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
Hello,
in kamailio/openser, the equvalents of the operations are (see inline):
On 02/24/2009 11:17 AM, olivier taylor wrote:
What I do in SER 0.96
Before any lookup("aliases")
avp_write("$ruri/username","s:RealRuri"); //to save the
Ruri
$avp(s:RealRuri) = $rU;
and just before sending the call to the user :
avp_pushto("$ruri/username", "s:RealRuri");
$rU = $avp(s:RealRuri);
You can then modify the Ruri as you want with prefix or strip if needed
strip/prefix are still there, however there are other features that can
help:
- string concatenation of variables using + (e.g., $var(x) = "001" + $rU;)
- transformations:
http://www.kamailio.org/dokuwiki/doku.php/transformations:devel
Cheers,
Daniel
Hope it helps
Olivier
Antonio Reale a écrit :
--
Daniel-Constantin Mierla
http://www.asipto.com
Hi Iñaki,
did you find a way to solve this problem. I have the same problem with
devices that can't read the To: value.
For customers with 10/100 assigned number I routed the entire root number to
the customer's IP (loosing auth...), but for users with only one
supplementary number it is not practicable.
Has someone a suggestion for this problem?
Thanks to all.
Regards.
Antonio.
------------------------------------------------------------------------
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -----Original Message-----
From: users-bounces(a)lists.openser.org
[mailto:users-bounces@lists.openser.org]
On Behalf Of Iñaki Baz Castillo
Sent: Wednesday, April 02, 2008 6:30 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] [OT] How to handle different DID's in
incoming
calls for a registered client?
El Wednesday 02 April 2008 16:21:59 Jesus Rodriguez escribió:
_______________________________________________
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Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
Yes, in theses cases you have to relay on To:
header value. The
problem is that not all "multiport" devices are "multi account" as
Asterisk is and you need to use aliases to send all different DIDs to
the same SIP account... some of these "multiport" devices can read the
To: value and send the call to the right place even registering only
one account.
Ok, thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
26 Feb
26 Feb
11:32 a.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
Thanks Olivier, thanks Daniel,
in this way it works. My idea was that rewrite RURI after aliases lookup
(that also rewrite the RURI) was not the best solution.
Thank you very much.
Regards,
Antonio.
------------------------------------------------------------------------
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
--
Daniel-Constantin Mierla
http://www.asipto.com
-----Original Message-----
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: Wednesday, February 25, 2009 11:36 AM
To: olivier(a)sunforyou.be
Cc: Antonio Reale; users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different
DID's in incoming calls for a registered client?
Hello,
in kamailio/openser, the equvalents of the operations are (see inline):
On 02/24/2009 11:17 AM, olivier taylor wrote:
number
What I do in SER 0.96
Before any lookup("aliases")
avp_write("$ruri/username","s:RealRuri"); //to save the
Ruri
$avp(s:RealRuri) = $rU;
and just before sending the call to the user :
avp_pushto("$ruri/username", "s:RealRuri");
$rU = $avp(s:RealRuri);
You can then modify the Ruri as you want with prefix or strip if needed
strip/prefix are still there, however there are other features that can
help:
- string concatenation of variables using + (e.g., $var(x) = "001" + $rU;)
- transformations:
http://www.kamailio.org/dokuwiki/doku.php/transformations:devel
Cheers,
Daniel
>
> Hope it helps
>
> Olivier
>
>
> Antonio Reale a écrit :
>> Hi Iñaki,
>> did you find a way to solve this problem. I have the same problem with
>> devices that can't read the To: value.
>> For customers with 10/100 assigned number I routed the entire root to
>> the customer's IP (loosing auth...), but for users with only one
>> supplementary number it is not practicable.
>> Has someone a suggestion for this problem?
>> Thanks to all.
>> Regards.
>>
>> Antonio.
>>
>>
>>> -----Original Message-----
>>> From: users-bounces(a)lists.openser.org
>>>
>> [mailto:users-bounces@lists.openser.org]
>>
>>> On Behalf Of Iñaki Baz Castillo
>>> Sent: Wednesday, April 02, 2008 6:30 PM
>>> To: users(a)lists.openser.org
>>> Subject: Re: [OpenSER-Users] [OT] How to handle different DID's in
>>>
>> incoming
>>
>>> calls for a registered client?
>>>
>>> El Wednesday 02 April 2008 16:21:59 Jesus Rodriguez escribió:
>>>
>>>> Yes, in theses cases you have to relay on To: header value. The
>>>> problem is that not all "multiport" devices are "multi
account" as
>>>> Asterisk is and you need to use aliases to send all different DIDs to
>>>> the same SIP account... some of these "multiport" devices can
read
the
> To: value
and send the call to the right place even registering only
> one account.
>
Ok, thanks.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
_______________________________________________
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Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
1:56 p.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
3:38 p.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
On 02/26/2009 02:56 PM, olivier taylor wrote:
It's a pleasure, I just gave an idea...
Daniel did the translation.
Just take care, lookup("location") also modify the ruri.
... also it can
add extra branches, so, in order to update the URI for
each branch you have to use branch_route.
Cheers,
Daniel
Olivier
Antonio Reale a écrit :
--
Daniel-Constantin Mierla
http://www.asipto.com
Thanks Olivier, thanks Daniel,
in this way it works. My idea was that rewrite RURI after aliases lookup
(that also rewrite the RURI) was not the best solution.
Thank you very much.
Regards,
Antonio.
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
------------------------------------------------------------------------
_______________________________________________
Kamailio (OpenSER) - Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
http://lists.openser-project.org/cgi-bin/mailman/listinfo/users -----Original Message-----
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: Wednesday, February 25, 2009 11:36 AM
To: olivier(a)sunforyou.be
Cc: Antonio Reale; users(a)lists.kamailio.org
Subject: Re: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different
DID's in incoming calls for a registered client?
Hello,
in kamailio/openser, the equvalents of the operations are (see inline):
On 02/24/2009 11:17 AM, olivier taylor wrote:
number
What I do in SER 0.96
Before any lookup("aliases")
avp_write("$ruri/username","s:RealRuri"); //to save the
Ruri
$avp(s:RealRuri) = $rU;
and just before sending the call to the user :
avp_pushto("$ruri/username", "s:RealRuri");
$rU = $avp(s:RealRuri);
You can then modify the Ruri as you want with
prefix or strip if needed
strip/prefix are still there, however there are other features that can
help:
- string concatenation of variables using + (e.g., $var(x) = "001" + $rU;)
- transformations:
http://www.kamailio.org/dokuwiki/doku.php/transformations:devel
Cheers,
Daniel
> Hope it helps
>
> Olivier
>
>
> Antonio Reale a écrit :
>
>> Hi Iñaki,
>> did you find a way to solve this problem. I have the same problem with
>> devices that can't read the To: value.
>> For customers with 10/100 assigned number I routed the entire root
>> to
>> the customer's IP (loosing auth...), but for users with only one
>> supplementary number it is not practicable.
>> Has someone a suggestion for this problem?
>> Thanks to all.
>> Regards.
>>
>> Antonio.
>>
>>
>>
>>> -----Original Message-----
>>> From: users-bounces(a)lists.openser.org
>>>
>>>
>> [mailto:users-bounces@lists.openser.org]
>>
>>
>>> On Behalf Of Iñaki Baz Castillo
>>> Sent: Wednesday, April 02, 2008 6:30 PM
>>> To: users(a)lists.openser.org
>>> Subject: Re: [OpenSER-Users] [OT] How to handle different DID's in
>>>
>>>
>> incoming
>>
>>
>>> calls for a registered client?
>>>
>>> El Wednesday 02 April 2008 16:21:59 Jesus Rodriguez escribió:
>>>
>>>
>>>> Yes, in theses cases you have to relay on To: header value. The
>>>> problem is that not all "multiport" devices are "multi
account" as
>>>> Asterisk is and you need to use aliases to send all different DIDs to
>>>> the same SIP account... some of these "multiport" devices can
read
>>>>
the
>>> To: value and send the call to the right place even registering only
>>> one account.
>>>
>>>
>> Ok, thanks.
>>
>>
>> --
>> Iñaki Baz Castillo
>> ibc(a)in.ilimit.es
>>
>> _______________________________________________
>> Users mailing list
>> Users(a)lists.openser.org
>> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>>
>>
> _______________________________________________
> Kamailio (OpenSER) - Users mailing list
> Users(a)lists.kamailio.org
> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
>
>
>
------------------------------------------------------------------------
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--
Daniel-Constantin Mierla
http://www.asipto.com
27 Feb
27 Feb
4:12 p.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
2009/2/26 Daniel-Constantin Mierla <miconda(a)gmail.com>om>:
... also it can add extra branches, so, in order to update the URI for each
branch you have to use branch_route.
You're right. Fixed...
Cheers,
Daniel
Thanks again.
Regards,
Antonio.
24 Feb
24 Feb
10:20 a.m.
New subject: [Kamailio-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?
2 Apr
2 Apr
6:29 p.m.
El Wednesday 02 April 2008 16:01:47 Iñaki Baz Castillo escribió:
Imagine a user "clientX@sip_proxy" using an
Asterisk and registering with:
REGISTER sip:sip_proxy SIP/2.0
To: <sip:clientX@sip_proxy>
Contact: <sip:s@ip_asterisk>
It will appear in location table with "Contact=sip:clientX@IP".
Sorry, I mean:
It will appear in location table with "Contact=sip:s@ip_asterisk".
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
5536
days inactive
5867
days old
17 comments
7 participants
participants (7)
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Antonio Reale -
Daniel-Constantin Mierla -
hh174 -
Iñaki Baz Castillo -
Jesus Rodriguez -
jh＠tutpro.com -
olivier taylor