12 Feb
2008
12 Feb
'08
4:51 p.m.
Hi All
I am trying to achieve the following
1. User Calls a number
* If the number is 00+all numbers go to PSTN -> Else check if the
user is existent in our SIP Proxy user pool and act accordingly
What I achieved so far is the following
2. User Calls a number
* Route checks if the number is online if yes the call is routed
to him (route[1]) if not it send the call to PSTN if it is 00+[0-9]
route [2]. This works fine for 00+[0-9] number however if 123456 is
called and the user 123456 is not online, the call goes down through the
conditional statements because it does not meet 00+[0-9] and I get an
error "To Many Hops"
Seems I messed up somewhere, can anyone help me fix this
http://pastebin.com/m50328c12
Thanks
12 Feb
12 Feb
5:01 p.m.
On Tuesday 12 February 2008 15:51:34 Ali Jawad wrote:
Hi All
I am trying to achieve the following
1. User Calls a number
* If the number is 00+all numbers go to PSTN -> Else check if the
user is existent in our SIP Proxy user pool and act accordingly
What I achieved so far is the following
2. User Calls a number
* Route checks if the number is online if yes the call is routed
to him (route[1]) if not it send the call to PSTN if it is 00+[0-9]
route [2]. This works fine for 00+[0-9] number however if 123456 is
called and the user 123456 is not online, the call goes down through the
conditional statements because it does not meet 00+[0-9] and I get an
error "To Many Hops"
Use:
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # If user is online call him:
route(CALL_USER);
else
route(VOICEMAIL); # Or send 404
}
In this way you will avoid sending out a call for an online user that is not
registered.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
5:12 p.m.
Thank you for your reply
I commented out my section and replaced it with
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # The user is online
{
route(1);
exit;
}
else
{
route(2);
exit;
}
}
Now if I call a PSTN number I get too many hops, and if I do a local call to a user online
it does not ring at the other end.
Thank you.
-----Original Message-----
From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf
Of Iñaki Baz Castillo
Sent: Tuesday, February 12, 2008 5:02 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Routing Problem "To Many Hops"
On Tuesday 12 February 2008 15:51:34 Ali Jawad wrote:
Hi All
I am trying to achieve the following
1. User Calls a number
* If the number is 00+all numbers go to PSTN -> Else check if the
user is existent in our SIP Proxy user pool and act accordingly
What I achieved so far is the following
2. User Calls a number
* Route checks if the number is online if yes the call is routed
to him (route[1]) if not it send the call to PSTN if it is 00+[0-9]
route [2]. This works fine for 00+[0-9] number however if 123456 is
called and the user 123456 is not online, the call goes down through the
conditional statements because it does not meet 00+[0-9] and I get an
error "To Many Hops"
Use:
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # If user is online call him:
route(CALL_USER);
else
route(VOICEMAIL); # Or send 404
}
In this way you will avoid sending out a call for an online user that is not
registered.
--
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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5:19 p.m.
Dear Inaki
Thanks for your help..I had the if statements wrong..can you please help me on how to send
a 404 statement ?
Thank you.
-----Original Message-----
From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf
Of Ali Jawad
Sent: Tuesday, February 12, 2008 5:12 PM
To: Iñaki Baz Castillo; users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Routing Problem "To Many Hops"
Thank you for your reply
I commented out my section and replaced it with
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # The user is online
{
route(1);
exit;
}
else
{
route(2);
exit;
}
}
Now if I call a PSTN number I get too many hops, and if I do a local call to a user online
it does not ring at the other end.
Thank you.
-----Original Message-----
From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf
Of Iñaki Baz Castillo
Sent: Tuesday, February 12, 2008 5:02 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Routing Problem "To Many Hops"
On Tuesday 12 February 2008 15:51:34 Ali Jawad wrote:
Hi All
I am trying to achieve the following
1. User Calls a number
* If the number is 00+all numbers go to PSTN -> Else check if the
user is existent in our SIP Proxy user pool and act accordingly
What I achieved so far is the following
2. User Calls a number
* Route checks if the number is online if yes the call is routed
to him (route[1]) if not it send the call to PSTN if it is 00+[0-9]
route [2]. This works fine for 00+[0-9] number however if 123456 is
called and the user 123456 is not online, the call goes down through the
conditional statements because it does not meet 00+[0-9] and I get an
error "To Many Hops"
Use:
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # If user is online call him:
route(CALL_USER);
else
route(VOICEMAIL); # Or send 404
}
In this way you will avoid sending out a call for an online user that is not
registered.
--
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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5:28 p.m.
On Tuesday 12 February 2008 16:19:38 Ali Jawad wrote:
Dear Inaki
Thanks for your help..I had the if statements wrong..can you please help me
on how to send a 404 statement ?
http://www.openser.org/docs/modules/1.3.x/sl.html
But agian the same: Please, read the doc, read the cook-book, the
pseudovariables page, the modules doc...
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
13 Feb
13 Feb
7:50 a.m.
Iñaki Baz Castillo wrote:
On Tuesday 12 February 2008 16:19:38 Ali Jawad wrote:
To be fair to Ali and all others in his predicament, OpenSER
documentation - as much as has been put into the effort - *is* rather
recondite and abstruse. It provides a lot of specification on how to
accomplish particular tasks, but not really a whole lot of high-level
conceptual overview or theoretical introduction to the mechanics of the
engine, or why you would want to do various particular things.
OpenSER is wonderful, but it's one of the harder things to "get into" in
the open-source VoIP space because its documentation is rather lopsided
in this regard.
[And yes, I realise that such a gripe is best addressed by making a
thoughtful and original contribution to the effort - that's why there's
a wiki, right?]
--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
Dear Inaki
Thanks for your help..I had the if statements wrong..can you please help me
on how to send a 404 statement ?
http://www.openser.org/docs/modules/1.3.x/sl.html
But agian the same: Please, read the doc, read the cook-book, the
pseudovariables page, the modules doc... 
11:16 a.m.
On Wednesday 13 February 2008, Alex Balashov wrote:
To be fair to Ali and all others in his predicament,
OpenSER
documentation - as much as has been put into the effort - *is* rather
recondite and abstruse. It provides a lot of specification on how to
accomplish particular tasks, but not really a whole lot of high-level
conceptual overview or theoretical introduction to the mechanics of the
engine, or why you would want to do various particular things.
OpenSER is wonderful, but it's one of the harder things to "get into" in
the open-source VoIP space because its documentation is rather lopsided
in this regard.
Hi Alex,
yes, you're right. I've added a link to the Rome admin course materials on the
Wiki which has some information about the general configuration, and will ask
Bogdan about to provide some other slides about the internal structure from
the programming course.
[And yes, I realise that such a gripe is best
addressed by making a
thoughtful and original contribution to the effort - that's why there's
a wiki, right?]
Exactly. ;-)
Henning
12:31 p.m.
Hi Henning,
Just let me know what you have in mind, and I'll try to help.
Regards,
Bogdan
Henning Westerholt wrote:
OpenSER is
wonderful, but it's one of the harder things to "get into" in
the open-source VoIP space because its documentation is rather lopsided
in this regard.
Hi Alex,
yes, you're right. I've added a link to the Rome admin course materials on the
Wiki which has some information about the general configuration, and will ask
Bogdan about to provide some other slides about the internal structure from
the programming course.
12:47 p.m.
On Wednesday 13 February 2008, Bogdan-Andrei Iancu wrote:
Hi Henning,
Just let me know what you have in mind, and I'll try to help.
Hi Bogdan,
i was refering to your slides from the OpenSER development course. Could you
perhaps provide them on voice-systems.ro for downloading like the admin
course slides?
Thank you,
Henning
1:11 p.m.
Well...... :) as you remember, the course was more with the code in
front, rather then slides....
Anyhow, I will dig in to see if I found something that can be used.
Regards,
Bogdan
Henning Westerholt wrote:
On Wednesday 13 February 2008, Bogdan-Andrei Iancu
wrote:
Hi Henning,
Just let me know what you have in mind, and I'll try to help.
Hi Bogdan,
i was refering to your slides from the OpenSER development course. Could you
perhaps provide them on voice-systems.ro for downloading like the admin
course slides?
Thank you,
Henning
12 Feb
12 Feb
5:21 p.m.
On Tuesday 12 February 2008 16:12:14 Ali Jawad wrote:
Thank you for your reply
I commented out my section and replaced it with
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # The user is online
{
route(1);
exit;
}
else
{
route(2);
exit;
}
}
Now if I call a PSTN number I get too many hops, and if I do a local call
to a user online it does not ring at the other end. Thank you.
Ali, OpenSer scripting is not easy, you should read more doc and understand it
better before trying to do all your requeriments.
If a call does "too many hops" it mean that you send it out (t_relay) without
chaning the RURI, so it comes back to OpenSer, again and again.
If you call a registered user and doesn't ring it's because the INVITE doesn't
arrive to the phone, does it? have you verified it? This is a low level
software, you cannot speak about "a phone doesn't ring", the question is:
does the INVITE arrive to the phone? you can test it with ngrep, for example.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
5:25 p.m.
Thanks again ..I refered to the documentation and it worked..I have tried to find
lookup("location")) to see what it does but I could not find it. In which part
should I look for it ?
Thanks for your patience and help. It is working now but I am trying to understand my
script better.
-----Original Message-----
From: users-bounces(a)lists.openser.org [mailto:users-bounces@lists.openser.org] On Behalf
Of Iñaki Baz Castillo
Sent: Tuesday, February 12, 2008 5:21 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Routing Problem "To Many Hops"
On Tuesday 12 February 2008 16:12:14 Ali Jawad wrote:
Thank you for your reply
I commented out my section and replaced it with
if (does_uri_exist()) { # The called number is a local user
if (lookup("location")) # The user is online
{
route(1);
exit;
}
else
{
route(2);
exit;
}
}
Now if I call a PSTN number I get too many hops, and if I do a local call
to a user online it does not ring at the other end. Thank you.
Ali, OpenSer scripting is not easy, you should read more doc and understand it
better before trying to do all your requeriments.
If a call does "too many hops" it mean that you send it out (t_relay) without
chaning the RURI, so it comes back to OpenSer, again and again.
If you call a registered user and doesn't ring it's because the INVITE doesn't
arrive to the phone, does it? have you verified it? This is a low level
software, you cannot speak about "a phone doesn't ring", the question is:
does the INVITE arrive to the phone? you can test it with ngrep, for example.
--
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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5:36 p.m.
Hi Ali,
-----Ursprüngliche Nachricht-----
Von: users-bounces(a)lists.openser.org
[mailto:users-bounces@lists.openser.org] Im Auftrag von Ali Jawad
Gesendet: Dienstag, 12. Februar 2008 16:25
An: Iñaki Baz Castillo; users(a)lists.openser.org
Betreff: Re: [OpenSER-Users] Routing Problem "To Many Hops"
Thanks again ..I refered to the documentation and it
worked..I have tried to find lookup("location")) to see what
it does but I could not find it. In which part should I look for it ?
Have a look at the module doc.
lookup() is from the registrar module:
http://www.openser.org/docs/modules/1.3.x/registrar.html#AEN342
br,
janosch
Thanks for your patience and help. It is working now but I am
trying to understand my script better.
-----Original Message-----
From: users-bounces(a)lists.openser.org
[mailto:users-bounces@lists.openser.org] On Behalf Of Iñaki
Baz Castillo
Sent: Tuesday, February 12, 2008 5:21 PM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Routing Problem "To Many Hops"
On Tuesday 12 February 2008 16:12:14 Ali Jawad wrote:
Thank you for your reply
I commented out my section and replaced it with
if (does_uri_exist()) { # The called number is
a local user
if (lookup("location")) #
The user is online
{
route(1);
exit;
}
else
{
route(2);
exit;
}
}
Now if I call a PSTN number I get too many hops, and if I
do a local call
to a user online it does not ring at the other
end. Thank you.
Ali, OpenSer scripting is not easy, you should read more doc
and understand it
better before trying to do all your requeriments.
If a call does "too many hops" it mean that you send it out
(t_relay) without
chaning the RURI, so it comes back to OpenSer, again and again.
If you call a registered user and doesn't ring it's because
the INVITE doesn't
arrive to the phone, does it? have you verified it? This is a
low level
software, you cannot speak about "a phone doesn't ring", the
question is:
does the INVITE arrive to the phone? you can test it with
ngrep, for example.
--
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
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
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6133
days inactive
6134
days old
12 comments
6 participants
participants (6)
-
Alex Balashov -
Ali Jawad -
Bogdan-Andrei Iancu -
Henning Westerholt -
Iñaki Baz Castillo -
Siller Janosch