3 Apr
2008
3 Apr
'08
10:50 a.m.
Hi everybody,
I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server.
Suppose that an incoming calls arrives to Ser from Asterisk and the user
is offline;
Then Openser sends back the Invite to Asterisk that should activate the
voicemail application but
unfortunately it detects a Loop (482) and rejects the invite.
I think I have to mangle the SIP message in Openser before the send to
Asterisk again.
This is what I currently do:
if(!lookup("location"))
{
#send to voicemail if active but not registerd
xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
if($avp(s:vmail) == "1")
{
route(11);
}
else
{
route(20);
}
}
else
{
xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
route(9);
}
exit;
....
route[11]
{
xlog("L_INFO", "Forwarding request to VM\n");
prefix("vm");
sethostport("asterisk-gw:5060");
#append_branch();
if (!t_relay())
{
xlog("L_INFO", "Unable TO Forward the request to VM\n");
route(20);
exit;
}
What can I do to avoid Loop Detection?
Thanks in advance,
Cosimo Fadda
___________
11:37 a.m.
El Thursday 03 April 2008 07:50:17 Cosimo Fadda escribió:
Hi everybody,
I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server.
Suppose that an incoming calls arrives to Ser from Asterisk and the user
is offline;
Then Openser sends back the Invite to Asterisk that should activate the
voicemail application but
unfortunately it detects a Loop (482) and rejects the invite.
I think I have to mangle the SIP message in Openser before the send to
Asterisk again.
Asterisk FAILS detecting loop-back since your case is NOT a loop-back.
There is a patch to solve it but it's stopped. Anyway I tryed it with a
concrete SVN version and it works:
http://bugs.digium.com/view.php?id=7403
Also, I wrote (in Spanish) how to apply it:
http://blog.aliax.net/2007/08/asterisk-parche-chansip-para-permitir.html
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
12:21 p.m.
On 04/03/08 11:37, Iñaki Baz Castillo wrote:
El Thursday 03 April 2008 07:50:17 Cosimo Fadda
escribió:
Is is working out of the box with the SVN version? Or the patch have to
be applied to SVN? I have been interested in this patch, but lost the
row of events over the time ...
Thanks,
Daniel
Hi everybody,
I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server.
Suppose that an incoming calls arrives to Ser from Asterisk and the user
is offline;
Then Openser sends back the Invite to Asterisk that should activate the
voicemail application but
unfortunately it detects a Loop (482) and rejects the invite.
I think I have to mangle the SIP message in Openser before the send to
Asterisk again.
Asterisk FAILS detecting loop-back since your case is NOT a loop-back.
There is a patch to solve it but it's stopped. Anyway I tryed it with a
concrete SVN version and it works:
http://bugs.digium.com/view.php?id=7403
Also, I wrote (in Spanish) how to apply it:
http://blog.aliax.net/2007/08/asterisk-parche-chansip-para-permitir.html
--
http://www.asipto.com
12:24 p.m.
El Thursday 03 April 2008 09:21:31 Daniel-Constantin Mierla escribió:
AFAIK it's not included in the trunk version at all. There is just a patch
working wiht a specific SVN version. Some time ago I tried to apply that
patch to the current svn version and I couldn't apply it because important
changes in chan_sip.c and Asterisk code (BTW I didn't spend too much time
trying it).
Best regards.
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
Asterisk FAILS
detecting loop-back since your case is NOT a loop-back.
There is a patch to solve it but it's stopped. Anyway I tryed it with a
concrete SVN version and it works:
http://bugs.digium.com/view.php?id=7403
Is is working out of the box with the SVN version? Or the patch have to
be applied to SVN? I have been interested in this patch, but lost the
row of events over the time ...
1:41 p.m.
Iñaki Baz Castillo ha scritto:
El Thursday 03 April 2008 07:50:17 Cosimo Fadda
escribió:
I'll have a look to the patch, even if I can't apply it to a running
system..
Hi everybody,
I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail server.
Suppose that an incoming calls arrives to Ser from Asterisk and the user
is offline;
Then Openser sends back the Invite to Asterisk that should activate the
voicemail application but
unfortunately it detects a Loop (482) and rejects the invite.
I think I have to mangle the SIP message in Openser before the send to
Asterisk again.
Asterisk FAILS detecting loop-back since your case is NOT a loop-back.
There is a patch to solve it but it's stopped. Anyway I tryed it with a
concrete SVN version and it works:
http://bugs.digium.com/view.php?id=7403
Also, I wrote (in Spanish) how to apply it:
http://blog.aliax.net/2007/08/asterisk-parche-chansip-para-permitir.html
Juha Heinanen ha scritto:
Cosimo Fadda writes:
My first thought was to send a 302 from Openser to Asterisk, but I
don't
know how..
Thanks everybody,
Cosimo Fadda
_______________
Then Openser sends back the Invite to Asterisk
that should activate the
voicemail application but
how about sending back 302 to voicemail uri.
-- juha
1:50 p.m.
El Thursday 03 April 2008 10:49:19 Juha Heinanen escribió:
Cosimo Fadda writes:
And add the mandatory "Contact" header with the URI of the voicemail URI (I
don't remember the function adding headers to the reply but it does exist).
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
My first thought was to send a 302 from Openser
to Asterisk, but I don't
know how..
try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily").
2 p.m.
On 04/03/08 13:50, Iñaki Baz Castillo wrote:
--
http://www.asipto.com
El Thursday 03 April 2008 10:49:19 Juha Heinanen
escribió:
append_to_reply(...) in textops module
http://www.openser.org/docs/modules/1.3.x/textops.html#AEN276
Cheers,
Daniel
Cosimo Fadda writes:
And add the mandatory "Contact" header with the URI of the voicemail URI (I
don't remember the function adding headers to the reply but it does exist).
My first thought was to send a 302 from Openser
to Asterisk, but I don't
know how..
try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily").
2:04 p.m.
Thank you everybody,
I'll try it immediatly!
Regards,
Cosimo
OT:
Greetings to Daniel from Dario Busso
Daniel-Constantin Mierla ha scritto:
On 04/03/08 13:50, Iñaki Baz Castillo wrote:
--
*I*_______________________
El Thursday 03 April 2008 10:49:19 Juha Heinanen
escribió:
append_to_reply(...) in textops module
http://www.openser.org/docs/modules/1.3.x/textops.html#AEN276
Cheers,
Daniel
Cosimo Fadda writes:
And add the mandatory "Contact" header with the URI of the
voicemail URI (I
don't remember the function adding headers to the reply but it does exist).
My first thought was to send a 302 from Openser
to Asterisk, but I don't
know how..
try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily").
4 Apr
4 Apr
12:24 p.m.
Hi everybody,
thanks to your tips now the voicemail route works without loop detection
from asterisk;
But maybe I'm still making some little mistake: this is the route I'm using:
route[11]
{
xlog("L_INFO", "Forwarding request to VM\n");
#remove_hf("Contact");
append_to_reply("Contact: <sip:vm$rU@asteriskgw>");
remove_hf("Contact");
sl_send_reply("302", "Moved Temporalily ");
exit;
}
I've tried both removing the original "Contact" before and after the
append, but I always have this 302 Message:
<-- SIP read from OPENSERIP:5060:
SIP/2.0 302 Moved Temporalily
Via: SIP/2.0/UDP asteriskgw:5060;branch=z9hG4bK1262b98d;rport=5060
From: "0MXXXXXX" <sip:0MXXXXX@asteriskgw>;tag=as13654251
To: <sip:cosimo@OPENSERIP>;tag=26501d74ed63701b338d7e605938b7cb.11ac
Call-ID: 52e6481a1dd8674c559c477c36558c19@asteriskgw
CSeq: 102 INVITE
Contact: <sip:vmcosimo@asteriskgw>Contact: sip:cosimo@OPENSERIP
Server: OpenSER (1.3.1-notls (i386/linux))
Content-Length: 0
There are two Contact headers!!
This is not a problem for me since Asterisk ignore the second, but I'd
like to avoid this behavior..
Maybe I have to rewrite the contact without appending another one.
Thanks in advance,
Cosimo
Cosimo Fadda ha scritto:
Thank you everybody,
I'll try it immediatly!
Regards,
Cosimo
OT:
Greetings to Daniel from Dario Busso
Daniel-Constantin Mierla ha scritto:
____________________
On 04/03/08 13:50, Iñaki Baz Castillo wrote:
El Thursday 03 April 2008 10:49:19 Juha Heinanen
escribió:
append_to_reply(...) in textops module
http://www.openser.org/docs/modules/1.3.x/textops.html#AEN276
Cheers,
Daniel
Cosimo Fadda writes:
> My first thought was to send a 302 from Openser to Asterisk, but I don't
> know how..
try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily").
And add the mandatory "Contact" header with the URI of the
voicemail URI (I
don't remember the function adding headers to the reply but it does exist).
12:54 p.m.
El Friday 04 April 2008 09:24:21 Cosimo Fadda escribió:
There are two Contact headers!!
This is not a problem for me since Asterisk ignore the second, but I'd
like to avoid this behavior..
Maybe I have to rewrite the contact without appending another one.
Opsss, that's an issue yes!
Sincerely, I think is not possible to avoid it since it's created by OpenSer
when creating the response :(
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
4:16 p.m.
El Friday 04 April 2008 12:21:12 Juha Heinanen escribió:
Cosimo Fadda writes:
You said:
"try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily")."
Do you mean that if OpenSer receives a request, changes the RURI and isntead
of forwarding the request creates a reply "302" then the Contact header will
be the new RURI? Why it shouldn't work?
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
But maybe I'm still making some little
mistake: this is the route I'm
using:
did you try my proposal? i said to call rewrite_uri and then
sl_send_reply. i didn't say anything about manipulating contact
header. 
4:22 p.m.
Iñaki Baz Castillo writes:
Do you mean that if OpenSer receives a request,
changes the RURI and
isntead of forwarding the request creates a reply "302" then the
Contact header will be the new RURI? Why it shouldn't work?
all i tried to say is that rewriting request uri (using rewrite_uri or
by some other means) and then calling sl_send_reply("302", "Moved
Temporalily") might work. have you tried and it does not work for you?
-- juha
4:47 p.m.
El Friday 04 April 2008 13:22:42 Juha Heinanen escribió:
Iñaki Baz Castillo writes:
Sorry Juha, my question should be:
"Why it should work?"
since I can't understand how rewriting the RURI and sending a 302 can generate
the reply with "Contact = new RURI".
I've never read doc aobut doing it.
Do you mean that if OpenSer receives a request,
changes the RURI and
isntead of forwarding the request creates a reply "302" then the
Contact header will be the new RURI? Why it shouldn't work?
all i tried to say is that rewriting request uri (using rewrite_uri or
by some other means) and then calling sl_send_reply("302", "Moved
Temporalily") might work. have you tried and it does not work for you?
No, I haven't tryed, but it seems to work as Cosimo has replied, so you are
completely right ;)
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
6:53 p.m.
Iñaki Baz Castillo writes:
"Why it should work?"
since I can't understand how rewriting the RURI and sending a 302 can
generate
the reply with "Contact = new RURI".
I've never read doc aobut doing it.
indeed, it is not documented in sl/README. you could issue a tracker
request on it.
-- juha
7:15 p.m.
Hi,
Here is an example ;)
http://openser.svn.sourceforge.net/viewvc/openser/trunk/examples/redirect.c…
Regards,
Bogdan
Juha Heinanen wrote:
Iñaki Baz Castillo writes:
"Why it should work?"
since I can't understand how rewriting the RURI and sending a 302 can
generate
the reply with "Contact = new RURI".
I've never read doc aobut doing it.
indeed, it is not documented in sl/README. you could issue a tracker
request on it.
-- juha
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
4:29 p.m.
Hi everybody and thank you for your interest,
Juha Heinanen ha scritto:
Cosimo Fadda writes:
I finally followed your proposal and now it works,
route[11]
{
xlog("L_INFO", "Forwarding request to VM\n");
prefix("vm");
rewritehost("asteriskgw");
sl_send_reply("302", "Moved Temporalily ");
exit;
}
<-- SIP read from openser:5060:
SIP/2.0 302 Moved Temporalily
Via: SIP/2.0/UDP 217.199.19.212:5060;branch=z9hG4bK5f777042;rport=5060
From: "0MXXXXXX" <sip:0MXXXXXX@asteriskgw>;tag=as1c030a33
To: <sip:matteo@openser>;tag=26501d74ed63701b338d7e605938b7cb.54a5
Call-ID: 7eb5e5ea03c57c900b45052c28701977@asteriskgw
CSeq: 102 INVITE
Contact: sip:vmmatteo@asteriskgw
Server: OpenSER (1.3.1-notls (i386/linux))
Content-Length: 0
Iñaki Baz Castillo ha scritto:
But maybe I'm still making some little
mistake: this is the route I'm
using:
did you try my proposal? i said to call rewrite_uri and then
sl_send_reply. i didn't say anything about manipulating contact
header.
El Friday 04 April 2008 12:21:12 Juha Heinanen
escribió:
I haven't found anything in the docs about this.
Thank you again,
Regards,
Cosimo
____________
Cosimo Fadda writes:
You said:
"try rewriting the uri and then calling sl_send_reply("302", "Moved
Temporalily")."
Do you mean that if OpenSer receives a request, changes the RURI and isntead
of forwarding the request creates a reply "302" then the Contact header will
be the new RURI? Why it shouldn't work?
But maybe I'm still making some little
mistake: this is the route I'm
using:
did you try my proposal? i said to call rewrite_uri and then
sl_send_reply. i didn't say anything about manipulating contact
header.
11 Apr
11 Apr
2 a.m.
Here is how I solve this.
In Asterisk, I add a SIP header just before sending it to OpenSER using the
function: SipAddHeader(P-hint: From Asterisk).
Then, in OpenSER I have the following:
# check to see if user is online
if (!lookup("location")) {
# SIP from Asterisk or not a call attempt
if (search("(P-hint):.From Asterisk") ||
!(method==INVITE)) {
sl_send_reply("480","User Temporarily
Unavailable");
exit;
} else {
route(2);
exit;
};
};
route[2] {
# Let Asterisk deal with voicemail/forward
sl_send_reply("181", "Forward to Voicemail");
prefix("vm");
setflag(9); # forward to Asterisk
route(1);
}
On Thu, Apr 3, 2008 at 12:50 AM, Cosimo Fadda <cosimo.fadda(a)klarya.it>
wrote:
Hi everybody,
I'm using Opeser as Proxy and one Asterisk as Gateway and Voicemail
server.
Suppose that an incoming calls arrives to Ser from Asterisk and the user
is offline;
Then Openser sends back the Invite to Asterisk that should activate the
voicemail application but
unfortunately it detects a Loop (482) and rejects the invite.
I think I have to mangle the SIP message in Openser before the send to
Asterisk again.
This is what I currently do:
if(!lookup("location"))
{
#send to voicemail if active but not registerd
xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
if($avp(s:vmail) == "1")
{
route(11);
}
else
{
route(20);
}
}
else
{
xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu
IP=$si ID=$ci\n");
route(9);
}
exit;
....
route[11]
{
xlog("L_INFO", "Forwarding request to VM\n");
prefix("vm");
sethostport("asterisk-gw:5060");
#append_branch();
if (!t_relay())
{
xlog("L_INFO", "Unable TO Forward the request to VM\n");
route(20);
exit;
}
What can I do to avoid Loop Detection?
Thanks in advance,
Cosimo Fadda
___________
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
10:33 a.m.
El Thursday 10 April 2008 23:00:28 Adrian A escribió:
Here is how I solve this.
In Asterisk, I add a SIP header just before sending it to OpenSER using the
function: SipAddHeader(P-hint: From Asterisk).
Then, in OpenSER I have the following:
# check to see if user is online
if (!lookup("location")) {
# SIP from Asterisk or not a call attempt
if (search("(P-hint):.From Asterisk") ||
!(method==INVITE)) { sl_send_reply("480","User Temporarily
Unavailable");
exit;
} else {
route(2);
exit;
};
};
route[2] {
# Let Asterisk deal with voicemail/forward
sl_send_reply("181", "Forward to Voicemail");
prefix("vm");
setflag(9); # forward to Asterisk
route(1);
}
I can't understand why that should work, in fact Asterisk is receiving the
same INVITE he sent (well, RURI modified but the Asterisk bug is that it
doesn't recogniza a spiral).
--
Iñaki Baz Castillo
ibc(a)in.ilimit.es
12 Apr
12 Apr
2:04 a.m.
If the INVITE comes in from Asterisk, OpenSER replies with 480 and lets
Asterisk deal with sending call to voicemail (e.g. dialstatus =
unavailable). This eliminates the loop because the INVITE does not come back
to Asterisk.
On Fri, Apr 11, 2008 at 12:33 AM, Iñaki Baz Castillo <ibc(a)in.ilimit.es>
wrote:
El Thursday 10 April 2008 23:00:28 Adrian A escribió:
Here is how I solve this.
In Asterisk, I add a SIP header just before sending it to OpenSER using
the
function: SipAddHeader(P-hint: From Asterisk).
Then, in OpenSER I have the following:
# check to see if user is online
if (!lookup("location")) {
# SIP from Asterisk or not a call attempt
if (search("(P-hint):.From Asterisk") ||
!(method==INVITE)) { sl_send_reply("480","User Temporarily
Unavailable");
exit;
} else {
route(2);
exit;
};
};
route[2] {
# Let Asterisk deal with voicemail/forward
sl_send_reply("181", "Forward to Voicemail");
prefix("vm");
setflag(9); # forward to Asterisk
route(1);
}
I can't understand why that should work, in fact Asterisk is receiving the
same INVITE he sent (well, RURI modified but the Asterisk bug is that it
doesn't recogniza a spiral).
--
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
2:59 a.m.
2008/4/12, Adrian A <adrianvoip(a)gmail.com>om>:
If the INVITE comes in from Asterisk, OpenSER replies
with 480 and lets
Asterisk deal with sending call to voicemail (e.g. dialstatus =
unavailable). This eliminates the loop because the INVITE does not come back
to Asterisk.
Ok ok, I understand what you mean. But that is just useful for
redirection to voicemail and so.
For example, imagine you have Asterisk as PSTN gateway. Imagine
Asterisk receives a call from PSTN and routes it to OpenSer, and
OpenSer destination user has a redirection to a mobile number. Then
OpenSer would route the invite back to Asterisk who will detect a loop
(an spiral in fact, but Asterisk is buggy here).
Regards.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
11:05 a.m.
On Apr 12, 2008, at 1:59 AM, Iñaki Baz Castillo wrote:
Ok ok, I understand what you mean. But that is just
useful for
redirection to voicemail and so.
For example, imagine you have Asterisk as PSTN gateway. Imagine
Asterisk receives a call from PSTN and routes it to OpenSer, and
OpenSer destination user has a redirection to a mobile number. Then
OpenSer would route the invite back to Asterisk who will detect a loop
(an spiral in fact, but Asterisk is buggy here).
What we've been doing for the last 4 years in this is that OpenSER
never, ever replies back to Asterisk with the same INVITE message.
Asterisk has all the redirection intelligence. When OpenSER detects
that a user is offline it replies with some 4xx error message,
Asterisk detects that, goes "ooh, user is offline, let's see what i
have to do in this case" and does it's thing. Unfortunately we've
never had the opportunity to handle the cases where the users UA or
OpenSER replies back with a 3xx type message to tell Asterisk where to
redirect to. That might have something to do with us being a little
strange though.
--
Andreas Sikkema
13 Apr
13 Apr
7:18 a.m.
That's what the call forwarding in SIP UAs use to "forward" a call to
another destination, 302 redirect messages. It works fine with Asterisk.
Unless you need openser to handle accounting or such, redirects would be a
valid option.
Mike
-----Original Message-----
From: users-bounces(a)lists.openser.org
[mailto:users-bounces@lists.openser.org] On Behalf Of Andreas Sikkema
Sent: Saturday, April 12, 2008 1:05 AM
To: users(a)lists.openser.org
Subject: Re: [OpenSER-Users] Avoid Loops between Openser and Asterisk
Unfortunately we've
never had the opportunity to handle the cases where the users UA or
OpenSER replies back with a 3xx type message to tell Asterisk where to
redirect to. That might have something to do with us being a little
strange though.
--
Andreas Sikkema
8:19 p.m.
Hi everybody,
Andreas Sikkema ha scritto
Asterisk has all the redirection intelligence. When
OpenSER detects
that a user is offline it replies with some 4xx error message,
Asterisk detects that, goes "ooh, user is offline, let's see what i
have to do in this case" and does it's thing.
Not very sure that Asterisk
has all the redirection intelligence, in
particular if the SIP users are not registered on Asterisk the variable
Dialstatus gives you the 'state' of the proxy rather than the user; I
mean there is a difference between this two cases:
Dial(SIP/alice,,)
Dial(SIP/PROXY-1/alice,,}
where in the first case alice is registered with Asterisk, in the second
is regisered with Proxy-1.
Dial() returns DIALSTATUS=CONGESTION for pretty much every call setup
problem; there is also no way to look at a SIP response in the dialplan,
even though you can see every other SIP header with
${SIP_HEADER(<header_name>) you cannot see and treat accordingly the
actual response code.
So, in my opinion is better to put the intelligence on Openser rather
than on Asterisk,
Regards,
Cosimo
____________________
6073
days inactive
6083
days old
26 comments
9 participants
participants (9)
-
Adrian A -
Andreas Sikkema -
Bogdan-Andrei Iancu -
Cosimo Fadda -
Daniel-Constantin Mierla -
Iñaki Baz Castillo -
Iñaki Baz Castillo -
jh＠tutpro.com -
Mike Fedyk