4 Apr
2008
4 Apr
'08
11:24 a.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).