Hi Iqbal,
Asterisk is not sending ser anything because nothing is reaching
asterisk. I have used sip debug but no sip messages are received by
asterisk. Asterisk and ser are one the same machine with ser on port
5060 and asterisk on 5064.
Therefore it is SER sending the 404. The call scenario is as follows:
314 (caller) SER 2092(callee)
INVITE ->
<- 407
ACK ->
INVITE ->
<- 100
INVITE ->
<- 100
<- 486 Busy
ACK ->
<- 404
ACK ->
The t_relay_to_udp mustn't be working to forward to asterisk....Then
route 1 is invoked.....
Aisling.
-----Original Message-----
From: Iqbal [mailto:iqbal@gigo.co.uk]
Sent: 04 October 2005 12:02
To: Greger V. Teigre
Cc: Aisling; serusers(a)lists.iptel.org
Subject: Re: [Serusers] voicemail route
if ser is sending you the 404, then what is asterisk sending ser
what I normally do is sip debug on asterisk and print it out, then ngrep
on ser also...or just ngrep both paths, and draw it out sip_scenario is
good, but a bit OTT sometimes
Iqbal
Greger V. Teigre wrote:
Aisling,
I think the only way you can get further on this is to use ngrep and
create a complete trace of the call. Then you have to match each of
your log messages to each SIP message. sip_scenario can help you in
drawing out who sent what. Remember that once you relay to Asterisk,
Asterisk will get in the loop and these messages should also be
relayed properly. My guess is that this has something to do with the
OK or ACK at the end of call. Most likely you forget about a SIP
message when reading your logs... ;-) (I've done it myself so many
times)
g-)
----- Original Message -----
*From:* Aisling <mailto:ashling.odriscoll@cit.ie>
*To:* serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org>
*Sent:* Monday, October 03, 2005 09:05 PM
*Subject:* [Serusers] voicemail route
Hello everyone,
I am using the onsip call features ser.cfg and am adapting it for
asterisk voicemail. This is what I currently have changed:
1) In the usr_preferences table in the ser database have an entry
for
user 2092.
Insert into usr_preferences (username, attribute, value) values
("2092", "voicemail", "y");
2) In Route[3] (used for call invite handling)
if(avp_db_load("$ruri/username","s:voicemail")){
if(avp_check("s:voicemail", "eq/y/i")){
setflag(18);
};
};
This will check if the user wants to use voicemail according to
the
preference that is set for them in the usr_preferences table. I
they
don't want to use voicemail set value to "n"
3) In failure route[1]
if (call fwd on no answer is enabled{
} else if(isflagset(18) && t_check_status("408")){
route(x);
break;
};
4) route[x]
{
acc_db_request("missed called", "missed_calls"); revert_uri();
rewritehostport("x.x.x.x:5064"); #port where asterisk is listening
append_branch();
t_relay_to_udp(x.x.x.x", "5064");
break();
}
I am getting a 404 sent back to the phone..I suspect this is
something got to do with route 1 as I have used loads of log
messages and I can see the flag being set, route x being called
but after the failure route, the code jumps to route 1...This is
probably because in route 3 it says t_on_failure("1") followed by
route 4 followed by route 1...I just don't know what to do about
it....Does anyone have any suggestions?
Kindest Regards,
Aisling.
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information contained in this communication is not a proper and complete
record of the message as transmitted by the sender nor for any delay in
its receipt.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the person to
whom it is addressed. Its contents may be protected by legal and/or professional
privilege. Should it be received by you in error please contact the sender at the above
quoted email address. Any unauthorised form of reproduction of this message is strictly
prohibited. The Institute does not guarantee the security of any information
electronically transmitted and is not liable if the information contained in this
communication is not a proper and complete record of the message as transmitted by the
sender nor for any delay in its receipt.