[Serusers] voicemail route
Iqbal
iqbal at gigo.co.uk
Tue Oct 4 13:01:38 CEST 2005
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 at cit.ie>
> *To:* serusers at lists.iptel.org <mailto:serusers at 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.
>
> -------------------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.
>
> ------------------------------------------------------------------------
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Serusers mailing list
>serusers at lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>
More information about the sr-users
mailing list