Aisling
I am using the following to get to voicemail.
failure_route[1] {
xlog("L_ALERT", "%Tf %mf ****** Failure Route 1: <%rm> <%rr>
<%rs>\n");
if(t_check_status("487")) {
break;
};
if(method=="INVITE" && (t_check_status("486|408|404|480"))) {
if(avp_db_load("$ruri", "s:mailbox"))
avp_pushto("$ruri/username",
"s:mailbox");
prefix("V");
rewritehostport("66.236.248.133:5060");
append_branch();
xlog("L_ALERT", "****** Transfering to
Voicemail\n");
t_on_reply("1");
t_relay();
};
}
This works for uid busy and uid not registered but doesn't work for ser's 30
sec time out 408 "modparam("tm", "fr_inv_timer", 30)" So if someone can
help me figure out why I would appreciate it.
Rick
_____
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Aisling
Sent: Tuesday, October 04, 2005 5:56 AM
To: 'Greger V. Teigre'; serusers(a)lists.iptel.org
Subject: RE: [Serusers] voicemail route
Ok Greger, thanks for the reply.
Just a quick question, for the forward to voicemail on busy scenario, I can
see the callee phone sending the 486 busy message to SER and SER sends an
ACK back. At this stage I then see SER send a 404 to the caller.I presume
the correct sequence instead for this would be SER forwarding the INVITE to
Asterisk?
Many thanks.
-----Original Message-----
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: 04 October 2005 06:55
To: Aisling; serusers(a)lists.iptel.org
Subject: Re: [Serusers] voicemail route
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
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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proper and complete record of the message as transmitted by the sender nor
for any delay in its receipt.
hi
I had installed SER. I get following error when I try to configure jain sip communicator to use SER.
SIP/2.0 403 No relaying
Call-ID: aa7c1df636a1640571eb10e920781a2b(a)192.168.1.169
CSeq: 1 REGISTER
From: "Rajesh" <sip:admin@192.168.1.207:20318;transport=udp>;tag=6972371
To: "Rajesh" <sip:admin@192.168.1.207:20318;transport=udp>;tag=b51ece8fd8c195776737473dd6552d43.1682
Via: SIP/2.0/UDP 192.168.1.169:20318;branch=z9hG4bKcba4c28b1d1942c2fe848a76e481591d;rport=20318;received=68.225.20.250
P-Behind-NAT: Yes
Server: Sip EXpress router (0.9.0udpfifo (i386/linux))
Warning: 392 195.37.77.99:5060 "Noisy feedback tells: pid=8440 req_src_ip=68.225.20.250 req_src_port=20318 in_uri=sip:192.168.1.207:5060;transport=udp out_uri=sip:192.168.1.207:5060;transport=udp via_cnt==1"
Content-Length: 0
Please help me resolve this.
Thanks
Rajesh
---------------------------------
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---------------------------------
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I have activated the "failed_transactions" flag in SER 0.9.4.
Now the "acc" module logs the 487 errors for INVITEs, but it DOESN'T log
the 404 or 480 errors. Why?
Perhaps they are considered "missed calls" (i don't log missed calls)?
But then the 487 should be considered missed call too...
Thanks.
--
___________________________________________________
__
|- giannici(a)neomedia.it
|ederico Giannici http://www.neomedia.it
___________________________________________________
Are there any SER Distrobutions already set up on an ISo File with all the
appropriate scrips. Something similar to Asterisk@Home. Please let me know
if one exists and where I can find it.
Thanks
Dear all,
I'm working to separate on two different servers registrar and mediaproxy.
Mediaproxy server (with both mediaproxy and ser proxy process) will be
the front-end and all registers and calls from Intenet will be directed
to this server and then routed to internal registrar.
Now, I'm working with natted clients. I've started with only one server
(with only one ser) and it works fine. When I've tried to split the
architecture on two server I can't fine a good configuration.
In particular, when a user with a public IP calls a natted user, the RTP
doesn't use mediaproxy
Anyone can help me? do you have any working configurations?
thanks,
Andrea
Hi,
I notice that whenever SER is binding UAs, everyone who register / invite at
that time will just get stucked for a few seconds.
SER will not response when it's binding UAs
Is this normal? Anything can be done to improve this situation? Thanks in
advance
Regards,
Chia
Rajeev,
SER itself cannot do conferencing. In fact it doesn't understand
anything about RTP streams at all.
If you want to do conferencing or voice mail or other sorts of things
then you need to look at integrating SER with other software
applications, such as Asterisk PBX or SEMS.
Regards,
Paul
On 10/4/05, rajeev.khetan(a)cem-solutions.net
<rajeev.khetan(a)cem-solutions.net> wrote:
> Hi,
>
> I want to know wheather ser proxy support conferencing ?.
> If it support plz. send the config file for it.
> Thanks in advance.
>
> Regards,
> Rajeev
>
>
> _______________________________________________
> Serdev mailing list
> serdev(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serdev
>
Hello,
How to load radius authentication modules into ser. I have compiled ser as
described in the radius howto and I can not load modules in it. I have tried
lots of options but without success. When added a line (example: loadmodule
"/usr/local/lib/ser/modules/auth_radius.so") in ser.cfg there was an error.
Please Help.
Hi!
I tried to run binary from package ser-0.9.4_linux_i368.tar.gz and I get:
ser: relocation error: ser: symbol regexec, version GLIBC_2.3.4 not
defined in file libc.so.6 with link time reference
What's wrong? Is it my fault?
Regards,
Christian
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.
