Hi all,
Something slightly off topic. Last weekend I hacked a quick and dirty
SIP conferencing bot. The idea is similar to Gtalk conference bot
(http://www.google.com/search?q=confbot), which we use daily, except
that this will be SIP (of course!), and it supports voice conferencing
as well (hell yeah!). I'm also thinking since the bot support some
remote commands, it may be useful for somekind of testing as well.
The initial version is done, see pjsip-apps/src/confbot in SVN trunk,
it kinda works though it will crash if something bad happens since I
haven't put in any error checkings. I'm sure there are a lot of other
issues as well.
The problem is, turns out there's few pjsip based applications
(http://www.pjsip.org/apps.htm) that support IM. Only Sipek supports
it, AFAIK. So I was thinking if I announce this confbot project early
we can work together to create the complete pjsip based solution. It
would be nice if I don't have to use pjsua for everyday's chatting!
So Saúl, Klaus, if you're listening.. ;-)
Cheers
Benny
PS:
I can only work on this project on weekends. ;-)
Sounds cool!
So Saúl, Klaus, if you're listening.. ;-)
I'm currently adding IM capabilities to YASS, I'll keep you posted!
Regards,
--
/Saúl
http://www.saghul.net | http://www.sipdoc.net
On Mon, Aug 24, 2009 at 5:37 PM, Saúl Ibarrasaghul@gmail.com wrote:
So Saúl, Klaus, if you're listening.. ;-)
I'm currently adding IM capabilities to YASS, I'll keep you posted!
Cool. I have YASS trunk version installed, I'll just need to "svn up"
whenever it's ready. ;-)
Btw, you acc.send_pager() patch has helped a lot!
Cheers
Benny
QjSimple supports IM, but the GUI is not nice at all
klaus
Benny Prijono wrote:
Hi all,
Something slightly off topic. Last weekend I hacked a quick and dirty
SIP conferencing bot. The idea is similar to Gtalk conference bot
(http://www.google.com/search?q=confbot), which we use daily, except
that this will be SIP (of course!), and it supports voice conferencing
as well (hell yeah!). I'm also thinking since the bot support some
remote commands, it may be useful for somekind of testing as well.
The initial version is done, see pjsip-apps/src/confbot in SVN trunk,
it kinda works though it will crash if something bad happens since I
haven't put in any error checkings. I'm sure there are a lot of other
issues as well.
The problem is, turns out there's few pjsip based applications
(http://www.pjsip.org/apps.htm) that support IM. Only Sipek supports
it, AFAIK. So I was thinking if I announce this confbot project early
we can work together to create the complete pjsip based solution. It
would be nice if I don't have to use pjsua for everyday's chatting!
So Saúl, Klaus, if you're listening.. ;-)
Cheers
Benny
PS:
I can only work on this project on weekends. ;-)
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
On Mon, Aug 24, 2009 at 10:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
QjSimple supports IM, but the GUI is not nice at all
Aha! I missed that. I just tried it and it's quite pleasant actually.
I'm leaving the bot running at sip:bot@pjsip.org for the night, if you
want to try it. Lets see how long it survives. :)
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
Cheers
Benny
Already crashed?
U 2009/08/25 01:10:23.254021 83.136.32.159:5060 -> 208.109.222.137:5060
MESSAGE sip:bot@pjsip.org SIP/2.0.
Record-Route:
sip:83.136.32.159;lr=on;ftag=d8ae96649be045e3869bcbcf0c084938;nat=caller.
Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bKc6d6.e2e8f9a.0.
Via: SIP/2.0/UDP
80.109.241.212:63133;received=80.109.241.212;rport=63133;branch=z9hG4bKPj39a7fabba711414d826d36d4751bdeaa.
Max-Forwards: 69.
From: sip:klaus.darilion@labs.nic.at;tag=d8ae96649be045e3869bcbcf0c084938.
To: sip:bot@pjsip.org.
Call-ID: 1fe50d2fdf9d4681adec75b7b9314e8e.
CSeq: 975 MESSAGE.
Accept: text/plain, application/im-iscomposing+xml.
Contact: sip:klaus.darilion@80.109.241.212:63133.
User-Agent: QjSimple 0.6.3 (pjproject 1.3/win32).
Content-Type: text/plain.
Content-Length: 6.
.
--help
U 2009/08/25 01:10:23.412709 208.109.222.137:5060 -> 83.136.32.159:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 83.136.32.159;branch=z9hG4bKc6d6.e2e8f9a.0.
Via: SIP/2.0/UDP
80.109.241.212:63133;received=80.109.241.212;rport=63133;branch=z9hG4bKPj39a7fabba711414d826d36d4751bdeaa.
From: sip:klaus.darilion@labs.nic.at;tag=d8ae96649be045e3869bcbcf0c084938.
To: sip:bot@pjsip.org;tag=0b1d2af7948181256520c4e3a2271928-df62.
Call-ID: 1fe50d2fdf9d4681adec75b7b9314e8e.
CSeq: 975 MESSAGE.
Server: OpenSER (1.3.2-tls (i386/linux)).
Content-Length: 0.
regards
klaus
Benny Prijono wrote:
On Mon, Aug 24, 2009 at 10:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
QjSimple supports IM, but the GUI is not nice at all
Aha! I missed that. I just tried it and it's quite pleasant actually.
I'm leaving the bot running at sip:bot@pjsip.org for the night, if you
want to try it. Lets see how long it survives. :)
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
Cheers
Benny
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
On Tue, Aug 25, 2009 at 12:11 AM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Already crashed?
No actually. The registration has timed out and it's not retried. :)
It's ON again.
Benny Prijono schrieb:
On Mon, Aug 24, 2009 at 10:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
QjSimple supports IM, but the GUI is not nice at all
Aha! I missed that. I just tried it and it's quite pleasant actually.
I'm leaving the bot running at sip:bot@pjsip.org for the night, if you
want to try it. Lets see how long it survives. :)
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
regards
klaus
Cheers
Benny
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Klaus Darilion schrieb:
Benny Prijono schrieb:
On Mon, Aug 24, 2009 at 10:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
QjSimple supports IM, but the GUI is not nice at all
Aha! I missed that. I just tried it and it's quite pleasant actually.
I'm leaving the bot running at sip:bot@pjsip.org for the night, if you
want to try it. Lets see how long it survives. :)
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
I found the problem: My proxy rejects the reply MESSAGE as it contains
an incorrect Route header:
Route: sip:pjsip.org;transport=tcp;lr
Looks like the request is not loose-routed by your openser proxy - the
Route header should be removed by the proxy.
U 2009/08/25 15:55:28.601970 208.109.222.137:5060 -> 83.136.32.159:5060
MESSAGE sip:klaus.darilion@labs.nic.at SIP/2.0.
Via: SIP/2.0/UDP 208.109.222.137;branch=z9hG4bK431a.08417c55.0;i=e01.
Via: SIP/2.0/TCP
192.168.0.13:56553;received=81.178.58.134;rport=56553;branch=z9hG4bKPjRITWFFl4nxPTIeMVPgRzOR-L9G.YZ.E4.
Max-Forwards: 69.
From: sip:bot@pjsip.org;tag=fgPeRepTOf4gmTxmewPBXpqjDeqtcOwh.
To: sip:klaus.darilion@labs.nic.at.
Call-ID: XXHLHUweLWupu1fjYc.Z.Neq3WFAIihc.
CSeq: 15965 MESSAGE.
Accept: text/plain, application/im-iscomposing+xml.
Contact: sip:bot@81.178.58.134:56553;transport=tcp.
User-Agent: PJSIP ConfBot.
Route: sip:pjsip.org;transport=tcp;lr.
Content-Type: text/plain.
Content-Length: 14.
P-hint: outbound.
.
Nobody is here
U 2009/08/25 15:55:28.603016 83.136.32.159:5060 -> 208.109.222.137:5060
SIP/2.0 403 out-of-dialog loose_route not allowed.
Via: SIP/2.0/UDP 208.109.222.137;branch=z9hG4bK431a.08417c55.0;i=e01.
Via: SIP/2.0/TCP
192.168.0.13:56553;received=81.178.58.134;rport=56553;branch=z9hG4bKPjRITWFFl4nxPTIeMVPgRzOR-L9G.YZ.E4.
From: sip:bot@pjsip.org;tag=fgPeRepTOf4gmTxmewPBXpqjDeqtcOwh.
To:
sip:klaus.darilion@labs.nic.at;tag=5fcf32020f171aefa0445747f7988cba.e12a.
Call-ID: XXHLHUweLWupu1fjYc.Z.Neq3WFAIihc.
CSeq: 15965 MESSAGE.
X-Info: Registration marked for NAT traversal.
Server: Kamailio (1.5.0-notls (i386/linux)).
Content-Length: 0.
.
regards
Klaus
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to you?
I am not getting anything back myself.
I am using the SIP MESSAGE script http://sipsimpleclient.com/wiki/sip_message
, which quits after sending the message while I have another instance
registered on the same SIP account listening in for incoming MESSAGE
requests.
Is this setup suppose to work?
Adrian
On Aug 25, 2009, at 4:12 PM, Benny Prijono wrote:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org
to you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
--
Adrian
As stated in the other email, the bug is in the pjsip.org SIP proxy.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Okay, that is something that I don't understand. The MESSAGE sent by
the bot does have Route: sip:pjsip.org;transport=tcp, since I wanted
to use TCP. I also have "pjsip.org" as my alias in my openser.cfg. And
I have this as well:
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
So it should remove my Route, shouldn't it? (openser newbie here)
Anyway I now hide the Route header from the bot, it shouldn't be there anymore.
Thanks
Benny
On Tue, Aug 25, 2009 at 3:30 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
As stated in the other email, the bug is in the pjsip.org SIP proxy.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
On Tue, Aug 25, 2009 at 3:30 PM, Adrian Georgescuag@ag-projects.com wrote:
I am not getting anything back myself.
I am using the SIP MESSAGE script
http://sipsimpleclient.com/wiki/sip_message, which quits after sending the
message while I have another instance registered on the same SIP account
listening in for incoming MESSAGE requests.
Is this setup suppose to work?
I think so. The bot will just send to the From address of the MESSAGE.
Another thing worth mentioning (once we sort out this initial
setback), to successfully join the chat, the bot will subscribe to
your presence and it will only join you if you're online.
Cheers
Benny
Benny Prijono schrieb:
Okay, that is something that I don't understand. The MESSAGE sent by
the bot does have Route: sip:pjsip.org;transport=tcp, since I wanted
to use TCP. I also have "pjsip.org" as my alias in my openser.cfg. And
I have this as well:
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
So it should remove my Route, shouldn't it? (openser newbie here)
No. This just adds a record-route header.
You should have somewhere in the beginning your config:
...
if (loose_route()) {
.....
t_relay();
exit;
}
...
The loose_route function is not easy to understand. Basically this
function checks if a route-set is announced by the client - i.e. if the
request contains a route header - as this would indicate that all the
routing information is already available and the request can be relayed
(as it is usually the case with in-dialog requests). If Route header is
present, the top-most route entry is removed.
But there is one exception: If there is only 1 Route header, and the
Route header points to the proxy itself (Route URI domain equals one of
the aliases), and there is no totag (as with the MESSAGE), then the
function returns FALSE and the Route header is removed.
regards
klaus
Anyway I now hide the Route header from the bot, it shouldn't be there anymore.
Thanks
Benny
On Tue, Aug 25, 2009 at 3:30 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
As stated in the other email, the bug is in the pjsip.org SIP proxy.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
On Tue, Aug 25, 2009 at 4:08 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Benny Prijono schrieb:
Okay, that is something that I don't understand. The MESSAGE sent by
the bot does have Route: sip:pjsip.org;transport=tcp, since I wanted
to use TCP. I also have "pjsip.org" as my alias in my openser.cfg. And
I have this as well:
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
So it should remove my Route, shouldn't it? (openser newbie here)
No. This just adds a record-route header.
You should have somewhere in the beginning your config:
...
if (loose_route()) {
.....
t_relay();
exit;
}
...
The loose_route function is not easy to understand.
Nor that I ever even try to understand it. :)
Turns out my "if(loose_route)" was put inside "if (has_totag())", I
wonder why. Must be left over from a very old openser example.
Thanks
Benny
Basically this function
checks if a route-set is announced by the client - i.e. if the request
contains a route header - as this would indicate that all the routing
information is already available and the request can be relayed (as it is
usually the case with in-dialog requests). If Route header is present, the
top-most route entry is removed.
But there is one exception: If there is only 1 Route header, and the Route
header points to the proxy itself (Route URI domain equals one of the
aliases), and there is no totag (as with the MESSAGE), then the function
returns FALSE and the Route header is removed.
regards
klaus
Anyway I now hide the Route header from the bot, it shouldn't be there
anymore.
Thanks
Benny
On Tue, Aug 25, 2009 at 3:30 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
As stated in the other email, the bug is in the pjsip.org SIP proxy.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to
you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Attached you can see how I do it.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 4:08 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Benny Prijono schrieb:
Okay, that is something that I don't understand. The MESSAGE sent by
the bot does have Route: sip:pjsip.org;transport=tcp, since I wanted
to use TCP. I also have "pjsip.org" as my alias in my openser.cfg. And
I have this as well:
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
So it should remove my Route, shouldn't it? (openser newbie here)
No. This just adds a record-route header.
You should have somewhere in the beginning your config:
...
if (loose_route()) {
.....
t_relay();
exit;
}
...
The loose_route function is not easy to understand.
Nor that I ever even try to understand it. :)
Turns out my "if(loose_route)" was put inside "if (has_totag())", I
wonder why. Must be left over from a very old openser example.
Thanks
Benny
Basically this function
checks if a route-set is announced by the client - i.e. if the request
contains a route header - as this would indicate that all the routing
information is already available and the request can be relayed (as it is
usually the case with in-dialog requests). If Route header is present, the
top-most route entry is removed.
But there is one exception: If there is only 1 Route header, and the Route
header points to the proxy itself (Route URI domain equals one of the
aliases), and there is no totag (as with the MESSAGE), then the function
returns FALSE and the Route header is removed.
regards
klaus
Anyway I now hide the Route header from the bot, it shouldn't be there
anymore.
Thanks
Benny
On Tue, Aug 25, 2009 at 3:30 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
As stated in the other email, the bug is in the pjsip.org SIP proxy.
regards
klaus
Benny Prijono schrieb:
On Tue, Aug 25, 2009 at 1:18 PM, Klaus
Darilionklaus.mailinglists@pernau.at wrote:
Hint: send IM "--join" to join the chatroom (or "--help" for other
commands), or just call it to join the voice conf. You don't need to
register to pjsip.org SIP service to join the conf.
either it does not work or I am too stupid.
I do not get any responses (IMs) when sending "--help"
Obviously you must allow MESSAGE to get through from bot@pjsip.org to
you?
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
Visit our blog: http://blog.pjsip.org
pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
...
if ( is_method("CANCEL") && !t_check_trans() ) {
xlog("L_WARN","$ci CANCEL without matching transaction ... ignore and discard.\n");
exit;
}
route(8); # disable mediaproxy (needs to see valid CANCEL requests)
remove_hf("Proxy-Authorization");
remove_hf("Authorization");
if ( is_method("CANCEL") ) {
xlog("L_INFO","$ci CANCEL with matching transaction ... t_relay.\n");
t_relay();
exit;
}
if (loose_route()) {
xlog("L_INFO","$ci beginning loose_route processing...\n");
xlog("L_DBG","[$Tf] $rm $ru (From: $fu -> To: $tu) entering route(32): loose_route handling\n");
if (!has_totag()) {
xlog("L_WARN","$ci loose_route request without to-tag, 403...\n");
sl_send_reply("403", "out-of-dialog loose_route not allowed");
exit;
}
... NAT handling...
xlog("L_INFO","$ci t_relay request to $bR\n");
if (!t_relay()) {
sl_reply_error();
}
exit;
}
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# non loose-route, but stateful ACK; must be an ACK after a 487
xlog("L_INFO","$ci local end-to-end ACK for an existent INVITE transaction detected ...t_relay()\n");
t_relay();
} else {
xlog("L_WARN","$ci ACK without matching transaction ... ignore and discard.\n");
}
exit;
}
if ( has_totag() ) {
xlog("L_ERR","$ci in-dialog request was not catched by loose_route block, 403... \n");
xlog("L_ERR","$ci this is the complete message:\n");
xlog("L_ERR","$ci $mb\n");
sl_send_reply("403","in-dialog request without loose_route is not allowed, this is a bug in the client or in this proxy");
exit;
}
route(7); # proxy authentication
...
Checkout latest trunk (or tag 0.5.3) now YASS supports MESSAGE :)
--
/Saúl
http://www.saghul.net | http://www.sipdoc.net
Just tried the bot, and after joining it stopped responding :-O is it dead?
--
/Saúl
http://www.saghul.net | http://www.sipdoc.net