[Serusers] HELP: SER-0.9 Record Route Problem Between SER andAsterisk
Java Rockx
javarockx at gmail.com
Thu Feb 24 13:10:53 CET 2005
Here is my ser.cfg which I've removed all the non-BYE related route blocks.
Many Thanks!
Paul
listen=24.11.12.24
listen=10.255.255.1
mhomed=1
# loadmodule stuff goes here
# modparam stuff goes here
route {
# ------------------------------------------------------------------------
# Sanity Check Section
# ------------------------------------------------------------------------
# ------------------------------------------------------------------------
# Record Route Section
# ------------------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
if (loose_route()) {
if (method=="BYE") {
setflag(1); # enable accounting for BYE messages. We do not
# enable accounting for Record-Routed INVITE messages
# because these are re-INVITEs and we already captured
# the original INVITE during the call setup
};
# I'm using mediaproxy - so tear down the RTP stuff
if (method=="BYE" || method=="CANCEL")) {
end_media_session();
};
if (method=="INVITE") {
# re-INVITE messages must be handled differently because
# our transparent RTP proxy needs to test the caller and
# callee for NATed IP addresses otherwise we could lose
# audio on one or both directions
route(8);
} else {
route(1);
};
break;
};
# Only handle message destined for out served domains. Other messages
are just relayed
if (!uri==myself) {
route(1);
break;
};
# ------------------------------------------------------------------------
# Message Handler Logic
# ------------------------------------------------------------------------
if (method=="BYE") {
# NOTE: all BYE messages should be record-routed so they should never
# hit this code block since they should be handled in the
# loose_route section above, so this is just a safety net
end_media_session();
route(1);
} else if (method=="CANCEL") {
# I'm not sure if any CANCELs messages would ever hit this block
# but it's here for good measure :-)
end_media_session();
route(1);
} else if (method=="INVITE") {
setflag(1); # enable accounting for *original* INVITE messages
route(6);
} else if (method=="NOTIFY") {
route(2);
} else if (method=="OPTIONS") {
route(3);
} else if (method=="REFER") {
route(6);
} else if (method=="REGISTER") {
route(4);
} else if (method=="SUBSCRIBE") {
route(5);
} else {
# all other messages come here for default handling
route(1);
};
}
route[1] {
# ------------------------------------------------------------------------
# default message handler
# ------------------------------------------------------------------------
# A note on when we need to call lookup("location")
#
# This was posted to serusers on 02/23/2005 by Daniel-Constantin Mierla
#
# lookup("location") has to be used for any request that has the domain
# part of R-URI pointing to your SIP server, should be delivered to an
# end-user and does not have to follow any Route header -- it does not
# matter the type of method. Could be said that only REGISTERs are just
# for servers, the others are either mixed (e.g, OPTIONS) or only for
# end-users.
if (!search(^"Route:") && !search(^"Record-Route:")) {
lookup("location");
};
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# ------------------------------------------------------------------------
# NOTIFY Message Handler
# ------------------------------------------------------------------------
}
route[3] {
# ------------------------------------------------------------------------
# OPTIONS Message Handler
# ------------------------------------------------------------------------
}
route[4] {
# ------------------------------------------------------------------------
# REGISTER Message Handler
# ------------------------------------------------------------------------
}
route[5] {
# ------------------------------------------------------------------------
# SUBSCRIBE Message Handler
# ------------------------------------------------------------------------
}
route[6] {
# ------------------------------------------------------------------------
# INVITE Message Handler
# ------------------------------------------------------------------------
}
route[7] {
# voicemail route
rewritehostport("10.255.255.2:5060");
append_branch();
if (!isflagset(15)) {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
end_media_session();
sl_reply_error();
};
}
route[8] {
# ------------------------------------------------------------------------
# re-INVITE Message Handler
#
# This route is a stripped down version of route[6]. Here we only
# lookup('location') in order to get the NAT flag from the location
# table because we need to know wheather or not to enable RTP proxying
}
onreply_route[1] {
}
failure_route[1] {
}
On Wed, 23 Feb 2005 22:08:12 -1000, Richard <richard at o-matrix.org> wrote:
> It looks like a ser related problem. A ser config would help to
> troubleshoot.
>
> Richard
>
> > -----Original Message-----
> > From: serusers-bounces at iptel.org [mailto:serusers-bounces at lists.iptel.org] On
> > Behalf Of Java Rockx
> > Sent: Wednesday, February 23, 2005 6:44 PM
> > To: serusers at lists.iptel.org
> > Subject: [Serusers] HELP: SER-0.9 Record Route Problem Between SER
> > andAsterisk
> >
> > Hi All.
> >
> > I'm using ser-0.9
> >
> > Can anyone take a quick look at this short SIP conversation and tell
> > me if they think the problem is with my ser.cfg or a bug in Asterisk
> > 1.0.2.
> >
> > We use a 3rd party for PSTN gateway functionality. This 3rd party uses
> > a Sonus box behind a SIP proxy. Our SER proxy talks directly to their
> > SIP proxy as needed to complete PSTN calls.
> >
> > The problem is that when a PSTN caller dials a SIP phone and gets sent
> > to voice mail (Asterisk) because of a no answer or busy condition,
> > Asterisk hangs up after the caller leaves a message. When Asterisk
> > hangs up, the BYE from Asterisk is sent to SER, however, SER
> > incorrectly forwards the BYE directly to their Sonus gateway, rather
> > than the their SIP proxy. This causes our PSTN gateway provider to
> > have "open" billing records in their system.
> >
> > If you look at the BYE message from Asterisk to SER you can see that
> > route headers are missing (I think). The final BYE should have been
> > sent to 68.86.100.20, but it was sent to 68.86.100.30 instead.
> >
> > I am record_route()ing all messages except for REGISTER and I have the
> > mhomed=1 parameter set.
> >
> > Can anyone help me put the blame on either my ser.cfg or Asterisk?
> >
> > Regards,
> > Paul
> >
> >
> > IP LEGEND
> > -----------
> > 68.86.100.30 - 3rd Party Sonus PSTN Gateway
> > 68.86.100.20 - 3rd Party SIP Proxy
> > 24.11.12.24 - Sip Express Router (eth0)
> > 10.255.255.1 - Sip Express Router (eth1)
> > 10.255.255.2 - Asterisk PBX
> >
> > NOTE: I have Asterisk connected to the SER server with a crossover cable.
> >
> >
> >
> > U 2005/02/23 22:24:18.848582 68.86.100.20:5060 -> 24.11.12.24:5060
> > INVITE sip:4075551212 at 24.11.12.24:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > Max-Forwards: 4.
> > Contact: sip:3211231234 at 68.86.100.30:5060.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > Content-Type: application/sdp.
> > Content-Length: 312.
> > .
> > v=0.
> > o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
> > s=sip call.
> > c=IN IP4 66.236.245.98.
> > t=0 0.
> > m=audio 16814 RTP/AVP 18 0 4 8 101.
> > a=rtpmap:18 G729/8000.
> > a=fmtp:18 annexb=no.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:4 G723/8000.
> > a=fmtp:4 annexa=yes.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> >
> > #
> > U 2005/02/23 22:24:18.860022 24.11.12.24:5060 -> 68.86.100.20:5060
> > SIP/2.0 100 trying -- your call is important to us.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > Content-Length: 0.
> > .
> >
> > #
> > U 2005/02/23 22:24:18.860259 10.255.255.1:1033 -> 10.255.255.2:5060
> > INVITE sip:699 at 10.255.255.2:5060 SIP/2.0.
> > Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Via: SIP/2.0/UDP 10.255.255.1;branch=z9hG4bKb929.21080974.0.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > Max-Forwards: 3.
> > Contact: sip:3211231234 at 68.86.100.30:5060.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > Content-Type: application/sdp.
> > Content-Length: 312.
> > .
> > v=0.
> > o=NexTone-MSW 1234 187 IN IP4 66.236.245.98.
> > s=sip call.
> > c=IN IP4 24.11.12.24.
> > t=0 0.
> > m=audio 36574 RTP/AVP 18 0 4 8 101.
> > a=rtpmap:18 G729/8000.
> > a=fmtp:18 annexb=no.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:4 G723/8000.
> > a=fmtp:4 annexa=yes.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> >
> > #
> > U 2005/02/23 22:24:18.871131 10.255.255.2:5060 -> 10.255.255.1:1033
> > SIP/2.0 100 Trying.
> > Via: SIP/2.0/UDP
> > 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
> > 3.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > User-Agent: Asterisk PBX.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> > Contact: <sip:699 at 10.255.255.2>.
> > Content-Length: 0.
> > .
> >
> > #
> > U 2005/02/23 22:24:18.879160 10.255.255.2:5060 -> 10.255.255.1:1033
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP
> > 10.255.255.1;branch=z9hG4bKb929.21080974.0;received=10.255.255.1;rport=103
> > 3.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > User-Agent: Asterisk PBX.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> > Contact: <sip:699 at 10.255.255.2>.
> > Content-Type: application/sdp.
> > Content-Length: 362.
> > .
> > v=0.
> > o=root 550 550 IN IP4 10.255.255.2.
> > s=session.
> > c=IN IP4 10.255.255.2.
> > t=0 0.
> > m=audio 17900 RTP/AVP 97 18 3 4 2 0 8 101.
> > a=rtpmap:97 iLBC/8000.
> > a=rtpmap:18 G729/8000.
> > a=rtpmap:3 GSM/8000.
> > a=rtpmap:4 G723/8000.
> > a=rtpmap:2 G726-32/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> >
> > #
> > U 2005/02/23 22:24:18.883882 24.11.12.24:5060 -> 68.86.100.20:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP 68.86.100.20:5060;branch=z9hG4bKed82c1ba766-c3014f40.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=7a874abcac87c7752a3d3c5c7ba10dc0.
> > Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 INVITE.
> > User-Agent: Asterisk PBX.
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER.
> > Contact: <sip:699 at 10.255.255.2>.
> > Content-Type: application/sdp.
> > Content-Length: 363.
> > .
> > v=0.
> > o=root 550 550 IN IP4 10.255.255.2.
> > s=session.
> > c=IN IP4 24.11.12.24.
> > t=0 0.
> > m=audio 36574 RTP/AVP 97 18 3 4 2 0 8 101.
> > a=rtpmap:97 iLBC/8000.
> > a=rtpmap:18 G729/8000.
> > a=rtpmap:3 GSM/8000.
> > a=rtpmap:4 G723/8000.
> > a=rtpmap:2 G726-32/8000.
> > a=rtpmap:0 PCMU/8000.
> > a=rtpmap:8 PCMA/8000.
> > a=rtpmap:101 telephone-event/8000.
> > a=fmtp:101 0-16.
> > a=silenceSupp:off - - - -.
> >
> > #
> > U 2005/02/23 22:24:19.097436 68.86.100.20:5060 -> 24.11.12.24:5060
> > ACK sip:699 at 10.255.255.2 SIP/2.0.
> > Via: SIP/2.0/UDP
> 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 ACK.
> > Max-Forwards: 4.
> > Contact: sip:3211231234 at 68.86.100.30:5060.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Content-Length: 0.
> > .
> >
> > #
> > U 2005/02/23 22:24:19.098087 10.255.255.1:1033 -> 10.255.255.2:5060
> > ACK sip:699 at 10.255.255.2 SIP/2.0.
> > Record-Route: <sip:10.255.255.1;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Record-Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-17BD;lr=on>.
> > Via: SIP/2.0/UDP 10.255.255.1;branch=0.
> > Via: SIP/2.0/UDP
> 68.86.100.20:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> > Via: SIP/2.0/UDP
> 68.86.100.30:5060;branch=ef5ce1f8a400adab242436d0564cb045.
> > To: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > From: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 1 ACK.
> > Max-Forwards: 3.
> > Contact: sip:3211231234 at 68.86.100.30:5060.
> > Record-Route: <sip:68.86.100.20:5060;lr>.
> > Content-Length: 0.
> > .
> >
> > ###
> > U 2005/02/23 22:24:25.104860 10.255.255.2:5060 -> 10.255.255.1:1033
> > BYE sip:3211231234 at 68.86.100.30:5060 SIP/2.0.
> > Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport.
> > Route: <sip:24.11.12.24;r2=on;ftag=27DECB5C-
> > 17BD;lr=on>,<sip:68.86.100.20:5060;lr>,<sip:3211231234 at 68.86.100.30:5060>.
> > From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Contact: <sip:699 at 10.255.255.2>.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 102 BYE.
> > User-Agent: Asterisk PBX.
> > Content-Length: 0.
> > .
> >
> > #
> > U 2005/02/23 22:24:25.108961 24.11.12.24:5060 -> 68.86.100.30:5060
> > BYE sip:3211231234 at 68.86.100.30:5060 SIP/2.0.
> > Max-Forwards: 10.
> > Record-Route: <sip:24.11.12.24;r2=on;ftag=as588114d9;lr=on>.
> > Record-Route: <sip:10.255.255.1;r2=on;ftag=as588114d9;lr=on>.
> > Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
> > Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> > From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Contact: <sip:699 at 10.255.255.2>.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 102 BYE.
> > User-Agent: Asterisk PBX.
> > Content-Length: 0.
> > Route: <sip:3211231234 at 68.86.100.30:5060>.
> > .
> >
> > #
> > U 2005/02/23 22:24:25.175832 68.86.100.30:5060 -> 24.11.12.24:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP 24.11.12.24;branch=z9hG4bK231b.07949ea5.0.
> > Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> > From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 102 BYE.
> > Content-Length: 0.
> > .
> >
> > #
> > U 2005/02/23 22:24:25.176182 10.255.255.1:1033 -> 10.255.255.2:5060
> > SIP/2.0 200 OK.
> > Via: SIP/2.0/UDP 10.255.255.2:5060;branch=z9hG4bK2da77693;rport=5060.
> > From: 4075551212 <sip:4075551212 at 68.86.100.30:5060>;tag=as588114d9.
> > To: sip:3211231234 at 66.236.245.98;tag=27DECB5C-17BD.
> > Call-ID: 9028535-3318204258-749010 at 68.86.100.30.
> > CSeq: 102 BYE.
> > Content-Length: 0.
> > .
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
More information about the sr-users
mailing list