[SR-Users] Routing PSTN calls from Asterisk through Kamailio to UAC & loose_route
Anthony Messina
amessina at messinet.com
Thu Mar 26 01:18:11 CET 2015
I've been working on integration of Asterisk and Kamailio, currently on the
same host with different ports, and have come across a problem with calls that
originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a
SIP UAC. In short, when the SIP UAC (10.1.1.9) sends the BYE, loose_route()
is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead
of Asterisk (10.1.1.1:5080). I am using the stock WITHINDLG route
configuration.
RR module settings are as follows:
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 1)
The BYE from the SIP UAC contains the following Route header which only
contains the contents of Kamailio's Record-Route header. I have attached the
full sip trace for review as well.
Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
What would be the best method to resolve this issue in either Asterisk or
Kamailio? Should I manually add a Record-Route header for the Asterisk
host:port to Kamailio config? Is there something to be done in Asterisk?
Thanks. -A
--
Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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INVITE sip:sipuac1 at 10.1.1.9:5060;transport=udp SIP/2.0.
Record-Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;rport=5080;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>.
Contact: <sip:9f7900bd-ef13-477c-a490-2e293b886505 at example.com>.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2657 INVITE.
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE.
Supported: 100rel, timer, replaces, norefersub.
Session-Expires: 1800.
Min-SE: 90.
Alert-Info: info=<Bellcore-dr2>.
Max-Forwards: 69.
User-Agent: Asterisk PBX 13.2.0.
Content-Type: application/sdp.
Content-Length: 319.
Route: <sip:10.1.1.1:5080>.
.
v=0.
o=- 1141388262 1141388262 IN IP4 10.1.1.1.
s=Asterisk.
c=IN IP4 10.1.1.1.
t=0 0.
m=audio 30520 RTP/AVP 9 0 3 97 101.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:97 iLBC/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=maxptime:30.
a=sendrecv.
a=rtcp:30521.
#
U 2015/03/25 17:42:39.882274 10.1.1.9:5060 -> 10.1.1.1:5060
SIP/2.0 100 Trying.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2657 INVITE.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rport=5080.
Content-Length: 0.
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
.
#
U 2015/03/25 17:42:39.911087 10.1.1.9:5060 -> 10.1.1.1:5060
SIP/2.0 180 Ringing.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2657 INVITE.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rport=5080.
Record-Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
Content-Length: 0.
Call-Info: <sip:example.com>;appearance-index=1.
Allow-Events: talk, hold, conference.
Contact: sip:sipuac1 at 10.1.1.9:5060.
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
.
#
U 2015/03/25 17:42:50.012951 10.1.1.1:5060 -> 10.1.1.9:5060
....
#
U 2015/03/25 17:42:51.696604 10.1.1.9:5060 -> 10.1.1.1:5060
SIP/2.0 200 OK.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2657 INVITE.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rport=5080.
Record-Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
Content-Length: 203.
Session-Expires: 1800;refresher=uac.
Require: timer.
Call-Info: <sip:example.com>;appearance-index=1.
Allow-Events: talk,hold,conference.
Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO.
Content-Type: application/sdp.
Supported: replaces.
Contact: sip:sipuac1 at 10.1.1.9:5060.
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
.
v=0.
o=MxSIP 0 1634248351 IN IP4 10.1.1.9.
s=SIP Call.
c=IN IP4 10.1.1.9.
t=0 0.
m=audio 3000 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
#
U 2015/03/25 17:42:51.700088 10.1.1.1:5060 -> 10.1.1.9:5060
ACK sip:sipuac1 at 10.1.1.9:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.f27b2296b5fa7279ff08c135da90b4e4.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;rport=5080;branch=z9hG4bKPj04521f53-6781-40c2-8d20-4a0c767dfd3b.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2657 ACK.
Max-Forwards: 69.
User-Agent: Asterisk PBX 13.2.0.
Content-Length: 0.
.
#
U 2015/03/25 17:43:11.690086 10.1.1.9:5060 -> 10.1.1.1:5060
BYE sip:9f7900bd-ef13-477c-a490-2e293b886505 at example.com SIP/2.0.
Via: SIP/2.0/UDP 10.1.1.9:5060;branch=z9hG4bKfcf67cc54.
Max-Forwards: 70.
Content-Length: 0.
To: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
From: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 440935283 BYE.
Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>.
Supported: timer.
Call-Info: <sip:example.com>;appearance-index=1.
Supported: replaces.
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
.
#
U 2015/03/25 17:43:11.690728 10.1.1.1:5060 -> 10.1.1.9:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 10.1.1.9:5060;rport=5060;branch=z9hG4bKfcf67cc54.
To: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
From: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 440935283 BYE.
Server: Kamailio.
Content-Length: 0.
.
#
U 2015/03/25 17:43:20.015806 10.1.1.1:5060 -> 10.1.1.9:5060
....
#
U 2015/03/25 17:43:26.453279 10.1.1.1:5060 -> 10.1.1.9:5060
BYE sip:sipuac1 at 10.1.1.9:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK405d.f9fa0b6296dde81fdf69545e10f428f0.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;rport=5080;branch=z9hG4bKPjfc3f7cf9-24b0-4e14-bb50-7c112912c211.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2658 BYE.
Reason: Q.850;cause=16.
Max-Forwards: 69.
User-Agent: Asterisk PBX 13.2.0.
Content-Length: 0.
.
#
U 2015/03/25 17:43:26.501109 10.1.1.9:5060 -> 10.1.1.1:5060
SIP/2.0 481 Call Does Not Exist.
Call-ID: 7dc52f67-590e-4482-9065-00bf56512104.
CSeq: 2658 BYE.
From: "UNKNOWN" <sip:3125551212 at example.com>;tag=2400d939-de0b-4456-9e01-f9a3302f3e25.
To: <sip:2200 at example.com>;tag=3617a1de20d55ba.
Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK405d.f9fa0b6296dde81fdf69545e10f428f0.0.
Via: SIP/2.0/UDP 10.1.1.1:5080;received=10.1.1.1;branch=z9hG4bKPjfc3f7cf9-24b0-4e14-bb50-7c112912c211;rport=5080.
Content-Length: 0.
User-Agent: Aastra 480i Cordless/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45.
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