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$00d939-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
INVITE sip:sipuac1@10.1.1.9:5060;transport=udp SIP/2.0. Record-Route: <sip:10.1.1.1;lr=on;ftag$00d939-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.1.1.1;rportP80;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed. From: "UNKNOWN" <sip:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>. Contact: <sip:9f7900bd-ef13-477c-a490-2e293b886...@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:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0. Via: SIP/2.0/UDP 10.1.1.1:5080;received.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rportP80. 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:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0. Via: SIP/2.0/UDP 10.1.1.1:5080;received.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rportP80. Record-Route: <sip:10.1.1.1;lr=on;ftag$00d939-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@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:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK6f4d.055e1daca77b51d0ceb990515cc44e56.0. Via: SIP/2.0/UDP 10.1.1.1:5080;received.1.1.1;branch=z9hG4bKPj522c1fb9-73b7-4ebb-bf20-44a92ddd13ed;rportP80. Record-Route: <sip:10.1.1.1;lr=on;ftag$00d939-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@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@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.1.1.1;rportP80;branch=z9hG4bKPj04521f53-6781-40c2-8d20-4a0c767dfd3b. From: "UNKNOWN" <sip:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. 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-a490-2e293b886...@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:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. From: <sip:2...@example.com>;tag617a1de20d55ba. Call-ID: 7dc52f67-590e-4482-9065-00bf56512104. CSeq: 440935283 BYE. Route: <sip:10.1.1.1;lr=on;ftag$00d939-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;rportP60;branch=z9hG4bKfcf67cc54. To: "UNKNOWN" <sip:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. From: <sip:2...@example.com>;tag617a1de20d55ba. 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@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.1.1.1;rportP80;branch=z9hG4bKPjfc3f7cf9-24b0-4e14-bb50-7c112912c211. From: "UNKNOWN" <sip:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. Call-ID: 7dc52f67-590e-4482-9065-00bf56512104. CSeq: 2658 BYE. Reason: Q.850;cause. 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:3125551...@example.com>;tag$00d939-de0b-4456-9e01-f9a3302f3e25. To: <sip:2...@example.com>;tag617a1de20d55ba. Via: SIP/2.0/UDP 10.1.1.1;branch=z9hG4bK405d.f9fa0b6296dde81fdf69545e10f428f0.0. Via: SIP/2.0/UDP 10.1.1.1:5080;received.1.1.1;branch=z9hG4bKPjfc3f7cf9-24b0-4e14-bb50-7c112912c211;rportP80. 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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