sr-users September 2007

sr-users@lists.kamailio.org

158 participants
247 discussions

by Morten Isaksen

Hi! I have a strange problem with a missing RTP stream between OpenSER and Asterisk. I am not sure if it is OpenSER og Asterisk related. I have this setup Phone A (172.17.96.17) -- \ Openser -- Asterisk -- PSTN / (192.168.0.6) (192.168.0.3) Phone B (172.17.96.10) -- (172.17.64.1) I also have a Mediaproxy running on OpenSER and I force every call to use the Mediaproxy. I call from Phone A or B to the PSTN works fine and from PSTN to Phone A or B it also works. I have the dialplan logic on my Asterisk server so I want calls from Phone A to Phone B to pass the Asterisk server. And this is were I have the problem. When the call is established the RTP stream is missing between Mediaproxy and Asterisk. I only have a RTP stream between the phones and Mediaproxy. As far as I can see the SIP signalling is correct. The SIP traces is listed below. Can you spot the problem in this? I will buy a beer (or 5) at OpenSER training in Rome to anyone who can help me solve this problem. SIP trace between the phones and OpenSER: U 172.17.96.17:5060 -> 172.17.64.1:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Max-Forwards: 70. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 366. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.96.17. t=0 0. m=audio 10000 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 407 Proxy Authentication Required. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=e7a9ec63e83496804ae077e38a2ef69c.8352. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Proxy-Authenticate: Digest realm="172.17.96.17", nonce="46f2861ed3a501550cd33c28b12936d24c42199c". Server: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. Warning: 392 172.17.64.1:5060 "Noisy feedback tells: pid=15649 req_src_ip=172.17.96.17 req_src_port=5060 in_uri=sip:43336000@172.17.64.1:5060;transport=UDP out_uri=sip:43336000@172.17.64.1:5060;transport=UDP via_cnt==1". . # U 172.17.96.17:5060 -> 172.17.64.1:5060 ACK sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=e7a9ec63e83496804ae077e38a2ef69c.8352. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 1 ACK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccac7-4b008a75. Max-Forwards: 70. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Content-Length: 0. . # U 172.17.96.17:5060 -> 172.17.64.1:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Max-Forwards: 70. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Type: application/sdp. Content-Length: 366. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.96.17. t=0 0. m=audio 10000 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 INVITE sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as28ccd5e9>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Contact: <sip:43336999@192.168.0.3:1265>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Thu, 20 Sep 2007 14:34:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 182. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 172.17.64.1. t=0 0. m=audio 60024 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 SIP/2.0 180 Ringing. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1,SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.1,SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Record-Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Contact: <sip:081700@172.17.96.10:5060;transport=UDP>. Content-Length: 171. Content-Type: application/sdp. . v=0. o=- 926679040 1 IN IP4 172.17.96.10. s=ATI-SIP Call. c=IN IP4 172.17.96.10. t=0 0. m=audio 50600 RTP/AVP 8 13. a=rtpmap:8 PCMA/8000. a=ptime:10. a=rtpmap:13 CN/8000. # U 172.17.64.1:5060 -> 172.17.96.12:5060 CANCEL sip:081700@172.17.96.12:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.0. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. To: <sip:081700@172.17.64.1>. CSeq: 102 CANCEL. User-Agent: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 ACK sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as28ccd5e9>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bKcf8.161cb842.3. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK78618ac6;rport=1265. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 69. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.17:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3:5060>. Content-Type: application/sdp. Content-Length: 238. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 172.17.64.1. t=0 0. m=audio 60022 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 172.17.96.17:5060 -> 172.17.64.1:5060 ACK sip:43336000@192.168.0.3:5060 SIP/2.0. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 ACK. Via: SIP/2.0/UDP 172.17.96.17:5060;rport;branch=z9hG4bK-3e125d66-77bcd96f-26ad0a63. Max-Forwards: 70. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Route: <sip:172.17.64.1;lr;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Length: 0. . # U 172.17.96.10:5060 -> 172.17.64.1:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 70. Supported: 100rel,replaces,timer. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Content-Length: 0. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3:5060>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 172.17.64.1:5060 -> 172.17.96.10:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3:1265>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 172.17.64.1:5060 -> 172.17.96.17:5060 BYE sip:081701@172.17.96.17:5060 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=as119e7abe>. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK4aff.f48c6cd7.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0716a581;rport=5060. Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Contact: <sip:43336000@192.168.0.3>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. User-Agent: Asterisk PBX. Max-Forwards: 69. Content-Length: 0. . # U 172.17.96.17:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK4aff.f48c6cd7.0. Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;branch=z9hG4bK0716a581. Supported: replaces,100rel. Content-Length: 0. . And the SIP trace between OpenSER and Asterisk: # U 192.168.0.6:5060 -> 192.168.0.3:5060 INVITE sip:43336000@172.17.64.1:5060;transport=UDP SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Max-Forwards: 69. Supported: replaces,100rel. Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Type: application/sdp. Content-Length: 365. . v=0. o=081701 1041391881 1041391881 IN IP4 172.17.96.17. s=-. c=IN IP4 172.17.64.1. t=0 0. m=audio 60022 RTP/AVP 8 8 18 4 98 2 101. a=rtpmap:8 PCMA/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:4 g723/8000. a=rtpmap:98 g726-16/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:18 annexb=no. a=fmtp:4 annexa=no. a=fmtp:101 0-15. # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Length: 0. . # U 192.168.0.3:5060 -> 172.17.64.1:5060 INVITE sip:081700@172.17.64.1 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 70. Date: Thu, 20 Sep 2007 14:34:26 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Content-Type: application/sdp. Content-Length: 182. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 192.168.0.3. t=0 0. m=audio 10810 RTP/AVP 8 0. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=silenceSupp:off - - - -. # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Server: OpenSer (1.1.0-notls (i386/linux)). Content-Length: 0. Warning: 392 172.17.64.1:5060 "Noisy feedback tells: pid=15647 req_src_ip=192.168.0.3 req_src_port=1265 in_uri=sip:081700@172.17.64.1 out_uri=sip:081700@172.17.96.12:5060;transport=UDP via_cnt==1". . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 180 Ringing. User-Agent: ATI-RG613TX/2-3_59. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAAGNwAN2gAZewxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 180 Ringing. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 SIP/2.0 200 OK. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK6ab42128;rport=1265. From: 081701 <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 INVITE. Supported: 100rel,replaces,timer. Record-Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Contact: <sip:081700@172.17.96.10:5060;transport=UDP>. Content-Length: 170. Content-Type: application/sdp. . v=0. o=- 926679040 1 IN IP4 172.17.96.10. s=ATI-SIP Call. c=IN IP4 172.17.64.1. t=0 0. m=audio 60024 RTP/AVP 8 13. a=rtpmap:8 PCMA/8000. a=ptime:10. a=rtpmap:13 CN/8000. # U 192.168.0.3:5060 -> 172.17.64.1:5060 ACK sip:081700@172.17.96.10:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK78618ac6;rport. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. From: "081701" <sip:43336999@192.168.0.3>;tag=as28ccd5e9. To: <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. Contact: <sip:43336999@192.168.0.3>. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 102 ACK. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d63-77bccb03-1bf1c302. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336000@192.168.0.3>. Content-Type: application/sdp. Content-Length: 238. . v=0. o=root 21472 21472 IN IP4 192.168.0.3. s=session. c=IN IP4 192.168.0.3. t=0 0. m=audio 12684 RTP/AVP 8 0 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. # U 192.168.0.6:5060 -> 192.168.0.3:5060 ACK sip:43336000@192.168.0.3:5060 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. To: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 2 ACK. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.17:5060;rport=5060;branch=z9hG4bK-3e125d66-77bcd96f-26ad0a63. Max-Forwards: 69. Contact: <sip:081701@172.17.96.17:5060;transport=UDP>. Route: <sip:172.17.64.1;lr;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. Proxy-Authorization: Digest username="081701",realm="172.17.96.17",nonce="46f2861ed3a501550cd33c28b12936d24c42199c",uri="sip:43336000@172.17.64.1:5060;transport=UDP",response="4bf0194044fb4a759ec6134d68bef752",algorithm=MD5. Content-Length: 0. . # U 192.168.0.6:5060 -> 192.168.0.3:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=0. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 69. Supported: 100rel,replaces,timer. Route: <sip:172.17.64.1;lr;ftag=as28ccd5e9>. Content-Length: 0. . # U 172.17.64.1:5060 -> 192.168.0.3:5060 BYE sip:43336999@192.168.0.3:1265 SIP/2.0. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. User-Agent: ATI-iMG616RF+/3-5_78. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK26ae.adc1e03.0. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. Max-Forwards: 69. Supported: 100rel,replaces,timer. Content-Length: 0. P-hint: rr-enforced. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=0;received=192.168.0.6. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 192.168.0.3:5060 -> 172.17.64.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.17.64.1;branch=z9hG4bK26ae.adc1e03.0;received=172.17.64.1. Via: SIP/2.0/UDP 172.17.96.10:5060;branch=z9hG4bKAABFNwxx. Record-Route: <sip:172.17.64.1;lr=on;ftag=AAA5NwAN2ggCmwxx>. From: 081700 <sip:081700@172.17.64.1>;tag=AAA5NwAN2ggCmwxx. To: <sip:43336999@192.168.0.3>;tag=as28ccd5e9. Call-ID: 473e63475049c3e6481c39555040cf75(a)192.168.0.3. CSeq: 1 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Contact: <sip:43336999@192.168.0.3>. Content-Length: 0. X-Asterisk-HangupCause: Normal Clearing. . # U 192.168.0.3:5060 -> 192.168.0.6:5060 BYE sip:081701@172.17.96.17:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK0716a581;rport. Route: <sip:172.17.64.1;lr=on;ftag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63>. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Contact: <sip:43336000@192.168.0.3>. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. User-Agent: Asterisk PBX. Max-Forwards: 70. Content-Length: 0. . # U 192.168.0.6:5060 -> 192.168.0.3:5060 SIP/2.0 200 OK. From: "43336000"<sip:43336000@172.17.64.1:5060>;tag=as119e7abe. To: "081701"<sip:081701@172.17.96.17>;tag=100c3d98-116011ac-13c4-3e125d63-6075d01c-3e125d63. Call-ID: 100c7470-116011ac-13c4-3e125d63-539dcd67-3e125d63(a)172.17.96.17. CSeq: 102 BYE. Via: SIP/2.0/UDP 192.168.0.3:5060;rport=5060;branch=z9hG4bK0716a581. Supported: replaces,100rel. Content-Length: 0. . -- Morten Isaksen http://www.misak.dk/blog/

17 years, 2 months

[OpenSER-Users] Forward REGISTER to Asterisk

by Halomoan Chow

Dear All I want user who register with my openser also register with my asterisk for later lookup. I tried t_replicate function from TM module but no luck. I end up with Unauthorized message. Could any help me how to make this work? What do I miss ? Thank so much Regards, Halomoan Zhou

17 years, 2 months

Re: [OpenSER-Users] test operator]

by Klaus Darilion

Bogdan-Andrei Iancu schrieb: > Hi Klaus, > > the [..] operator is used to convert generic numerical value (maybe > result of arithmetic ops to logical value. > > in your example, the == is already a logical operator, so no need to any > conversion. Does ($rd == "mydomain") works too? is this somewhere documented? The corebook only describes assignments, but not conditions/comparisons. regards klaus

17 years, 2 months

[OpenSER-Users] tcp receiver with tcp socket?

by Klaus Darilion

Hi! I have configured openser 1.2 to only listen to udp: Starting Open SIP Router: Listening on udp: 1.2.3.4 [1.2.3.4]:5060 Aliases: *: mydomain.at:* $ openser-1.2-SVN/scripts/openserctl ps Process:: ID=0 PID=29582 Type=attendant Process:: ID=1 PID=29583 Type=receiver child=0 sock= 1.2.3.4:5060 Process:: ID=2 PID=29585 Type=receiver child=1 sock= 1.2.3.4:5060 Process:: ID=3 PID=29586 Type=receiver child=2 sock= 1.2.3.4:5060 Process:: ID=4 PID=29587 Type=receiver child=3 sock= 1.2.3.4:5060 Process:: ID=5 PID=29588 Type=receiver child=4 sock= 1.2.3.4:5060 Process:: ID=6 PID=29589 Type=receiver child=5 sock= 1.2.3.4:5060 Process:: ID=7 PID=29590 Type=receiver child=6 sock= 1.2.3.4:5060 Process:: ID=8 PID=29591 Type=receiver child=7 sock= 1.2.3.4:5060 Process:: ID=9 PID=29592 Type=timer Process:: ID=10 PID=29593 Type=tcp receiver Process:: ID=11 PID=29594 Type=tcp receiver Process:: ID=12 PID=29595 Type=tcp receiver Process:: ID=13 PID=29596 Type=tcp receiver Process:: ID=14 PID=29597 Type=tcp receiver Process:: ID=15 PID=29598 Type=tcp receiver Process:: ID=16 PID=29599 Type=tcp receiver Process:: ID=17 PID=29600 Type=tcp receiver Process:: ID=18 PID=29601 Type=tcp main process So, where does all the "tcp receiver" come from? If they are DB related maybe we should change the "Type". This proxy uses mysql for presence - this I guess all the TCP connections are for presence. But why do I have 19 mysql connections? $ netstat -anp|grep openser tcp 0 0 1.2.3.4:48896 1.2.3.68:3306 ESTABLISHED 29598/openser tcp 0 0 1.2.3.4:48897 1.2.3.68:3306 ESTABLISHED 29599/openser tcp 0 0 1.2.3.4:48898 1.2.3.68:3306 ESTABLISHED 29600/openser tcp 0 0 1.2.3.4:48899 1.2.3.68:3306 ESTABLISHED 29601/openser tcp 0 0 1.2.3.4:48900 1.2.3.68:3306 ESTABLISHED 29582/openser tcp 0 0 1.2.3.4:48889 1.2.3.68:3306 ESTABLISHED 29591/openser tcp 0 0 1.2.3.4:48888 1.2.3.68:3306 ESTABLISHED 29590/openser tcp 0 0 1.2.3.4:48891 1.2.3.68:3306 ESTABLISHED 29593/openser tcp 0 0 1.2.3.4:48890 1.2.3.68:3306 ESTABLISHED 29592/openser tcp 0 0 1.2.3.4:48893 1.2.3.68:3306 ESTABLISHED 29595/openser tcp 0 0 1.2.3.4:48892 1.2.3.68:3306 ESTABLISHED 29594/openser tcp 0 0 1.2.3.4:48895 1.2.3.68:3306 ESTABLISHED 29597/openser tcp 0 0 1.2.3.4:48894 1.2.3.68:3306 ESTABLISHED 29596/openser tcp 0 0 1.2.3.4:48883 1.2.3.68:3306 ESTABLISHED 29585/openser tcp 0 0 1.2.3.4:48882 1.2.3.68:3306 ESTABLISHED 29583/openser tcp 0 0 1.2.3.4:48885 1.2.3.68:3306 ESTABLISHED 29587/openser tcp 0 0 1.2.3.4:48884 1.2.3.68:3306 ESTABLISHED 29586/openser tcp 0 0 1.2.3.4:48887 1.2.3.68:3306 ESTABLISHED 29589/openser tcp 0 0 1.2.3.4:48886 1.2.3.68:3306 ESTABLISHED 29588/openser udp 0 0 1.2.3.4:5060 0.0.0.0:* 29582/openser Thus, why 19 DB connections for 8 UDP childrens? Why 8 tcp receivers in "openserctl ps"? thanks klaus

17 years, 2 months

[OpenSER-Users] test operator

by Klaus Darilion

Hi! from the core cookbook of 1.2: > If you want to evaluate arithmetic operations and test the value in > conditions, use the test operator ' [ ... ] ' . This is required to > convert the result to internal convention for drop, true and false. > > if ( [ $var(a)&4 ] ) { > xlog("bit 3 is set in var a"); > } Is there somewhere a detailed description of the test operator? Can it only be used to test for 0/1 or can I also do comparison like e.g. if ([ $rd == "mydomain.at" ]) ? regards klaus

17 years, 2 months

[OpenSER-Users] Unregister a user when not responding to SIP pings

by Iñaki Baz Castillo

Hi, many times a softphone or PC crashes without sending the un-REGISTER, so the user remains in location table. OpenSer allows sending SIP ping to users, is not possible to delete the location entry for a user who has left responding to those SIP pings? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

17 years, 2 months

[Serusers] Ser web configuration

by Tuhin

Please help me to configure serweb by telling me the procedure.

17 years, 2 months

[OpenSER-Users] Do we like the same books?

by rizwan_hisham

I just joined Shelfari to connect with other book lovers. Come see the books I love and see if we have any in common. Then pick my next book so I can keep on reading. Click below to join my group of friends on Shelfari! http://www.shelfari.com/Register.aspx?ActivityId=16332510&InvitationCode=3d… rizwan_hisham Shelfari is a free site that lets you share book ratings and reviews with friends and meet people who have similar tastes in books. It also lets you build an online bookshelf, join book clubs, and get good book recommendations from friends. You should check it out. -------- You have received this email because rizwan_hisham (rizwanhasham(a)gmail.com) directly invited you to join his/her community on Shelfari. It is against Shelfari's policies to invite people who you don't know directly. Follow this link (http://www.shelfari.com/actions/emailoptout.aspx?email=Users@openser.org&ac…) to prevent future invitations to this address. If you believe you do not know this person, you may view (http://www.shelfari.com/rizwan_hisham) his/her Shelfari page or report him/her in our feedback (http://www.shelfari.com/Feedback.aspx) section. Shelfari, 616 1st Ave #300, Seattle, WA 98104

17 years, 2 months

[Serusers] Mediaproxy in public IP e two client behind same Nat

by pistara_bros＠tin.it

Hi, I want add others information about my test, providing also the log of MediaProxy WEB GUI. Windows Client used : X-Lite Version 3.0 build 29712 First scenarious : Client_A and Client_B running on Windows XP in the same LAN behind same NAT. Client_A call Client_B and During the conversation MediaProxy WEB GUI (http://smile.ing.uniroma1. it/serweb/mediaproxy/) show me following dump: Call | Caller |Called |Via address |status |Duration|.... ------------------------------|---------------- |---------------------------|-------------------| F:j2me@151. 100.122.144 | - | - |151.100.122.144:60012|inactive| 0'59'' |.... To:enzo@151.100.122.144| - | - | | | | And after 60 seconds the session expired and MediaProxy WEB GUI don't show it. Result: the RTP packet dont' pass through the Mediaproxy, but the packet go directly from one computer to another of my Lan. Second scenarious: Windows Client used : X-Lite Version 3.0 build 29712 Linux Client used : X-Lite Version 2.0 build 1105d Client_A runnning on WindowsXP and Client_B running on Linux Fedora in the same LAN behind same NAT. Client_A call Client_B and During the conversation MediaProxy WEB GUI (http://smile.ing.uniroma1.it/serweb/mediaproxy/) show me following dump: Call | Caller |Called --------------------------------|-------------------------- |--------------------------- F:j2me@151.100.122.144 | 85.39.206.210: 62969 |85.39.206.210:62947 To: enzo(a)151.100.122.144 | | |Via address |status |Duration|.... |151.100.122.144:60010 |active | 1'25'' |.... Result: the RTP packet pass through the Mediaproxy, that acts as "man in the middle" anf forward RTP packet versus clients of my Lan. Why MediaProxy, in windowz -> windowz client communications doesn't work????? Orazio >Hi to All, >I'm investigating again about Mediaproxy and SER , but in >this test i'm using the >the testbed of smile.ing.uniroma1.it whic use >SER and Mediaproxy in a public ip 151.100.122.144 >I'm using Client_A (X- >lite) running on WindowsXP 192.168.123.10 >and Client_B (X-lite) running >on WindowsXP 192.168.123.168 >My Lan use a router with a public ip >83.211.41.253 >So the clients are behind the same Nat. >Using ethereal i >see that during the conversation the RTP packet dont' pass through the >Mediaproxy in public ip (151.100.122.144) but the packet go directly >from one computer to another of my Lan. >So the question is : >what SER >architecture (with Mediaproxy or with SEMS or other..) i have to use in >order to >have a proxy RTP that work as man in the middle and manage all >RTP Traffic between the two clients? >Please help me. >thanks.

17 years, 2 months

[OpenSER-Users] Free BSD 6.0 and SEAS problem.

by Kuddusi

Hi, I am using default seas configuration. When I tried to start openser seas module can not be loaded althoug module is compiled without problem. I tried 1.2.0 and 1.2.2 and the same problem occurred. It seems that it could not find "dprintf" function. How can I solve this problem? root@~: openser -ddddddddddd 0(43499) loading module /usr/local/lib/openser/modules/tm.so 0(43499) DEBUG: register_pv: tm 0(43499) xl_add_extra_spec: extra items list is not initialized 0(43499) xl_add_extra_spec: inserting extra item [T_branch_idx] at [0] 0(43499) xl_add_extra_spec: inserting extra item [T_reply_code] at [1] 0(43499) loading module /usr/local/lib/openser/modules/seas.so 0(43499) ERROR: load_module: could not open module </usr/local/lib/openser/modules/seas.so>: /usr/local/lib/openser/modules/seas.so: Undefined symbol "dprintf" and config is below Thanks, Kuddusi CIFTCIBASI. # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) children=4 # Uncomment these lines to enter debugging mode fork=no log_stderror=yes # reply_to_via=1 listen = tcp:84.51.32.24:5060 listen = udp:84.51.32.24:5060 port=5060 # ------------------ module loading ---------------------------------- #set module path #mpath="/usr/local/lib/openser/modules/" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/seas.so" modparam("seas", "listen_sockets","84.51.32.24:5080") modparam("seas", "stats_socket","84.51.32.24:5088") #modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ if(!as_relay_t("app_server_one")){ t_reply("500","No App Server"); } }

17 years, 2 months

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sr-users September 2007