September 2007 - sr-users

Re: [OpenSER-Users] Missing RTP stream
by Morten Isaksen 20 Sep '07

20 Sep '07

Hi! I can see in the mediaproxy log the it is initialized to proxy the call, but I newer get a "session xxxxx: called signed in from xxx" from Asterisk. session.py shows that the the connection between mediaproxy and Asterisk is missing. I will try to take a look at the sip debug from asterisk and try to change the NAT settings in Asterisk. Thanks for your input. On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > You stated that you've forced every call through mediaproxy. … [View More]Are you > positive ? > > Have you taken a look at the mediaproxy logs (and/or sessions.py when > the call is up) ? They might provide some useful information to you. > > Ditto for Asterisk "sip set debug on" (note that the sip debug command > format is a moving target). > > Have you looked at the "nat=" settings in sip.conf as well ? At times, > they tie closely with "canreinvite=". > > Norm > > > Morten Isaksen wrote: > > Hi! > > > > canreinvite is set to no and the OpenSER/mediaproxy is "hiding" the > > clients IP-addresses from Asterisk, so I am pretty sure that this is > > not the issue. > > > > On 9/20/07, Norman Brandinger <norm(a)goes.com> wrote: > > > >> Hi Morten, > >> > >> Admittedly, I haven't looked closely at your trace. However, based on > >> the description you gave, the first place to look is at the "canrevite" > >> setting in Asterisk sip.conf. You might want to try "canreinvite=no" > >> after reading up on this particular setting. > >> > >> Regards, > >> Norm > >> > >> > >> Morten Isaksen wrote: > >> > >>> 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: > >>> > >>> > > > > > > > > -- Morten Isaksen http://www.misak.dk/blog/ [View Less]

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[OpenSER-Users] Missing RTP stream
by Morten Isaksen 20 Sep '07

20 Sep '07

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 … [View More]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/ [View Less]

2 2

[OpenSER-Users] Forward REGISTER to Asterisk
by Halomoan Chow 20 Sep '07

20 Sep '07

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

4 4

Re: [OpenSER-Users] test operator]
by Klaus Darilion 20 Sep '07

20 Sep '07

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

2 1

[OpenSER-Users] tcp receiver with tcp socket?
by Klaus Darilion 20 Sep '07

20 Sep '07

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=… [View More]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 [View Less]

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[OpenSER-Users] test operator
by Klaus Darilion 20 Sep '07

20 Sep '07

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 == "… [View More]

2 1

[OpenSER-Users] Unregister a user when not responding to SIP pings
by Iñaki Baz Castillo 20 Sep '07

20 Sep '07

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

4 10

[Serusers] Ser web configuration
by Tuhin 20 Sep '07

20 Sep '07

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

1 0

[OpenSER-Users] Do we like the same books?
by rizwan_hisham 20 Sep '07

20 Sep '07

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, … [View More]

1 0

[Serusers] Mediaproxy in public IP e two client behind same Nat
by pistara_bros＠tin.it 20 Sep '07

20 Sep '07

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|....… [View More] ------------------------------|---------------- |---------------------------|-------------------| 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. [View Less]

1 0

[OpenSER-Users] Free BSD 6.0 and SEAS problem.
by Kuddusi 20 Sep '07

20 Sep '07

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) … [View More]

2 2

[OpenSER-Users] Do we like the same books?
by voipexpert 20 Sep '07

20 Sep '07

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=16229378&InvitationCode=84… voipexpert 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 … [View More]

1 0

[Serusers] Creating a test scenario to test NATing..
by Piyush.Bansal＠relianceada.com 20 Sep '07

20 Sep '07

hi all, I wanna simulate a test scenario in which the sip clients are behind NAT (whether same or different NAT, in some private network), and the SER and MediaProxy are running on some public IP. My question is, in this case the client is not able to communicate with the server through the Firewall / NAT. My SIP messages are being dropped at the Firewall itself. Can anybody tell me, what could be the reason behind that????? I'm totally stuck at this point..... plz revert asap... thanx in … [View More]

1 0

[OpenSER-Users] openser + freeradius + cdrtool in freebsd
by Sam Wun 20 Sep '07

20 Sep '07

Hi, I found openser and freeradius in the freebsd ports tree, but the cdrtools is a tool for CD ISO in freebsd. Is this the right cdrtools for freesradius? In the CDRtools website, it said: http://download.dns-hosting.info/CDRTool/ "CDRTool provides mediation, accounting and tracing for Call Detail Records generated by OpenSER by using RADIUS protocol and OpenSER siptrace facility. " It seems that the CDrtools in freebsd ports tree is not the one for freeradius. Thanks W

1 0

Re: [OpenSER-Users] OpenSER with MySQL Cluster
by Christian Schlatter 19 Sep '07

19 Sep '07

Kerker Staffan wrote: > hi > ok, thanks for the update. i kind of dropped that > track as i had all those errors. > > with your experience, is that a better solution than > using the t_replicate? i'm only synchronizing two servers. Well, there is a trade-off between the complexity of a mysql cluster deployment and the possible inconsistencies introduced by t_replicate. With mysql cluster, you can always be sure that both of your proxies have the same, consistent view of … [View More]the registration state. This comes at the prize of deploying and maintaining the db cluster. t_replicate is a lightweight replication mechanism that doesn't ensure registration state consistency. I like the simplicity of this approach. Personally, I prefer having a consistent registration state. I also think that the mysql guys have done a great job in making the cluster deployment and management as easy as possible. /Christian > > best regards, > /staffan > > >> -----Original Message----- >> From: Christian Schlatter [mailto:cs@unc.edu] >> Sent: den 19 september 2007 15:42 >> To: Kerker Staffan >> Cc: Juha Heinanen; users(a)openser.org >> Subject: Re: [OpenSER-Users] OpenSER with MySQL Cluster >> >> >> Kerker Staffan wrote: >>> hi >>> i'm curious about this one. we used to run a dual Openser >> installation >>> with MySQL cluster, but this proved to very unstable (on >> the cluster >>> side). >>> is this working fine now? >>> >>> for instance, if one of the mysql cluster nodes was >> disconnected, this >>> could lead to complete cluster shut down, and nodes didn't handle >>> reconnect very good. >>> >>> so, we actually switched to the register replication instead... >> We're running a mysql 5.0.22 cluster for our two redundant >> openser proxies. This setup works very stable for about a year now. >> >> In the beginning our cluster was shutting down itself quite >> regularly e.g. after a network failure. Since then we >> >> - are running the management nodes on hosts that are >> connected to the data nodes through different network paths >> (this is essential, a cluster node needs to be able to solve >> the splitbrain problem) >> >> - have added 'StopOnError = 0' to the data node settings >> (this causes a data nodes to try to reconnect after it lost >> communication to its neighbors, per default, a data node just >> shuts itself down) >> >> - added enough DataMemory and IndexMemory for the data nodes >> (DataMemory = 1024M, IndexMemory = 64M) >> >> >> Overall I'm quite pleased with our mysql cluster solution, >> although I'd >> prefer a solution where the UAs do register with both of our >> proxies, so >> that there is no need to replicate registration state between >> the proxies. >> >> Our mysql cluster basically only contains the 'location' >> table since we >> store all account data on redundant LDAP/H350 directories. >> >> >> /Christian >> >> >> >>> best regards >>> /Staffan >>> >>> >>>> -----Original Message----- >>>> From: users-bounces(a)openser.org >>>> [mailto:users-bounces@openser.org] On Behalf Of Christian Schlatter >>>> Sent: den 18 september 2007 23:37 >>>> To: Juha Heinanen >>>> Cc: users(a)openser.org >>>> Subject: Re: [OpenSER-Users] OpenSER with MySQL Cluster >>>> >>>> >>>> Juha Heinanen wrote: >>>> ... >>>>> unfortunately you then cannot support presence. using >>>> another proxy >>>>> for presence is not a solution either, because same users >> need both >>>>> presence and other sip methods. >>>> What exactly prevents me from using a dedicated presence >>>> server? And why can't I support presence with db_mode=3? >>>> >>>> E.g. if I just want to deploy a simple PA using the PRESENCE >>>> module, I don't see any dependencies on USRLOC. >>>> >>>> I'm not sure if PUA_USRLOC depends on db_mode<3, but at least >>>> the documentation doesn't mention it. And it looks to me like >>>> I could use >>>> pua_set_publish() from PUA_USRLOC to send a PUBLISH to my >>>> dedicated presence server ;-) ... I haven't tried that though. >>>> >>>>> by the way, looks like presence related requests are by >> far the most >>>>> common ones if all sip UAs have presence turned on. it means that >>>>> presence would also create biggest db activity. >>>> Yes, I know, SIP presence is very chatty. >>>> >>>> >>>> /Christian >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>> >> [View Less]

1 0

[Serusers] Mediaproxy in public IP e two client behind same Nat
by pistara_bros＠tin.it 19 Sep '07

19 Sep '07

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 … [View More]

1 0

[OpenSER-Users] UAC module with Asterisk
by Halomoan Chow 19 Sep '07

19 Sep '07

Dear All Now I want to try UAC module to solve the proxy authentication to Asterisk. Until now it still fail even the username/password and realm are already correct 100% Then I found this statement "For sending authentication credentials to the termination provider openser has the "uac" module. But be aware that this module does not always works as it does not increase the cseq when sending the INVITE with credentials (search the archive for details). E.g. the uac module wont work if the … [View More]

1 0

[Serusers] MeadiaProxy + SER + X-lite in the same Lan.
by pistara_bros＠tin.it 19 Sep '07

19 Sep '07

I want correct my previous posted sentences : "My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call." with "My guess is mediaproxy makes a test on the IPs (they are 192.168.123.168 and 192.168.123.10) and refuses to proxy the call." Does Anyone encounter this problem? could you share your opinion, with me?. Thanks. Orazio

1 0

[OpenSER-Users] memory allocation issue
by Benko 19 Sep '07

19 Sep '07

Hello! I wanted to upgrade a server from 1.1.0 to 1.2.2 today but despite extensive testing beforehand i have some trouble: After starting with the new openser-binary, it didn't allow any registrations and answered with "500 server error occured". Logging said: Sep 18 17:23:12 [/sbin/openser] ERROR: sip_msg_cloner: cannot allocate \ memory_ Sep 18 17:23:12 [/sbin/openser] ERROR: new_t: out of mem:_ Sep 18 17:23:12 [/sbin/openser] ERROR: t_newtran: new_t failed_Sep 18 and 17:23:12 [/sbin/… [View More]

2 3

[OpenSER-Users] OpenSER with MySQL Cluster
by Vasile Zaharia 19 Sep '07

19 Sep '07

Hi *, Normally, OpenSER should work just fine with a MySQL Cluster, but I would like to know if anyone has tried this configuration yet. Are there any configuration settings I should be aware of? Thanks, Sica. ____________________________________________________________________________________ Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos & more. http://mobile.yahoo.com/go?refer=1GNXIC

8 16

[OpenSER-Users] Configuration of Permissions module
by Sergio Gutierrez 19 Sep '07

19 Sep '07

Hi everybody. Does anybody has any example about how to configure the permissions module to use address validation from database configuration tables? In our setup, we observe an inconsistent behaviour when defining subnets; fifo command subnet_dump does not show the expected subnet. The only case wich works well is wehn mask is set to 32. Thanks in advance for your attention. Sergio Gutierrez EPM Telecomunicaciones Medellin - Colombia

1 0

[OpenSER-Users] example web_im send_im.html/php within openser 1.2.1 doensn't work
by CHEN XUEQIN 19 Sep '07

19 Sep '07

Hi all: I put example script file send_im.html/send_im.php in example/web_im direcotry into apache webroot. Then I launch firefox, fill in SIP Address and Message, click to send. But firefox don't response. The status bar indicated wait something. The fact is: 1. openser load mi_fifo.so correctly 2. the fifo file /tmp/openser_fifo, with the permission 0666, everyone can read and write 3. LOG message " DEBUG:mi_fifo:mi_fifo_server: entered consume", missing log "**** done … [View More]

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Re: [OpenSER-Users] OpenSER with MySQL Cluster
by Helge Waastad 19 Sep '07

19 Sep '07

Hi My experiences with MySQL clusters is that you have to expand the index memory etc as far as you dear to enable adding new tables/columns etc. I guess yopu would not like to do that in a production evnvironment anyway... br hw

1 0

[OpenSER-Users] user meeting (beer drinking in Vienna)
by Klaus Darilion 19 Sep '07

19 Sep '07

Hi! I proudly announce the first ser/openser/asterisk beer drinking evening in Vienna. When: Thursday (thirsty day) 20. September 2007, 19:00 CEST Where: Vienna, a bar in an inner district - exact location to be announced If you want to join please send a short reply so that I know for how many people I have to arrange a table. cu Klaus

6 6

[Serusers] help needed in configuring ENUM with ENUM module
by geethanjali.csg＠wipro.com 19 Sep '07

19 Sep '07

Hi, I am trying to configure ser to use enum module. But I am getting the following error. 0(6192) get_record: lookup(5.6.7.8.e164.arpa, 35) failed 0(6192) enum_query(): No NAPTR record found for 5.6.7.8.e164.arpa. Can someone kindly give me some pointers to fix this error. I am trying this from past few days. Thanks, Geethanjali C S Wipro Technologies The information contained in this electronic message and any attachments to this message are intended for the exclusive … [View More]

3 5

[OpenSER-Users] mediaproxy work for muti domains
by liu york 18 Sep '07

18 Sep '07

Dear all on my mind，mediaproxy can work for muti sip domains， like it's feature list―― -Ability to have different proxies for different SIP domains - Ability to use multiple proxy servers per SIP domain Now，using multiple proxy servers per SIP domain is ok for me，but i cant make it work with different SIP domains.Can somebody tell me how to do? mediaproxy.ini like below start = yes socket = /var/run/proxydispatcher.sock group = openser defaultProxy = domain://sip1.gteling.com the zone file … [View More]

2 1

Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Iñaki Baz Castillo 18 Sep '07

18 Sep '07

El Martes, 18 de Septiembre de 2007, Gregorio escribió: > Hi Elias et all, Please Gregorio, por future new questions to the list use the "New message" button in your Microsoft Office Outlook instead of replying to any other mail in the list and deleting body and subject. The mail you have sent has been created by pressing "Reply" over a mail with subject "Per branch timer and CANCEL on non final reply", so your mail appears in the same thread because it has a header: In-Reply-To: … [View More]

1 0

[OpenSER-Users] OSPToolkit Warning Messages
by Edson 18 Sep '07

18 Sep '07

Hi... I'm trying to compile OSP support for eather 1.2.2 or TRUNK. I have OpenSSL 0.9.8d on a OpenSUSE 10.2. I was wondering if the warning that pops are something that desire attention, since are related to the X509 part: ospopenssl.c: In function âbio_dump_cbâ: ospopenssl.c:597: warning: format â%08Xâ expects type âunsigned intâ, but argument 3 has type âstruct BIO *â ospopenssl.c:597: warning: format â%08lXâ expects type âlong unsigned intâ, but argument 4 has type âconst char *â … [View More]

1 0

[OpenSER-Users] snmpstats agent failing to connect to master agent
by Gabi Mahu 18 Sep '07

18 Sep '07

Hello all, I am currently trying to get the snmpstats module up and running, for monitoring purposes. I followed the instructions in the module documentation, I studied Net-SNMP references and I searched web resources for several days now, but still I couldn't succeed in starting the openser module properly. I have no idea what the problem could be. The NET-SNMP part of the equation seams to work properly: I am using the 5.4.1 version. I managed to play around with the tools from the … [View More]package, in local and remote scenarios too. I was able to send and receive traps, get and set different values in the MIB tree, load new MIBs... On the openser part, I'm not doing something special. I have a config file that does nothing, except load the snmpstats module and set the limits for the dialogs and the unparsed data traps. That's it. I tried with fork on yes and on no, because I found out from my web searches that this could be a problem, but all I got was some logging in the no fork scenario, saying that the snmp agent from snmpstats could not connect to the master agent, the snmpd that runs on the same machine. The "check for a proper configuration" step in the module documentation failed also, before this (I got no output in the snmpd window after openser was started, until that point, everything was like in the documentation), so this couldn't work actually, but i still did it anyway. The problem must be in the configuration files, but it escapes me. This is the content of the files (yes, both in the same location...): snmpd.conf ----------------------------------- com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view systemview included .1.3.6.1.2.1.1 view systemview included .1.3.6.1.2.1.25.1.1 access notConfigGroup "" any noauth exact systemview none none syslocation localhost (edit /etc/snmp/snmpd.conf) syscontact Root <root@localhost> (configure /etc/snmp/snmp.local.conf) pass .1.3.6.1.4.1.4413.4.1 /usr/bin/ucd5820stat rocommunity public localhost trapsink localhost:162 public trap2sink localhost:162 public authtrapenable 1 master agentx ---------------------------------------------- snmpstats.conf ------------------------------- agentXSocket tcp:localhost:705 ---------------------------------------------- And the way I start snmpd now is: snmpd -f -Lo -M /usr/local/share/mibs -m ALL -d -x tcp:localhost:705 I did try several versions including the one found in the module documentation. This is the context of my problem. Hopping that there is a small, but obvious mistake, in my config files, due to my lack of attention or something, and not a more serious thing, I thank you for your time. Gabriel Mahu --------------------------------- Check out the hottest 2008 models today at Yahoo! Autos. [View Less]

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Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 18 Sep '07

18 Sep '07

Hi Gregorio, can you send over the logs but with the 'debug' variable set to 7 ? I couldn't see the SIP messages in the logs you sent me. Elias Gregorio wrote: > > Hi Elias et all, > > > > I still get the parsing exception in the SEAS module when SIP MESSAGE > are sent from Openser to WeSIP. If I could know which header fail, > perhaps I could find a temporal solution and go on withw the > development, Elias can you tell me something about this? > > … [View More]

1 0

[OpenSER-Users] OpenXCAP problem
by mika.saari＠wipsl.com 18 Sep '07

18 Sep '07

Hi, I have "successfully" installed OpenXCAP with latest OpenSER SVN. The OpenXCAP server manages to contact the SQL server. When I am using EyeBeam 1.5 with setting "http://openxcap.mydomain.com:8000/xcap-root/" I get next error to the syslog. Sep 18 14:17:21 openxcap openxcap[9902]: [-] error: Invalid Document Selector /xcap-root/resource-lists/users/testacc/buddies-resource-list.xml (the Document Selector context must be 'users' or 'global') What kind of misconfiguration I have done ? … [View More]

2 2

[OpenSER-Users] Per branch timer and CANCEL on non final reply
by Madalina Pitis 18 Sep '07

18 Sep '07

Hello, I have the following scenario: I receive an INVITE and do parallel forking, having two branches. What I would like to do then is set a timer for the first branch, and send a CANCEL on that branch in one of the following cases: 1. no final response is received on the first branch and the timer expires. In this case it is important that the timer is per branch, so not both branches expire, and not both are canceled. 2. a 181 Call is being forwarded reply is received on the … [View More]

2 1

[OpenSER-Users] Openser with b2bua[query on media]
by Kaeman Chris 17 Sep '07

17 Sep '07

Hello, I have few questions before i proceed to get the integration of openser with the b2bua; I am confused that i need to use the mediaproxy module of the openser if i am taking up the media from the b2bua as i need the b2bua as openser is not performing topology hiding. Cheers, KChris --------------------------------- Try the revolutionary next-gen Yahoo! Mail. Click here.

1 0

[OpenSER-Users] [CDRTool] Call limit per user/domain and fromuser/authuser
by Iñaki Baz Castillo 17 Sep '07

17 Sep '07

Hi, I've started reading CDRTool documentation (very complete and long). Let me two basic questions about it: 1) Does CDRTool allow limiting simultaneous calls from/to a user/domain? I mean something as "fixed number of lines" per client. 2) I plan to use this scenario: sip-users <-> OpenSer <-> Asterisk B2BUA <-> CDRTool <-> SIP_to_PSTN provider So the sip-user will just authenticate against OpenSer, which will route call to Asterisk, and Asterisk as B2BUA will … [View More]

1 0

[OpenSER-Users] Searchable Mailing List Archives
by Norman Brandinger 17 Sep '07

17 Sep '07

Everyone: I've set up a mirror of the OpenSER mailing lists on Nabble. One of the main reasons for this was to obtain search capabilities. If you're interested in this, please take a look at: http://www.nabble.com/OpenSER---the-Open-Source-SIP-Server-f26014.html Their home page is http://www.nabble.com Regards, Norm p.s I would like to thank the folks at Nabble for importing the historical archives into their systems.

2 1

RE: [OpenSER-Users] Openser and Voice master
by Dan 17 Sep '07

17 Sep '07

Hi Ram, I had tried with many providers having Voice Master but have not been able to work with any of them. People have tried opening username and password account, only prefix based and only IP based authentication. But none have worked, VM keep rejecting calls. As far as I understand VM checks IP Address/username in contact field as well, which I am unable to change. I can change username and or IP Address in from field using uac_replace_from but could not find anything to change the … [View More]

1 0

[Serusers] MeadiaProxy + SER + X-lite in the same Lan.
by pistara_bros＠tin.it 17 Sep '07

17 Sep '07

Hi to all, I'm newbie using SER and need your help. This is my testbed: SER running on 192.168.123.186 MediaProxy running on 192.168.123.186 Client_A X-lite running under WinowsXP on 192.168.123.168 Client_B X-lite running under WinowsXP on 192.168.123.10 I want to route RTP packets between 2 computers in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly … [View More]

1 0

[Serusers] SER + RADIUS authentication example?
by Mike Trest - Personal 17 Sep '07

17 Sep '07

Hi, I have build my SER 2.0.0 with radius client to leverage existing authentication application. Can someone point me to any HOWTO use radius clent call authentication? Or, can some thoughtful person share their own ser.cfg example? Thanks ..mike.. --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000771-1, 09/01/2007 Tested on: 9/7/2007 9:44:06 AM avast! - copyright (c) 1988-2007 ALWIL Software. http://www.avast.com

2 1

[Serusers] UA With Presence Capabilities
by Abdul Qadir 17 Sep '07

17 Sep '07

Hi, I want to try SER 2.0.0 RC 1 with presence capabilities. Can any one tell me some clients with presence support which I can configure and test with SER presence module. Thanking you in advance. Best Regards, Abdul Qadir --------------------------------- Building a website is a piece of cake. Yahoo! Small Business gives you all the tools to get online.

2 1

[Serusers] Communication Problem btw SER and User Agent
by Adeyeye O. Michael 17 Sep '07

17 Sep '07

Hi, Here is an indication that my SER is running: micadeyeye@asmicom:/usr/src/Research-Progress$ ser Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 196.24.224.76 [196.24.224.76]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 196.24.224.76 [196.24.224.76]:5060 Aliases: tcp: asmicom.local:5060 tcp: localhost:5060 udp: asmicom.local:5060 udp: localhost:5060 ----------------------------------------… [View More]

2 1

[Serusers] TLS problem ser 2_0_0 - ERROR: tls_init.c:366: Unable to set the memory allocation functions
by Tomasz Zieleniewski 17 Sep '07

17 Sep '07

Hi, I did everything according to the tls module description when I start SER I get the following error: What do I have to do to make things work? Thanks in advance. Tomasz Aug 24 18:42:39 sen ser[26840]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos support: off, compression: on Aug 24 18:42:39 sen ser[26840]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8c 05 Sep 2006" (0x0090803f), kerberos support: off, zlib … [View More]

2 1

[Serusers] Loadbalancing for interco
by inge 17 Sep '07

17 Sep '07

Hi all, Is there a way to have something like a loadbalancing on SER for outgoing calls ? I want to distribute the calls between two gateways. Ideally, with a coefficient (ie. 60% for the first and 40% for the second). Thanks for your support. Adrien .L

5 10

[OpenSER-Users] Quick question about RTPproxy and Mediaproxy
by Arya 17 Sep '07

17 Sep '07

Hello for those of you that use openser as a SIP proxy/registrar do you also have RTPproxy or Mediaproxy configured? -- Thank You

2 1

[OpenSER-Users] SIP web dialer
by Schumann Sebastian 17 Sep '07

17 Sep '07

Hello all, I am currently trying to make my openser Registrar/Presence solution work with several clients. Can anyone recommend web applets which are capable of SIP (at least to register/unregister and make calls) based on Java or (not prefered but also possible) ActiveX? Furthermore, some functins to support SIMPLE/XCAP acc. RFC would be perfect. I searched a lot but no satisfying solution was found yet. OpenSource is preferred of course, but other solutions are possible. My thesis … [View More]

1 0

[OpenSER-Users] snmpstats agent failing to connect to master agent
by Gabi Mahu 17 Sep '07

17 Sep '07

Hello all, I am currently trying to get the snmpstats module up and running, for monitoring purposes. I followed the instructions in the module documentation, I studied Net-SNMP references and I searched web resources for several days now, but still I couldn't succeed in starting the openser module properly. I have no idea what the problem could be. The NET-SNMP part of the equation seams to work properly: I am using the 5.4.1 version. I managed to play around with the tools from the … [View More]package, in local and remote scenarios too. I was able to send and receive traps, get and set different values in the MIB tree, load new MIBs... On the openser part, I'm not doing something special. I have a config file that does nothing, except load the snmpstats module and set the limits for the dialogs and the unparsed data traps. That's it. I tried with fork on yes and on no, because I found out from my web searches that this could be a problem, but all I got was some logging in the no fork scenario, saying that the snmp agent from snmpstats could not connect to the master agent, the snmpd that runs on the same machine. The "check for a proper configuration" step in the module documentation failed also, before this (I got no output in the snmpd window after openser was started, until that point, everything was like in the documentation), so this couldn't work actually, but i still did it anyway. The problem must be in the configuration files, but it escapes me. This is the content of the files (yes, both in the same location...): snmpd.conf ----------------------------------- com2sec notConfigUser default public group notConfigGroup v1 notConfigUser group notConfigGroup v2c notConfigUser view systemview included .1.3.6.1.2.1.1 view systemview included .1.3.6.1.2.1.25.1.1 access notConfigGroup "" any noauth exact systemview none none syslocation localhost (edit /etc/snmp/snmpd.conf) syscontact Root <root@localhost> (configure /etc/snmp/snmp.local.conf) pass .1.3.6.1.4.1.4413.4.1 /usr/bin/ucd5820stat rocommunity public localhost trapsink localhost:162 public trap2sink localhost:162 public authtrapenable 1 master agentx ---------------------------------------------- snmpstats.conf ------------------------------- agentXSocket tcp:localhost:705 ---------------------------------------------- And the way I start snmpd now is: snmpd -f -Lo -M /usr/local/share/mibs -m ALL -d -x tcp:localhost:705 I did try several versions including the one found in the module documentation. This is the context of my problem. Hopping that there is a small, but obvious mistake, in my config files, due to my lack of attention or something, and not a more serious thing, I thank you for your time. Gabriel Mahu ____________________________________________________________________________________ Fussy? Opinionated? Impossible to please? Perfect. Join Yahoo!'s user panel and lay it on us. http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7 [View Less]

1 0

[Serusers] SER File transfer
by Allan Chao ( 趙廷睿 ) 17 Sep '07

17 Sep '07

Dear All, How should i do if i want to transfer a "file " to my friend (just like msn) in SIP protocol , does there any spec(oma, 3gpp etc.) to metion it ? allan

1 0

[OpenSER-Users] Openser as Loadbalancer with NAT support example
by Tung Tran 16 Sep '07

16 Sep '07

Hi all, I am studying how to setup the Loadbancer with OpenSER and NAT support, If anyone has done it please share your configuration, Thanks so much Regards Tung

1 0

[OpenSER-Users] rtpproxy recording
by nhadie 16 Sep '07

16 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] rtpproxy recording
by nhadie 16 Sep '07

16 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] RE: Trunk Compile Error 2
by Edson 16 Sep '07

16 Sep '07

Just an update info: # make modules=modules/auth_radius modules;echo $? make: *** [modules] Error 2 2 # And yes, I need Radius support... Edson >-----Original Message----- >From: Edson [mailto:4lists@gmail.com] >Sent: sexta-feira, 14 de setembro de 2007 23:01 >To: 'users(a)openser.org' >Subject: Trunk Compile Error 2 > >I'm trying to compile the trunk version. > >After updating I run >make distclean >and >make exclude_modules="auth_diameter h350 ldap … [View More]

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[Serusers] Per-cent Sign in Attributes
by Martin Hoffmann 15 Sep '07

15 Sep '07

Salut, sorry for getting lazy now. In the config snipped | append_to_reply("%$digest_challenge"); what is the point of the per-cent sign in front of the attribute name? Thanks, Martin, who though he had figured all this out. Apparently not.

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[OpenSER-Users] Problem with SRV lookups
by Tim Madorma 15 Sep '07

15 Sep '07

Hi, I have a simple application where I am trying to an SRV record to load balance towards a few different destination servers. I set up my DNS server to resolve to four different destination A record URIs - something like: _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 crunch.com. _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 spinner.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 homer.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 bart.com. In my configuration file, I am … [View More]putting a domain name in my call to seturi as follows: seturi("sip:feature_fs@aimcidfilter.com:6010"); and then I call route to send the INVITE. I was expecting to see a NAPTR and SRV record attempts, but when I sniffed DNS, I only saw A record attempts. Any advice would be very helpful. thanks, Tim ------------------------------------ My configuration file looks as follows: # # TWC OpenSER SIP Proxy for Caller ID - openser_cid.cfg # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 # One for the ENUM server and one for other DNS lookups user="sipproxy" group="sipproxy" disable_dns_blacklist=yes # FOR TESTING ONLY dns_try_ipv6=no # FOR TESTING ONLY # Local IP Address and port to listen to # Set the following IP addresses to the value of the local e1000g0 interface listen=udp:65.185.232.62:5060 # Alias IP address/port pair values will be in the Req URI when the # OpenSER is being load balanced with a BIG IP server # Set the IP addresses to the value of the associated blade # of the BIG IP that is load balancing for the SIP Proxy server. alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { route(1); }; if (method=="OPTIONS") { sl_send_reply("200", "OK"); exit; } if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") { xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n"); seturi("sip:feature_fs@aimcidfilter.com:6010"); setflag(1); route(1); } else { sl_send_reply("404", "Not Found"); }; } ##################################################### # Default Message Handler ##################################################### route[1] { # Send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } [View Less]

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[Serusers] SER Implementation
by Tuhin 15 Sep '07

15 Sep '07

Hi, I am from Bangladesh. I have downloaded and configured SER for proxy purpose. My server is working fine in test mode. Now I need some more features using SER. I wants to implement CDR and credit control module with this. My OS is Redhat Ent 4. What are the applications I need to to do so, pls let me know. Rgds Tuhin

1 0

[OpenSER-Users] Trunk Compile Error 2
by Edson 15 Sep '07

15 Sep '07

I'm trying to compile the trunk version. After updating I run make distclean and make exclude_modules="auth_diameter h350 ldap mi_xmlrpc osp postgres unixodbc" modules install but after I always get: make[1]: Leaving directory `/usr/local/src/VoIP/openser/openser/trunk/modules/auth_db' make: *** [modules] Error 2 What is this "Error 2" ? In this same machine I can compile, without a glance, branche 1.2 with this same 'make' command. Thanks for any help.... Edson

1 0

[OpenSER-Users] registration and redirection
by Mike Clark 14 Sep '07

14 Sep '07

Ok, I am brand new to openser and don't really have ti routing anything yet. But I did have a dumb or at least naive question. Can openser be configured as a very simple load balancer and redirect registration requests, in a round robin manner, to a set of Asterisk servers? I know having openser be the registrar is the best solution in the long run, but this may solve an issue quickly. I looked at the dispatch module and I know it will do want I want for invites, but didn't know if … [View More]

1 0

[OpenSER-Users] memory error sems with openser
by laurent schweizer 14 Sep '07

14 Sep '07

Hello, I user openser with sems , and each 2 week a memory error. the problem is that I don't have more trafic that usualy. Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmUtils.cpp)(write_to_socket)(614): write to unix socket: completed Sep 13 13:39:46 dns2 ./openser[3673]: t_uac: short of cell shmem Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmEventQueue.cpp)(processEvents)(71): event processed Sep 13 13:39:46 dns2 ./openser[3673]: unix_server_loop: Command 't_uac_dlg' failed with return value -… [View More]

4 3

[Serusers] SER+Mediaproxy + X-LITE in the same LAN.
by pistara_bros＠tin.it 14 Sep '07

14 Sep '07

Hi all, i'm newbie in SER, I want to route RTP packets between 2 computers with x-lite client voip in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another, no packet trough Mediaproxy. This is my environment: SER and Mediaproxy on the same PC (192.168.123.186) X-lite client_A running on 192.168.123.10 X-lite client_B running on … [View More]192.168.123.168 Client_A call Client_B that accept, but mediaproxy refuses to proxy the call. This is my ser.cfg # ----------- global configuration parameters ------------------------ debug=3 # debug level listen=192.168.123.186 port=5060 log_facility=LOG_LOCAL6 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser: heslo@localhost/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local1/lib/ser/modules/mysql.so" loadmodule "/usr/local1/lib/ser/modules/sl.so" loadmodule "/usr/local1/lib/ser/modules/tm.so" loadmodule "/usr/local1/lib/ser/modules/rr.so" loadmodule "/usr/local1/lib/ser/modules/maxfwd.so" loadmodule "/usr/local1/lib/ser/modules/usrloc.so" loadmodule "/usr/local1/lib/ser/modules/registrar.so" loadmodule "/usr/local1/lib/ser/modules/textops.so" loadmodule "/usr/local1/lib/ser/modules/uri.so" loadmodule "/usr/local1/lib/ser/modules/uri_db.so" loadmodule "/usr/local1/lib/ser/modules/domain.so" loadmodule "/usr/local1/lib/ser/modules/auth.so" loadmodule "/usr/local1/lib/ser/modules/auth_db.so" #MEDIA PROXY module needed loadmodule "/usr/local1/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local1/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- modparam ("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") #SQL settings modparam("auth_db", "db_url", "mysql://ser: heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://ser: heslo@localhost/ser") #MEDIAPROXY settings #--------------------------- # configurazione moduli #--------------------------- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy"," natping_interval", 30) modparam("mediaproxy","mediaproxy_socket"," /var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics"," /usr/local1/etc/ser/sip-clients") modparam("mediaproxy"," rtp_asymmetrics","/usr/local1/etc/ser/rtp-clients") modparam ("registrar", "nat_flag", 6) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol #if (method=="INVITE" && client_nat_test("3")) { # record_route_preset("192.168.123.186:5060;nat=yes"); #} else if (method!="REGISTER") { # record_route(); #}; if (method=="INVITE" ) { if (client_nat_test("3")) { record_route_preset("192.168.123.186:5060; nat=yes"); } else record_route_preset ("192.168.123.186"); } else if (method!="REGISTER") { record_route(); }; if (method==" BYE" || method=="CANCEL") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply ("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; if (!uri==myself) { route(4); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method==" REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact: [ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri! =myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (! isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search ("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } And this is Mediaproxy Log: Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands on `192. 168.123.186:25060' Remote commands are allowed from: 192.168.123.186, 127.0.0.1 Using IP address `192.168.123.186' for the RTP/RTCP proxy request B390Opu7cB 192.168.123.168:25500:audio 192.168.123.168 sip. antech.it local 192.168.123.186 remote unknown-agent info=from: orazio@sip.antech.it,to:j2me@sip.antech.it, fromtag:As94,totag: session B390Opu7cB: started. listening on 192.168.123.186:52010 execution time: 1.71 ms lookup B390Opu7cB 192.168.123.186:25500:audio 192.168.123.186 sip.antech.it local sip.antech.it unknown unknown-agent info=from:orazio@sip.antech.it,to:j2me@sip.antech.it,fr omtag:As94, totag:oePf execution time: 0.57 ms session B390Opu7cB: caller signed in from 192.168.123.168:25500 (RTP) (will return to 192.168.123.168: 25500) status execution time: 0.14 ms session B390Opu7cB: 2/0/0 packets, 344/0/0 bytes (caller/called/relayed) session B390Opu7cB: ended (did timeout). delete B390Opu7cB info= execution time: 0.14 ms and this is dump of session.py [root@sip mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------------- 192.168.123.168:25500 - 192.168.123.186:52010 - ?.?.?.?:? inactive 0'32" GSM Audio 344/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.9.0 Is mediaproxy working just between different networks or it's an error in my configuration file? Please help me. Any suggestions? Orazio [View Less]

1 0

[Serusers] how to record conversation using rtpproxy
by nhadie 14 Sep '07

14 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] domain name in route header
by jh＠tutpro.com 14 Sep '07

14 Sep '07

klaus, if you want to support domains in route headers, then you need to do naptr and srv lookups for the domain when handling loose routing. it is not enough to compare the domain to the ones domain module knows about. domain in route header may namely be anything as long as it resolves to your proxy. for this reason, i don't like the idea of domains in route headers. it hurts performance far too much. -- juha

1 0

[OpenSER-Users] help with outbound proxy
by Dawid Jarosławski 14 Sep '07

14 Sep '07

Hello, I'm trying to make outbound proxy for my network using OpenSER. Currently I've configured OpenSER as sip registrar/rtpproxy and it works fine even with NAT traversal. The problem is that I don't how shuould I reconfigure my config to make it work as outbound proxy. I've tried many changes with no results. The first problem is how to correctly handle the register method. My config is based on: http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html Can anyone help me? Best regards, Dawid Jaroslawski

2 5

[OpenSER-Users] Patch for openserunix.c
by Patrick Miccio 14 Sep '07

14 Sep '07

Hi everyone, I use 1.1.1 notls, in my testing environment I came across some strange errors when using a home made script in combination with the openserunix util. First of all there were thousands of files like"/tmp/OpenSER.XXXXXX" but looking around with google I found a patch for that: https://sourceforge.net/tracker/?func=detail&atid=743022&aid=1708301&group_… Then with the home made script I was still getting lots of socket errors in syslog, a colleague of mine fixed the … [View More]

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[OpenSER-Users] [presence] "in-dialog" SUBSCRIBE without "Route" record
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I have presence in OpenSer same box. modparam("presence","server_address","sip:85.92.0.210") I've a problem because I get "in-dialog" SUBSCRIBE messages without "Route" header: When my SIP client (Twinkle or X-Lite) starts up it sends a SUBSCRIBE NOT-in-dialog: ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:08.294012 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:ibc@sip.domain.net SIP/2.0 … [View More]Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKcsmbires Max-Forwards: 70 To: <sip:ibc@sip.domain.net> From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 # U 2007/09/10 17:54:08.296369 85.92.0.210:5060 -> 222.21.235.18:5060 SIP/2.0 202 OK Via: SIP/2.0/UDP 192.168.1.58;rport=5060;branch=z9hG4bKcsmbires;received=222.21.235.18 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Expires: 30 Contact: <sip:85.92.0.210> Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 # U 2007/09/10 17:54:08.297746 85.92.0.210:5060 -> 222.21.235.18:5060 NOTIFY sip:angel@222.21.235.18:5060 SIP/2.0 Via: SIP/2.0/UDP 85.92.0.210;branch=z9hG4bKfaad.3f308931.0 To: sip:angel@sip.domain.net;tag=wddxn From: sip:ibc@sip.domain.net;tag=10.14184.1189439648.1 CSeq: 2 NOTIFY Call-ID: bdrmoffsihrozyj(a)192.168.1.58 Content-Length: 342 User-Agent: OpenSER (1.2.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:85.92.0.210> Subscription-State: active;expires=30 Content-Type: application/pidf+xml ---------------------------------------------------------------------------------------------------------- And after 60 minutes (presence refresh in X-Lite) the following SUBSCRIBE from X-Lite is this sent. Note that: - This is a IN-DIALOG SUBSCRIBE (has totag and same callerd ID as before). - This in-dialog SUBSCRIBE hasn't "Route" header !!!!!!!!!!! ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:35.390759 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:85.92.0.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKpgezzksl Max-Forwards: 70 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 638 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- So, because this second SUBSCRIBE hasn't a "Route" header the funcion "loose_route()" doesn't match it and it's routed as NOT-in-dialog message. So, what to do??? I've a in-dialog message without "Route" header, so it's not routed by "loose_route()". I've tryied with this: if (loose_route() || has_totag()) { but it generates a loop. Does OpenSer store the SUBSCRIBE dialog info for 60 minutes? could it recognize this establishd dialog? or should I route this in-dialog message as NOT in-dialog? Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es [View Less]

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[OpenSER-Users] About presence server in other host
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I will separate the presence server in a different host, and I'd like to know your opinions: - For SUBSCRIBE messages main OpenSer will "rewriteuri(presence_serve)" and will NOT apply "record_route()" in order to avoid in-dialog NOTIFY to pass through main OpenSer. Is it ok? - "server_address" for Presence module will have: "sip:presence-server:5060" (of course presence module will just exist in presence-server). - Any other suggestion? In which cases is necessary to do this … [View More]

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[OpenSER-Users] record_route_preset("x.x.x.x:5060; nat=yes"); and search("^Route:.*; nat=yes")
by Halomoan Chow 14 Sep '07

14 Sep '07

Dear All I don't understand why in the Openser+MediaProxy i need record_route_preset("x.x.x.x:5060;nat=yes"); and search("^Route:.*;nat=yes") What is the purpose while I can do client_nat_test to see if the request is come from a nat-ed client. If I remove search("^Route:.*;nat=yes") from below statement, I can do re-INVITE without any problem because it still have client_nat_test to find out the nat in between. if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); … [View More]

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Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 14 Sep '07

14 Sep '07

On Thu, Sep 13, 2007 at 04:51:22PM +0200, Klaus Darilion wrote: > no - two: Right, my mistake. > Two identical NAPTR. One in RFC 3761 syntax and one in 2916 syntax. > Looks like enum_query() supports both syntax. Yes, it is my problem. I am sending this information for other users who will crawl this archive. :) > Solution 1: fix the NAPTR I sent request to their DNS administrators, it is not my site. > Solution 2: use enum_query("e164.arpa.","+sip"); (not sure about this … [View More]

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Re[2]: [OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello Bogdan-Andrei, Friday, September 14, 2007, 10:15:55 AM, you wrote: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: method: <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: uri: <sip:820999@voip.md> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: version: <SIP/2.0> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[… [View More]11708]: parse_headers: flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: Found param type 235, <rport> = <n/a>; state=17 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: end of header reached, state=5 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: Via found, flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: this is the first via Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: After parse_msg... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: preparing to run routing scripts... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=100 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=9 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:820999@voip.md} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:820999@voip.md] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: to body [sip:820999@voip.md ] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_body : content_length=0 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: add_param: tag=gd60cu Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=29 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:checksip@sheik:37176} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=200 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: find_first_route: No Route headers found Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: loose_route: There is no Route HF Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: rewrite_uri: Rewriting Request-URI with 'sip:820999@172.17.6.103' Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: t_newtran: msg id=106 , global msg id=105 , T on entrance=0xffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]:check_via_address(Y.Y.Y.Y, sheik, 0) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBG:sl:run_sl_callbacks: callback id 0 entered Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: ================================= Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: trace off... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:destroy_avp_list: destroying list (nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: receive_msg: cleaning up Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e44354 next=0xb5e3bac8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e3bac8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:1,tl=0xb5e3bb34 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e44344 next=0xb5e3bab8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e3bab8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: method: <BYE> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: uri: <sip:X.X.X.X:5060;r2=on;lr=on;ftag=1455B87C-17C9> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: version: <SIP/2.0> > Hi, > Have you checked the openser logs to get more details about the error? > regards, > bogdan > Ion Minzu wrote: >> Hello , >> >> I have an working openser, i installed and configured in Nagios check_sip >> plugin, the problem is that openser answer to OPTIONS message with the error >> 500. >> Help me >> >> # >> U Y.Y.Y.Y:49813 -> X.X.X.X:5060 >> OPTIONS sip:820999@voip.md SIP/2.0 >> Via: SIP/2.0/UDP sheik:49813;rport >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Contact: sip:checksip@sheik:49813 >> Content-length: 0 >> Max-Forwards: 70 >> User-agent: check_sip 1.01 >> Accept: text/plain >> >> # >> U X.X.X.X:5060 -> Y.Y.Y.Y:49813 >> SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). >> Via: SIP/2.0/UDP sheik:49813;rport=49813;received=Y.Y.Y.Y. >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md;tag=420087ba42f57437a013c6a41fc4e4c8.b0df >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Server: OpenSer (1.1.1-notls (i386/linux)). >> Content-Length: 0. >> >> >> [View Less]

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[OpenSER-Users] need help
by luhan 14 Sep '07

14 Sep '07

hi master-hand: i had rencontred a problem.when i configured my Openser work together with wesip,use two linux OS,one use to run openser,other run wesip, this is my configure file: openser: ------------------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=8 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = … [View More]0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" "openser.cfg" 154L, 3866C 14,1 Top # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { 61,1 49% }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(2); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ if(!as_relay_t("app_server_one")){ t_reply("500","Application Server error"); } 121,3-17 98% } ----------------------------- and this is wesip: ----------------------------- <Server port="8005" shutdown="SHUTDOWN" debug="0"> <Service name="Tomcat-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="localhost:5060" pathName="com.voztele" protocol="tcp" > <ExtraProperties> <Property key="javax.sip.SER_ADDRESS" value="192.168.7.15" /> <Property key="javax.sip.SER_PORT" value="5080" /> <Property key="STATISTICS" value="true" /> <Property key="javax.sip.STACK_NAME" value="app_server_one" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="localhost" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> ----------------------------------------------------------------------- at the debug model: found those tips: WARNING: using only the first listen address (no fork) 0(0) init_unixsock_socket: No unix domain socket will be opened ....... bind (Cannot assign requested address) -1(0) unable to open server sockets on dispatcher -1(0) ERROR:init_mod_child: Error while initializing module seas, rank 1 -1(0) main_dontfork: init_child failed would you help me! thank you very much. All my best regards. ------------------------------ (luhan china)卢涵技术部 E-mail: luhan(a)bjybkl.com phone: 010-59828003 mobile: 13810262476 ------------------------------ [View Less]

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[OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello , I have an working openser, i installed and configured in Nagios check_sip plugin, the problem is that openser answer to OPTIONS message with the error 500. Help me # U Y.Y.Y.Y:49813 -> X.X.X.X:5060 OPTIONS sip:820999@voip.md SIP/2.0 Via: SIP/2.0/UDP sheik:49813;rport From: sip:checksip@sheik:49813;tag=gd60cu To: sip:820999@voip.md Call-ID: dailgg@sheik CSeq: 1 OPTIONS Contact: sip:checksip@sheik:49813 Content-length: 0 Max-Forwards: 70 User-agent: check_sip 1.01 Accept: text/plain … [View More]

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[OpenSER-Users] Makefile "bug" in the latest 1.2.2 tarball...
by Geir O. Jensen 14 Sep '07

14 Sep '07

Hey, Downloaded the 1.2.2-tls tarball, extracted and did a make all - install. Everything seemed to work fine untill I tried to start it, whereupon it complained that tls was not built in. Looking at the Makefile I spotted the #TLS=1 - whence uncommented and rebuilt made everything a.o.k. I'm guessing if I did an export TLS=1 before make all it would also work? But when you give out a tarball that's supposed to include support for TLS I think I shouldn't need to do anything else than make … [View More]

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[OpenSER-Users] How to automate the new user registration process?
by lu 14 Sep '07

14 Sep '07

Hi all, Is there a simple way/opensource tool that can handle all the new VoIP users registration to the Server side automatically without human intervention/manual entry? iam using Openser 1.2 with Mysql 5.0 database. cheers.

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[OpenSER-Users] Do most SIP devices support SRV DNS lookup?
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I'd like to know about experiences using SRV DNS: - Do most SIP device support it? - Will I have unexpected surprises with some SIP phones or softphones because don't support it? - How often do the SIP devices supporting SRV have it enabled by default? Of course I will test it by myself, but I'd like to be warned about it :) Thanks for any response. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] tls errors
by Jeremy George 14 Sep '07

14 Sep '07

Hello All, I could use some help with tls. Any pointers to a possible solution would be greatly appreciated. Please let me know if any further information would be helpful. Thanks. Everything seems to register correctly. Openserctl ul show seems correct (see below.) For a short time all is well. I can call from/to Polycom hard phones. (Eyebeam never seems to make calls, though it does register correctly.) Then calls stop working, and I begin getting the following error messages. … [View More] Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Sep 12 16:17:16 spongy openser[30644]: ERROR: tcpconn_connect: tcp_blocking_connect failed Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_send: connect failed Sep 12 16:17:16 spongy openser[30644]: msg_send: ERROR: tcp_send failed Sep 12 16:17:16 spongy openser[30644]: ERROR:tm:t_forward_nonack: sending request failed Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:w_t_relay: t_forward_nonack failed I thought the errors might be caused by an underlying tcp configuration issue on the machine, so I reconfigured the proxy and uas to use tcp (but not tls) and I had no trouble. Googling and looking through the archives seems to indicate I'm not the first person to have this problem. http://www.ip-phone-forum.de/archive/index.php/t-100539.html (I don't know if there is an answer here; unfortunately, I can't read German :) Spongy% sudo openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ main.c compiled on 11:33:05 Sep 1 Spongy% sudo openserctl ul show Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=5 max_slot=1 AOR:: didama Contact:: sip:didama@75.56.10.117:37522;transport=tls Q= Expires:: 227 Callid:: acae15de-44fdabfd-e712aa40(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: jeremy Contact:: sip:jeremy@75.56.10.115:5060;transport=udp Q= Expires:: 85 Callid:: 00036b09-50520002-40d733bc-0d83cc25(a)75.56.10.115 Cseq:: 140 User-agent:: Cisco-CP7960G/8.0 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:128.91.3.19:5060 Methods:: 4294967295 AOR:: jessica Contact:: sip:jessica@75.56.10.116:55250;transport=tls Q= Expires:: 1196 Callid:: ef97ec67-46682f92-799b9cc1(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: peggy Contact:: sip:peggy@75.56.10.113:1210;transport=TLS;rinstance=dd48392409b03c0a Q= Expires:: 3567 Callid:: M2NhMmI1OGZkZThhYjUzMmZiZTcxY2M0MjhlODE1ZDQ. Cseq:: 4 User-agent:: eyeBeam release 1013t stamp 43069 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 5951 AOR:: helen Contact:: sip:helen@75.56.10.116:55250;transport=tls Q= Expires:: 1197 Callid:: f556d6ba-86bff9c9-f62d758c(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 Contact:: sip:helen@75.56.10.117:37522;transport=tls Q= Expires:: 228 Callid:: dd7e9645-83cfde68-f0f8341f(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 - Jeremy [View Less]

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[OpenSER-Users] 513 Massage too big
by Arya 14 Sep '07

14 Sep '07

Hello everyone I have a problem connecting to OpenSER from my public IP. I can connect and make calls to OpenSER inside my LAN but when I connect from outside LAN I get the 513 Massage too big massage. I than changed the openSER default config's section that checks a message size to a bigger number and I got the I get the "483 - too many Hops" error. I changed the config back to default and tested it with ngrep and I tried connecting to the server from a server in another state. The SIP … [View More]server is on 192.168.1.109 and the computer trying to connect is on 192.168.1.100 And ngrep gave me these massages: Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 67.186.60.123:29296 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. -- Thank You [View Less]

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Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 13 Sep '07

13 Sep '07

Hello, On Wed, Sep 12, 2007 at 05:32:15PM +0200, Klaus Darilion wrote: > 1. I have forgotten to tell you to use the "-W byline" switch too in > ngrep (much more readable SIP logs) I can resend you new logs if needed. > 2. Nevertheless, I have found the error: In the openser proxy your > create a second branch: Take a look at the logs - there are 2 INVITEs > sent from openser to ser - the only different in these INVITEs is the > branch parameter in the topmost Via header. … [View More]

2 2

[Serusers] FW: problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh … [View More]

1 0

[OpenSER-Users] Installing OpenSER in multiprocessor machine
by John Barry 13 Sep '07

13 Sep '07

Hello, I would like to know if there are any especial considerations when compiling, configuring, and running OpenSER v1.2.2 in a multiprocessor machine. I am using a quad-processor machine (Intel Xeon @ 3.00GHz) with Red Hat Enterprise Linux AS release 4 (2.6.9-42.0.3.ELsmp #1 SMP x86_64 x86_64 x86_64 GNU/Linux). I am also using MySQL DB for authentication in the same machine. I will appreciate any advice or information. Bests, JB … [View More]

2 1

[Serusers] problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh … [View More]

1 0

Re: [Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Jiri Kuthan 13 Sep '07

13 Sep '07

At 11:55 12/09/2007, Frank Wang wrote: >I downloaded ser 2.0.0-rc1 source code from <http://www.iptel.org/ser_2_0_0_release_candidate_1>http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. > >When I running SER, I got follow messages: > >[root@sipmcu ser]# ../../sbin/ser -f ser.cfg >Listening on > udp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 > … [View More]

2 2

Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 13 Sep '07

13 Sep '07

Hi Gregorio, posting the SIP message that caused the problem would help me reproduce it, also some OpenSER logs would help. It seems to be related to the binary-encoding of the message. try out sipSpy to capture messages if you like. remove any field which you find confidential (lest Max-Forwards...) from the SIP message send it to me privately if you want. regards, Elias Gregorio wrote: > > Hi everyone, > > > > Hi! > > Just installed WeSIP beta-0.1.4 and … [View More]openser 1.2.0 in the same machine, > configuration looks alright because some SIP Servlets receive > messages. Now I am getting an exception in the SEAS module when I > receive MESSAGE, I think this problem is similar to a previous one in > the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser > 1.2.0 again]). Does someone know anything about it? > > > > This is the exception: > > > > Thanks for helping. > > > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > [View Less]

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[OpenSER-Users] redirect server and handling 302 messages
by erik.linder＠accenture.com 12 Sep '07

12 Sep '07

Hello, I'm trying to make openser process 302 messages though I have problems(part of .cfg file below). I'll briefly explain the call flow I'd like to have: UA1 ---INVITE---> Session border controller ---INVITE--->SIP PROXY1 SIP PROXY1---INVITE--->REDIRECT SERVER---302 MOVED--->SIP PROXY1 SIP PROXY1---INVITE--->SIP PROXY2---INVITE---> Session border controller---INVITE---> UA2 This is how I want it to work. However managed to get a weird call … [View More]flow were the 302 is forwarded back to the session border controller by the proxy1 which then trigger an INVITE sent to the SIP proxy2. But that is not what I want. My 302 message looks like this 20:12:34.314705 IP 192.168.200.101.sip > 192.168.200.102.sip: SIP, length: 488 E.....@.@.%....e...f........SIP/2.0 302 Moved temporarily Via: SIP/2.0/UDP 192.168.200.102;branch=z9hG4bK1658.2543f3d6.0 Via: SIP/2.0/UDP 192.168.200.10:5060;branch=z9hG4bKpis9m130dgj04b08v7k1.1 To: "22222"<sip:22222@btsip.bt.com>;tag=cc0e227ef05fff2dd833db745f3905a6.25f 2 From: "111"<sip:11111@btsip.bt.com>;tag=SDvto0501-df614a6e Call-ID: SDvto0501-515b91532ff7927a8345f39add52762f-06ahc50 CSeq: 1 INVITE Contact: , sip:22222@mydomain.com;maddr=192.168.200.103:5060;side=term> Content-Length: 0 192.168.200.101 is redirect server 192.168.200.102 is the proxy1 192.168.200.103 is proxy2 192.168.200.10 is session border controller 1. I would like to have proxy1 process the 302 message but there seems (from other mails on the mailing list) that openser does not allow routing depending on maddr in the Contact field. Sic :-( <- how can this routing decision been taken , using rewritehostport after get_redirect("*") is kind of ignored. The only workaround I have is to change the mydomain.com to the IP of proxy2 to get the automatic redirect 2. How can I block the redirect to the session border controller and send an invite right away to the proxy2? Here are the relevant part of the logic I use in proxy1 to handle the 302 I'm using this to trigger the forward to the redirect server: lookup("aliases"); #dest is remote if (!lookup("location")) { t_on_failure("3"); xlog("L_INFO","redirect invite to SLS\n"); #change orig to term in request URI $avp(s:request_uri_to_use)=$ru; avp_subst("$avp(s:request_uri_to_use)", "/orig/term/"); avp_pushto("$ru","$avp(s:request_uri_to_use)"); #forward("udp:192.168.200.101:5060"); #exit; rewritehostport("192.168.200.101:5060"); if(!t_relay()) { sl_reply_error(); } return; and this to try to handle the 302 that is received but it does not behave like I want: failure_route[3] { xlog("L_WARN","redirect FAILUREROUTE: time_t=$Ts messageid=$mi remark=here we are $bR\n"); if(t_check_status("3[0-9][0-9]")) { get_redirects("*"); xlog("L_WARN","^method=$rm ^ruri=$ru ^messageid=$mi^remark=2status is 3xx\n--$br--\nbR((($bR)))\n"); #rewritehost("$(ru{uri.maddr})"); serialize_branches(0); } xlog("L_WARN","nextbranch\n"); #next_branches(); #t_on_failure("3"); #t_relay(); return; #t_reply("404", "No branches left"); } Help would be appreciated Cheers erik This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. [View Less]

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[OpenSER-Users] authorization problems with multiple servers
by Ben Smithurst 12 Sep '07

12 Sep '07

Hi We have a problem with clients authorizing (whether for INVITE or REGISTER) against a pair of openser servers, served via either round robin A records, or multiple equal preference SRV records. The problem in a nutshell is client sends INVITE to server 1 server 1 replies with authorization challenge client sends INVITE with auth to server 2 server 2 doesn't like the 'nonce' value Now, this only happens via our nat proxy, as far as we can tell, so I think the problem lies there (and I … [View More]

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[OpenSER-Users] WeSIP with openser 1.2.0 again
by Gregorio 12 Sep '07

12 Sep '07

Hi everyone, Hi! Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, configuration looks alright because some SIP Servlets receive messages. Now I am getting an exception in the SEAS module when I receive MESSAGE, I think this problem is similar to a previous one in the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser 1.2.0 again]). Does someone know anything about it? This is the exception: Thanks for helping. java.lang.… [View More]ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) [View Less]

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[OpenSER-Users] Wanted: VoIP Engineer for Warsaw !
by laurent schweizer 12 Sep '07

12 Sep '07

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>and AVM/FRITZ!Box <http://www.fritz-shop.ch/>. Due to our rapid growth, for our new Polish subsidiary we are currently seeking for: VOIP SPECIALIST Place of work: Warsaw *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - … [View More]

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[OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 12 Sep '07

12 Sep '07

Hello, I have problems calling from my phones connected to openser to grandstream phones on another university to ser. Problem is, that my openser generates a "CANCEL" SIP packet sent to grandstream phone, but no CANCEL received from my phone. Wireshark shows something like this: 1189587144.843962 158.195.15.100 -> 158.197.16.77 SIP/SDP Status: 200 OK, with session description 1189587144.845528 158.197.16.77 -> 158.195.15.100 SIP Request: CANCEL sip:+421xxxx44903@domain.sk 1189587144.… [View More]

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[Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Frank Wang 12 Sep '07

12 Sep '07

I downloaded ser 2.0.0-rc1 source code from http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. When I running SER, I got follow messages: [root@sipmcu ser]# ../../sbin/ser -f ser.cfg Listening on udp: 10.0.0.21 [10.0.0.21]:5060 tcp: 10.0.0.21 [10.0.0.21]:5060 Aliases: tcp: sipmcu.leadtek.com.cn:5060 udp: sipmcu.leadtek.com.cn:5060 *: mytestser.org:* WARNING:… [View More]

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Fwd: [OpenSER-Users] Transaction Deletion
by Brandon Armstead 12 Sep '07

12 Sep '07

---------- Forwarded message ---------- From: Brandon Armstead <brandon.armstead(a)gmail.com> Date: Sep 7, 2007 12:06 PM Subject: Re: [OpenSER-Users] Transaction Deletion To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Hello Bodgan, There is no noticable delay however the only thing I see is that one of the RR'ed proxy's responds sends out 200 OK rather than the original. You can see an example sipcap of this at this url: http://pastebin.ca/685897 . If you could let me … [View More]know if you see anything that looks rather, odd, towards the end you can see the last few transactions and where there is no response to 216.82.224.203's 200 OK, where it flips from 216.82.224.202 to 216.82.224.203 from the RR. Also then later the PSTN follows with a BYE, due to no response, once again thanks for your help! Regards, Brandon On 9/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > Hi Brandon, > > Is there any noticeable delay between the 200OK and ACK ? > > regards, > bogdan > > Brandon Armstead wrote: > > Hello Users, > > > > I have a weird scenario where the PSTN is responding with 200 OK, > > however Openser Never forwards on the ACK, after several > > retransmissions of 200OK to Openser, PSTN responds with BYE, anyone > > know what might be going on? I've played with the timers to no avail, > > any help is appreciated, thank you! The log below is as follows: > > > > 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 > > 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 > > 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : removing 0x2b78e4e73338 from table > > 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > delete transaction 0x2b78e4e73338 > > 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : done > > 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:parse_to:end of header reached, state=10 > > 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_field: <To> [26]; uri=[sip:openser] > > 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to > > body [<sip:openser> ] > > 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_body : content_length=0 > > 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:destroy_avp_list: destroying list (nil) > > 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:forward_reply: found module tm, passing reply to it > > 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: start=0xffffffffffffffff > > 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > add_param: tag=gK0288a919 > > 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:parse_to:end of header reached, state=29 > > 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > get_hdr_field: <To> [53]; uri=[ sip:17144828306@openser] > > 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to > > body [<sip:17144828306@openser>] > > 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: hash 53903 label 1526431020 branch 0 > > 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: no matching transaction exists > > 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: failure to match a transaction > > 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: end=(nil) > > > > It seems like its not matching the transaction, because a timer is > > removing the transaction? Any ideas? Thanks! > > -- > > Brandon Armstead > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- Brandon Armstead -- Brandon Armstead [View Less]

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Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 12 Sep '07

12 Sep '07

Arya, it means you have a routing error in the script - you need to debug your script to see where the routing logic is broken - try using xlog() messages to see the execution path of your script. I cannot help you with more than this. regards, bogdan Arya wrote: > any idea how I can fix the problem? > > On 9/11/07, *Arya* <arya6000(a)gmail.com <mailto:arya6000@gmail.com>> > wrote: > > I used ngrep but it just shows that the massage keeps bouncing > … [View More] server to client, and eventually it says massage too big > > > On 9/11/07, * Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > use ngrep or tcpdump to get a network trace from the proxy > server, on > the loopback interface. > > Regards, > Bogdan > > Arya wrote: > > Can you tell me how I can trace it? could this be an issue > with my > > router? its the Linksys Wr54G and its set on DMZ to the server > > > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro> > > <mailto: bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>> wrote: > > > > Hi Arya, > > > > I suspect you have a routing problem that cause the > request to loop on > > openser (via loopback interface) - openser keeps sending the > > request to > > itself. That's why you get a "Message to big" or "too > many hops".... > > > > run a trace on lo to see if this is what's happening. > > > > regards, > > bogdan > > > > Arya wrote: > > > Hello everyone > > > > > > I have a problem connecting to OpenSER from my public IP. > > > > > > I can connect and make calls to OpenSER inside my LAN > but when I > > > connect from outside LAN I get the 513 Massage too big > massage. > > I than > > > changed the openSER default config's section that > checks a message > > > size to a bigger number and I got the I get the "483 - > too many > > Hops" > > > error. > > > > > > I changed the config back to default and tested it with > ngrep and I > > > tried connecting to the server from a server in another > state. > > > > > > The SIP server is on 192.168.1.109 > <http://192.168.1.109> <http://192.168.1.109 > <http://192.168.1.109>> > > <http://192.168.1.109> and the > > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> > > < http://192.168.1.100> <http://192.168.1.100 < > http://192.168.1.100>> > > > > > > And ngrep gave me these massages: > [View Less]

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[OpenSER-Users] 'while' statement in config file
by Daniel-Constantin Mierla 12 Sep '07

12 Sep '07

Hello, although is more part of devel, the decision will affect users of future stable releases, so I posted on both lists. Maybe some of you noticed that in the ongoing process of improving the pseudo-variables system, the AVPs and HDRs can be accessed with dynamic index. That means next lines are correct in the config file. $var(i) = 1; xlog("$(avp(i:20)[$var(i)])"); if($(avp(i:20)[$var(i)]) == 3) { } In the past, the index was static, as a number. The question I want to address is … [View More]

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[OpenSER-Users] add/remove headers and append_branch
by Madalina Pitis 12 Sep '07

12 Sep '07

Hello, I have a problem trying to remove a header using remove_hf() after an append_branch(). I am using 1.2 version of OpenSer. The scenario with the problem is as described in the folowing lines. When I receive an INVITE I try a parallel fork (one leg towards PSTN, the other towards IP), at which time I append a "Redirection" header. The PSTN leg will be answered with a 486 response, due to the presence of the "Redirection" header. When the IP leg expires, I receive a 408 response, at which … [View More]time I do a serial fork, and resend the message towards the PSTN destination, this time without the "Redirection" header. The problem is that I cannot seem to be able to remove (or find) the header. I tried everywhere (before appending the new branch, after, in branch_route, before relaying), as you will see below. I tried searching the mailing lists but nothing turned up. I would greatly appreciate any help, for I am running out of ideas. Here is a simplified version of my configuration script, containing only the significant bits of the routing blocks: #--------------------------------------------------- # main routing logic #--------------------------------------------------- route { # check maxfwd and message lenght # do loose routing if (method=="INVITE") { # rewrite ruri rewriteuri(<PSTN destination>); # signal forking append_hf("Redirection: count=5\r\n"); append_branch(); revert_uri(); t_on_failure("1"); if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } else if (method=="ACK" || method=="BYE" || method=="CANCEL") { # default message handler } else { # forbidden message handler } } #--------------------------------------------------- # Failure route relay #--------------------------------------------------- route[1] { xlog("acc_info", " Failure route relay"); # try removing it before t_relay if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before t_relay"); remove_hf("Redirection"); } if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } #--------------------------------------------------- # Failure route handling #--------------------------------------------------- failure_route[1] { xlog("acc_info", " failure route handler"); if(t_check_status("408")) { t_on_branch("1"); rewriteuri(<PSTN destination>); # try removing it before append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before append branch"); remove_hf("Redirection"); } append_branch(); # try removing it after append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove after append branch"); remove_hf("Redirection"); } route(1); } } [View Less]

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[OpenSER-Users] append_hf() and failure_route issue
by Alex Hermann 11 Sep '07

11 Sep '07

Hello all, I'm having some trouble with failure routes and custom SIP headers. In FAILURE_ROUTE I set the ruri to a new destination. This destination needs a custom SIP header. When I add a SIP header with append_hf() (from within the failure_route), this header is correctly sent to the first IP of the destination on t_relay(). However, when this destination fails with 503 or timeout, the built-in DNS based failover doesn't add the custom SIP header to the new requests for subsequent … [View More]

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Re: [OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by Iñaki Baz Castillo 11 Sep '07

11 Sep '07

First of all sorry for this post not being in the thread it belongs to (I've not the initial mail). > But, I don't want to use the Asterisk dtmf features, > I want to use the SIP INVITE/REFER features from the > UAC (actually polycom and snom phones). > > Can someboy shed me some light on this? You receive a call from Asterisk UAC and want to transfer (blind or attender) to other UACregistered in OpenSer. Of course you don't need Asterisk native transfer, SIP allows blind … [View More]

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Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 11 Sep '07

11 Sep '07

Hi, use ngrep or tcpdump to get a network trace from the proxy server, on the loopback interface. Regards, Bogdan Arya wrote: > Can you tell me how I can trace it? could this be an issue with my > router? its the Linksys Wr54G and its set on DMZ to the server > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Arya, > > I suspect you have a routing problem that cause the request to … [View More]

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[OpenSER-Users] Openser and Voice master
by Dan 11 Sep '07

11 Sep '07

Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan

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[OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by perl＠libkvm.org 11 Sep '07

11 Sep '07

Hello, I am configuring a platform unig OpenSER 1.2 with Asterisk 1.2. I have all the SIP phones registered on OpenSER. I use the realtime setup on Asterisk using a view on MySQL. I need to be able to process the attended transfer between two UAC registered on OpenSER, having the first received a call from the Asterisk side, so that the second UAC gets the call transferred. I also need to do this as a blind transfer. But, I don't want to use the Asterisk dtmf features, I want to use the SIP … [View More]

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[Serusers] ERROR: warning_builder: buffer size exceeded
by Kostas Marneris 10 Sep '07

10 Sep '07

Hello all, I have occasionally the following warning mesgs at my SER log : Sep 10 14:34:27 serhost ser[17957]: ERROR: warning_builder: buffer size exceeded Sep 10 14:34:27 serhost ser[17957]: WARNING: warning skipped -- too big I noticed that the R-URI in that case is : R-URI=<sip:XXXXXXXXXX@mydomain.com;user=phone;transport=UDP> when usually the R-URI is : <sip:XXXXXXXXXX@mydomain.com;transport=UDP> <sip:XXXXXXXXXXXXX@mydomain.com> Anyone seen those mesgs before ? Can … [View More]

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[OpenSER-Users] Problem with rtpproxy
by Jeferson Prevedello 10 Sep '07

10 Sep '07

Hi Guys, When I originate a call for an user behind of a NAT the call is without audio in the two ways. I believe that the problem is related to rtpproxy, however did not find nothing of wrong in my configuration. Some tip? openser.cfg: # ------------- !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "unix:/… [View More]

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[Serusers] Need SIP proxy to translate UDP<->TCP
by Thermal Wetland 10 Sep '07

10 Sep '07

Hello, I would like to hire someone to setup a SIP UDP<->TCP proxy using SER. SER will sit between an Asterisk box and Microsoft Exchange 2007. Asterisk speaks UDP & the Exchange box speaks TCP. I will make a fresh fully patched CentOS 5 box available for the work. Payment could be through PayPal or Rent a Coder or similar. I would expect an SER expert to be able to do this in less than 2 hours. Please let me know if you are interested - thermalwetland at gmail dot com. Thanks, Thermal

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Re: [OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Hi Bogdan I got it fixed. The problem is on this line: if(!t_relay("0x05")) if I change to if(!t_relay()) everything is ok. Don't know what is 0x05 mean. Regards, Halomoan On 9/10/07, Halomoan Chow <halomoan(a)gmail.com> wrote: > > Hi > > My Openser version is openser-1.2.0-notls and the Mediaproxy is > mediaproxy-1.9.0 > I was using the openser.cfg that was generated from > http://www.sipwise.com/wizard/ for Openser + Mediaproxy only. > > Just only … [View More]

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Re: [Serusers] [Serdev] Release of SER 2.0 (and future 2.1)
by Greger V. Teigre 10 Sep '07

10 Sep '07

Ok, I conclude then that final release of SER 2.0 is aimed at October 15. That should allow us to align SER, serctl, SERWeb, rtpproxy, and SEMS. SER, SERWeb, and SEMS will have independent releases, and they will be aligned to be interoperable. SER, rtpproxy, and serctl will have a common release. The release will include the new buildsystem that was tested in release candidate 1. The config file and features available work with sercmd, serctl, as well as with SERWeb. SEMS support is … [View More]

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[OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Dear All I'm running into segmentation fault error on Openser 1.2.x with Mediaproxy. What is the first thing I should take a look into to find out which line or which module causing this segmentation fault? Thank you. Regards, Halomoan

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sr-users September 2007