sr-users June 2006

sr-users@lists.kamailio.org

203 participants
495 discussions

by Kenny Chua

Thanks Andreas, that was really the problem. Really got to watch out for codecs. --------------------------------- Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail Beta.

18 years, 4 months

Re: [Users] keep alive using RE-INVITE

by Javier Ramirez

Bogdan, I don't understand how to make this with openser. but I understand the keepalive mechanism can you help me? best regards Javier Ramirez ----- Original Message ----- From: Bogdan-Andrei Iancu To: Javier Ramirez Sent: Wednesday, June 21, 2006 1:35 PM Subject: Re: [Users] keep alive using RE-INVITE Javier, I guess the RFC3261 or google (SIP + reINVITE) are the best source for looking into. regards, bogdan Javier Ramirez wrote: > Bogdan, > I read that a lot of person use the RE-INVITE to keepalive the call > but I don´t understand this. > please, can you help me ? > best regards > Javier Ramirez > > > ----- Original Message ----- > *From:* Bogdan-Andrei Iancu <mailto:bogdan@voice-system.ro> > *To:* Javier Ramirez <mailto:jvr_78@yahoo.com.ar> > *Cc:* users(a)openser.org <mailto:users@openser.org> > *Sent:* Thursday, June 08, 2006 7:14 AM > *Subject:* Re: [Users] keep alive using RE-INVITE > > Hi Javier, > > this is quite impossible to do - by sending re-INVITEs from a middle > point, you will break the CSEQ consistency between the end points. > Within the dialog, the cseq must increase from message to message > - if > you send a re-invite to point A with cseq n+1, after a while point > B may > sent a request with the same cseq since it was not aware that some > party > in the middle changed the number. > > the best way to do it is if you have a end2end keepalive mechanism as > Session-Timer. > > regards, > bogdan > > Javier Ramirez wrote: > > > Hi ! > > I looking for a mechanism to keep alive, and I see that is used the > > RE-INVITE, but > > How to use the Re-INVITE to a keep alive mechanism ? > > exist another way to implement another keep alive mechanism ? > > > > best regards > > Javier ramirez > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > __________ NOD32 1.1586 (20060608) Information __________ > > This message was checked by NOD32 antivirus system. > http://www.eset.com > __________ NOD32 1.1611 (20060620) Information __________ This message was checked by NOD32 antivirus system. http://www.eset.com

18 years, 4 months

[Users] FW: ACK problems

by Glenn Dalgliesh

It seems that for some reason openser(555.111.555.111:5060) will not forward the ACK packet in this scenario. Config at bottom. ACKs from other clients seem to get forwarded just fine. Have tried both 1.0.1 and CVS-head same results.. Thanks 111.222.111.222:5060 Openser (Statefull) sending us a call 555.111.555.111:5060 Openser (Statefull) 77.159.77.105:9193 Eyebeam Client ACK is in SIP Message 9 which openser should then send to 77.159.77.105:9193 SIP MESSAGE 1 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 1 21/Jun/06 16:13:12.8910 TimeFromPreviousSipFrame=0.0000 TimeFromStart=0.0000 INVITE sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:9999999019@111.222.111.222:5060;nat=yes;ftag=783386-fdb1078445624;lr=on > Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: Sansay VSX 2.1 Contact: <sip:9999992264@666.111.82.202:5060;transport=udp> Max-Forwards: 69 Content-Type: application/sdp Content-Length: 283 v=0 o=sansay-VSX 10 10 IN IP4 206.57.80.14 s=session controller c=IN IP4 111.222.111.222 t=0 0 m=audio 63226 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=ptime:20 ________________________________________ SIP MESSAGE 2 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 2 21/Jun/06 16:13:12.8933 TimeFromPreviousSipFrame=0.0023 TimeFromStart=0.0023 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: OpenSer (1.0.1 (i386/linux)) Content-Length: 0 Warning: 392 555.111.555.111:5060 "Noisy feedback tells: pid=16606 req_src_ip=111.222.111.222 req_src_port=5060 in_uri=sip:9999999019@555.111.555.111:5060 out_uri=sip:9999999019@77.159.77.105:9193 via_cnt==2" ________________________________________ SIP MESSAGE 3 555.111.555.111:5060() -> 77.159.77.105:9193() UDP Frame 3 21/Jun/06 16:13:12.8934 TimeFromPreviousSipFrame=0.0001 TimeFromStart=0.0024 INVITE sip:9999999019@77.159.77.105:9193 SIP/2.0 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;nat=yes;ftag=783386-fdb1078445624;lr=on > Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060> Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Server: Sansay VSX 2.1 Contact: <sip:9999992264@666.111.82.202:5060;transport=udp> Max-Forwards: 68 Content-Type: application/sdp Content-Length: 283 P-hint: usrloc applied v=0 o=sansay-VSX 10 10 IN IP4 206.57.80.14 s=session controller c=IN IP4 111.222.111.222 t=0 0 m=audio 63226 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=ptime:20 ________________________________________ SIP MESSAGE 4 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 4 21/Jun/06 16:13:12.9413 TimeFromPreviousSipFrame=0.0479 TimeFromStart=0.0503 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 ________________________________________ SIP MESSAGE 5 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 5 21/Jun/06 16:13:12.9426 TimeFromPreviousSipFrame=0.0013 TimeFromStart=0.0516 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 0 ________________________________________ SIP MESSAGE 6 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 7 21/Jun/06 16:13:13.9012 TimeFromPreviousSipFrame=0.9586 TimeFromStart=1.0102 Extra Information: Packet is not a complete SIP message ________________________________________ SIP MESSAGE 7 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 8 21/Jun/06 16:13:15.1016 TimeFromPreviousSipFrame=1.2003 TimeFromStart=2.2106 SIP/2.0 200 OK Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 8 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 9 21/Jun/06 16:13:15.1037 TimeFromPreviousSipFrame=0.0021 TimeFromStart=2.2127 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 9 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 10 21/Jun/06 16:13:15.2282 TimeFromPreviousSipFrame=0.1245 TimeFromStart=2.3372 ACK sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:111.222.111.222;ftag=783386-fdb1078445624;lr=on> Via: SIP/2.0/UDP 111.222.111.222;branch=0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK78338620-bdb1078445624 Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 ACK Server: Sansay VSX 2.1 Max-Forwards: 69 Content-Length: 0 ________________________________________ SIP MESSAGE 10 77.159.77.105:9193() -> 555.111.555.111:5060() UDP Frame 11 21/Jun/06 16:13:15.5921 TimeFromPreviousSipFrame=0.3639 TimeFromStart=2.7011 SIP/2.0 200 OK Via: SIP/2.0/UDP 555.111.555.111;branch=z9hG4bKdbc7.30da3132.0 Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 11 555.111.555.111:5060() -> 111.222.111.222:5060() UDP Frame 12 21/Jun/06 16:13:15.5942 TimeFromPreviousSipFrame=0.0020 TimeFromStart=2.7032 SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.111.222;branch=z9hG4bKdbc7.ff2338c7.0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK7833861-bdb1078445624 Record-Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> Record-Route: <sip:9999999019@111.222.111.222:5060;lr;nat=yes;ftag=783386-fdb1078445624> Contact: <sip:9999999019@77.159.77.105:9193> To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: eventlist User-Agent: eyeBeam release 3010n stamp 19039 Content-Length: 235 v=0 o=- 519699122 519699160 IN IP4 77.159.77.105 s=eyeBeam c=IN IP4 77.159.77.105 t=0 0 m=audio 6902 RTP/AVP 0 101 a=alt:1 1 : BA47AC39 000000CF 77.159.77.105 6902 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv ________________________________________ SIP MESSAGE 12 111.222.111.222:5060() -> 555.111.555.111:5060() UDP Frame 13 21/Jun/06 16:13:15.6761 TimeFromPreviousSipFrame=0.0819 TimeFromStart=2.7850 ACK sip:9999999019@555.111.555.111:5060 SIP/2.0 Record-Route: <sip:111.222.111.222;ftag=783386-fdb1078445624;lr=on> Via: SIP/2.0/UDP 111.222.111.222;branch=0 Via: SIP/2.0/UDP 666.111.82.202:5060;rport=5060;branch=z9hG4bK78338620-bdb1078445624 Route: <sip:555.111.555.111;ftag=783386-fdb1078445624;lr=on> From: <sip:9999992264@666.111.82.202>;tag=783386-fdb1078445624 To: <sip:9999999019@111.222.111.222:5060>;tag=6963193a Call-ID: 100-1-1150920078(a)666.111.82.202 CSeq: 1 ACK Server: Sansay VSX 2.1 Max-Forwards: 69 Content-Length: 0 -*-*-*-*-*-*-*-*-*-*-*- 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) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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=="REGISTER") { # Uncomment this if you want to use digest authentication #if (!www_authorize("openser.org", "subscriber")) { # www_challenge("openser.org", "0"); # exit; #}; save("location"); exit; }; 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(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

18 years, 4 months

FW: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723

by Richard C. Thompson

As far as I know mediaproxy does just that, it proxies the call but doesn't change the codec so if it comes in g729 it goes out the same way. It doesn't do transcoding. Most of my calls are g729 and you can see these calls by doing a sessions.py in the mediaproxy server Richard C. Thompson VP Technology, CCNA Virtual Communications, LLC -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov Sent: Wednesday, June 21, 2006 4:31 AM To: serusers(a)iptel.org Subject: Re: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 How about making clients change the default codec to smth else? On 6/21/06, Benjamin.George(a)t-systems.com <Benjamin.George(a)t-systems.com> wrote: > > > > Hi, > > In my understanding, you can't configure media proxy for g729 as it won't > support g729 codec. It is a patented codec and SER media proxy as well as > SEMS supports only the patent free codecs like g711a/u or iLBC, etc. As a > result of this you will get some choppy sound only if you use g729. So I > feel, you have to modify the code in order to incorporate g729 with media > proxy. > > Regards, > > Benjamin. > > > ________________________________ > > > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of ravi > reddy > Sent: Wednesday, June 21, 2006 1:02 PM > To: serusers(a)lists.iptel.org > Subject: [Serusers] How to change Mediaproxy codec g-711a to g-729 or g-723 > > > > > Hi , > I am using ser-0.9.6 and mediaproxy-1.4.2 and the are working > fine til now but when i use the sip-sip phone (or) sip - pstn in the peak > hours i cant hear voice very clearly its choppy sound , > > when i tried to call grandstream sip phone to pstn i made > grandstream phone to use only g-729 but mediaproxy is showing session in > active > > can any body tell me how to change mediaproxy's g-711a to some thing like > g-729 or g-723 so that i can hear clear voice in peak hours too > > > thanks you very much > > regards > ravi. > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

18 years, 4 months

FW: [Serusers] calls status at SER

by Richard C. Thompson

I use a mediaproxy server with ser so if you do a sessions.py it will show you all the current calls Richard C. Thompson VP Technology, CCNA Virtual Communications, LLC _____ From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Atif Rasheed Sent: Wednesday, June 21, 2006 8:26 AM To: serusers(a)iptel.org Subject: [Serusers] calls status at SER Hi all, We are doing stateful routing from SER. I need to know how can we check simultanous calls going through SER, total no. of calls processed etc. Like we check current channels in asterisk or we see statistics in gnugk. appriciate the help. Thank you -- Atif Rasheed

18 years, 4 months

[Users] mediaproxy.ini (more than 1 IP)

by Andrew And

When I am using mediaproxy I need to write the server address IP in ProxyIP of the file mediaproxy.ini. But when I have 2 or more alias in my interface of the server how can I put more than 1 IP?

18 years, 4 months

[Serusers] location information

by Alois Schafferhofer

Hi, I have a problem with sending position coordinates with PUBLISH. If I send the x and y coordinates <x>01524.3441</x><y>04704.1020</y> via PUBLISH, SER cuts the last two digits (<x>1524.34</x><y>4704.1</y>). Maybe, someone could tell me in which class or in which method this happens and what I have to do. Best regards, Alois /usr/sbin/ser[11700]: handle publish request /usr/sbin/ser[11700]: get_pres_uri: _puri=sip:pda_user@fh-joanneum.at /usr/sbin/ser[11700]: handle_publish -4- p_uri= p_uri.len=23 /usr/sbin/ser[11700]: handle_publish -5- presentity=0x4069d1d8 /usr/sbin/ser[11700]: parse_pidf: sipuri=0x8216278:pres:pda_user@fh-joanneum.at contact=0x8216098:sip:pda_user@10.0.0.3 basic=0x82160b8:online location=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: site=(nil):(null) floor=(nil):(null) room=(nil):(null) /usr/sbin/ser[11700]: parse_pidf: x=0x821be28:01524.3441 y=0x821be38:04704.1020 radius=(nil):(null) /usr/sbin/ser[11700]: find_presence_tuple: _p=0x4069d1d8 _p->tuples=0x4069d488 /usr/sbin/ser[11700]: publish_presentity_pidf: -1- /usr/sbin/ser[11700]: publish_presentity: -2- /usr/sbin/ser[11700]: publish_presentity: -3-: changed=1 /usr/sbin/ser[11700]: db_update_presentity starting: use_place_table=1 presid=4 /usr/sbin/ser[11700]: db_update_presentity: tuple- >contact=sip:pda_user@10.0.0.3 len=21 basic=1 expires=1150916054 priority=0.500000 /usr/sbin/ser[11700]: db_update_presentity: cleaning contact from table /usr/sbin/ser[11700]: db_update_presentity: room= loc= /usr/sbin/ser[11700]: publish_presentity: -4- /usr/sbin/ser[11700]: publish_presentity: event_package=1 -1- /usr/sbin/ser[11700]: handle_publish -8- paerrno=0 /usr/sbin/ser[11700]: handled publish /usr/sbin/ser[11708]: timer_presentity: _p=0x4069d1d8 pda_user(a)fh-joanneum.at flags=1 watchers=0x4069d3a8 /usr/sbin/ser[11708]: notifying esca(a)fh-joanneum.at _p->flags=1 _w- >event_package=1 _w->accept=2 _w->status=1 PUBLISH sip:pda_user@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK874819433..Max-Forwards: 70..From: <sip:pda_user@fhjoanneum .at>;tag=540877786..To: <sip:pda_user@fh-joanneum.at>..Call-ID: 996402808-569651771- 701518565..CSeq: 10 PUBLISH..Contact: <sip:pda_user@10.0.0.3>..Con tent-Type: application/cpim-pidf+xml..Content-Length: 396..Expires: 50..Event: presence..User- Agent: SIP .NET 1.0 evaluation version, www.independents oft.com....<?xml version="1.0" encoding="utf-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:pda_user@fh-joanneum.at"><tuple id="9da019 ef-d0ea-4de0-8ad3-edacae0ca89c"><status><basic>online</basic><geopriv><locationinfo>< civilAddress><x>01524.3441</x><y>04704.1020</y></civilAddress></ location-info></geopriv></status><contact priority="0.8">sip:pda_user@10.0.0.3</contact></tuple></presence>.. 10.15.200.57:5060 -> 10.15.200.136:5060 NOTIFY sip:esca@10.15.200.136 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK766a.c7db8933.0..To: <sip:esca@fhjoanneum. at>;tag=z9hG4bK29818902. .From: <sip:pda_user@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-f2af..CSeq: 10 NOTIFY..Call-ID: 603541788327@10.15.200.136..Content-Length: 464..User-Agent: Sip EXpress router(0.9.3 (i386/linux))..Event: presence..Content-Type: application/pidf+xml..Subscription-State: active;expires=3479. ...<?xml version="1.0"?>..<!DOCTYPE presence PUBLIC "//IETF//DTD RFCxxxx PIDF 1.0//EN" "pidf.dtd">..<presence entity="sip:pda_user@fh-joanneum.at">..< tuple id="9r28r49">tid6e65cf14">.. <contact priority="0.500000">sip:pda_user@10.0.0.3</contact>.. <status>.. <basic>open</basic>.. <geopriv>< location-info><civilAddress> <x>1524.34</x>.. <y>4704.1</y>.. </civilAddress></location-info></geopriv>.. </status>.. </status>..</tupl e>..</presence>..

18 years, 4 months

[Users] 499 Not Acceptable here

by Kenny Chua

Hi all, I'm somehow getting this message when I try to dial from SIP 2 SIP. It says 499 Not acceptable here. Any help will be appreciated. Also, I was wondering how do I setup rtpproxy on another machine. How am I suppose to link it from my openser machine to the other server. RECEIVE TIME: 18379350 RECEIVE << 70 .xxx.xxx.xxx:5060 SIP/2.0 488 Not Acceptable Here Call-ID: AD7821E1-85B3-47DB-9EC8-ACF50039FC3E(a)42.77.149.86 From: "Test "<sip:200@test.org>;tag=1917233790 To: <sip:203@test.org>;tag=2d5d4 CSeq: 50639 INVITE Via: SIP/2.0/UDP 130.70.11.16:5060;rport;branch=z9hG4bK9847221F640D42E0B8F2F08835F57F28 Record-Route: <sip:192.168.1.5> Content-Length: 0 Openser.cfg debug=3 fork=yes log_stderror=no listen=192.168.1.5 alias=test.org port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); return; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); return; }; if (method=="INVITE") { if (!proxy_authorize("test.org","subscriber")) { proxy_challenge("test.org","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); return; }; if (method=="ACK") { route(1); return; } else if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); return; }; if (uri=~"^sip:00[0-9]*@") { # International PSTN xlog("International PSTN\n"); strip(2); route(4); route(5); return; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); } } route[5] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- xlog("Routing to PSTN gateway\n"); rewritehostport("pstn.net:5060"); avp_write("i:45", "inv_timeout"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Yahoo! Sports Fantasy Football 06 - Go with the leader. Start your league today!

18 years, 4 months

RE: [Users] Out of Memory

by Douglas Garstang

Oh, well you know what, I don't have access to system with a publically available ftp or http server. I'm trying to use pastbin.com, but they seem to be having a problem right now. > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, June 21, 2006 10:14 AM > To: Douglas Garstang > Cc: Users(a)openser.org > Subject: Re: [Users] Out of Memory > > > As written in the wiki, if you do not know how to interpret > the result, > you should post them on a ftp or http server and send the link to the > list - also, important is to mention in which context did you get the > mem dump - is the shutdown dump, a run time dump... > > regards, > bogdan > > > Douglas Garstang wrote: > > >Once I get the memory status, how do I interpret the results? > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Wednesday, June 21, 2006 9:43 AM > >>To: Douglas Garstang > >>Cc: Users(a)openser.org > >>Subject: Re: [Users] Out of Memory > >> > >> > >>Hi Douglas, > >> > >>it was signal SIGUSR1 and not SIGHUP - I strongly advice you > >>to go through > >> http://openser.org/dokuwiki/doku.php?id=memory > >>note you have to recompile and reinstall in order to be able to do > >>memory debug. > >> > >>also does not matter how much mem. your system has - openser > >>will use by > >>default only 1M per process. Refer to the wiki for how to > increase it. > >> > >>Regards, > >>Bogdan > >> > >> > >> > >

18 years, 4 months

[Users] user 'admin' and mysql

by Mark Kent

Hello, I just noticed that openser_mysql.sh creates the username "admin" with the default openserrw password in the subscriber table. This seems to introduce a security hole where a well-known username and password pair would exist on most virgin openser installations. Is there a good reason to have that entry in the "subscriber" table? Is it used anywhere? Now I know that we're supposed to change the mysql access passwords, but I have to admit that I didn't think to change a password actually emebedded IN the data of the mysql database. Did I miss a critical security note somewhere alerting me to this default user? Thanks, -mark

18 years, 4 months

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sr-users June 2006