sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

by Marco F. Delaurenti

Hi all, I've a working openser and a cisco router (17xx). I'd like to authenticate the cisco router as a sip ua. If I disable www_authorize everything is working fine and the router register to the server. If I enable it and configure the router with sip-ua authentication username xxx password xxx realm xxx I see the router trying to register, but no way to make it working. I tried I think every combination of realm in www_authorize, www_challenge and so on. If I debug a successful authentication, I see a query on the db, "select ... from subscriber". If I debug the cisco auth, the server doesn't even try the query. Has somebody any hint please? TIA, -- "Work and play are words used to describe the same thing under differing conditions." Mark Twain

17 years, 9 months

[Serusers] How to Register SER server

by jk

Hello, May be its a very simple question, I have a SIP account from one VoIP service provider. username:password@domain-name-of-voip-service-provider. How can I register my ser server to receive calls. -JK

17 years, 9 months

Re: [Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()

by Benko

On Tue, 17 Oct 2006 10:36:20 -0400 Nathan Hawkins <utsl(a)quic.net> wrote: > Comments inline. > My gut feeling > is you've found a firmware bug, or it's not interacting right with > the other phone. I'd try calling between two Polycom's and see if > that works as expected. > > Good luck! no, it doesn't make a difference, i just tried it with a polycom 601 and 501 involved. While i guess your assumption that this is incorrect behaviour of the polycom is true i'm not sure if this is the main reason for the one-way audio issue i initially posted. The sdp-part in the invite message on resuming the call on hold looks like it would have to correct information to go through the nat, the "c=a.b.c.d"-line is updated with the proxy ip and all other nat-parameters are appended too. Could this be a problem of rtpproxy or asterisk? The rtp-ports change a lot, seems some device sends the audio to a port that has not been recognized by the firewall. Hope someone has some additional input, i'll try to contact polycom but with their strict reseller policy i guess i'll not get an answer... best regards christian > > To clarify what i'm talking about i've attached the whole sip-debug > > of two calls, the first call is sent on hold when a second call > > comes and resumed after the second call has ended(1111111801 calls > > 2222222804 and it answers, then 0891234567 calls 2222222804 and > > 1111111801 is set on hold and resumed later). Openser runs on .98, > > which also serves the rtpproxy. .97 is a asterisk-gateway. > > > > Thanks for the input > > Christian > > > > > > > > > > > >> What firmware version are you running? > >> > >> Nathan > >> > >> Benko wrote: > >> > >>> Hmm, actually this parameter is set to 0(default). > >>> However, in practice it still uses the old standard(rfc2543). > >>> There's no media direction parameter in the sdp-message sent by > >>> the polycom for some reason, although the manual states to do so > >>> (firmware 2.0.1). Is it possible to force this somehow? > >>> > >>> regards > >>> christian > >>> > >>> On Sun, 15 Oct 2006 19:58:33 -0400 > >>> Nathan Hawkins <utsl(a)quic.net> wrote: > >>> > >>> > >>>> There's an option for the Polycom phones to switch the hold > >>>> behaviour. > >>>> > >>>> Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264 > >>>> rules for signalling hold instead of 0.0.0.0. > >>>> > >>>> Benko wrote: > >>>> > >>>>> Hello! > >>>>> > >>>>> I'm having a issue with NAT and rtpproxy. Usually my setup works > >>>>> fine with natted clients, the Connection Information is > >>>>> overwritten with the IP of the rtpproxy and audio passes through > >>>>> in both directions. However, today i came across a problem where > >>>>> the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the > >>>>> private ip after resuming a call that was on hold(actually, the > >>>>> other party is invited again) - and the force_rtp_proxy > >>>>> ()-command on openser left the ip untouched instead of > >>>>> overwriting it with the rtpproxy-ip. As a result the person that > >>>>> was on hold had audio but the polycom user (with the "wrong" ip) > >>>>> hadn't. > >>>>> > >>>>> The false ip left aside, is it expected behaviour of > >>>>> force_rtp_proxy to not touch 0.0.0.0? > >>>>> > >>>>> Just out of curiosity - does someone know the "on hold"-problem > >>>>> with polycoms? > >>>>> > >>>>> thx > >>>>> christian > >>>>> > >>>>> _______________________________________________ > >>>>> Users mailing list > >>>>> Users(a)openser.org > >>>>> http://openser.org/cgi-bin/mailman/listinfo/users > >>>>> > >> ------------------------------------------------------------------------ > >> > >> interface: bond0 (2XX.XXX.XXX.96/255.255.255.240) > >> filter: ip and ( port 5060 and host 81.XXX.XXX.196 ) > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> P-Key-Flags: keys="3". > >> User-Agent: snom190/3.60w. > >> Accept: application/sdp. > >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > >> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. > >> Supported: timer, 100rel, replaces. > >> Session-Expires: 3600. > >> Content-Type: application/sdp. > >> Content-Length: 268. > >> . > >> v=0. > >> o=root 1814496751 1814496751 IN IP4 10.1.99.169. > >> s=call. > >> c=IN IP4 10.1.99.169. > >> t=0 0. > >> m=audio 55428 RTP/AVP 8 0 18 101. > >> a=rtpmap:8 pcma/8000. > >> a=rtpmap:0 pcmu/8000. > >> a=rtpmap:18 g729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-15. > >> a=ptime:20. > >> a=sendrecv. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport=2051;received=81.XXX.XXX.196. > >> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 INVITE. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=2051 > >> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> out_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-mdyquvng2u65;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=2f7d8fa65b1e4a32386804888f034ca8.5f9f. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 1 ACK. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0892222222804@sip.de.mydomain.org;user=phone SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> P-Key-Flags: keys="3". > >> User-Agent: snom190/3.60w. > >> Accept: application/sdp. > >> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > >> SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer. > >> Supported: timer, 100rel, replaces. > >> Session-Expires: 3600. > >> Proxy-Authorization: Digest > >> username="11111111801",realm="sip.de.mydomain.org",nonce="4534dc22e41f8b3a51321f458eed7830772d2ea7",uri="sip:0892222222804@sip.de.mydomain.org;user=phone",response="139c40e2fecbadf4412628412607a7d7",algorithm=md5. > >> Content-Type: application/sdp. Content-Length: 268. > >> . > >> v=0. > >> o=root 1814496751 1814496751 IN IP4 10.1.99.169. > >> s=call. > >> c=IN IP4 10.1.99.169. > >> t=0 0. > >> m=audio 55428 RTP/AVP 8 0 18 101. > >> a=rtpmap:8 pcma/8000. > >> a=rtpmap:0 pcmu/8000. > >> a=rtpmap:18 g729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-15. > >> a=ptime:20. > >> a=sendrecv. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-x9wr16oqr17g;rport=2051;received=81.XXX.XXX.196. > >> From: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> To: <sip:0892222222804@sip.de.mydomain.org;user=phone>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> Server: OpenSer (1.1.0-tls (i386/linux)). > >> Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20909 > >> req_src_ip=81.XXX.XXX.196 req_src_port=2051 > >> in_uri=sip:0892222222804@sip.de.mydomain.org;user=phone > >> out_uri=sip:0892222222804@mgw.de.mydomain.org;user=phone > >> via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;received=81.XXX.XXX.196;branch=z9hG4bK-x9wr16oqr17g;rport=2051. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From: > >> "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 INVITE. > >> User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0892222222804@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 303. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36578 RTP/AVP 0 18 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>. Contact: > >> <sip:0891111111@2XX.XXX.XXX.97:5060>. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102 > >> INVITE. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Date: Tue, 17 Oct 2006 13:30:30 GMT. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Content-Type: application/sdp. > >> Content-Length: 396. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 18 4 3 97 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:4 G723/8000. > >> a=rtpmap:3 GSM/8000. > >> a=rtpmap:97 iLBC/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=direction:active. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0892222222804@2XX.XXX.XXX.97 SIP/2.0. > >> Via: SIP/2.0/UDP > >> 10.1.99.169:2051;branch=z9hG4bK-kqgbv28tcm8k;rport. Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=lxlk4gyz7h>. From: "801" > >> <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. To: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 2 ACK. > >> Max-Forwards: 70. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 100 Trying. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq: > >> 102 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0062c4dd;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. CSeq: > >> 102 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. Allow: INVITE, ACK, > >> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, > >> UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type: > >> application/sdp. Content-Length: 185. . > >> v=0. > >> o=- 1161091665 1161091665 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK72b6.93580965.2. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK77d83684;rport=5060. From: > >> "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. To: > >> <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. CSeq: 102 > >> ACK. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> INVITE sip:2222222804@10.1.99.161:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>. Contact: > >> <sip:0891234567@2XX.XXX.XXX.97:5060>. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102 > >> INVITE. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Date: Tue, 17 Oct 2006 13:30:38 GMT. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Content-Type: application/sdp. > >> Content-Length: 396. > >> . > >> v=0. > >> o=root 6068 6068 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36584 RTP/AVP 0 18 4 3 97 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:4 G723/8000. > >> a=rtpmap:3 GSM/8000. > >> a=rtpmap:97 iLBC/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=direction:active. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 100 Trying. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 180 Ringing. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Allow-Events: > >> talk,hold,conference. Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 1 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Max-Forwards: 70. > >> Content-Type: application/sdp. > >> Content-Length: 181. > >> . > >> v=0. > >> o=- 1161091665 1161091666 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 0.0.0.0. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 1 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKaf85a48130E1CF6C. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 1 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 2 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 181. . v=0. o=- 1161091665 1161091666 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 0.0.0.0. > >> t=0 0. > >> m=audio 2266 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bKf40b74c7C2612792;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 2 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server: > >> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20907 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bKf40b74c7C2612792. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 2 INVITE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 232. > >> . > >> v=0. > >> o=root 6068 6069 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK4ec65e93B39DC51E. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 2 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization: > >> Digest username="2222222804", realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="a690d1208542f7132ef16747f0abb98a", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK0378aca9;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> CSeq: 102 INVITE. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. Contact: > >> <sip:2222222804@10.1.99.161:5060>. Record-Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. Allow: INVITE, ACK, > >> BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, > >> UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Content-Type: > >> application/sdp. Content-Length: 185. . > >> v=0. > >> o=- 1161091674 1161091674 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> m=audio 2268 RTP/AVP 0 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> ACK sip:2222222804@81.XXX.XXX.196:5060 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bK7e63.43f60273.2. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK4365390f;rport=5060. From: > >> "0891234567" <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. > >> To: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> Contact: <sip:0891234567@2XX.XXX.XXX.97>. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 102 > >> ACK. User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> BYE sip:0891234567@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKc2fe28198F44BA84. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> To: "0891234567" > >> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Route: > >> <sip:2XX.XXX.XXX.98;lr=on;ftag=as2ff3bab6>. CSeq: 1 BYE. > >> Call-ID: 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. > >> Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKc2fe28198F44BA84. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=4245106-795C21FB>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=4245106-795C21FB. > >> To: "0891234567" > >> <sip:0891234567@sip.de.mydomain.org>;tag=as2ff3bab6. Call-ID: > >> 3a690e9411ee7c8d4523e07315bc2864(a)sip.de.mydomain.org. CSeq: 1 BYE. > >> User-Agent: MyMediaGW. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, > >> REFER, SUBSCRIBE, NOTIFY. Contact: <sip:0891234567@2XX.XXX.XXX.97>. > >> Content-Length: 0. > >> X-Asterisk-HangupCause: Normal Clearing. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 3 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc2bd45af3e3355df978e228223446d25d5b", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="4fafb927667083777a520a6f717fc8d3", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> a=sendrecv. > >> m=audio 2266 RTP/AVP 0 8 18 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:18 G729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 407 Proxy Authentication Required. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 3 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Proxy-Authenticate: Digest realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932". Server: OpenSer > >> (1.1.0-tls (i386/linux)). Content-Length: 0. Warning: 392 > >> 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK924b3fa8460CEBC3. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 3 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Max-Forwards: 70. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> INVITE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 4 INVITE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Supported: > >> 100rel,replaces. Allow-Events: talk,hold,conference. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5. > >> Max-Forwards: 70. Content-Type: application/sdp. Content-Length: > >> 247. . v=0. o=- 1161091665 1161091667 IN IP4 10.1.99.161. > >> s=Polycom IP Phone. > >> c=IN IP4 10.1.99.161. > >> t=0 0. > >> a=sendrecv. > >> m=audio 2266 RTP/AVP 0 8 18 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:18 G729/8000. > >> a=rtpmap:101 telephone-event/8000. > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 100 trying -- your call is important to us. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;branch=z9hG4bK3fa5014e6D19C349;rport=5060;received=81.XXX.XXX.196. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> CSeq: 4 INVITE. Call-ID: > >> 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. Server: > >> OpenSer (1.1.0-tls (i386/linux)). Content-Length: 0. > >> Warning: 392 2XX.XXX.XXX.98:5060 "Noisy feedback tells: pid=20903 > >> req_src_ip=81.XXX.XXX.196 req_src_port=5060 > >> in_uri=sip:0891111111@2XX.XXX.XXX.97:5060 > >> out_uri=sip:0891111111@2XX.XXX.XXX.97:5060 via_cnt==1". . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;rport=5060;received=81.XXX.XXX.196;branch=z9hG4bK3fa5014e6D19C349. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 4 INVITE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Type: application/sdp. > >> Content-Length: 303. > >> . > >> v=0. > >> o=root 6068 6070 IN IP4 2XX.XXX.XXX.97. > >> s=session. > >> c=IN IP4 2XX.XXX.XXX.98. > >> t=0 0. > >> m=audio 36580 RTP/AVP 0 18 8 101. > >> a=rtpmap:0 PCMU/8000. > >> a=rtpmap:18 G729/8000. > >> a=fmtp:18 annexb=no. > >> a=rtpmap:8 PCMA/8000. > >> a=rtpmap:101 telephone-event/8000. > >> a=fmtp:101 0-16. > >> a=silenceSupp:off - - - -. > >> a=nortpproxy:yes. > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> ACK sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKca5c1daDE079395. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 4 ACK. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > >> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: > >> PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. Proxy-Authorization: > >> Digest username="2222222804", realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="e6f5f00560921aceca9a0329333b720f", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 81.XXX.XXX.196:5060 -> 2XX.XXX.XXX.98:5060 > >> BYE sip:0891111111@2XX.XXX.XXX.97:5060 SIP/2.0. > >> Via: SIP/2.0/UDP 10.1.99.161:5060;branch=z9hG4bKcda270c0BA153C5B. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as7545c33a>. > >> CSeq: 5 BYE. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> Contact: <sip:2222222804@10.1.99.161:5060>. > >> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.0.1.0291. > >> Proxy-Authorization: Digest username="2222222804", > >> realm="sip.de.mydomain.org", > >> nonce="4534dc3073903ac300e605f4fd2834968e7ef932", > >> uri="sip:0891111111@2XX.XXX.XXX.97:5060", > >> response="d1e585adb491da5b0643fcdb7a5d8cef", algorithm=MD5. > >> Max-Forwards: 70. Content-Length: 0. . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP > >> 10.1.99.161:5060;received=81.XXX.XXX.196;branch=z9hG4bKcda270c0BA153C5B. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=DFB31535-E70817A>. > >> From: <sip:2222222804@sip.de.mydomain.org>;tag=DFB31535-E70817A. > >> To: "Benko" <sip:0891111111@sip.de.mydomain.org>;tag=as7545c33a. > >> Call-ID: 69f1042b0b11bac145d95d053db3b93f(a)sip.de.mydomain.org. > >> CSeq: 5 BYE. User-Agent: MyMediaGW. > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >> Contact: <sip:0891111111@2XX.XXX.XXX.97>. > >> Content-Length: 0. > >> X-Asterisk-HangupCause: Normal Clearing. > >> . > >> > >> # > >> U 2XX.XXX.XXX.98:5060 -> 81.XXX.XXX.196:2051 > >> BYE sip:11111111801@81.XXX.XXX.196:2051 SIP/2.0. > >> Record-Route: <sip:2XX.XXX.XXX.98;lr=on;ftag=as0fd69c4e>. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> Contact: <sip:0892222222804@2XX.XXX.XXX.97>. Call-ID: > >> 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE. > >> User-Agent: MyMediaGW. > >> Max-Forwards: 69. > >> Content-Length: 0. > >> . > >> > >> # > >> U 81.XXX.XXX.196:2051 -> 2XX.XXX.XXX.98:5060 > >> SIP/2.0 200 OK. > >> Via: SIP/2.0/UDP 2XX.XXX.XXX.98;branch=z9hG4bKf278.897153c6.0. > >> Via: SIP/2.0/UDP > >> 2XX.XXX.XXX.97:5060;branch=z9hG4bK13f9319f;rport=5060. From: > >> <sip:0892222222804@sip.de.mydomain.org;user=phone>;tag=as0fd69c4e. > >> To: "801" <sip:11111111801@sip.de.mydomain.org>;tag=lxlk4gyz7h. > >> Call-ID: 3c27c91d975e-nqjv7jntspdo@snom190. CSeq: 102 BYE. > >> Contact: <sip:11111111801@10.1.99.169:2051;line=9t7zelja>. > >> User-Agent: snom190/3.60w. > >> Content-Length: 0. > >> . > >> > >> exit > >> 78 received, 0 dropped > >> >

17 years, 9 months

[Serusers] route logic

by vijay tiwari

hi all i have facing a problem that i am not able to send the call on other sip proxy where in route logic give the ip address of particullar sip proxy to send the call on it. thanks vijay --------------------------------- Stay in the know. Pulse on the new Yahoo.com. Check it out.

17 years, 9 months

Re: [Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()

by Benko

On Tue, 17 Oct 2006 08:08:32 -0400 Nathan Hawkins <utsl(a)quic.net> wrote: > I'm testing Polycom 601's with 2.0.1 firmware and Mediaproxy. I > haven't noticed any problems with hold at all, so I haven't looked > into whether it's using 0.0.0.0 or not. I supposed it's possible that > that parameter doesn't actually work, and Mediaproxy just works with > it. > > I think though, that what you're describing below, where it resumes > from hold with 0.0.0.0 as the address, is a firmware bug. The Polycom > should definitely not be doing that. That's going to break all kinds > of things. hmm, guess i missunderstood - should the media direction parameter also be marked when "on hold" is "started" or only at the end when we resume the call that was on hold? Currently, Polycom sets IP 0.0.0.0 at the beginning of the hold, no media path direction is set. When the call is resumed later, the polycom sets the correct "sendrecv"-parameter (Which i didn't recognize earlier, sorry). However, if i understood rfc3264 the polycom should mark the media stream as sendonly on the beginning of the hold... To clarify what i'm talking about i've attached the whole sip-debug of two calls, the first call is sent on hold when a second call comes and resumed after the second call has ended(1111111801 calls 2222222804 and it answers, then 0891234567 calls 2222222804 and 1111111801 is set on hold and resumed later). Openser runs on .98, which also serves the rtpproxy. .97 is a asterisk-gateway. Thanks for the input Christian > > What firmware version are you running? > > Nathan > > Benko wrote: > > Hmm, actually this parameter is set to 0(default). > > However, in practice it still uses the old standard(rfc2543). > > There's no media direction parameter in the sdp-message sent by the > > polycom for some reason, although the manual states to do so > > (firmware 2.0.1). Is it possible to force this somehow? > > > > regards > > christian > > > > On Sun, 15 Oct 2006 19:58:33 -0400 > > Nathan Hawkins <utsl(a)quic.net> wrote: > > > >> There's an option for the Polycom phones to switch the hold > >> behaviour. > >> > >> Set voIpProt.SIP.useRFC2543hold to 0, and it should use RFC3264 > >> rules for signalling hold instead of 0.0.0.0. > >> > >> Benko wrote: > >>> Hello! > >>> > >>> I'm having a issue with NAT and rtpproxy. Usually my setup works > >>> fine with natted clients, the Connection Information is > >>> overwritten with the IP of the rtpproxy and audio passes through > >>> in both directions. However, today i came across a problem where > >>> the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the > >>> private ip after resuming a call that was on hold(actually, the > >>> other party is invited again) - and the force_rtp_proxy > >>> ()-command on openser left the ip untouched instead of > >>> overwriting it with the rtpproxy-ip. As a result the person that > >>> was on hold had audio but the polycom user (with the "wrong" ip) > >>> hadn't. > >>> > >>> The false ip left aside, is it expected behaviour of > >>> force_rtp_proxy to not touch 0.0.0.0? > >>> > >>> Just out of curiosity - does someone know the "on hold"-problem > >>> with polycoms? > >>> > >>> thx > >>> christian > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users(a)openser.org > >>> http://openser.org/cgi-bin/mailman/listinfo/users >

17 years, 9 months

[Users] Dual phone (wifi SIP/GSM) compatible list ?

by Christophe Irles

Hi, Do you know dual phones (wifi SIP and GSM) compatible with openser ? For example is Nokia E61 compatible ? Any others ? Thanks in advance, Christophe

17 years, 9 months

[Users] polycom on hold, IN IP4 0.0.0.0 and force_rtp_proxy()

by Benko

Hello! I'm having a issue with NAT and rtpproxy. Usually my setup works fine with natted clients, the Connection Information is overwritten with the IP of the rtpproxy and audio passes through in both directions. However, today i came across a problem where the Polycom 501 sets a outgoing ip of 0.0.0.0 instead of the private ip after resuming a call that was on hold(actually, the other party is invited again) - and the force_rtp_proxy ()-command on openser left the ip untouched instead of overwriting it with the rtpproxy-ip. As a result the person that was on hold had audio but the polycom user(with the "wrong" ip) hadn't. The false ip left aside, is it expected behaviour of force_rtp_proxy to not touch 0.0.0.0? Just out of curiosity - does someone know the "on hold"-problem with polycoms? thx christian

17 years, 9 months

[Serusers] How-to write a custom module

by Steve Blair

Samuel: A while back you responded to one of my posts about writing modules for SER. I am currently writing a module and I have an issue I do not understand. I'm hoping you can help. I created a directory under the ser-0.9.7-pre3/modules source directory on my server and placed my code there along with a Makefile. If I make the code from this directory it compiles with two warnings including one about running make from the main directory. If I cd to the main directory and run make clean my new subdirectory is visited as expected. If I then run make my modules does not get compiled. I believe the new directory is not recognized but I cannot figure out why. Would you hapen to be able to explain this behavior? Thank you, Steve

17 years, 9 months

RE: [Serusers] :Check for URI (from incoming messages) w.r.t database

by Kamal.Mann＠t-systems.com

Got my answer Regards Kamal Mann -----Original Message----- From: Mann, Kamal Sent: Tuesday, October 17, 2006 11:37 AM To: serusers(a)lists.iptel.org Subject: [Serdev] :Check for URI (from incoming messages) w.r.t database Hi All I need to check for each URI with respect to his/her credentials stored in SER database. I am using avpops functions avp_check & avp_db_load. I am able to compile ser.cfg but after writing this piece of code ser stops responding to all kinda requests not even for register messages. Like --> if(avp_check("s:kamal","eq/$from/username")) //avp_db_load("$from/username","s:kamal") // also tried with scheme1: example { ****some more code**** break; } Is this way is correct or I need to use some another function?? I am a newbie to ser; I need your valuable suggestions for the same. Regards Kamal Mann

17 years, 9 months

[Serusers] Authenticate when routing calls to SIP-gateway. How?

by Evert

Hi all! I am experimenting a bit with SER, and am trying to get it to do some of the same things I now have Asterisk for. But I am having trouble with the following: I want to route all non-local calls to my SIP service provider. But for this to work I need to authenticate myself first towards my SIP service provider. How do I do this with SER? Regards, Evert

17 years, 9 months

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sr-users October 2006