What about the Contact header, Contact:<sip:vebinar-...@sip.myservice.com:5068> Can you verify is a valid one.
On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko <ovoshl...@gmail.com> wrote: > Hello. I use kamailio for calling to porvider. My providr seccefuully > registered from UAC module, but when I try to call through it? it back 401 > Unauthorised. I send second try with Digest Auth header at INVITE and it > receive me 401 too... > > I register this provider from asterisk and call succesfully Ok. So i get > dump from asterisk This is successfull INVITE: > > INVITE sip:89126975...@sip.provider.com SIP/2.0 > Via: SIP/2.0/UDP 17.4.28.7:50600;branch=z9hG4bK5f118681;rport > Max-Forwards: 70 > From: <sip:gw2@17.4.28.7:50600>;tag=as33192a38 > To: <sip:89126975...@sip.provider.com> > Contact: <sip:gw2@17.4.28.7:50600> > Call-ID: 021088c360a8dbf023bf35560a9daf1e@17.4.28.7:50600 > CSeq: 103 INVITE > User-Agent: Asterisk PBX 12.6.1 > Authorization: Digest username="gw2", realm="provider.com", > algorithm=MD5, uri="sip:89126975...@sip.provider.com", nonce="014d80ca", > response="67bad8a0c97afc2b6747b471a56bca9f" > Date: Wed, 29 Oct 2014 18:50:50 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 253 > > v=0 > o=root 1098729670 1098729671 IN IP4 17.4.28.7 > s=Asterisk PBX 12.6.1 > c=IN IP4 17.4.28.7 > t=0 0 > m=audio 10088 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > > Then I get dump from my kamailio (unsuccessfull INVITE) > > INVITE sip:89126975...@sip.provider.com SIP/2.0 > Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as4684d4b9;lr=on> > Via: SIP/2.0/UDP sip.myservice.com:5068 > ;branch=z9hG4bK600b.1d5ff0fd59d4f3d2a1a06d722c0daa92.2 > Via: SIP/2.0/UDP my.aterisk:50600;branch=z9hG4bK2b8d9b09;rport=50600 > Max-Forwards: 70 > From: <sip:g...@sip.myservice.com:5068>;tag=as4684d4b9 > To: <sip:89126975...@sip.provider.com > > Contact:<sip:vebinar-...@sip.myservice.com:5068> > Call-ID: 445a7b884aeeab125d91886210c9b...@sip.myservice.com:50600 > CSeq: 102 INVITE > User-Agent: SoftSwitch > Date: Wed, 29 Oct 2014 22:32:32 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 312 > Authorization: Digest username="gw2", realm="provider.com", > nonce="10129bde", uri="sip:89126975...@sip.provider.com ", > response="6d3411b24cbb57ad72271790ec01b453", algorithm=MD5 > > v=0 > o=root 468654998 468654998 IN IP4 1.2.3.4 > s=SoftSwitch > c=IN IP4 1.2.3.4 > t=0 0 > m=audio 30104 RTP/AVP 8 3 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > a=rtcp:30105 > > > I see difference between packetts only at SDP (not inportant things) and > at VIA and request route Headers. All other fields identical. > > So -why Asterisk call successull and Kamailio kall unsuccessfull? What the > differense? > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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