What about the Contact header, Contact:sip:Vebinar-gw2@sip.myservice.com:5068 Can you verify is a valid one.
On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko ovoshlook@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:89126975590@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:89126975590@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:89126975590@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:89126975590@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:gw2@sip.myservice.com:5068;tag=as4684d4b9 To: <sip:89126975590@sip.provider.com > Contact:sip:Vebinar-gw2@sip.myservice.com:5068 Call-ID: 445a7b884aeeab125d91886210c9beb7@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:89126975590@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?
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