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:

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
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)

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
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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