[SR-Users] UAC INVITE to Provider

[Yuriy Gorlichenko]

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 at 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 at 17.4.28.7:50600>;tag=as33192a38
To: <sip:89126975590 at sip.provider.com>
Contact: <sip:gw2 at 17.4.28.7:50600>
Call-ID: 021088c360a8dbf023bf35560a9daf1e at 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 at 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 at 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 at sip.myservice.com:5068>;tag=as4684d4b9
To: <sip:89126975590 at sip.provider.com >
Contact:<sip:Vebinar-gw2 at sip.myservice.com:5068>
Call-ID: 445a7b884aeeab125d91886210c9beb7 at 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 at 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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