[SR-Users] UAC INVITE to Provider

[Gonzalo Gasca]

What about the Contact header,
Contact:<sip:Vebinar-gw2 at sip.myservice.com:5068>
Can you verify is a valid one.




On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko <ovoshlook at 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 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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