[SR-Users] UAC INVITE to Provider

[Yuriy Gorlichenko]

Thanks for answer. Contact is Ok. It is just literal mistake at dump. This
error happens because CSeq not incremented by kamailio. Already talking
about this with Daniel at the another List.

2014-10-31 1:37 GMT+04:00 Gonzalo Gasca <gascagonzalo at gmail.com>:

> 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?
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20141031/3da25502/attachment.html>