[SIPForum-discussion] originator of call

[Gopalakrishnan A.N]

That means your service provider is routing the calls based on the public
IP. Make sure the public IP is configured in your FreePBX machine or if you
have router, make sure the public IP is forwarded to your FreePBX local IP
address.



On Wed, Jun 15, 2011 at 3:58 AM, Thompson, Dantley <
Dantley.Thompson at corp.earthlink.com> wrote:

> Anyone out there had issues terminating inbound to the "FreePBX" based
> off the Asterisk SIP stack...Getting 401 unauthorized inbound, but pbx
> vendor claims he is not challenging. Modified the headers to only use
> IP, and several other attempts at moving, adding, or deleting...Pretty
> much threw the book at this, but continue to struggle. Any advice would
> be greatly appreciated...
>
> Dantley Thompson | Systems Engineer
>
>
> -----Original Message-----
> From: discussion-bounces at sipforum.org
> [mailto:discussion-bounces at sipforum.org] On Behalf Of Rohan Almeida
> Sent: Tuesday, June 14, 2011 9:39 AM
> To: Ankit Agarwal; satyadarshi28 at gmail.com
> Cc: discussion at sipforum.org
> Subject: Re: [SIPForum-discussion] originator of call
>
> Hi,
>
> you can can check the Max-forwards header. if the header is less than
> default(70) then you can conclude that it has come from proxy.
>
> ------------------------------------------------------------------------
> ----------------------------------
>
>      INVITE sip:bob at biloxi.com SIP/2.0
>      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
>      Max-Forwards: 70
>      To: Bob <sip:bob at biloxi.com>
>      From: Alice <sip:alice at atlanta.com>;tag=1928301774
>      Call-ID: a84b4c76e66710 at pc33.atlanta.com
>      CSeq: 314159 INVITE
>      Contact: <sip:alice at pc33.atlanta.com>
>      Content-Type: application/sdp
>      Content-Length: 142
> ------------------------------------------------------------------------
> ----------------------------------
>
> Max-Forwards  =  70   in the above request -> no proxy
>
> ------------------------------------------------------------------------
> ----------------------------------
>
>      INVITE sip:bob at biloxi.com SIP/2.0
>      Via: SIP/2.0/UDP pc33.atlanta.com;branch=z9hG4bK776asdhds
>      Max-Forwards: 69
>      To: Bob <sip:bob at biloxi.com>
>      From: Alice <sip:alice at atlanta.com>;tag=1928301774
>      Call-ID: a84b4c76e66710 at pc33.atlanta.com
>      CSeq: 314159 INVITE
>      Contact: <sip:alice at pc33.atlanta.com>
>      Content-Type: application/sdp
>      Content-Length: 142
> ------------------------------------------------------------------------
> ----------------------------------
>
> Max-Forwards  =  69   in the above request   ->  proxy
>
>
> There is a catch here that if the the endpoint has changed the default
> 'max-forwards' value then you have to knw the new value for 'max
> forwards' header.
>
>
> Satyadarshi: as incdicated by Ankit knowing your exact requiremant
> will be more helpful.
>
> regards,
> Rohan Almeida
>
> On 6/7/11, Ankit Agarwal <clickankit4u at gmail.com> wrote:
> > Vijay,
> >
> > This is not mandate even because if endpoint is behind SBC which is
> capable
> > of doing Topology hiding then Invite will be having only one VIA which
> will
> > be of SBC not of endpoint.
> >
> > Satyadarshi: At the same time it could be better if you could tell
> your
> > requirement where you need to find out if endpoint is from proxy or
> > endpoint.
> >
> > Ankit Agarwal
> >
> >
> >
> >
> > On Mon, Jun 6, 2011 at 5:20 PM, Vijay Tiwari
> <vijay11tiwari at gmail.com>wrote:
> >
> >> Hello Satyadarshi
> >>
> >> If in the Invite message you have only one *Via header* it means its
> >> coming from EP.
> >>
> >> Thanks
> >>
> >> On Sun, Jun 5, 2011 at 12:12 AM, K Satyadarshi
> >> <satyadarshi28 at gmail.com>wrote:
> >>
> >>>  Hi Forum,
> >>>
> >>> Can some one please tell me which header in the sip message
> indicates
> >>> that
> >>> a particular sip message is comng from an EP and not from the proxy.
> >>>
> >>> --
> >>> K Satyadarshi
> >>> Email: satyadarshi28 at gmail.com
> >>> Cell:   +91-9449112443
> >>>
> >>>
> >>> _______________________________________________
> >>> This is the SIP Forum discussion mailing list
> >>> TO UNSUBSCRIBE, or edit your delivery options, please visit
> >>> http://sipforum.org/mailman/listinfo/discussion
> >>> Post to the list at discussion at sipforum.org
> >>>
> >>>
> >>
> >>
> >> --
> >> They can because they think they can.
> >>
> >>
> >> _______________________________________________
> >> This is the SIP Forum discussion mailing list
> >> TO UNSUBSCRIBE, or edit your delivery options, please visit
> >> http://sipforum.org/mailman/listinfo/discussion
> >> Post to the list at discussion at sipforum.org
> >>
> >>
> >
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>



-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:saigop at gtalk2voip.com
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20110617/46c9be17/attachment-0002.html>