OK. What you have told me makes me ask more questions> Call Flow:
PSTN number 1234567 calls subsciber 123 at 3335678. UA/subscriber is at IP x.x.x.x, sipx is at x.x.y.z Explain what you expected and what happened. What I am seeing is TCP which usually indicates the UA is a softphone. If so, WHAT is it? The UA MUST support REFER to do transfers (i.e. xlite cannot transfer calls). What do you mean sipxbridge is opne on the internet? Explain your NIC configuration in sipx. Be aware, 2NIC's is NOT supported at this time. If so, please report that back so we know precisely WHAT you are operating with. On Wed, Mar 28, 2012 at 5:41 PM, Josh Kennedy <[email protected]> wrote: > > Content-Type: text/plain; > charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > In-Reply-To: < > camgknjxljqwu+di_itlxswyfbbzdjcgofzuocu8qkvz+rqw...@mail.gmail.com> > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67050> > Message-ID: <[email protected]> > > > > I'm not completely sure what you mean by call flow, but I'll > answer with what I think you are asking, please let me know > if I'm off base. Also, the bridge connects straight out to > the provider without passing through a firewall. > > Phone dials external number, invite request goes to proxy, > proxy and phone authenticate, proxy sends invite to > registrar which responds with a redirect pointing to the > bridge, proxy sends invite to bridge, bridge responds with a > 404 to the proxy, proxy responds to phone with a 404. > > An odd thing I've noticed that differs between the failed > and successful calls is one of the sip via headers on the > invite between the proxy and bridge. The rport on failed > calls is the same port that the proxy listens on. On > successful calls the rport is a random high port number, not > sure if that is indicative of the problem or not. > > * Failed call header: > Via: SIP/2.0/TCP > 172.22.2.10;branch=z9hG4bK-XX-66c918`Ff7O4WerPX6DHjNI8eg;rpo > rt=5060 > > * Successful call header: > Via: SIP/2.0/TCP > 172.22.2.10;branch=z9hG4bK-XX-67b4Pdrdq8pBHUFbEnQ0H8w4eQ;rpo > rt=32769 > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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