It means they are not acking the call. I suspect this is because sipxbridge may not be involved in the call, and only sipxproxy is, which would be problematic for a lot of call scenarios (like transfers).
I'm confused though, because it seems you are breaking "rule #1" when using sipxbridge... you are having the calls sent to port 5060 instead of 5080. When you register with teliax, can you see on their portal what port you are registering on? Can you confirm they are sending to you on a specific port? If so, what port? You should peek at this: http://blog.myitdepartment.net/wp-content/uploads/2009/11/Call-Setup-Example-sipXecs-through-ITSP1.pdf Somehow I don't believe you are doing it quite like that. On Tue, Sep 7, 2010 at 5:18 PM, Stiles Watson <wat...@datatek-net.com>wrote: > Running > > - sipXecs v 4.2.1 > - ITSP is Teliax > - SIP ports 5060 & 5061 are routed to sipX server > - RTP ports 30000-31000 are routed to sipX server > - Polycom IP 335 hardphone > > I'm able to place incoming and outgoing calls through Teliax, but calls > consistently drop after 1 min. 29 sec. > > Teliax device config change attempts: > > - Enable DNIS (teliax sends number in sip invite instead of user) > - result: calls still drop after 1 min. 29 sec., but made call > routing easier via a custom DID! > - Entered public IP under "Your IP" > - This is optional and resulted in not being able to make inbound > calls (I read in the archives that this is recommended with Teliax - is > there a sipX config change needed to make this work?) > > sipX config for teliax SIP trunk Gateway: > > - Configuration > - Enabled: yes > - Name: teliax > - SBC Route: sipXbridge-1 > - Address: den.teliax.net (this has to match with the proxy setting > in your teliax account) > - Port: 0 > - Transport protocol: Auto > - Location: all > - Shared: yes > - Caller ID > - Default Caller ID: set this to the number from Teliax > - use default for all other settings > - Dial Plan > - Enabled and added both Local & Long Distance dial plans to this > gateway > - ITSP Account > - Username: use teliax username > - Authentication Username: same as Username > - Password: use teliax device password > - Register on init: yes > - ITSP server address: same as Config-->Address above > - Use public address for call setup: yes (I tried both yes and no, > calls completed either way and did not effect disconnect problem) > - Strip private headers: default > - Use default asserted identity: default > - Asserted identity: default > - Use default preferred identity: default > - Preferred identity: default > - User part of INVITE SIP URI is a phone number: NO > - ITSP Registrar Address: default > - ITSP Registrar Port: default > - Registration interval: default > - Session Timer Interval: default > - Method to use for SIP keepalive: Empty SIP message (also tried > None) > - Method to use for RTP keepalive: Replay last sent packet (also > tried None) > - Route by To Header: default > > Any thoughts as to why the calls would drop after 1 min. 29 sec.? > > Stiles > > _______________________________________________ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
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