Got it...its registered now. What is the best place to start for inbound and outbound routing...
Inbound to a Conf Bridge for example. I want to call into the SIP trunk from PSTN and hit a Conf Bridge. Outbound from registered endpoints and send the call out the SIP trunk. Any pointers to Documentation/wiki links will be appreciated. -----Original Message----- From: Tony Graziano [mailto:[email protected]] Sent: Monday, August 09, 2010 5:08 AM To: [email protected]; Ujjval Karihaloo; [email protected] Subject: Re: [sipx-users] SIP Trunk Setup questions You need to make sure you choose sbc route/sipxbrige1 to get these options. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected] <[email protected]> To: 'Ujjval Karihaloo' <[email protected]>; [email protected] <[email protected]> Sent: Mon Aug 09 02:05:05 2010 Subject: Re: [sipx-users] SIP Trunk Setup questions I think you are missing a few options on the page. When you open Device/gateway you will get a menu on the left hand side of the screen. Click on ITSP account. You can enter your username and password on that screen. Don't miss the Advanced Options on the upper right hand corner while in that screen, it will give you some additional choices! From: [email protected] [mailto:[email protected]] On Behalf Of Ujjval Karihaloo Sent: Sunday, August 08, 2010 7:56 PM To: [email protected] Subject: [sipx-users] SIP Trunk Setup questions Used the following instructions to install SIPX. http://sipx-wiki.calivia.com/index.php/Installing_sipXecs_on_Fedora_and_Cent os (BTW - Import Yum repository for CentOS does not work) How do I register my SIPX with a ITSP. I do not see any way to put in a username and passwd.only address (FQDN)... XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX Gateway : MyTestSIPTrunk <https://208.77.200.20:8443/sipxconfig/gateway/EditGateway,$gateway$GatewayN avigation.$DirectLink_0.sdirect?sp=1&state:gateway/EditGateway=BrO0ABXcnAAAA AQEADiRjb21tb24kQm9yZGVyABBpbml0aWFsU2Vzc2lvbklkdAAMengwYzMwejg3ZDVt> / SIP trunk To setup a new gateway fill in the parameters on this page, then setup PSTN Lines. No other settings need to be considered as all gateway parameters are auto-configured for a typical deployment. Consult the gateway vendor's manual or ask an expert for advice if other parameters need to be adjusted. Changes applied successfully. * Hide Advanced Settings <javascript:tapestry.form.submit('Form',%20'setting:toggle');> Enabled Name Address For a PSTN gateway: IP address of the gateway (example: 10.1.1.1) or the fully qualified hostname of the gateway (example: gateway.example.com). The gateway can be on any routed subnet without NAT. For an ITSP SIP Trunk: External IP address or fully qualified hostname of the Internet Telephony Service Provider (e.g. itsp.example.com). Note: An SBC Route needs to be defined in the field below. For a Direct SIP Trunk: To interconnect two VoIP systems using SIP enter the IP address or fully qualified name of the other system. Port Optional port if the gateway uses a non-standard port. Set to 0 to ignore this field (example: 5070). Transport protocol Set to force the SIP transport protocol. If set to auto the transport is determined through a DNS query. Location Restrict the gateway by selecting a specific location for which it can be used. A location is represented by a group of users and you need to create a branch for every location that needs to be distinguished. This setting allows routing of calls based on in which location or by which user the call originates (source routing). This is useful if users located in a branch office would like to have a gateway preference so that calls are routed through their local gateway, i.e. to preserve WAN bandwidth or to use Caller ID offered by an analog gateway based on the PSTN number assigned to it. Only if that gateway is not available call routing will fall back to other gateways specified for the corresponding dialing rule. Shared If checked this gateway can be used by any user in any location, even if a specific location is selected. This setting is checked by default so that users in an identified location still use their preferred gateway, but the gateway can also be used by other users in other locations. Description _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
