You need to configure a dial plan to send calls out to the PSTN. There is adequate detail in the wiki, search for DIAL PLAN.
You cannot assign a DID number to a conf bridge directly, but you can create a user without a phone, assign it the DID and forward all calls in the sipxconfig gui to the CONF directly, OR you can assign the DID to a separate Auto Attendant and let peopel choose the conference they want (1 for sales conf, 2 for management conf, etc.). On Mon, Aug 9, 2010 at 11:26 AM, Ujjval Karihaloo <[email protected]>wrote: > 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 > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] 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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