Morning Richard, Not exactly sure how it's done the Trixbox way but I'll explain a couple of scenario on similar configuration.
- DID Origination Provider doesn't require me to register with them --> Asterisk is set to send all unknown inbound calls to [inbound-calls] context I created with a simple line exten => 4162220000,1,Answer blah blah blah and any no-matching DIDs goes to exten => _X.,1,Answer blah blah [sent to simple echo-test] - DID orgination providers requiring me to trust a number of their orginating IPs but no registration --> A sip trunk is created as follow [Provider_IP] host=Provider_IP type=friend insecure=very context=my_custom_context canreinvite=no - DID orig. providers requiring a registration before they can send calls A context is set as yours below with the exception of having "insecure=very" and a specific context to route the call to. The last 2 scenario is easier to debug and see where the call is breaking. As you may set the debug level to 10, then disable all debug and just enable the specific peer sip debug peer SIP_Trunk My suggestion: try directing a particular context [in lieu of from-pstn] and set it to echo-test. It might just make a difference between the call ever reaching your asterisk box or not. HTH, have a great one! Claudius ________________________________________ From: Richard (Rogers @ work) [EMAIL PROTECTED] Sent: Wednesday, October 10, 2007 8:43 AM To: Claudius Fortis; [email protected] Subject: Re: [on-asterisk] Setup asterisk as a provider Hi Claudius, Just a bit more backgroung. I was told by Mike in this group that the following would show the CID. 1/ use this mapping format to map the DID say 4162220000 with SIP. [EMAIL PROTECTED] (no password required) 2/ Create an inbound route in freeePBX for DID 4162220000. When I called the DID, I got ringing tone but my phone/destination specified in the inbound route did not ring. I turned on sip debug and did not see anything related to this DID. Question is where did the call go? How come the sip debug did not reveal anything? Am I still missing steps? Thanks, Richard ----- Original Message ----- From: "Richard (Rogers @ work)" <[EMAIL PROTECTED]> To: "Claudius Fortis" <[EMAIL PROTECTED]>; <[email protected]> Sent: Wednesday, October 10, 2007 8:26 AM Subject: Re: [on-asterisk] Setup asterisk as a provider > Hi Claudius, thanks for your reply. More detail is provided below. > > Richard > ----- Original Message ----- > From: "Claudius Fortis" <[EMAIL PROTECTED]> > To: <[email protected]> > Sent: Tuesday, October 09, 2007 9:40 PM > Subject: RE: [on-asterisk] Setup asterisk as a provider > > > Without much details on what you're trying to accomplish or what's > failing, I thought I'd try > > - Are you thinking of allowing external users connect to your Asterisk box > via SIP or IAX protocol and make outbound calls? > --> If this is case, can local users register and call out > RK-> I bought a DID from a provider and their website allows me to map it > with SIP/IAX. It simply for incoming calls. > > - Shouldn't your register line be in this format? > --> UserName:[EMAIL PROTECTED] and sometimes you might need to add > /Username at the end of your register line > RK--> I am currently using this format below but I do not get the CID of > the originating caller for the incoming calls. > extension1:[EMAIL PROTECTED]/extension2 > Instead, I only get extension1 shown in the history. > > - "What I got was a ringing tone but the actual destination did not ring" > --> Is the destination, username/extension 4162220000 ? > RK-> Destination can be anything ranging from ring group to individual > extension as I configure that with inbounf route in freePBX. > > > Hoping for some more details, have a great one! > > > > Claudius > > ________________________________________ > From: Richard (Rogers @ work) [EMAIL PROTECTED] > Sent: October 9, 2007 1:32 PM > To: [email protected] > Subject: [on-asterisk] Setup asterisk as a provider > > Hi, > > I would like to setup my asterisk to accept SIP or IAX incoming calls and > then use the inbound route to route calls to various destinations. > > I setup a trunk with the following settings and use the following to > connect. > allow=ulaw > context=from-pstn > Secret=abcdef > Type=friend > Username=4162220000 > > [EMAIL PROTECTED] (no password) > > What I got was a ringing tone but the actual destination did not ring. > I also have an inbounf route created for the 4162220000. I also went to > the > general section to turn on the anonymous SIP incoming calls. > > Any suggestions? > > Thanks, > Richard > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] > --------------------------------------------------------------------- To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
