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  • SIP Trunk to Trixbox

    We have set up SIP trunks from Shoretel to Trixbox. Everything works great except caller id from Shoretel. Does Shoretel use standard cid on sip protocol. The caller id from the sip phones from Trixbox passes through to the Shoretel phones just fine. We can even transfer between the two systems but the CID stays from the original caller..very weird.
    We are just testing to see if we can use the Trixbox at remote locations to save money on switches. Anyone have any info regarding the CID on the SIP Trunk in ST?

    Thanks,
    Andy

    we got the CID fixed..had to check the box so it would allow CID on the Trunk..
    Last edited by ajmorris; 06-21-2007, 02:28 PM.

  • #2
    I would like to get configuration steps for setting up Shoretel to Trixbox freepbx.

    Comment


    • #3
      Ok, that makes three of us

      Comment


      • #4
        Make it 4. I would love this!
        ~ MiJa

        Comment


        • #5
          He come here for help and everyone wants help from him hahaha.

          Add me as # 5 too

          Comment


          • #6
            Asterisk -> ShoreTel working!

            Ok, here's my setup. (comments in parenthesis)
            added a SIP trunk group on the ST side with the following settings:
            Name: APC_SIP
            Teleworkers: off
            Enable SIP Info for G.711 DTMF Signaling: off
            Enable Digest Authentication: on
            User ID: siptrunk
            Password: ****
            Number of Digits from CO: 4 (same as my ST setup)
            DNIS: off
            DID: off
            Extension: on
            Translation Table: clicked <none>
            Tandem Trunking: off
            Destination: **** (our main Auto Attendant, this is only if it can't match the dialed extension)
            Outbound: off (i decided to start small so i turned outbound calls off. this means in can only make calls asterisk -> shoretel. But it's a start!)
            everything else is off.

            then I setup Asterisk with the following settings.
            Outgoing:
            Trunk Name: APC_SIP
            host=193.168.100.4 (the IP of my shoretel switch that has the SIP ports)
            secret=**** (same as digest password from the ST setup)
            type=peer
            username=siptrunk (same as digest user name from ST setup)

            Incoming: blank (again, i'll work on shoretel -> asterisk after i get asterisk->shoretel perfected)

            Registration String: siptrunk:****@193.168.100.41/siptrunk (**** is the digest password)

            I then brought up an asterisk soft phone and dialed my shoretel extension. my phone rang! the caller ID was the extension of my asterisk phone and the trunk was APC_SIP! However, when i picked up, there was no voice. Time to dive into the logs on both sides and figure out what happened.
            I'll post my results.
            ~ MiJa

            Comment


            • #7
              ok, i setup a SIP hard phone (linksys SPA-841) and made a test call with sound! I guess it was something in my PC sound setup (perhaps conflicting with personal call manager). Anyway that works, but i'm still stuck on shoretel->asterisk calls.

              Perhaps someone here knows the answer. How do i tell the Shoretel system to send certain calls over the SIP trunk? I want all calls to 1000-1099 to go to the asterisk box.
              ~ MiJa

              Comment


              • #8
                Asterisk &lt;-&gt; ShoreTel making progress, need ST expertise

                I'm kind of stumped with the outgoing trunk settings. When i created the trunk, ST forced me to enter an Access Code. I entered 9. Does that mean I have to dial 9 to access the trunk? Also, I added 1000-1099 to the "Off System Extensions" list, but when i dial 1001 it says "Cannot complete call (Reorder Tone)" and when i dial 91001 in PCM (personal call manager) it says "This input does not contain enough digits to form a valid North American phone number"
                When i dial 91001 from a ST phone it says "Intercept Tone".

                I checked the group permissions and gave my group access to the SIP trunk.
                I also checked the trunk test tool and there is was no activity on the SIP trunk when i tried the test calls.

                I'm sure a ShoreTel expert could point me in the right direction, and I know there are plenty of those hanging around this forum

                Comment


                • #9
                  SIP trunk to Asterisk - Working!

                  I deleted and recreate the trunk group in ShoreTel and it works (including caller id). Let me know if anyone else needs help with this.
                  thanks for the assistance.
                  ~ MiJa

                  Comment


                  • #10
                    About Trunks and Off system extensions

                    Just to answer some of MiJaMu's questions from earlier in case someone else needs the info:

                    Even though the GUI makes you choose a trunk access code when you create the trunk group, it doesn't matter once you create off system extensions. Also you need not grant anyone permission to use the trunk. In fact you probably shouldn't unless there are actual PSTN services available on the trunk.

                    Suppose all your phone lines (PRI, POTS, whatever) are busy, and someone picks up their phone, dials 9 and a number. The call is going to attempt to go out the SIP trunk. You can prevent this by just not granting anyone access to the trunk. With off system extensions, the call to the extension will still go out the correct trunk, whether or not the caller has permission to access the trunk.

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                    • #11
                      Asterisk admin

                      What are the incoming settings for the asterisk trunk? We cannot receive calls from the Shoretel.

                      Thanks in advance.

                      JEG

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                      • #12
                        re: asterisk incoming settings.

                        user-context: from-pstn

                        user details:
                        username=siptrunk (username I used when setting up the trunk)
                        type=peer
                        secret=whatever (same password as digest authentication in ST)
                        insecure=very
                        host=192.168.0.4 (the ip address of the shoretel vm server)
                        dtmfmode=info

                        register string:
                        siptrunk:[email protected]/siptrunk

                        this worked for me
                        ~ Mike
                        Last edited by MiJaMu; 03-11-2008, 02:09 PM.

                        Comment


                        • #13
                          Password

                          Do you have to use a password or can you leave it blank? SO if I dont set a password in ST on the sip trunk I would not have to put that info in the Asterisk.


                          Thanks
                          Ronnie

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                          • #14
                            i suppose you don't have to use a password, but it would be unsecure. Anyone could tap into the SIP trunk and make unauthorized calls.

                            Comment


                            • #15
                              Thanks!

                              I may yet hit you up for add'l details, but thank you so much for giving all that info. I'm very interested in giving it a shot, and I'll see what happens.

                              -Ken

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