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  • Asterisk SIP Trunk

    I have a problem with my ST v 7 deployment and SIP trunks. I have a Asterisk box that is connecting to ST using a SIP trunk. I can call both ways extension to extension but when I try to transfer the call homed on an asterisk extension and connected to a ST extension to another ST extension the transfer does not work.

    ST techs are saying this is because the Asterisk system isnt handling a refer right... does anyone know if this is true or not? I dont understand how the latest enterprise version of Asterisk would have such a limitation.

    Should this work? Does anyone have experience with it?

    Thanks

  • #2
    I could be wrong but, but I don't believe transfers (either re-invite, or refer) are supported on SIP trunks in ShoreTel 7.

    Have you tried it on a later version of ShoreTel like 7.5 or 8/8.1?

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    • #3
      Any update?

      Any update on this? I'm trying to use Asterisk as my PRI interface, instead of ShoreTel's T1 box. Outgoing calls (ShoreTel -> Asterisk -> PRI) work fine. But incoming calls work "somewhat". I can reach any extension (menu, individual or workgroup). However, if I try to leave voicemail for the workgroup, OR if I try to leave voicemail for an individual at a "remote" site (not the HQ site), I get the "No messages may be left for this mailbox." Voicemails for individuals at the HQ site work fine.

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      • #4
        Not certified to work, I doubt, Highly Doubt ST will ever cert the Astrix since its free ware open source

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        • #5
          Thanks for the reply. I, too, am confident that ShoreTel will never certify Asterisk. I'm just wondering does anybody have any thoughts on my situation?

          Additional info: while looking at the Vmail-021009.log file, I find that there is a "VM: No mailbox in open mailbox" each time the call fails. Something is getting lost in the translation.

          Any thoughts or leads will be greatly appreciated.

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          • #6
            Originally posted by jmadden View Post
            Thanks for the reply. I, too, am confident that ShoreTel will never certify Asterisk. I'm just wondering does anybody have any thoughts on my situation?

            Additional info: while looking at the Vmail-021009.log file, I find that there is a "VM: No mailbox in open mailbox" each time the call fails. Something is getting lost in the translation.

            Any thoughts or leads will be greatly appreciated.
            I am workign on a solution, wil be back

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            • #7
              Your efforts are very much appreciated! Many thanks!

              On a possibly-related note, I find that I cannot leave a voicemail for any workgroup (which are all the HQ site , where the PRI and Asterisk are located) The workgroups function normally, otherwise. While watching the Asterisk console, I see that it's being told to dial 101 (the voicemail extension) when nobody answers the workgroup (which Asterisk dutifully dials). But if I dial 101 from a phone, it forwards to 106, and I can leave a voicemail for that phone. But when I try to go through Asterisk, I'm told "No messages may be left for that mailbox", the same message I get when I dial through Asterisk to one of the extensions at a remote site (if nobody answers). But if I dial a user (through Asterisk) at the HQ site, their voicemail works fine. Are workgroup mailboxes somehow "different" than user mailboxes? It gets more "interesting" each day...
              Regards,
              Jimmy

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              • #8
                Can you pm me with you SIP trunk setup info on your asterisk as well as your SIP setup on ST

                Thanks, Asterisk is simple but it does some funky stuff in trunking

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                • #9
                  SIP trunks between Asterisk and ST going out of service after upgrade

                  We are experiencing a problem were the SIP trunks (7 of them) that connect the ST to the Asterisk. They go out of service, we then have to completely delete them and set them all back up. They will stay up for a while then go out of service again. This began happening after upgrading to a new build of 8.1. Any thoughts? Thanks!

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                  • #10
                    Asterisk SIP Trunk

                    Hi, My asterisk server is trying to register with ST but keeps receiving complains that ST switch "does not implement 'RIGESTER'
                    WARNING[2952]: chan_sip.c:13288 handle_response: Host '192.168.0.112' does not implement 'REGISTER'

                    Any thoughts?

                    Thanks.

                    Comment


                    • #11
                      Originally posted by euphraty View Post
                      Hi, My asterisk server is trying to register with ST but keeps receiving complains that ST switch "does not implement 'RIGESTER'
                      WARNING[2952]: chan_sip.c:13288 handle_response: Host '192.168.0.112' does not implement 'REGISTER'

                      Any thoughts?

                      Thanks.
                      More then likely you set up the SIP Server IP incorrectly. The Switch is not the Register Server / Responder, Thats the HQ Server

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                      • #12
                        I am trying to tie NEC Sphericall with Shoretel and having a similar issue. I am using 9.1 and I believe the problem may be that Shoretel doesn't support the SIP "diversion" option when sending to voicemail ext (1101 in our case). Example of Diversion header in SIP packet below. Are there any ST engineers here that can help?

                        Scott

                        Diversion: <sip:[email protected]>;origin=external;reason=deflection

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