Welcome to

Welcome to!

This site was created as a place to share stories, tips, and troubleshooting help with ShoreTel/Mitel systems. ShoreTel/Mitel is obviously the MOST exciting VoiP platform on the market right now, and we realized there was no centralized place to discuss this platform, but now there is. Please feel free to join and share your experiences.

Please Note: This site IS NOT owned, funded, or managed by ShoreTel/Mitel, Inc. although you may find ShoreTel/Mitel employees sharing there experiences and expertise. If you would like more information on ShoreTel/Mitel systems, contact BTX at [email protected]

As always please support the advertisers that help support our site.

Thank You,
See more
See less
  • Filter
  • Time
  • Show
Clear All
new posts

  • Cannot Place Outgoing Call from Asterisk through Shoretel.

    We have setup a conference bridge on an Asterisk beige box. Currently you can dial a DID or extension number and it will forward to one of a couple conference rooms.

    We would like the asterisk box to be able to place calls on the Shoretel, though the PRI or to local extensions. However we encounter an error when doing so.

    ocalhost*CLI> console dial 96002
      == Console is full duplex
        -- Executing [96002@default:1] Dial("OSS/dsp", "SIP/96002") in new stack
    [Sep  8 10:39:36] WARNING[3779]: chan_sip.c:2907 create_addr: No such host: 96002
    Really destroying SIP dialog '[email protected]' Method: INVITE
    [Sep  8 10:39:36] WARNING[3779]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
      == Everyone is busy/congested at this time (1:0/0/1)
      == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL'
    exten = _9.,1,Dial(SIP/${EXTEN})
    type = peer
    qualify = yes
    port = 5060
    host =
    context = from-internal
    canreinvite = no
    register => user:[email protected]/siptrunk
    Let me know if there is anything else that you need to help me troubleshoot this.

  • #2
    This has been resolved.

    The extensions.conf should have read:

    exten = _9.,1,Dial(SIP/${EXTEN:1}@my_provider_out)