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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.
extensions.conf:
sip.conf
Let me know if there is anything else that you need to help me troubleshoot this.
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.
Code:
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'
Code:
exten = _9.,1,Dial(SIP/${EXTEN})
Code:
[general] type = peer qualify = yes port = 5060 host = 10.0.10.3 context = from-internal canreinvite = no register => user:[email protected]/siptrunk
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