vicidial.org

Hello,
I am attempting to setup a test inbound call campaign using a SIP trunk, however, it's not actually landing the call.


Here is my Configuration:
Vicibox 5 from .iso | VERSION: 2.8-437a BUILD: 140425-1257 | Asterisk 1.8.27 | Single Server | No Digium/Sangoma Hardware | PHP MyAdmin (Not being used).

Here is our SIP account entry, we are using Skype's SIP service as a test ground:

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[skype]
type=friend
context=default
dtmfmode=rfc2833
host=sip.skype.-com
username={skype user}
fromuser={skype user}
secret={skype pass}
disallow=all
allow=ulaw
allow=alaw
allow=g729
nat=yes
fromdomain=sip.skype.-com
insecure=port,invite


(the '-' was added to bypass the forum spam filter)

Here is the Asterisk error we get:

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NOTICE[2186]: chan_sip.c:23806 handle_request_invite: Call from '99051000236681' (63.209.144.201:5060) to extension '14125671197' rejected because extension not found in context 'default'.

If any additional information is needed, we can provide it.

Switching the context to "trunkinbound" seems to make the system accept the call, however we don't know where this call is being routed to. Which DID could the system be referring to? There is no longer a "default" defined on the DID listing, and the only DID defined there is as follows:

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Did Extension: 99051000236681
Description: default
Active: Y
DID Route: EXTEN (Extension: 201, context is "default")
REcord Call: N
Voicemail is 201 (same as extension)
Phone extension is blank, same with User Agent.
Unavailable action is "VOICEMAIL"
In-Group ID is "NONE"


Any other information in the details panel seems to be irrelevant, but we can post it if needed.

your asterisk command line interface should indicate the DID that was called when an inbound call arrives. that DID is the one you will need to configure.

NEVER delete the default DID. If you did, you'll need to perform a db restore to get the old one back to be sure it's "perfect". the default did is "special" and should never be deleted.