Wow, Thank you Tony.
You know what...when this system ran 4.0.4(upgraded from 4.0.3) it did work
without assigning the did as alias to aa in dial plan. That is why i assumed
it sort of caught all dids not specified elsewhere.

Now i know.
Thank you and have a nice weekend.

*Vänliga Hälsningar/Best Regards*

/Ola Samuelson
/



2010-04-16 13:43, Tony Graziano skrev:
Its an either you direct all inbound calls to an internal destination or you
route calls by DID, which means the DID should be an alias of the AA in the
fial plan.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431

Email: tgrazi...@myitdepartment.net

LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427

Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/

----- Original Message -----
From: sipx-users-boun...@list.sipfoundry.org
<sipx-users-boun...@list.sipfoundry.org>
To: sipx-users@list.sipfoundry.org<sipx-users@list.sipfoundry.org>
Sent: Fri Apr 16 07:26:10 2010
Subject: [sipx-users] Catch all incoming with dial plan aa

Hi!I was under the impression that if i did not direct calls at SBC
level and did not assign dids as aliases to users,
that the autoattendant defined in the dialplan would catch all calls and
direct them
to which ever attendants that i had selected(based on schedule).

Is this not true?

Sending all calls via sbc to "operator" works but not assigning a
destination at sbc
sends a relayed error respons to itsp. i.e call is not caught by dial
plan autoattendant.

11:17:08.568740 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP
(17), length 504)
88.xxx.43.xxx.AZENT-BREDBAND-NET.host.songnetworks.se.5080>   xxx.xxx.se.sip:
SIP, length: 476
          SIP/2.0 404 Not found
          Via: SIP/2.0/UDP 80.83.208.29:5060;branch=z9hG4bK28297067
          From: "0701111111"<sip:0701111...@80.xxx.208.xxx>;tag=tv3130538c
          To:<sip:0775111...@88.xxx.43.xxx:5080>
          Call-ID:74f5fb97466189f947d1fdba73b1b...@80.xxx.208.xxx
          CSeq: 102 INVITE
          Server: sipXecs/4.2.0 sipXecs/sipxbridge (Linux)
          Supported: replaces
          Contact:<sip:~~id~bri...@88.xxx.43.xxx:5080;transport=udp>
          Reason: ~~id~bridge;cause=213;text="Relayed Error Response"
          Content-Length: 0


Before i get into details/logs/snapshots. Is this how it is supposed to
work with autoattendants in the dial plan?

*Vänliga Hälsningar/Best Regards*

/Ola Samuelson
/
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