[cisco-voip] SRST with FXO ports??

[Nate VanMaren]

If you have caller-id:

connection plar opx immediate XXX

The connection plar opx immediate option enables FXO ports to set up calls with no ring discrepancy for Caller ID between the caller and the called party. To implement the FXO Delayed Caller ID Delivery feature, you must have a configured network with a Cisco 2800 or Cisco 3800 series integrated services router running Cisco IOS Release 12.4(11)XW. The integrated services router must have at least one voice interface card. Cisco CallManager Release 4.2.3 SR1 or later releases must be installed on the network to support this feature


From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Lelio Fulgenzi
Sent: Tuesday, April 20, 2010 2:07 PM
To: Edward Countryman
Cc: Cisco-Voip-Puck
Subject: Re: [cisco-voip] SRST with FXO ports??

you've got yourself a case of TSD there my friend. (two-stage dialing)

you need need to configure PLAR on those ports so it goes to your auto-attendant.

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
Cooking with unix is easy. You just sed it and forget it.
                              - LFJ (with apologies to Mr. Popeil)


----- Original Message -----
From: "Edward Countryman" <Edward.Countryman at provena.org>
To: "Cisco-Voip-Puck" <cisco-voip at puck.nether.net>
Sent: Tuesday, April 20, 2010 3:57:42 PM GMT -05:00 US/Canada Eastern
Subject: [cisco-voip] SRST with FXO ports??


Looking for direction on a config issue, I cant seem to put my finger on the problem

I have a site with a 2811 (MGCP) with 4-POTS lines connected to a VIC2-4FXO card.

Outbound calling from the site in SRST mode works just fine, but when I call into one of the POTS lines while in SRST mode the router answers with a dial tone instead of ringing the default internal extension # 1120.  What am I missing??

dial-peer voice 10 pots
   trunkgroup 1
   answer-address 1120




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