Hey, Onur,
I will try it tomorrow when I go in to the office and post results here. Jane ________________________________ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Onur Tufekci Sent: Tuesday, June 10, 2008 4:08 PM To: OSL CCIE Voice Lab Exam Cc: <ccie_voice@onlinestudylist.com> Subject: Re: [OSL | CCIE_Voice] calling out to FXS port via SIP trunk - nocaller-id? Can you please try: Sip-ua Remote-party-id I am curious if it works. On Jun 9, 2008, at 11:11 PM, "Jane Ryer (jryer)" <[EMAIL PROTECTED]> wrote: I set up a SIP trunk from Call Manager to a router with an FXS port. When I call from the analog phone attached to the FXS port to an IP Blue phone registered to Call Manager, I do see the name and number for the FXS port (as set via station-id commands on the voice-port for the FXS port). However, if I call out from the IP Blue phone to the analog phone, all I see on the IP Blue phone is the number I dialed (4001) - no name. Is this to be expected with SIP trunks? Here is the relevant portion of my router config: voice-port 0/2/1 station-id name Analog Phone station-id number 2122214001 caller-id enable (not sure whether this accomplished anything or not - didn't work differently with or without it) ! dial-peer voice 4000 voip session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) incoming called-number 4... dtmf-relay rtp-nte (just realized that I put this command on this dial peer but not the one to CCM) codec g711ulaw no vad ! dial-peer voice 4001 pots destination-pattern 4001 port 0/2/1 ! dial-peer voice 1000 voip destination-pattern 1... session protocol sipv2 session target ipv4:10.x.x.x (IP address of my CCM) codec g711ulaw no vad ! Any insight would be appreciated. Is this supposed to work or not? Is it just a limitation of SIP? Or am I missing some configuration that is needed to pass the called name back? Thanks,
