Verizon IP Trunking

Question

I am starting this discussion area to help answer any questions related to Verizon's IP Trunking services for Cisco VoIP products.
I work with the Verizon engieering teams to help with Interop issues between Cisco and Verizon's VoIP network.
Please feel free to ask any questions related to design, implementation, and trobleshooting a Cisco deployment using Vz IP Trunking.
Thanks,
Charles

Answer 1

Hi Charles,
Is there a way to modify "screen=no" to "screen=yes" in Remote-Party-ID on the Cisco AS5400XM gateway for either specific dial peer or as a global setting?   
I’m running Cisco AS5400XM IOS version c5400-is-mz.124-7g.bin.
I receive the following SIP Invite from the remote SIP media/proxy (IP address of the origin is xxx.xxx.xxx.140, I don’t have ability to modify that INVITE):
Received:
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.140:5060;branch=z9hG4bK135397667
Call-ID: [email protected]
CSeq: 1 INVITE
From: "user_001" [email protected]>;tag=2359100369135397665
To: sip:[email protected];user=phone
Contact: sip:[email protected]:5060
User-Agent: Alcatel-Lucent 8460 ACS 8.5.0b5703
Max-Forwards: 70
P-Charging-Vector: icid-value=8165bb7afd93697f1d6433a60d767968
P-Asserted-Identity:
Allow-Events: edial-service-info
Supported: timer
X-owner-phone: true
Remote-Party-ID: "user_001"
Allow: ACK,BYE,CANCEL,INFO,REFER,MESSAGE,INVITE,NOTIFY
Content-Length: 208
Content-type: application/sdp
v=0
o=user_001 2359100369135397665 1 IN IP4 xxx.xxx.xxx.140
s=phone-call
c=IN IP4 xxx.xxx.xxx.140
t=0 0
m=audio 12026 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Then I use the following dial peer on AS5400XM to translate and forward this INVITE as follows:
dial-peer voice 8999 voip
description Verizon Testing
destination-pattern [phone]
session protocol sipv2
session target ipv4:63.79.178.193:5060
dtmf-relay rtp-nte
codec g711ulaw
So the INVITE from my Cisco AS5400XM to Verizon destination (63.79.178.193:5060) looks like this:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP  xxx.xxx.xxx.143:5060;branch=z9hG4bK1A1FD015E
From: "user_001" ;tag=B0052C54-164A
To:
Date: Wed, 09 Nov 2011 05:46:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,replaces
Min-SE:  1800
Cisco-Guid: 4280823408-164368865-3125598130-2410715365
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: "user_001" ;party=calling;screen=no;privacy=off
Timestamp: 1320817585
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 274
v=0
o=CiscoSystemsSIP-GW-UserAgent 2697 8709 IN IP4 xxx.xxx.xxx.143
s=SIP Call
c=IN IP4 xxx.xxx.xxx.143
t=0 0
m=audio 17446 RTP/AVP 0 101 19
c=IN IP4 xxx.xxx.xxx.143
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
Everything works out, besides the field
Remote-Party-ID: "user_001" ;party=calling;screen=no;privacy=off
Instead of that, according to Verizon - I need to send:
Remote-Party-ID: "user_001" ;party=calling;screen=yes;privacy=off
Only difference is I’m sending "screen=no", but I need to send "screen=yes"
Is there a way to modify "screen=no" to "screen=yes" on the gateway itself for either specific dial peer or as a global setting?   
Thanks you!