Hello,
I have a customer with a CYCM 14 cluster who has recently added a new site that has quite a few analogue phones. A Cisco VG410 48 FXS port gateway was installed to connect these analogue phones.
The VG410 is quite a new device and is not supported as an SCCP gateway on the current version of CUCM used by the customer. It would need to be upgraded to SU3 or later to support it.
This upgrade was not possible in the near future so the VG410 was set up with POTS and VOIP dial-peers and connected to CUCM using a SIP trunk.
This works ok for basic calling but quite a few features are lost including:
Ability to place analogue phones in Line Groups
Ability to (easily) control dialling privileges using CSS
Ability to present Calling Name to IP phones registered to the CUCM cluster.
Of these issues the Calling Name presentation is the main problem for the customer so I am trying to work out a way in which this can be achieved with the current configuration.
Below is an example of an Invite sent by the VG410 when an analogue phone dials an IP phone.
001439: Oct 28 2024 15:38:04 GMT: //3340/7508F47D8D95/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:5103101@10.243.101.201 SIP/2.0
Via: SIP/2.0/UDP 10.243.57.200:5060;branch=z9hG4bK9DE667
Remote-Party-ID: <sip:5108437@10.243.57.200>;party=calling;screen=no;privacy=off
From: <sip:5108437@10.243.57.200>;tag=FDC9F4-2246
To: <sip:5103101@10.243.101.201>
Date: Mon, 28 Oct 2024 15:38:04 GMT
Call-ID: 75094291-947911EF-8D9ACE2A-E62BE12A@10.243.57.200
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1963521149-2490962415-2375405098-3861635370
User-Agent: Cisco-SIPGateway/IOS-17.12.4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1730129884
Contact: <sip:5108437@10.243.57.200:5060>
Expires: 180
Allow-Events: telephone-event
Session-ID: cd87dd35ab0955ddb539011a6458a051;remote=00000000000000000000000000000000
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 321
v=0
o=CiscoSystemsSIP-GW-UserAgent 9923 2964 IN IP4 10.243.57.200
s=SIP Call
c=IN IP4 10.243.57.200
t=0 0
m=audio 16440 RTP/AVP 8 0 18 101
c=IN IP4 10.243.57.200
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Timestamp: 3812974684000
UTC Timestamp:3812974684000
My initial idea was to use SIP profiles on the VG410 to add the Calling Name to the Invite above. I am unsure which header(s) I would need to modify and, if I do that, whether the name will be passed through and used in the Invite that CUCM sends to the called IP phone.
I have tested using the SIP profile below which does not make the calling name appear on the IP phone.
voice class sip-profiles 1
rule 101 request ANY sip-header From modify "<sip:5108437@" "\"Test Calling Name\" <sip:5108437@"
I have also thought of using SIP Normalization on CUCM to do this but have not managed to test yet.
This is complicated by the fact that the customer site is 200 miles away from me and they are not very helpful when it comes to testing. I am currently unable to pull traces off CUCM as I cannot run RTMT of use an SFTP server.
I am hoping that someone can help me with this. Should SIP profiles on the VG410 allow me to do this or should I be looking at CUCM SIP Normalization or maybe something else?