Currently have a SIP trunk that requires authentication w/ ITSP. My sip-ua section is as follows:
sip-ua
credentials number AAAAAA username AAAAAA password 7 mypassword2 realm myitsp.net
authentication username AAAAAA password 7 mypassword
retry invite 2
registrar dns:voiceserver.myitsp.net expires 3600
sip-server dns:voiceserver.myitsp.net:5060
ITSP is migrating away from requiring authentication and will accept the call based on the source ip of the CUBE, so I made the following changes to my sip-ua(realized now I didn't remove the (retry invite 2" line):
sip-ua
no credentials number AAAAAA username AAAAAA password 7 mypassword2 realm myitsp.net
no authentication username AAAAAA password 7 mypassword
registrar ipv4:1.2.3.4 expires 3600
sip-server ipv4:1.2.3.4:5060
On my outbound dial-peers I updated the destination to the new ipv4 address I should be pointing to and added this(Gi0/2 has the public ip that points to ITSP):
session target ipv4:1.2.3.4
voice-class sip bind control source-interface gigabitEthernet 0/2
voice-class sip bind media source-interface gigabitEthernet 0/2
I updated the trusted list under voice services voip as well.
I cleared the sip connections to the old public ip, but wasn't able to get working inbound/outbound calls. Outbound calls weren't seen by the carrier (on my softphone I was able to dial but call wouldn't setup); inbound calls I'd see the incoming call and be able to answer it but on the calling phone it would either keep ringing or have no ringback. Then when I disconnected the call on the calling phone carrier would see the call cleared normally due to me hanging up. ITSP said this setup is identical to the existing setup so thinking current settings should work.
Took a packet capture on the CUBE and saw incoming INVITES from the carrier but no responses from the CUBE back to them, only outbound RTP. Looking at a debug for a test outbound call I seem to see the invite being sent, but they said they never received it(looking in more detail, the only invites I see on outbound calls are when I had removed the external phone number mask from the DN on the softphone):
INVITE sip:called_number@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP my_public_ip:5060;branch=z9hG4bK341CBE
Remote-Party-ID: "Test Phone" <sip:9009@my_public_ip>;party=calling;screen=yes;privacy=off
From: "Test Phone" <sip:9009@1.2.3.4>;tag=B91D8-523
To: <sip:called_number@1.2.3.4>
Date: Thu, 28 Aug 2025 01:38:20 GMT
Call-ID: 8A1332A2-8F2E6011-3113D78D-033C4684@my_public_ip
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4427858533-6500230536-9600012044-2256001809
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M5
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1756345100
Contact: <sip:9009@my_public_ip:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Content-Length: 0
My debugs got a bit jumbled up but I'll try to post them if I can get them straightened out.
