I am having an issue with SNR (to cell phone) when an external caller calls in via POTS (FXO) the call is directed to a hunt group, all the extensions ring and the call is received on my cell phone. When I try to answer the call the call does not connect and drops immediately, the other phones in the hunt group keep ringing.
Call flow - Inbound
PSTN (FXO) > Gateway 2851 (SIP) > CUCM
Call flow - Outbound
CUCM > Gateway 2851 (SIP) > SIP Trunk Provider
I've tried the following:
- Calling the users extension that SNR is enabled for and SNR is working fine. I can answer the call on my cell phone, and move it back to my desk phone.
- Call the hunt group DN that the external POTS line uses and I can I can answer the call on my cell phone, and move it back to my desk phone.
- Changed the FXO port on the gateway to the DN where SNR is enabled and the same problem exists.
I've gathered some packet captures from the firewall between the Gateway and the SIP trunk provider and found that the SIP invite appears to contain the wrong source number so I am assuming the provider is rejecting the call.
Here is an example of an internal extension calling the hunt pilot dn (1075) - This call is successful
The from sip: ending in 01 is the DID Number that is provided by the SIP provider. The To sip: ending in 44 is my cell phone number
Here is an example of an external caller, calling inbound on the FXO port and hitting the hunt DN (1075) - This call fails
The From sip: ending in 01 is the DID Number that is provided by the SIP provider. The To sip: ending in 44 is my cell phone number
The second line the from sip: ending in 04 is the external caller and the To sip: is the hunt DN which appears to be why the SIP trunk provider is rejecting the call.
After much searching I found this https://community.cisco.com/t5/collaboration-voice-and-video/configure-and-troubleshoot-call-forward-to-the-pstn-using-sip/ta-p/3118287 which indicates we can use the P-Asserted-Identity to resolve this. (The SIP trunk provider confirmed they do support P-Asserted_Identity)
After adding this to the outbound sip dial-peer I see that the P-Asserted-Identity is showing up but it doesn’t change the To sip field it still stays as the hunt pilot DN
This is and older environment CUCM is 8.6.2 and the gateway is a 2851. but tis what I have in the lab right now.
Adding P-Asserted Identity header to the SIP header will not change the To header. It’s expected that the ITSP should let the call pass if the PAI has a valid number from the assigned range even if the To field contains an out of range number. Can you share the configuration you’ve made for adding PAI to the SIP header?
In your packet capture you have two PAI.
P-Asserted-Identity: <sip:mydid@X.X.4.2>
Session-Expires: 1800
Content-Length: 0
P-Asserted-Identity:<sip:mydid@toronto.voip.ms>
Change the SIP profile to modify the PAI instead of adding a new one.
