Solved Unusual call behavior - Voice Agent Ending Call

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infograpes

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I'm seeing a weird error and not quite sure what to make of it.

Some users have reported getting the message "End destination cannot be reached, contact administrator", then the call ends. When they call back, they are able to reach someone.

When this happens, in the call log I see this activity:

Code:
02/20/2026 10:28:20 AM    000-818 
+1XXXXXX0222
Voice Agent
Inbound    Answered    -    00:00:00    00:00:05    0.00    Inbound: [v]CALLER XXXXX (+1XXXXXX0222) → Via trunk: SIPTRUNK.com Main (+1XXXXXX9683) → Ended by Voice Agent

02/20/2026 10:28:45 AM    000-819   
+1XXXXXX0222
Main Reception (8000)
Inbound    Answered    -    00:00:00    00:00:33    0.00    Inbound: [vCALLER XXXXX (+1XXXXXX0222) → Via trunk: SIPTRUNK.com Main (+1XXXXXX9020) → Main Reception (8000) was replaced by Doe, Jane (XXXX)

The first entry above with the number ending in 9683 is not a number that I recognize and is not listed in the PBX at all. I did check the CDR with the SIP provider and they dialed the number ending in 9020 both times.

1771863446533.png

There is also a corresponding entry in the activity log at the same time of the first call entry:

Code:
02/20/2026 10:28:20.814 AM    C:2072 Failed to build divert target EndCall.DisabledByFwdRule; fail reason = NotAvailable

The C:2072 number changes each iteration of the issue (I'm assuming this is a call number?), but the rest of the message stays the same.stays the same.


The call flow is:

Caller dials 1XXXXXX9020 -> Digital Receptionist
Digital Receptionist xfers to Ring group
Ring group xfers to Queue if no answer or extension unavailable.

I'm not sure what else to include as this is the first time I've been stumped and not been able to find more info on this behavior.

I appreciate any insight or assistance.
 
Hello @infograpes

Can you export your trunks configuration and post the section with the "ParameterIn" entries here? Are you using the default template for SIPTRUNK.com?
 
Yes, I believe we are using the default template for SIPTRUNK.COM.

I'll paste the whole section with "ParameterIn" and "ParameterOut"

<field name="ParameterIn" custom="" parameter="ToUserPart">$CalledNum</field>
<field name="ParameterIn" custom="" parameter="FromDisplayName">$CallerName</field>
<field name="ParameterIn" custom="" parameter="ContactUser">$CallerNum</field>
<field name="ParameterOut" custom="" parameter="RequestLineURIUser">$CalledNum</field>
<field name="ParameterOut" custom="" parameter="RequestLineURIHost">$GWHostPort</field>
<field name="ParameterOut" custom="" parameter="ContactUser">$AuthID</field>
<field name="ParameterOut" custom="" parameter="ContactHost">$ContactUri</field>
<field name="ParameterOut" custom="" parameter="ToDisplayName">$CalledName</field>
<field name="ParameterOut" custom="" parameter="ToUserPart">$CalledNum</field>
<field name="ParameterOut" custom="" parameter="ToHostPart">$GWHostPort</field>
<field name="ParameterOut" custom="" parameter="FromDisplayName">$OriginatorCallerId</field>
<field name="ParameterOut" custom="" parameter="FromUserPart">$OriginatorCallerId</field>
<field name="ParameterOut" custom="" parameter="FromHostPart">$GWHostPort</field>
<field name="ParameterOut" custom="" parameter="RemotePartyIDCallingPartyDisplayName">$OriginatorCallerId</field>
<field name="ParameterOut" custom="" parameter="RemotePartyIDCallingPartyUserPart">$OriginatorCallerId</field>
<field name="ParameterOut" custom="" parameter="RemotePartyIDCallingPartyHostPart">$GWHostPort</field>


I have also spoken with the carrier and they advised that they are receiving the following information when one of the calls comes through:

Code:
02/23/2026 10:44:53.461 -05:00: XX.XXX.XX.XXX:5060 -> XXX.XXX.X.XX:5060
INVITE sip:[email protected]:5060;rinstance=29a223c8bfea6d67 SIP/2.0
To: sip:[email protected]
From: "CALLER" sip:[email protected]
Call-ID: [email protected]
CSeq: 487571 INVITE
Contact: "MERCY ." sip:[email protected]:5060
Diversion: sip:[email protected];reason=unknown;screen=yes;privacy=unknown
Privacy: off
P-Asserted-Identity: "CALLER" sip:[email protected]

"Do note that the Request URI header does contain the proper destination number 1XXXXXX9020. To accept these sorts of forwarded calls, I recommend checking with 3CX support to assist in configuring your trunk to have it read off of the RURI header for inbound calls. "
 
Last edited:
You are actually using an outdated version of the SIPTRUNK template. You need to delete your existing trunk and recreate it using the updated template.
The calls you are having issues with are forwarded calls (from SIPTRUNKs end) which do not contain the correct number in the To:UserPart header.
The new template reads the number from the Request URI header.
 
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Okay, I will work on that this evening. Thanks for the insight. Re-creating the trunk configuration did cross my mind, but as a nuclear option.
 
It looks like that has resolved the behavior. Thank you for your help!
 
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