Call over VPN drops when party picks up

Asterisk Asterisk Support
1

I have a vpn on my android phone to my corporate LAN, where my Asterisk is. My SIP client on my android phone registers, and when I make a call, I hear ring tone, but as soon as the other party picks up, the call drops.
Below some SIP debug. Not sure what exactly to provide as these debug logs are so vast.

<— SIP read from UDP:10.32.0.17:50237 —>
INVITE sip:5200@10.1.1.184;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.32.0.17:50237;branch=z9hG4bK-524287-1—313609e32ad11637;rport
Max-Forwards: 70
Contact: sip:5203@10.32.0.17:50237;transport=UDP
To: sip:5200@10.1.1.184;transport=UDP
From: sip:5203@10.1.1.184;transport=UDP;tag=3312b067
Call-ID: WC7ttNXpj0ilnElXzecZVQ…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r28827
Allow-Events: presence, kpml
Content-Length: 240
<— Reliably Transmitting (no NAT) to 10.32.0.17:50237 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.32.0.17:50237;branch=z9hG4bK-524287-1—313609e32ad11637;received=10.32.0.17;rport=50237
From: sip:5203@10.1.1.184;transport=UDP;tag=3312b067
To: sip:5200@10.1.1.184;transport=UDP;tag=as19940c80
Call-ID: WC7ttNXpj0ilnElXzecZVQ…
CSeq: 1 INVITE
Server: FPBX-12.0.25(11.9.0)
Scheduling destruction of SIP dialog ‘WC7ttNXpj0ilnElXzecZVQ…’ in 13120 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 10.32.0.17:50237:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.32.0.17:50237;branch=z9hG4bK-524287-1—313609e32ad11637;received=10.32.0.17;rport=50237
From: sip:5203@10.1.1.184;transport=UDP;tag=3312b067
To: sip:5200@10.1.1.184;transport=UDP;tag=as19940c80
Call-ID: WC7ttNXpj0ilnElXzecZVQ…
CSeq: 1 INVITE

2

Everything with the same Call-ID as the first INVITE.

You should also take them from the log file, not the screen, as there are no time stamps on the screen.

As the Zoiper seems to be using late offer SDP and you cut off the log before the INVITE was accepted, there is very little to go on.

However, the fact that the Zoiper hasn’t ACKed the 401 doesn’t seem consistent with the call failing on answer, as it looks like there is no way for Asterisk to tell the phone what is happening.

I’d assume you have a routing problem.