I have two sip accounts registered, one using the default port 5060 the other 5077.
SIP SHOW REGISTRY indicates I’m properly logged in for both.
Everything works as it is supposed to, but when sending bye using the ‘5077’ account, the command is refused by the other side…
[i]== Spawn extension (ivrnaam, s, 2) exited non-zero on 'SIP/213.75.12.218-0066c 9e0’
set_destination: Parsing sip:0844203851@213.75.12.218:5077 for address/port to send to
set_destination: set destination to 213.75.12.218, port 5077
Reliably Transmitting (no NAT) to 213.75.12.218:5077:
BYE sip:0486462691@10.37.204.56:5060 SIP/2.0
Via: SIP/2.0/UDP 81.207.117.81:5061;branch=z9hG4bK1aec6eff;rport
Route: sip:0486462691@10.37.204.56:5060;transport=udp;maddr=10.37.204.56
From: sip:0844203851@10.37.205.10;user=phone;tag=as625fde3c
To: sip:0486462691@213.75.12.218;tag=test_tag_0034585387
Contact: sip:0844203851@81.207.117.81:5061
Call-ID: 2042364511-943831570764511-11-5012065622
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
<-- SIP read from 213.75.12.218:5077:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 81.207.117.81:5061;branch=z9hG4bK1aec6eff;rport
From: sip:0844203851@10.37.205.10;user=phone;tag=as625fde3c
To: sip:0486462691@213.75.12.218;tag=test_tag_0034585387
Call-ID: 2042364511-943831570764511-11-5012065622
CSeq: 102 BYE
user-agent: Asterisk PBX
Content-Length: 0 [/i]
When the other side drops the call, the call is properly ended.
It’s the same provider for both accounts, one is working properly the other one isn’t…
Any suggestions? Or is this truly a bug?