Hi,
I have a defined trunk on my VitalPBX with which I can make calls, but incoming calls give an error “The person trying to reach currently unavailable”.
This is the log that shows when I call from another number :
<— SIP read from UDP:204.11.192.23:5060 —>
INVITE sip:s@192.168.1.250:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.23:5060;branch=z9hG4bK-b69a6ddea39e8f59ef05aa7e4247e132
f: <sip:15554443333@66.193.176.35>;tag=3767460349-566550
t: <sip:19998887777@ss.callcentric.com>
i: 16032729-3767460349-566515@msw1.telengy.net
CSeq: 1 INVITE
Max-Forwards: 13
m: <sip:ca41cd8a1a64adfbca2002e8dbb1c10a@204.11.192.23:5060;transport=udp>
c: application/sdp
l: 316
v=0
o=NexTone-MSW 772718 873874 IN IP4 204.11.192.23
s=sip call
c=IN IP4 204.11.192.23
t=0 0
m=audio 49836 RTP/AVP 0 18 101
a=ptime:20
a=sendrecv
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off – – – –
a=setup:actpass
<————->
— (10 headers 15 lines) —
Sending to 204.11.192.23:5060 (NAT)
Sending to 204.11.192.23:5060 (NAT)
Using INVITE request as basis request – 16032729-3767460349-566515@msw1.telengy.net
Found peer ‘callcentric2’ for ‘15554443333’ from 204.11.192.23:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Capabilities: us – (ulaw), peer – audio=(ulaw|g729)/video=(nothing)/text=(nothing), combined – (ulaw)
Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1 (telephone-event|), combined – 0x1 (telephone-event|)
> 0x7fcd20080e60 — Strict RTP learning after remote address set to: 204.11.192.23:49836
Peer audio RTP is at port 204.11.192.23:49836
Looking for s in trk-3-in (domain 192.168.999.250)
<— Reliably Transmitting (NAT) to 204.11.192.23:5060 —>
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 204.11.192.23:5060;branch=z9hG4bK-b69a6ddea39e8f59ef05aa7e4247e132;received=204.11.192.23;rport=5060
From: <sip:15554443333@66.193.176.35>;tag=3767460349-566550
To: <sip:19998887777@ss.callcentric.com>;tag=as4f68163d
Call-ID: 16032729-3767460349-566515@msw1.telengy.net
CSeq: 1 INVITE
Server: VitalPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<————>
Scheduling destruction of SIP dialog ‘16032729-3767460349-566515@msw1.telengy.net’ in 32000 ms (Method: INVITE)
<— SIP read from UDP:204.11.192.23:5060 —>
ACK sip:s@192.168.999.250:5060 SIP/2.0
v: SIP/2.0/UDP 204.11.192.23:5060;branch=z9hG4bK-b69a6ddea39e8f59ef05aa7e4247e132
f: <sip:15554443333@66.193.176.35>;tag=3767460349-566550
t: <sip:19998887777@ss.callcentric.com>;tag=as4f68163d
i: 16032729-3767460349-566515@msw1.telengy.net
CSeq: 1 ACK
Max-Forwards: 15
l: 0
<————->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘16032729-3767460349-566515@msw1.telengy.net’ Method: ACK
Really destroying SIP dialog ‘7e08ba3f404e51d331eb8493215e8466@callcentric.com’ Method: BYE
<— SIP read from UDP:204.11.192.23:5060 —>
ACK sip:19998887777@alpha2.callcentric.com:5070 SIP/2.0
v: SIP/2.0/UDP 204.11.192.23:5060;branch=z9hG4bK-2866057de6c225b9de3092e73df8d6bc
f: <sip:15554443333@66.193.176.35>;tag=3767460349-566550
t: <sip:19998887777@ss.callcentric.com>;tag=as4f68163d
i: 16032729-3767460349-566515@msw1.telengy.net
CSeq: 1 ACK
Max-Forwards: 13
l: 0
<————->
vitalpbx*CLI> sip set debug off
SIP Debugging Disabled
vitalpbx*CLI>
Any idea?
Thanks
