Hi All Complete newbie to Asterisk and FreePBX.
Have set up two systems of AsteriskNow and FreePbx 2.6 and on one system incoming call is working fine but on other system it is picking up the outgoing peer and then hanging up. If I remove the outgoing details the incoming call works fine. Both systems have the same Trunks, Inbound and Outbound Routs the only difference is the DID. They are both sitting behind a NAT
Below is the log file from the setup not working
<------------->
[Jan 12 03:48:19] VERBOSE[2207] logger.c: — (12 headers 15 lines) —
[Jan 12 03:48:19] VERBOSE[2207] logger.c: Sending to 202.86.49.35 : 5060 (NAT)
[Jan 12 03:48:19] VERBOSE[2207] logger.c: Using INVITE request as basis request - [email protected]
[Jan 12 03:48:19] VERBOSE[2207] logger.c: Found peer ‘Freshtel_Out’
[Jan 12 03:48:19] VERBOSE[2207] logger.c:
<— Reliably Transmitting (NAT) to 202.86.49.35:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 202.86.49.35:5060;branch=z9hG4bKomeg6v208g0gcfs9l6o0.1;received=202.86.49.35
From: “Brendan” sip:*99883***@sip.freshtel.net;tag=b2b.33cce61
To: “99017***” sip:**9901***[email protected];transport=UDP;tag=as62ba89c8
Call-ID: [email protected]
CSeq: 100 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="015b865b"
Content-Length: 0
And this from the working one
[Jan 12 03:11:43] VERBOSE[2525] logger.c: — (12 headers 15 lines) —
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Sending to 202.86.49.35 : 5076 (NAT)
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Using INVITE request as basis request - [email protected]
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found no matching peer or user for ‘202.86.49.35:5076’
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found RTP audio format 18
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found RTP audio format 101
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found RTP audio format 8
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found RTP audio format 0
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found audio description format G729 for ID 18
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found audio description format telephone-event for ID 101
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found audio description format PCMA for ID 8
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Found audio description format PCMU for ID 0
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Peer audio RTP is at port 202.86.49.35:56164
[Jan 12 03:11:43] VERBOSE[2525] logger.c: Looking for s in from-sip-external (domain 150.101.103.94)
[Jan 12 03:11:43] VERBOSE[2525] logger.c: list_route: hop: sip:*[email protected]:5060;transport=udp
[Jan 12 03:11:43] VERBOSE[2525] logger.c:
<— Transmitting (NAT) to 202.86.49.35:5076 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.86.49.35:5060;branch=z9hG4bKlk41s420581gigc6j201.1;received=202.86.49.35
From: “Brendan” sip:*99883***@sip.freshtel.net;tag=b2b.2c395e4d
To: “99017***” sip:**99017***@sip.freshtel.net;transport=UDP
Call-ID: [email protected]
CSeq: 100 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:[email protected]
Content-Length: 0
Any help would be much appreciated
Thanks
Brendan