Had another post about this, using Vonage sip trunks and Asterisk. Outbound works fine when nat is turned off in the sip,conf and inbound dies. As soon as I turn nat on in the sip.conf inbound works but outbound get congestion recordings.
With nat turned off and outbound calls are working great I see my inbound call hit asterisk in my sip debug, can anyone make sense of this and help me out.
sip debug
[code]<-- SIP read from 216.115.20.41:5061:
INVITE sip:15125354348@68.201.250.45:5060 SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5061
Via: SIP/2.0/UDP 216.115.30.26:5060
Via: SIP/2.0/UDP 208.49.157.18:5060;branch=z9hG4bK1BC6
Record-Route: sip:15125354348@216.115.20.41:5061
Record-Route: sip:15125354348@216.115.30.26:5060
From: “Cell Phone TX” sip:12817824574@208.49.157.18;tag=1283130081
To: sip:15125354348@inbound4.vonage.net
Call-ID: ACDD9234-DD2F11DA-8C3ECBFB-40BC89E2@208.49.157.18
CSeq: 101 INVITE
Contact: sip:12817824574@208.49.157.18:5060
Max-Forwards: 13
Content-Type: application/sdp
Content-Length: 361
v=0
o=CiscoSystemsSIP-GW-UserAgent 7930 1850 IN IP4 208.49.157.18
s=SIP Call
c=IN IP4 208.49.157.18
t=0 0
m=audio 17418 RTP/AVP 0 18 2 100 101
c=IN IP4 208.49.157.18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
— (14 headers 15 lines)—
Using INVITE request as basis request - ACDD9234-DD2F11DA-8C3ECBFB-40BC89E2@208.49.157.18
Sending to 216.115.20.41 : 5061 (non-NAT)
Found peer 'vonage_out_4351’
Reliably Transmitting (no NAT) to 216.115.20.41:5061:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 216.115.20.41:5061;received=216.115.20.41
Via: SIP/2.0/UDP 216.115.30.26:5060
Via: SIP/2.0/UDP 208.49.157.18:5060;branch=z9hG4bK1BC6
From: “Cell Phone TX” sip:12817824574@208.49.157.18;tag=1283130081
To: sip:15125354348@inbound4.vonage.net;tag=as3e81e086
Call-ID: ACDD9234-DD2F11DA-8C3ECBFB-40BC89E2@208.49.157.18
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:15125354348@68.201.250.45
Proxy-Authenticate: Digest realm=“asterisk”, nonce=“2aa91e83”
Content-Length: 0
Scheduling destruction of call ‘ACDD9234-DD2F11DA-8C3ECBFB-40BC89E2@208.49.157.18’ in 15000 ms
e[Kasterisk1*CLI> sip no debug
<-- SIP read from 216.115.20.41:5061:
ACK sip:15125354348@inbound4.vonage.net SIP/2.0
Via: SIP/2.0/UDP 216.115.20.41:5061
From: “Cell Phone TX” sip:12817824574@208.49.157.18;tag=1283130081
To: sip:15125354348@inbound4.vonage.net;tag=as3e81e086
Call-ID: ACDD9234-DD2F11DA-8C3ECBFB-40BC89E2@208.49.157.18
CSeq: 101 ACK
Content-Length: [/code]
Like I said before I am getting a recording saying that the number I am trying to reach is unavailable at this time and gives an error code of I 104.
Thanks so much,
Nick