Incoming calls getting recording I 104


#1

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