In a 1.2 release of asterisk, I've had no problem connecting to a Broadvoice
SIP peer, to allow routing outgoing calls from Asterisk to Broadvoice. Now,
with the same SIP configuration, I cannot establish the peer. I've enclosed a
SIP log in the hope that someone can help me analyze this failure. I'd guess
the issue is NAT related and wondering if someone can spot a problem in the
logs, below.
Some details to help read this log (I've changed these numbers for privacy
purposes):
. My Asterisk server is behind a firewall. It's internal address is
192.168.71.1.
. My public IP address is 123.123.123.123
. I am calling 2125551212
. My Broadvoice phone number is 9145551212
Here is the log:
== Using SIP RTP CoS mark 5
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/siegeld-00e08e00",
"dial-sip,[EMAIL PROTECTED]
broadvoice.com") in new stack
-- Executing [EMAIL PROTECTED]:1] Dial("SIP/siegeld-00e08e00", "SIP/[EMAIL
PROTECTED]") i\
n new stack
== Using SIP RTP CoS mark 5
Audio is at 192.168.71.7 port 18596
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 123.123.123.123:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport
Max-Forwards: 70
From: "David Siegel" <sip:[EMAIL PROTECTED]>;tag=as57923ac4
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Date: Mon, 30 Jun 2008 05:13:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 256
v=0
o=root 593017814 593017814 IN IP4 192.168.71.7
s=Asterisk PBX 1.6.0-beta9
c=IN IP4 192.168.71.7
t=0 0
m=audio 18596 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called [EMAIL PROTECTED]
stsca1*CLI>
<--- SIP read from UDP://123.123.123.123:5060 --->
SIP/2.0 100 Trying
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: "David Siegel" <sip:[EMAIL PROTECTED]>;tag=as57923ac4
To: <sip:[EMAIL PROTECTED]>
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
stsca1*CLI>
<--- SIP read from UDP://123.123.123.123:5060 --->
SIP/2.0 403 Forbidden
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: "David Siegel" <sip:[EMAIL PROTECTED]>;tag=as57923ac4
To: <sip:[EMAIL PROTECTED]>;tag=lmno
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Type: application/sdp
Content-Length: 188
v=0
o=1213832004 593017814 593017814 IN IP4 192.168.71.7
s=-
c=IN IP4 192.168.71.7
t=0 0
m=audio 18596 RTP/AVP 0 3 8
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
<------------->
--- (9 headers 9 lines) ---
Transmitting (NAT) to 123.123.123.123:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.71.7:5060;branch=z9hG4bK2c01fcfd;rport
Max-Forwards: 70
From: "David Siegel" <sip:[EMAIL PROTECTED]>;tag=as57923ac4
To: <sip:[EMAIL PROTECTED]>;tag=lmno
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta9
Content-Length: 0
---
[Jun 30 01:13:51] WARNING[3023]: chan_sip.c:14738 handle_response_invite:
Received response: "Forbidden" f\
rom '"David Siegel" <sip:[EMAIL PROTECTED]>;tag=as57923ac4'
-- SIP/sip.broadvoice.com-00e0ddb0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Goto("SIP/siegeld-00e08e00",
"s-CONGESTION,1") in new stack
-- Goto (macro-dial-sip,s-CONGESTION,1)
-- Executing [EMAIL PROTECTED]:1] PlayTones("SIP/siegeld-00e08e00",
"congestion") in new st\
ack
-- Auto fallthrough, channel 'SIP/siegeld-00e08e00' status is 'CONGESTION'
Really destroying SIP dialog '[EMAIL PROTECTED]' Method: INVI
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