[asterisk-users] need help debug asterisk-1.6 sip connection

Sun Nov 1 00:10:54 CDT 2009

I have a DID but for some reason is not working in asterisk-1.6
The same sip connection in asterisk-1.4 is working OK, but it doesn't work with asterisk-1.6

Here is my sip.conf section:
...
[actio-out]
type=friend
secret=password
user=48746612254
username=48746612254
fromuser=48746612254
authname=48746612254
callerpage=48746612254
fromdomain=sip.actio.pl
host=sip.actio.pl
insecure=very
nat=yes
qualify=yes
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
context=internal
canreinvite=no

Here is relevant section from asterisk-1.6 (failed connection) and asterisk-1.4 (working connection)

========== start asterisk-1.6 (not working) ==================

<------------->
--- (17 headers 18 lines) ---
   == Using SIP RTP CoS mark 5
Sending to 81.15.150.20 : 5060 (no NAT)
Using INVITE request as basis request - FFC94F46-C5D211DE-9310E4A5-81FB2A2A at 82.177.2.12~1o
Found peer 'actio-out' for '17804791270' from 81.15.150.20:5060

<--- Reliably Transmitting (NAT) to 81.15.150.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0;received=81.15.150.20
Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKba07785b184a5f79266bde33dccc8212;rport=5061
From: <sip:17804791270 at 81.15.150.20>;tag=26a9eb26114a01c9f4d1f64b72cc1d9e
To: <sip:48746612254 at 81.15.150.20>;tag=as52ab0bbb
Call-ID: FFC94F46-C5D211DE-9310E4A5-81FB2A2A at 82.177.2.12~1o
CSeq: 200 INVITE
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0da18b05"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'FFC94F46-C5D211DE-9310E4A5-81FB2A2A at 82.177.2.12~1o' in 13632 ms (Method: INVITE)
syscon2*CLI>
<--- SIP read from UDP://81.15.150.20:5060 --->
ACK sip:s at 68.148.245.78:61454 SIP/2.0
Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK4c28.d397a70c58c5c983c7d85bb171d8e3b2.0
From: <sip:17804791270 at 81.15.150.20>;tag=26a9eb26114a01c9f4d1f64b72cc1d9e
Call-ID: FFC94F46-C5D211DE-9310E4A5-81FB2A2A at 82.177.2.12~1o
To: <sip:48746612254 at 81.15.150.20>;tag=as52ab0bbb
CSeq: 200 ACK
User-Agent: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
syscon2*CLI>
<--- SIP read from UDP://81.15.150.20:5060 --->

================= end asterisk-1.6 (not working) =====================

========== start asterisk-1.4 (working) ==================
<------------->
--- (17 headers 18 lines) ---
Sending to 81.15.150.20 : 5060 (no NAT)
Using INVITE request as basis request - F203CDEF-C5D411DE-932AE4A5-81FB2A2A at 82.177.2.12~1o
Found peer 'actio-out'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 2
Found RTP audio format 100
Peer audio RTP is at port 81.15.150.20:46648
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G723 for ID 4
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 99
Found audio description format G726-32 for ID 2
Found unknown media description format X-NSE for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 81.15.150.20:46648
Looking for s in from_poland (domain 68.148.245.78)
list_route: hop: <sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr>
syscon4*CLI>
<--- Transmitting (NAT) to 81.15.150.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20
Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061
Record-Route: <sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr>
From: <sip:17804791270 at 81.15.150.20>;tag=1f5a641fc6ffb42064d4123781f0e7bb
To: <sip:48746612254 at 81.15.150.20>
Call-ID: F203CDEF-C5D411DE-932AE4A5-81FB2A2A at 82.177.2.12~1o
CSeq: 200 INVITE
User-Agent: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s at 10.0.0.109>
Content-Length: 0

<------------>
     -- Executing [s at from_poland:1] Answer("SIP/48746612254-00789120", "") in new stack
Audio is at 10.0.0.109 port 13414
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP

<--- Reliably Transmitting (NAT) to 81.15.150.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.15.150.20;branch=z9hG4bK5978.8397fd91b29a224fb6158a2eb64d4489.0;received=81.15.150.20
Via: SIP/2.0/UDP 81.15.150.20:5061;branch=z9hG4bKa185fc54438defa99101bdc43db8e8c7;rport=5061
Record-Route: <sip:81.15.150.20;ftag=1f5a641fc6ffb42064d4123781f0e7bb;lr>
From: <sip:17804791270 at 81.15.150.20>;tag=1f5a641fc6ffb42064d4123781f0e7bb
To: <sip:48746612254 at 81.15.150.20>;tag=as31531c77
Call-ID: F203CDEF-C5D411DE-932AE4A5-81FB2A2A at 82.177.2.12~1o
CSeq: 200 INVITE
User-Agent: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:s at 10.0.0.109>
Content-Type: application/sdp
Content-Length: 202

v=0
o=root 2881 2881 IN IP4 10.0.0.109
s=session
c=IN IP4 10.0.0.109
t=0 0
m=audio 13414 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

========== end asterisk-1.4 (working) ==================

sip show peer is showing registration OK on both version, but 1.6 is not connecting IN to my asterisk.

-- 
Joseph