[Asterisk-Users] SIP Changes???

Fri May 28 07:37:30 MST 2004

Hello Olle!

> Please add a SIP debug of the call so we can see what happens, who 
> refuses what call.

Situation: I'm behind an NAT firewall and get an incoming call from my 
SIP provider. I have the following entries in sip.conf:

register => 1838933:8mtbdz at sipgate.de/1838933

[sipgate.de]
type=user
context=in-sip
nat=1
language=de
disallow=all
allow=gsm
allow=alaw
allow=ulaw

Unforunately not using an URL as name for the section as recommended 
does not work. Registration with my provider will fail because no 
section can be found so I used this one where this failure does not appear:

   == Parsing '/etc/asterisk/sip.conf':   == Parsing 
'/etc/asterisk/sip.conf': Found
May 28 16:47:01 WARNING[1114610608]: chan_sip.c:2191 sip_register: Host 
'sipgate-in' not found at line 28

Here you are with my complete debugging information for an incoming call:

---------------------------------------------
INVITE sip:<MyNumber>@172.20.0.2 SIP/2.0
Max-Forwards: 20
Record-Route: <sip:<MyNumber>@217.10.79.9;ftag=40b74c2525f79;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.ccdaac71.0
To: <sip:<MyNumber>@sipgate.de>
From: <sip:<CallerID>@sipgate.de>;tag=40b74c2525f79
CSeq: 1 INVITE
Call-ID: 40b74c2525f79.fifouacctd
Content-Length: 155
User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux))
Contact: <sip:caller at 217.10.79.9:5060>
Content-Type: application/sdp
Sipgate-Authentication: accepted

v=0
o=click-to-dial 0 0 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
b=CT:1000
t=0 0
m=audio 40814 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=direction:active

14 headers, 9 lines
Using latest request as basis request
Sending to 217.10.79.9 : 5060 (non-NAT)
Found RTP audio format 0
Peer RTP is at port 0.0.0.0:0
Found description format PCMU
Capabilities: us - 0x40e(GSM|ULAW|ALAW|ILBC), peer - 
audio=0x4(ULAW)/video=0x0(EMPTY), combined - 0x4(ULAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 
0x0(EMPTY)
Found peer 'sipgate-out'
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0;received=217.10.79.9
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.ccdaac71.0
From: <sip:<CallerID>@sipgate.de>;tag=40b74c2525f79
To: <sip:<MyNumber>@sipgate.de>;tag=as0ce31626
Call-ID: 40b74c2525f79.fifouacctd
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:<MyNumber>@172.20.0.2>
Proxy-Authenticate: Digest realm="voyager.localserver.de", nonce="2cb0b193"
Content-Length: 0

  to 217.10.79.9:5060
Scheduling destruction of call '40b74c2525f79.fifouacctd' in 15000 ms
zion*CLI>

Sip read:
ACK sip:<MyNumber>@172.20.0.2 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKc5ab.dcdaac71.0
From: <sip:<CallerID>@sipgate.de>;tag=40b74c2525f79
Call-ID: 40b74c2525f79.fifouacctd
To: <sip:<MyNumber>@sipgate.de>;tag=as0ce31626
CSeq: 1 ACK
User-Agent: Sip EXpress router(0.8.12-tcp_nonb (i386/linux))
Content-Length: 0

8 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK1eb6d6a5
From: <sip:<MyNumber>@sipgate.de>;tag=as166899a7
To: <sip:<MyNumber>@sipgate.de>
Call-ID: 6d9b5d33669387922bdf14387c820bd1 at 127.0.0.2
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:<MyNumber>@172.20.0.2>
Event: registration
Content-Length: 0

  (no NAT) to 217.10.79.9:5060
zion*CLI>

Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK1eb6d6a5
From: <sip:<MyNumber>@sipgate.de>;tag=as166899a7
To: <sip:<MyNumber>@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.94c0
Call-ID: 6d9b5d33669387922bdf14387c820bd1 at 127.0.0.2
CSeq: 109 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", 
nonce="40b74d5fe384afdade9e26b4da34a52421ad4140"
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=15717 
req_src_ip=172.20.0.2 req_src_port=5060 in_uri=sip:sipgate.de 
out_uri=sip:sipgate.de via_cnt==1"

10 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK2a07a252
From: <sip:<MyNumber>@sipgate.de>;tag=as166899a7
To: <sip:<MyNumber>@sipgate.de>
Call-ID: 6d9b5d33669387922bdf14387c820bd1 at 127.0.0.2
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="<MyNumber>", realm="sipgate.de", 
algorithm="MD5", uri="sip:sipgate.de", 
nonce="40b74d5fe384afdade9e26b4da34a52421ad4140", 
response="17d50f31e37949b4dd8e65e91f6c5002", opaque=""
Expires: 120
Contact: <sip:<MyNumber>@172.20.0.2>
Event: registration
Content-Length: 0

  (no NAT) to 217.10.79.9:5060
zion*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.0.2:5060;branch=z9hG4bK2a07a252
From: <sip:<MyNumber>@sipgate.de>;tag=as166899a7
To: <sip:<MyNumber>@sipgate.de>;tag=b11cb9bb270104b49a99a995b8c68544.2876
Call-ID: 6d9b5d33669387922bdf14387c820bd1 at 127.0.0.2
CSeq: 110 REGISTER
Contact: <sip:<MyNumber>@172.20.0.2>;q=0.00;expires=120
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=15717 
req_src_ip=172.20.0.2 req_src_port=5060 in_uri=sip:sipgate.de 
out_uri=sip:sipgate.de via_cnt==1"

10 headers, 0 lines
Destroying call '6d9b5d33669387922bdf14387c820bd1 at 127.0.0.2'
Destroying call '40b74c2525f79.fifouacctd'
zion*CLI>

Sip read:

0 headers, 0 lines
zion*CLI>
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