[asterisk-users] NAT problem

[Jose Limeres]

Hi,

I am experiencing a hard to solve problem with my VoIP provider. I can make
calls without any problem but I can not receive any. Actually, calls arive
to * but the phone just does not  ring. I believe must be a problem with NAT
but  I think I have a good config:
- Extensions have nat=always and qualify=yes
- Have introduced in sip.conf  Externip and localnet
- ADSL modem/router is redirected to my * server
- With sip debug I can see the call arrives
Am I misssing something that someone else can see?

Appreciate any hint. Thanks
====================================
ASTERISK VERSION:
Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q

SIP DEBUG CAPTURE

<-- SIP read from 62.22.20.194:5060:
INVITE sip:34700758288001 at 87.218.175.120:5060 SIP/2.0
Record-Route: <sip:62.22.20.194;ftag=08ff6000ff05ff10ff00000e0c4effff;lr>
Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0
Via: SIP/2.0/UDP
62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
From: <sip:690351498 at 62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
To: <sip:34700758288001 at 62.22.20.194:5060;user=phone>
Call-ID: d2c76000bf05c0108000000e0c4ef4b3 at siphit-1
CSeq: 1 INVITE
Contact: <sip:690351498 at 62.22.20.207;user=phone>
Max-Forwards:  9
User-Agent: MERA MSIP v.1.0.2
Cisco-Guid: 908093991-393679323-3151091529-1429652222
Content-Type: application/sdp
Content-Length:   216

v=0
o=- 1153435071 1153435071 IN IP4 62.22.20.207
s=-
c=IN IP4 62.22.20.207
t=0 0
m=audio 59320 RTP/AVP 18 4 101
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

--- (14 headers 10 lines)---
Using INVITE request as basis request -
d2c76000bf05c0108000000e0c4ef4b3 at siphit-1
Sending to 62.22.20.194 : 5060 (non-NAT)
Found peer 'Peoplecall'
Reliably Transmitting (NAT) to 62.22.20.194:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0;received=62.22.20.194
Via: SIP/2.0/UDP
62.22.20.207:5060;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
From: <sip:690351498 at 62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
To: <sip:34700758288001 at 62.22.20.194:5060;user=phone>;tag=as476d14de
Call-ID: d2c76000bf05c0108000000e0c4ef4b3 at siphit-1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:34700758288001 at 87.218.175.74>
Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0"
Content-Length: 0


---
Scheduling destruction of call
'd2c76000bf05c0108000000e0c4ef4b3 at siphit-1' in 15000 ms
asterisk1*CLI>
<-- SIP read from 62.22.20.194:5060:
ACK sip:34700758288001 at 87.218.175.120:5060 SIP/2.0
Via: SIP/2.0/UDP 62.22.20.194;branch=z9hG4bK90bf.b9c560e1.0
From: <sip:690351498 at 62.22.20.207;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
Call-ID: d2c76000bf05c0108000000e0c4ef4b3 at siphit-1
To: <sip:34700758288001 at 62.22.20.194:5060;user=phone>;tag=as476d14de
CSeq: 1 ACK
User-Agent: OpenSer (1.0.0 (i386/linux))
Content-Length: 0


--- (8 headers 0 lines)---
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 62.22.20.194:5060:
REGISTER sip:sip.peoplecall.com SIP/2.0
Via: SIP/2.0/UDP 87.218.175.74:5060;branch=z9hG4bK4a6abe4f;rport
From: <sip:34700758288001 at sip.peoplecall.com>;tag=as79fdfc26
To: <sip:34700758288001 at sip.peoplecall.com>
Call-ID: 1a0c3f3d4abe13fd462e52f7222cfc2e at 127.0.0.1
CSeq: 421 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="34700758288001",
realm="sip.peoplecall.com", algorithm=MD5,
uri="sip:sip.peoplecall.com",
nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6",
response="ee782a37bae7eed1a0a881147c733ede", opaque=""
Expires: 120
Contact: <sip:34700758288001 at 87.218.175.74>
Event: registration
Content-Length: 0


---
asterisk1*CLI>
<-- SIP read from 62.22.20.194:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.104:5060;branch=z9hG4bK4a6abe4f;rport=5060
From: <sip:34700758288001 at sip.peoplecall.com>;tag=as79fdfc26
To: <sip:34700758288001 at sip.peoplecall.com>;tag=555271b30cfd40f8a3b4837b054360a3.975d
Call-ID: 1a0c3f3d4abe13fd462e52f7222cfc2e at 127.0.0.1
CSeq: 421 REGISTER
Contact: <sip:34700758288001 at 192.168.1.104:5060>;expires=120
Server: OpenSer (1.0.0 (i386/linux))
Content-Length: 0


--- (9 headers 0 lines)---
Scheduling destruction of call
'1a0c3f3d4abe13fd462e52f7222cfc2e at 127.0.0.1' in 32000 ms
Destroying call 'd2c76000bf05c0108000000e0c4ef4b3 at siphit-1'
asterisk1*CLI> sip no debug
SIP Debugging Disabled
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