[asterisk-users] Problem with NAT

[Marco Mouta]

Did you port forwar in your router  RTP ports ? 10000-20000 to your *Box ?

On 7/21/06, Jose Limeres <jlimeres at gmail.com> wrote:
> 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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-- 
Com os melhores cumprimentos,

Marco Mouta