[asterisk-users] Problem with NAT

[Jose Limeres]

Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.


On 21/07/06, Marco Mouta <marco.mouta at gmail.com> wrote:
>
> 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
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
> --
> Com os melhores cumprimentos,
>
> Marco Mouta
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060721/d3ef80bb/attachment.htm