[asterisk-users] Problem with NAT
Jose Limeres
jlimeres at gmail.com
Fri Jul 21 04:31:53 MST 2006
Here is my SIP.conf. (just replaced psswds with *)
Thanks.
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external
callerid = Unknown
tos=0x68
register=34700758288001:********@sip.peoplecall.com/34700758288001
externip=boratelecom.dyndns.org
localnet=192.168.1.0/255.255.255.0
[01]
username=01
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=01 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=01 <01>
[199]
username=199
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5061
nat=never
mailbox=199 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=199 <199>
[501]
username=501
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=501 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=501 <501>
[502]
username=502
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=502 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=502 <502>
[503]
username=503
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=503 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=503 <503>
[504]
username=504
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=504 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=504 <504>
[99]
username=99
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5062
nat=never
mailbox=99 at device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=PSTN incoming <99>
[Peoplecall]
username=34700758288001
type=peer
secret=****
qualify=yes
nat=yes
host=sip.peoplecall.com
fromuser=34700758288001
fromdomain=sip.peoplecall.com
dtmfmode=rfc2833
disallow=all
allow=g729
[PSTN]
username=asterisk
type=peer
secret=****
port=5061
insecure=very
host=192.168.1.106
fromuser=asterisk
dtmfmode=rfc2833
context=from-internal
auth=md5
On 21/07/06, Marco Mouta <marco.mouta at gmail.com> wrote:
>
> Could you post your sip.conf?
>
> On 7/21/06, Jose Limeres <jlimeres at gmail.com> wrote:
> > 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
> > > >
> > > > _______________________________________________
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> > > >
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> > > >
> > > >
> > > >
> > >
> > >
> > > --
> > > Com os melhores cumprimentos,
> > >
> > > Marco Mouta
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
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> > >
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> >
> >
> > _______________________________________________
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> >
> >
>
>
> --
> Com os melhores cumprimentos,
>
> Marco Mouta
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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