[asterisk-users] Problem with NAT

Fri Jul 21 06:17:55 MST 2006

Yes you may be right and I going to investigate it bit I thought that using
the context  from -sip-external was enough. Specially when I have defined in
extensions.conf that calls belonging to this contyext should be sent to the
extension I want to ring.

Anyhow, will try  defining one unique context for this provider PeopleCall.
Thanks.

On 21/07/06, Marco Mouta <marco.mouta at gmail.com> wrote:
>
> Hi,
>
> I think i found your error. you are missing a context for your peer
> PeopleCall , this way no context for incoming calls!
>
> Am I wrong?
>
> Hope it helps,
> Marco Mouta
>
> On 7/21/06, Jose Limeres <jlimeres at gmail.com> wrote:
> > 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
> >
> You need a context for incoming calls from Peoplecall !
> context=from-PeopleCall ; just as an example and write your dialplan
> for this context in extensions.conf
> > [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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> > > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > > >
> > > > > >
> > > > > >
> > > > >
> > > > >
> > > > > --
> > > > > Com os melhores cumprimentos,
> > > > >
> > > > > Marco Mouta
> > > > > _______________________________________________
> > > > > --Bandwidth and Colocation provided by Easynews.com --
> > > > >
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> > > >
> > >
> > >
> > > --
> > > Com os melhores cumprimentos,
> > >
> > > Marco Mouta
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
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> >
>
>
> --
> Com os melhores cumprimentos,
>
> Marco Mouta
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
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