Here is my SIP.conf. (just replaced psswds with *)<br>
Thanks.<br>
<br>
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<pre>[general]
port = 5060
bindaddr = <a href="http://0.0.0.0">0.0.0.0</a>
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external
callerid = Unknown
tos=0x68
register=<a href="http://34700758288001:********@sip.peoplecall.com/34700758288001">34700758288001:********@sip.peoplecall.com/34700758288001</a>
externip=<a href="http://boratelecom.dyndns.org">boratelecom.dyndns.org</a>
localnet=<a href="http://192.168.1.0/255.255.255.0">192.168.1.0/255.255.255.0</a>
[01]
username=01
type=friend
secret=****
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
nat=always
mailbox=01@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@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@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@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@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@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@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=<a href="http://sip.peoplecall.com">sip.peoplecall.com</a>
fromuser=34700758288001
fromdomain=<a href="http://sip.peoplecall.com">sip.peoplecall.com</a>
dtmfmode=rfc2833
disallow=all
allow=g729
[PSTN]
username=asterisk
type=peer
secret=****
port=5061
insecure=very
host=<a href="http://192.168.1.106">192.168.1.106</a>
fromuser=asterisk
dtmfmode=rfc2833
context=from-internal
auth=md5
</pre><br>
<br>
<br><br><div><span class="gmail_quote">On 21/07/06, <b class="gmail_sendername">Marco Mouta</b> <<a href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Could you post your sip.conf?<br><br>On 7/21/06, Jose Limeres <<a href="mailto:jlimeres@gmail.com">jlimeres@gmail.com</a>> wrote:<br>> Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.<br>>
<br>><br>><br>> On 21/07/06, Marco Mouta <<a href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</a>> wrote:<br>> > Did you port forwar in your router RTP ports ? 10000-20000 to your *Box ?<br>
> ><br>> > On 7/21/06, Jose Limeres <<a href="mailto:jlimeres@gmail.com">jlimeres@gmail.com</a>> wrote:<br>> > > Hi,<br>> > ><br>> > > I am experiencing a hard to solve problem with my VoIP provider. I can
<br>> make<br>> > > calls without any problem but I can not receive any. Actually, calls<br>> arive<br>> > > to * but the phone just does not ring. I believe must be a problem with<br>> NAT<br>
> > > but I think I have a good config:<br>> > > - Extensions have nat=always and qualify=yes<br>> > > - Have introduced in sip.conf Externip and localnet<br>> > > - ADSL modem/router is redirected to my * server
<br>> > > - With sip debug I can see the call arrives<br>> > > Am I misssing something that someone else can see?<br>> > ><br>> > > Appreciate any hint. Thanks<br>> > > ==============================
<br>> > > ======<br>> > > ASTERISK VERSION:<br>> > > Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1q<br>> > ><br>> > > SIP DEBUG CAPTURE<br>> > > <-- SIP read from <a href="http://62.22.20.194:5060">
62.22.20.194:5060</a>:<br>> > > INVITE sip:34700758288001@87.218.175.120:5060 SIP/2.0<br>> > > Record-Route: <sip:<br>> > > <a href="http://62.22.20.194">62.22.20.194</a>;ftag=08ff6000ff05ff10ff00000e0c4effff;lr>
<br>> > > Via: SIP/2.0/UDP<br>> > > <a href="http://62.22.20.194">62.22.20.194</a>;branch=z9hG4bK90bf.b9c560e1.0<br>> > > Via: SIP/2.0/UDP<br>> > ><br>> <a href="http://62.22.20.207:5060">
62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff<br>> > ><br>> > > From:<br>> > ><br>> <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff
<br>> > > To: <<br>> > > sip:34700758288001@62.22.20.194:5060;user=phone><br>> > > Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> > > CSeq: 1 INVITE<br>> > > Contact: <
<br>> > > <a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone><br>> > > Max-Forwards: 9<br>> > > User-Agent: MERA MSIP v.1.0.2<br>> > > Cisco-Guid: 908093991-393679323-3151091529-1429652222
<br>> > > Content-Type: application/sdp<br>> > > Content-Length: 216<br>> > ><br>> > ><br>> > > v=0<br>> > > o=- 1153435071 1153435071 IN IP4 <a href="http://62.22.20.207">
62.22.20.207</a><br>> > > s=-<br>> > > c=IN IP4<br>> > > <a href="http://62.22.20.207">62.22.20.207</a><br>> > > t=0 0<br>> > > m=audio 59320 RTP/AVP 18 4 101<br>> > > a=rtpmap:18 G729/8000
<br>> > > a=rtpmap:4 G723/8000<br>> > > a=rtpmap:101 telephone-event/8000<br>> > > a=fmtp:101 0-15<br>> > ><br>> > > --- (14 headers 10 lines)---<br>> > > Using INVITE request as basis request -
<br>> > > d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> > > Sending to <a href="http://62.22.20.194">62.22.20.194</a> : 5060 (non-NAT)<br>> > > Found peer 'Peoplecall'<br>> > ><br>> > > Reliably Transmitting (NAT) to
<a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> > > SIP/2.0 407 Proxy Authentication Required<br>> > > Via: SIP/2.0/UDP<br>> > > <a href="http://62.22.20.194">62.22.20.194</a>;branch=
z9hG4bK90bf.b9c560e1.0;received=<br>> > > <a href="http://62.22.20.194">62.22.20.194</a><br>> > > Via: SIP/2.0/UDP<br>> > ><br>> <a href="http://62.22.20.207:5060">62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff
<br>> > > From: <<br>> > ><br>> <a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff<br>> > > To: <<a href="mailto:sip:34700758288001@62.22.20.194">
sip:34700758288001@62.22.20.194</a><br>> > > :5060;user=phone>;tag=as476d14de<br>> > > Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> > > CSeq: 1 INVITE<br>> > > User-Agent: Asterisk PBX
<br>> > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> > > Contact: <<br>> > > <a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74</a>
><br>> > > Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0"<br>> > ><br>> > > Content-Length: 0<br>> > ><br>> > ><br>> > > ---<br>
> > > Scheduling destruction of call<br>> > > 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1 ' in 15000<br>> ms<br>> > > asterisk1*CLI><br>> > > <-- SIP read from<br>> > >
<a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> > > ACK sip:34700758288001@87.218.175.120:5060 SIP/2.0<br>> > > Via: SIP/2.0/UDP <a href="http://62.22.20.194">62.22.20.194</a>;branch=<br>> > >
z9hG4bK90bf.b9c560e1.0<br>> > > From:<br>> > ><br>> <<a href="mailto:sip:690351498@62.22.20.207">sip:690351498@62.22.20.207</a>;user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff<br>> > >
<br>> > > Call-ID: d2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> > > To:<br>> > ><br>> <sip:34700758288001@62.22.20.194:5060;user=phone>;tag=as476d14de<br>> > > CSeq: 1 ACK<br>
> > > User-Agent: OpenSer (1.0.0 (i386/linux))<br>> > > Content-Length: 0<br>> > ><br>> > ><br>> > ><br>> > > --- (8 headers 0 lines)---<br>> > > REGISTER 13 headers, 0 lines
<br>> > > Reliably Transmitting (no NAT) to <a href="http://62.22.20.194:5060">62.22.20.194:5060</a><br>> > > :<br>> > > REGISTER sip: <a href="http://sip.peoplecall.com">sip.peoplecall.com</a> SIP/2.0
<br>> > > Via: SIP/2.0/UDP<br>> > > <a href="http://87.218.175.74:5060">87.218.175.74:5060</a>;branch=z9hG4bK4a6abe4f;rport<br>> > > From: < <a href="mailto:sip:34700758288001@sip.peoplecall.com">
sip:34700758288001@sip.peoplecall.com</a><br>> > > >;tag=as79fdfc26<br>> > > To: <<a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a>><br>> > > Call-ID:
<br>> > > <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a><br>> > > CSeq: 421 REGISTER<br>> > > User-Agent: Asterisk PBX<br>> > > Max-Forwards: 70
<br>> > > Authorization: Digest username="34700758288001", realm="<br>> > > <a href="http://sip.peoplecall.com">sip.peoplecall.com</a>", algorithm=MD5, uri="sip:<a href="http://sip.peoplecall.com">
sip.peoplecall.com</a><br>> > > ", nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6",<br>> > > response="ee782a37bae7eed1a0a881147c733ede", opaque=""<br>> > ><br>
> > > Expires: 120<br>> > > Contact: <<a href="mailto:sip:34700758288001@87.218.175.74">sip:34700758288001@87.218.175.74</a>><br>> > > Event: registration<br>> > ><br>> > > Content-Length: 0
<br>> > ><br>> > ><br>> > > ---<br>> > > asterisk1*CLI><br>> > > <-- SIP read from <a href="http://62.22.20.194:5060">62.22.20.194:5060</a>:<br>> > > SIP/2.0 200 OK
<br>> > ><br>> > > Via: SIP/2.0/UDP<br>> > > <a href="http://192.168.1.104:5060">192.168.1.104:5060</a>;branch=z9hG4bK4a6abe4f;rport=5060<br>> > > From: <<br>> > > <a href="mailto:sip:34700758288001@sip.peoplecall.com">
sip:34700758288001@sip.peoplecall.com</a>>;tag=as79fdfc26<br>> > > To: < <a href="mailto:sip:34700758288001@sip.peoplecall.com">sip:34700758288001@sip.peoplecall.com</a><br>> > > >;tag=555271b30cfd40f8a3b4837b054360a3.975d
<br>> > > Call-ID: <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a><br>> > ><br>> > > CSeq: 421 REGISTER<br>> > > Contact:<br>> > > <
sip:34700758288001@192.168.1.104:5060>;expires=120<br>> > > Server: OpenSer (1.0.0 (i386/linux))<br>> > > Content-Length: 0<br>> > ><br>> > ><br>> > > --- (9 headers 0 lines)---
<br>> > > Scheduling destruction of call '<br>> > > <a href="mailto:1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1">1a0c3f3d4abe13fd462e52f7222cfc2e@127.0.0.1</a>' in 32000 ms<br>> > > Destroying call
<br>> 'd2c76000bf05c0108000000e0c4ef4b3@siphit-1<br>> > > '<br>> > > asterisk1*CLI> sip no debug<br>> > > SIP Debugging Disabled<br>> > ><br>> > > _______________________________________________
<br>> > > --Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br>> > ><br>> > > asterisk-users mailing list<br>> > > To UNSUBSCRIBE or update options visit:
<br>> > ><br>> > ><br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>> > ><br>> > ><br>> > >
<br>> ><br>> ><br>> > --<br>> > Com os melhores cumprimentos,<br>> ><br>> > Marco Mouta<br>> > _______________________________________________<br>> > --Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br>> ><br>> > asterisk-users mailing list<br>> > To UNSUBSCRIBE or update options visit:<br>> ><br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">
http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>> ><br>><br>><br>> _______________________________________________<br>> --Bandwidth and Colocation provided by <a href="http://Easynews.com">
Easynews.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>><br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br>><br>><br>><br><br><br>--<br>Com os melhores cumprimentos,<br><br>Marco Mouta<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com
</a> --<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br>