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.<br>
<br>
Anyhow, will try defining one unique context for this provider PeopleCall.<br>Thanks.<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;">
Hi,<br><br>I think i found your error. you are missing a context for your peer<br>PeopleCall , this way no context for incoming calls!<br><br>Am I wrong?<br><br>Hope it helps,<br>Marco Mouta<br><br>On 7/21/06, Jose Limeres <
<a href="mailto:jlimeres@gmail.com">jlimeres@gmail.com</a>> wrote:<br>> Here is my SIP.conf. (just replaced psswds with *)<br>> Thanks.<br>><br>> [general]<br>><br>> port = 5060<br>> bindaddr = <a href="http://0.0.0.0">
0.0.0.0</a><br>> disallow=all<br>> allow=ulaw<br>> allow=alaw<br>><br>><br>> context = from-sip-external<br>> callerid = Unknown<br>> tos=0x68<br>><br>> register=<a href="http://34700758288001:********@sip.peoplecall.com/34700758288001">
34700758288001:********@sip.peoplecall.com/34700758288001</a><br>><br>> externip=<a href="http://boratelecom.dyndns.org">boratelecom.dyndns.org</a><br>> localnet=<a href="http://192.168.1.0/255.255.255.0">192.168.1.0/255.255.255.0
</a><br>><br>> [01]<br>> username=01<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>> record_in=Adhoc<br>> qualify=yes<br>> port=5060<br>> nat=always<br>> mailbox=01@device<br>
> host=dynamic<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=01 <01><br>><br>> [199]<br>> username=199<br>> type=friend<br>> secret=****<br>> record_out=Adhoc
<br>> record_in=Adhoc<br>> qualify=no<br>> port=5061<br>> nat=never<br>> mailbox=199@device<br>> host=dynamic<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=199 <199>
<br>><br>> [501]<br>> username=501<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>> record_in=Adhoc<br>> qualify=yes<br>> port=5060<br>> nat=always<br>> mailbox=501@device<br>> host=dynamic
<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=501 <501><br>><br>> [502]<br>> username=502<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>
> record_in=Adhoc<br>> qualify=yes<br>> port=5060<br>> nat=always<br>> mailbox=502@device<br>> host=dynamic<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=502 <502>
<br>><br>> [503]<br>> username=503<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>> record_in=Adhoc<br>> qualify=yes<br>> port=5060<br>> nat=always<br>> mailbox=503@device<br>> host=dynamic
<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=503 <503><br>><br>> [504]<br>> username=504<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>
> record_in=Adhoc<br>> qualify=yes<br>> port=5060<br>> nat=always<br>> mailbox=504@device<br>> host=dynamic<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=504 <504>
<br>><br>> [99]<br>> username=99<br>> type=friend<br>> secret=****<br>> record_out=Adhoc<br>> record_in=Adhoc<br>> qualify=no<br>> port=5062<br>> nat=never<br>> mailbox=99@device<br>> host=dynamic
<br>> dtmfmode=rfc2833<br>> context=from-internal<br>> canreinvite=no<br>> callerid=PSTN incoming <99><br>><br>> [Peoplecall]<br>> username=34700758288001<br>> type=peer<br>> secret=****<br>
> qualify=yes<br>> nat=yes<br>> host=<a href="http://sip.peoplecall.com">sip.peoplecall.com</a><br>> fromuser=34700758288001<br>> fromdomain=<a href="http://sip.peoplecall.com">sip.peoplecall.com</a><br>> dtmfmode=rfc2833
<br>> disallow=all<br>> allow=g729<br>><br>You need a context for incoming calls from Peoplecall !<br>context=from-PeopleCall ; just as an example and write your dialplan<br>for this context in extensions.conf<br>
> [PSTN]<br>> username=asterisk<br>> type=peer<br>> secret=****<br>> port=5061<br>> insecure=very<br>> host=<a href="http://192.168.1.106">192.168.1.106</a><br>> fromuser=asterisk<br>> dtmfmode=rfc2833
<br>> context=from-internal<br>> auth=md5<br>><br>><br>><br>><br>><br>><br>> On 21/07/06, Marco Mouta <<a href="mailto:marco.mouta@gmail.com">marco.mouta@gmail.com</a>> wrote:<br>> > 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
<br>> *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<br>> 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<br>> 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<br>> SIP/2.0<br>> > > > > Record-Route: <sip:<br>> > > > ><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>> > ><br>> <a href="http://62.22.20.207:5060">
62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff<br>> > > > ><br>> > > > > From:<br>> > > > ><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:<br>> 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>> > ><br>> <a href="http://62.22.20.207:5060">
62.22.20.207:5060</a>;branch=z9hG4bK-10ff6000ff05ff10ff00000e0c4effff<br>> > > > > From: <<br>> > > > ><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<br>> 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>> > ><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>> > ><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>> > > > > ",<br>> nonce="44c0059db2d71f523aeb30399a54a4a32d8aeed6",<br>> > > > > response="ee782a37bae7eed1a0a881147c733ede",
<br>> 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>> > > > ><br>> <a href="http://192.168.1.104:5060">192.168.1.104:5060</a>;branch=z9hG4bK4a6abe4f;rport=5060<br>> > > > > From: <<br>> > > > >
<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>> > > > > <<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<br>> 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>> > ><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>> > ><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>> > ><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>