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 <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]/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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> wrote: > Could you post your sip.conf? > > On 7/21/06, Jose Limeres <[EMAIL PROTECTED]> wrote: > > Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box. > > > > > > > > On 21/07/06, Marco Mouta <[EMAIL PROTECTED]> wrote: > > > Did you port forwar in your router RTP ports ? 10000-20000 to your *Box ? > > > > > > On 7/21/06, Jose Limeres <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]: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:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff > > > > To: < > > > > sip:[EMAIL PROTECTED]:5060;user=phone> > > > > Call-ID: [EMAIL PROTECTED] > > > > CSeq: 1 INVITE > > > > Contact: < > > > > sip:[EMAIL PROTECTED];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 - > > > > [EMAIL PROTECTED] > > > > 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:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff > > > > To: < sip:[EMAIL PROTECTED] > > > > :5060;user=phone>;tag=as476d14de > > > > Call-ID: [EMAIL PROTECTED] > > > > CSeq: 1 INVITE > > > > User-Agent: Asterisk PBX > > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > > > Contact: < > > > > sip:[EMAIL PROTECTED] > > > > > Proxy-Authenticate: Digest realm="asterisk", nonce="008d23b0" > > > > > > > > Content-Length: 0 > > > > > > > > > > > > --- > > > > Scheduling destruction of call > > > > '[EMAIL PROTECTED] ' in 15000 > > ms > > > > asterisk1*CLI> > > > > <-- SIP read from > > > > 62.22.20.194:5060: > > > > ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 > > > > Via: SIP/2.0/UDP 62.22.20.194;branch= > > > > z9hG4bK90bf.b9c560e1.0 > > > > From: > > > > > > <sip:[EMAIL PROTECTED];user=phone>;tag=08ff6000ff05ff10ff00000e0c4effff > > > > > > > > Call-ID: [EMAIL PROTECTED] > > > > To: > > > > > > <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED] > > > > >;tag=as79fdfc26 > > > > To: <sip:[EMAIL PROTECTED]> > > > > Call-ID: > > > > [EMAIL PROTECTED] > > > > 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:[EMAIL PROTECTED]> > > > > 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:[EMAIL PROTECTED]>;tag=as79fdfc26 > > > > To: < sip:[EMAIL PROTECTED] > > > > >;tag=555271b30cfd40f8a3b4837b054360a3.975d > > > > Call-ID: [EMAIL PROTECTED] > > > > > > > > CSeq: 421 REGISTER > > > > Contact: > > > > < sip:[EMAIL PROTECTED]:5060>;expires=120 > > > > Server: OpenSer (1.0.0 (i386/linux)) > > > > Content-Length: 0 > > > > > > > > > > > > --- (9 headers 0 lines)--- > > > > Scheduling destruction of call ' > > > > [EMAIL PROTECTED]' in 32000 ms > > > > Destroying call > > '[EMAIL PROTECTED] > > > > ' > > > > 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 > > > > > > > > > _______________________________________________ > > --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 > _______________________________________________ --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
