Erik. I already solved this problem and posted it.
I was reloading all the setting but, it wasn't changing the provider's ip info. After doing a restart now everything worked. Thanks any ways for your help. --- On Fri, 11/27/09, meetmecall <[email protected]> wrote: > From: meetmecall <[email protected]> > Subject: Re: [asterisk-users] can't call through voip provider > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Friday, November 27, 2009, 9:51 AM > It is not that easy to give the > answer. There are lots of itsp typical > ways of registration and you haven't provide the info > needed to help > you out. > > You need a register line in the general part of sip.conf. > It should > look something like (mine looks like this > > register => > <DID>:<SECRET>:<username>@ipness.net:6060 > > > And you need a sip entry in sip.conf. For me it looks > something like > > [<DID>] > type=friend > host=ipness.net > fromuser=<DID> > fromdomain=ipness.net > username=<username> > secret=<secret> > insecure=very > context=inbound > port=6060 > qualify=2000 > canreinvite=no > disallow=all > ;allow=ulaw > allow=alaw > > But your provider might need other settings. So ask your > provider. > > If you are on public IP and not behind NAT you should use > nat=no From > the sip message I make up that the > > You didn't provide debug info but copied and paste a sip > message. > > If you would like people to help you, you have to provide > proper info. > CLI output, sip.conf (without passwords and IP adress info) > and the > sip messages will be helpful. Are you aware of the > fact that you need > to open UDP ports and not TCP. > > Your provider should be able to tell you how to configure > such an > account on an asterisk box, or at least help you to figure > it out. A > serious ITSP must have customers using Asterisk. If you > have no idea > what you are doing my advice is to start reading Asterisk: > "The future > of telephony", freely available on http://www.asteriskdocs.org/ . > > VERY SERIOUS WARNING: Don't put the credentials of a sip > account in a > mail to a mailing list. People might use your account to > call satelite > lines for EUR 7,50 per minute. This kind of mistakes might > bankcrupt > you :-( > > I hope this helps. > > Erik > > > On 19 nov 2009, at 22:36, Landy Landy wrote: > > > Can someone please share with me a sip configuration > to connect an > > asterisk server to a voip provider since my > configuration isn't > > working for me. > > > > thanks. > > > > --- On Thu, 11/19/09, Landy Landy <[email protected]> > wrote: > > > >> From: Landy Landy <[email protected]> > >> Subject: Re: [asterisk-users] can't call through > voip provider > >> To: "Asterisk Users Mailing List - Non-Commercial > Discussion" <[email protected] > > >> > > >> Date: Thursday, November 19, 2009, 7:51 AM > >> > >>> > >>> Ok. I do NOT have ports 10000-20000 opened in. > I guess > >> I > >>> > >>>> > >>> I will open ports 5060 - 5070 and 10000 - > 100100 and > >> do > >>> some test tonight. I will keep you posted. > >>> > >> > >> I ran this test and there was no difference. > >> > >> I still can't get through. > >> > >> --- > >> Retransmitting #5 (NAT) to 190.80.153.193:5060: > >> INVITE sip:[email protected] > >> SIP/2.0 > >> Via: SIP/2.0/UDP > >> 190.80.153.193:5060;branch=z9hG4bK727987ef > >> Max-Forwards: 70 > >> From: "102" > >> <sip:[email protected]>;tag=as23e02274 > >> To: <sip:[email protected]> > >> Contact: <sip:[email protected]> > >> Call-ID: > [email protected] > >> CSeq: 102 INVITE > >> User-Agent: Asterisk PBX 1.6.1.5 > >> Date: Thu, 19 Nov 2009 12:50:38 GMT > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, > >> NOTIFY, INFO > >> Supported: replaces, timer > >> Content-Type: application/sdp > >> Content-Length: 475 > >> > >> v=0 > >> o=root 752676658 752676658 IN IP4 190.80.153.193 > >> s=Asterisk PBX 1.6.1.5 > >> c=IN IP4 190.80.153.193 > >> t=0 0 > >> m=audio 10026 RTP/AVP 0 3 8 112 5 10 7 111 9 101 > >> a=rtpmap:0 PCMU/8000 > >> a=rtpmap:3 GSM/8000 > >> a=rtpmap:8 PCMA/8000 > >> a=rtpmap:112 AAL2-G726-32/8000 > >> a=rtpmap:5 DVI4/8000 > >> a=rtpmap:10 L16/8000 > >> a=rtpmap:7 LPC/8000 > >> a=rtpmap:111 G726-32/8000 > >> a=rtpmap:9 G722/8000 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> a=sendrecv > >> > >> > >> I don't know why I don't see my provider's ip > address. > >> Isn't supposed to show in this debug? > >> > >> Here's my sip.conf file again maybe you can catch > an error > >> or something I'm missing. > >> > >> [voipprovider] > >> type=peer > >> host=208.78.163.3 > >> username=77000 > >> fromuser=77000 > >> secret=77000 > >> port=5060 > >> dtmfmode=rfc2833 > >> nat=route > >> insucure=port,invite > >> allow=all > >> careinvite=yes > >> > >> Please helppppppp. > >> > >> > >> > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
