[asterisk-users] can't call through voip provider

Fri Nov 27 10:21:56 CST 2009

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 <info at meetmecall.nl> wrote:

> From: meetmecall <info at meetmecall.nl>
> Subject: Re: [asterisk-users] can't call through voip provider
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com>
> 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 <landysaccount at yahoo.com>
> wrote:
> >
> >> From: Landy Landy <landysaccount at yahoo.com>
> >> Subject: Re: [asterisk-users] can't call through
> voip provider
> >> To: "Asterisk Users Mailing List - Non-Commercial
> Discussion" <asterisk-users at lists.digium.com
> 
> >> >
> >> 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:18292574075 at optimumwireless.myvnc.com
> >> SIP/2.0
> >> Via: SIP/2.0/UDP
> >> 190.80.153.193:5060;branch=z9hG4bK727987ef
> >> Max-Forwards: 70
> >> From: "102"
> >> <sip:77000 at 190.80.153.193>;tag=as23e02274
> >> To: <sip:18292574000 at optimumwireless.myvnc.com>
> >> Contact: <sip:77000 at 190.80.153.193>
> >> Call-ID:
> 034bf0572cffb96f621211a8439aa9d7 at 190.80.153.193
> >> 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.
> >>
> >>
> >>
> >>
> >> _______________________________________________
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> >>
> >
> >
> >
> >
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> 
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