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

[Landy Landy]

Sorry to bother you again with my problem but, is that I can't figure out what's going on with my setup. I have no idea of why my asterisk server is not communicating with my provider's. I've searched, googled, and can't find my solution. I've followed many tutorials but can't get anywhere.



--- 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, 5:53 PM
> Nothing. I don't know what in the
> world is going on with my setup.
> 
> Here's my FORWARD rules:
> eth0 = external nic, eth1 = lan
> 
>     0     0 ACCEPT 
>    udp  -- 
> eth0   eth1    0.0.0.0/0 
>           0.0.0.0/0   
>        udp dpts:5060:5070
>     0     0 ACCEPT 
>    udp  -- 
> eth0   eth1    0.0.0.0/0 
>           0.0.0.0/0   
>        udp dpts:10000:10100
>     1    62 ACCEPT 
>    udp  -- 
> eth1   eth0    0.0.0.0/0 
>           0.0.0.0/0   
>        udp dpts:5060:5070
>    36  2372 ACCEPT 
>    udp  -- 
> eth1   eth0    0.0.0.0/0 
>           0.0.0.0/0   
>        udp dpts:10000:10100
>     0     0 ACCEPT 
>    tcp  -- 
> eth0   eth1    0.0.0.0/0 
>           0.0.0.0/0   
>        tcp dpts:5060:5070
>     0     0 ACCEPT 
>    tcp  -- 
> eth0   eth1    0.0.0.0/0 
>           0.0.0.0/0   
>        tcp dpts:10000:10100
>     0     0 ACCEPT 
>    tcp  -- 
> eth1   eth0    0.0.0.0/0 
>           0.0.0.0/0   
>        tcp dpts:5060:5070
>     3   144 ACCEPT 
>    tcp  -- 
> eth1   eth0    0.0.0.0/0 
>           0.0.0.0/0   
>        tcp dpts:10000:10100
> 
> 
> and now the debug:
> 
> etransmitting #5 (NAT) to 190.80.152.200:5060:
> INVITE sip:18292574000 at optimumwireless.myvnc.com
> SIP/2.0
> Via: SIP/2.0/UDP
> 190.80.152.200:5060;branch=z9hG4bK794de7aa;rport
> Max-Forwards: 70
> From: "102"
> <sip:77000 at 190.80.152.200>;tag=as5084570c
> To: <sip:18292574000 at optimumwireless.myvnc.com>
> Contact: <sip:77000 at 190.80.152.200>
> Call-ID: 22569d3b767276276c6c65c84b314277 at 190.80.152.200
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.1.5
> Date: Thu, 19 Nov 2009 22:53:06 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 135722140 135722140 IN IP4 190.80.152.200
> s=Asterisk PBX 1.6.1.5
> c=IN IP4 190.80.152.200
> t=0 0
> m=audio 10074 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'm already frustrated with this.
> 
> 
> --- On Thu, 11/19/09, Warren Selby <wcselby at selbytech.com>
> wrote:
> 
> > From: Warren Selby <wcselby at selbytech.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, 5:11 PM
> > On Thu, Nov 19,
> > 2009 at 3:36 PM, Landy Landy <landysaccount at yahoo.com>
> > 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.
> > 
> > 
> > 
> > 
> > Who is your voipprovider?  Did they give you the
> settings
> > you're using in your sip.conf?  Also, you've got
> > some typos in your sip config (insucure = insecure,
> > careinvite = canreinvite).  You could try something
> like
> > this:
> > 
> > 
> > [voipprovider]
> > 
> > type=peer
> > 
> > host=208.78.163.3
> > 
> > username=77000
> > 
> > fromuser=77000
> > 
> > secret=77000
> > 
> > port=5060
> > 
> > dtmfmode=rfc2833
> > 
> > nat=yes
> > canreinvite=yes
> > 
> > insecure=very
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > 
> > 
> > 
> > 
> > 
> > -- 
> > Thanks,
> > --Warren Selby
> > http://www.selbytech.com
> > 
> > 
> > -----Inline Attachment Follows-----
> > 
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> 
>       
> 
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