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

[Landy Landy]

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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