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

Landy Landy landysaccount at yahoo.com
Thu Nov 19 15:36:38 CST 2009


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
> > should try that and see if it works.
> > 
> > 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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