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

Fri Nov 27 08:51:08 CST 2009

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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