[asterisk-users] configure my voip provider

[Baha @ SH]

Hi

I set the debug to 15, and changed to peer, I got this:

============================================================================
============================================

<--- SIP read from UDP:94.77.210.xxx:5060 --->
SIP/2.0 500 account has been moved to a remote system
Via: SIP/2.0/UDP 78.93.40.xxx:5060;branch=z9hG4bK637acf57
From: "8111488569" <sip:8111488569 at 78.93.40.xxx>;tag=as26c20707
To: <sip:0505103250 at 94.77.210.xxx>;tag=FF6EE3B3
Call-ID: 430bffa53f86bccf7566a3f9325a0497 at 78.93.40.xxx:5060
CSeq: 103 INVITE
Server: CommuniGatePro/5.3.12d
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
    -- Got SIP response 500 "account has been moved to a remote system" back
from 94.77.210.xxx:5060
Transmitting (no NAT) to 94.77.210.xxx:5060:
ACK sip:0505103250 at 94.77.210.xxx SIP/2.0
Via: SIP/2.0/UDP 78.93.40.xxx:5060;branch=z9hG4bK637acf57
Max-Forwards: 70
From: "8111488569" <sip:8111488569 at 78.93.40.xxx>;tag=as26c20707
To: <sip:0505103250 at 94.77.210.xxx>;tag=FF6EE3B3
Contact: <sip:8111488569 at 78.93.40.xxx:5060>
Call-ID: 430bffa53f86bccf7566a3f9325a0497 at 78.93.40.xxx:5060
CSeq: 103 ACK
User-Agent: FPBX-2.8.1(1.8.7.0)
Content-Length: 0

============================================================================
============================================

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Monday, March 05, 2012 4:51 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] configure my voip provider

Le 05/03/2012 07:37, Baha @ SH a écrit :
>
> Hello all,
>
> I am trying to configure my voip provider on asterisk box, I cannot 
> get the right configuration after trying many possibilities, always 
> getting circuits busy ,message, please let me know the meaning of the 
> debug message.
>
> And by the way, when I use a normal sip phone, I can dial normally, no 
> problems.
>
> First my config:
>

What is the [...] part?

> host=xx.xx.xx.xx
>
> username=8111488569
>
> secret=abcd
>
> fromuser=8111488569
>
> type=user
>

type=peer as you call them.

> allow=ulaw&alaw&gsm&g726
>
> canreinvite=no
>
> dtmfmode=inband
>
> qualify=1000
>
> insecure=very
>
> I also have registeration string:
>
> 8111488569:abcd at xx.xx.xx.xx/8111488569
>
> And my status on asterisk is registered.
>
> My debug:
>
> Mar 5 04:42:04
>
> 	
>
> VERBOSE
>
> 	
>
> [20042] pbx.c:
>
> 	
>
> -- Executing [s at macro-dialout-trunk:19] Dial("SIP/1001-00000004", 
> "SIP/8111488569/0505103250,300,") in new stack
>
> Mar 5 04:42:04
>
> 	
>
> VERBOSE
>
> 	
>
> [20042] netsock2.c:
>
> 	
>
> == Using SIP RTP TOS bits 184
>
> Mar 5 04:42:04
>
> 	
>
> VERBOSE
>
> 	
>
> [20042] netsock2.c:
>
> 	
>
> == Using SIP RTP CoS mark 5
>
> Mar 5 04:42:04
>
> 	
>
> ERROR
>
> 	
>
> [20042] netsock2.c:
>
> 	
>
> getaddrinfo("8111488569", "(null)", ...): Name or service not known
>
> Mar 5 04:42:04
>
> 	
>
> WARNING
>
> 	
>
> [20042] chan_sip.c:
>
> 	
>
> No such host: 8111488569
>

You call your account number, not your provider IP

Regards

-- 
Daniel

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