[asterisk-users] Asterisknow b5 - trouble registering at voip provider

[Erick Perez]

Hi, there.
I have asterisknow beta 5 with the following data:
Ip 192.168.0.60
mask 255.255.255.0
gw 192.168.0.1

the router (a linksys) has port forwarded the port udp 5060 and from
16384 to 16482 udp-tcp from the internet to the asterisk machine.
the only protocol allowed is g729. Which work fine for the ip phones I
already have setup in the LAN.

My problem is trying to register to a voip provider.
in the asterisknow gui I provide:
protocol sip
register (checked)
host sf2.clarocom.net
username (my phone number)
password (assigned password)

While executing "sip show claro91"
asterisk*CLI> sip show peer claro91
asterisk*CLI>

 * Name       : claro91
 Secret       : <Set>
 MD5Secret    : <Not set>
 Context      : DID_
 Subscr.Cont. : <Not set>
 Language     :
 AMA flags    : Unknown
 Transfer mode: open
 CallingPres  : Presentation Allowed, Not Screened
 Callgroup    : 1
 Pickupgroup  : 1
 Mailbox      :
 VM Extension : asterisk
 LastMsgsSent : 32767/65535
 Call limit   : 0
 Dynamic      : No
 Callerid     : "" <2029191>
 MaxCallBR    : 384 kbps
 Expire       : -1
 Insecure     : no
 Nat          : RFC3581
 ACL          : No
 T38 pt UDPTL : No
 CanReinvite  : No
 PromiscRedir : No
 User=Phone   : No
 Video Support: No
 Trust RPID   : No
 Send RPID    : No
 Subscriptions: No
 Overlap dial : No
 DTMFmode     : auto
 LastMsg      : 0
 ToHost       : sf2.clarocom.net
 Addr->IP     : 200.105.69.132 Port 5060
 Defaddr->IP  : 0.0.0.0 Port 5060
 Def. Username: 2029191
 SIP Options  : (none)
 Codecs       : 0x80100 (g729|h263)
 Codec Order  : (g729:20)
 Auto-Framing:  No
 Status       : Unmonitored
 Useragent    :
 Reg. Contact :
asterisk*CLI>
asterisk*CLI>


and when i try to call with my lan phones to the "outside" via the
claro91 trunk, I get

asterisk*CLI>
   -- Executing [966944780 at numberplan-custom-1:1]
Macro("SIP/6000-0820e870", "trunkdial|SIP/claro91/66944780") in new
stack
   -- Executing [s at macro-trunkdial:1] Dial("SIP/6000-0820e870",
"SIP/claro91/66944780") in new stack
   -- Called claro91/66944780
[May 13 17:37:40] WARNING[5522]: chan_sip.c:11860
handle_response_invite: Received response: "Forbidden" from '"Erick
Perez" <sip:6000 at 201.221.250.10>;tag=as7eabcb2e'
   -- SIP/claro91-082127d8 is circuit-busy
 == Everyone is busy/congested at this time (1:0/1/0)
   -- Executing [s at macro-trunkdial:2] Goto("SIP/6000-0820e870",
"s-CONGESTION|1") in new stack
   -- Goto (macro-trunkdial,s-CONGESTION,1)
   -- Executing [s-CONGESTION at macro-trunkdial:1]
NoOp("SIP/6000-0820e870", "") in new stack
 == Auto fallthrough, channel 'SIP/6000-0820e870' status is 'CONGESTION'
asterisk*CLI>


If I switch from my asterisknow box to the linksys box (that has two
rj11 ports) then the registration is fine.

I would like some guidance as to how to properly format the
registration string for my provider.

thanks,



-- 
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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
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