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 [EMAIL PROTECTED]:1]
Macro("SIP/6000-0820e870", "trunkdial|SIP/claro91/66944780") in new
stack
-- Executing [EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=as7eabcb2e'
-- SIP/claro91-082127d8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:2] Goto("SIP/6000-0820e870",
"s-CONGESTION|1") in new stack
-- Goto (macro-trunkdial,s-CONGESTION,1)
-- Executing [EMAIL PROTECTED]: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,
--
------------------------------------------------------------
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
------------------------------------------------------------
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
