Erick Perez
2007-May-13 15:39 UTC
[asterisk-users] Asterisknow b5 - trouble registering at voip provider
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@numberplan-custom-1:1] Macro("SIP/6000-0820e870", "trunkdial|SIP/claro91/66944780") in new stack -- Executing [s@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@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@macro-trunkdial:2] Goto("SIP/6000-0820e870", "s-CONGESTION|1") in new stack -- Goto (macro-trunkdial,s-CONGESTION,1) -- Executing [s-CONGESTION@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, -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------