Outgoing calls stop working after maintainance

[akoei]

Hi freephoneline, I got outgoing calls stopped working, after your maintainance:
"Failed to authenticate on INVITE to xxxxxxx"

My environment is asterisk 1.4, there is no any issue for incoming.
Attached is sip debug output, would you please take a look and point me out?

Thanks

Shawn
-- Executing [6473677222@default:1] Set("SIP/99-005e6a70", "CALLERID(all)=14164778887") in new stack
-- Executing [6473677222@default:2] Dial("SIP/99-005e6a70", "SIP/16473677222@fpl1|120") in new stack
Audio is at 209.197.184.137 port 10070
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 208.65.240.165:5060:
INVITE sip:16473677222@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 209.197.184.137:6631;branch=z9hG4bK2724c9a4;rport
From: "14164778887" <sip:14164778887@209.197.184.137:6631>;tag=as57d3b3a7
To: <sip:16473677222@voip.freephoneline.ca>
Contact: <sip:14164778887@209.197.184.137:6631>
Call-ID: 18d8b260749af6ae150dcceb0590e8eb@209.197.184.137
CSeq: 102 INVITE
User-Agent: Linksys/RT31P2-3.1.6(LI)
Max-Forwards: 70
Date: Thu, 28 Jun 2012 15:43:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 827 827 IN IP4 209.197.184.137
s=session
c=IN IP4 209.197.184.137
t=0 0
m=audio 10070 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 16473677222@fpl1
OpenWrt*CLI>
<--- SIP read from 208.65.240.165:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.197.184.137:6631;branch=z9hG4bK2724c9a4;rport=6631
To: <sip:16473677222@voip.freephoneline.ca>
From: "14164778887"<sip:14164778887@209.197.184.137:6631>;tag=as57d3b3a7
Call-ID: 18d8b260749af6ae150dcceb0590e8eb@209.197.184.137
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---
OpenWrt*CLI>
<--- SIP read from 208.65.240.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 209.197.184.137:6631;branch=z9hG4bK2724c9a4;rport=6631
Record-Route: <sip:208.65.240.165:5060;transport=udp;lr>
To: <sip:16473677222@voip.freephoneline.ca>
From: "14164778887"<sip:14164778887@209.197.184.137:6631>;tag=as57d3b3a7
Call-ID: 18d8b260749af6ae150dcceb0590e8eb@209.197.184.137
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="voip.freephoneline.ca",nonce="8c64d4b740d424f8d5499bf8eed0e7facacf"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 208.65.240.165:5060:
ACK sip:16473677222@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 209.197.184.137:6631;branch=z9hG4bK2724c9a4;rport
From: "14164778887" <sip:14164778887@209.197.184.137:6631>;tag=as57d3b3a7
To: <sip:16473677222@voip.freephoneline.ca>
Contact: <sip:14164778887@209.197.184.137:6631>
Call-ID: 18d8b260749af6ae150dcceb0590e8eb@209.197.184.137
CSeq: 102 ACK
User-Agent: Linksys/RT31P2-3.1.6(LI)
Max-Forwards: 70
Content-Length: 0


---
[Jun 28 11:43:23] NOTICE[659]: chan_sip.c:12377 handle_response_invite: Failed to authenticate on INVITE to '"14164778887" <sip:14164778887@209.197.184.137:6631>;tag=as57d3b3a7'
-- SIP/fpl1-005c4538 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

[seagame2001]

it works in TO now

[akoei]

it works in TO now

Do you use asterisk as well? I know ATA or s/w phone works

[seagame2001]

No I don't

I just smart phone so the app allow me to have multiple account so I did not use Asterisk

[akoei]

It suddenly stop working again, same auth fail error! The last successful outgoing is Wednesday morning, then I didn't have any outgoing call until last minute, I realize I can't make outgoing call...

Last time it just worked after my post, Freephoneline, please help!!!