similar to: What does the error "stale nonce' mean?

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from '<sip:reg-1@pbx.domain.com> I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same

Hi list, We have devices since more then 4 years which where running well with Asterisk. But with latest version (1.38 or more) we face problem with those devices when they try to register. We got [2011-05-14 17:18:06] WARNING[28559]: chan_sip.c:9950 register_verify: Failed to parse contact info <--- Transmitting (NAT) to XXX.XXX.XXX.XXX:5062 ---> SIP/2.0 400 Bad Request Followed by

Hi Guys, I am using the latest stable version of asterisk which I updated yesterday, but I keep getting this error message for some of my accounts. Can anyone explain to me what does this mean? "WARNING[3032]: chan_sip.c:4832 check_auth: Stale nonce received from". Thanks Joel

I'm getting dozens of these at a very high rate: [Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:121 at gnat.com>;tag=as5f1a9480' [Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:130 at

I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from '<sip:3034585725@voip.livewirenet.com;user=phone>' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable..

Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for

Scenario: [Asterisk Server] on routed/public IP \/ /\ \/ /\ \/ /\ \/ /\ \/ [Draytek Router] --> internal IP's \/ /\ \/ /\ \/ /\ \/ /\ \/ [internal Network 192.x.x.x] [IP10s] + [IP10s] + [Softphones] Everything is good as long as only *1* phone is registered from the internal network. The minute another phone goes online registration is dropped and the sip debug complains:

Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:18006733555@mantragroup.com>' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause

Hi, When I am debuggin any peer, I get swamped with those messages: chan_sip.c:8866 check_auth: Correct auth, but based on stale nonce received from xyz This seem to make up around 2Mbits/s of data (estimated after a quick tcpdump). Any ideas? This is while the server is idle (no calls, lots of registrations of course, but nothing worth 2Mbits/s) Regards, Mike

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hi all, There is a parameter called "nonce" included in every register request that a UA sends to asterisk. I have read sip debug a lot and only found out that the "nonce" parameter value which is used in register request was generated by asterisk server in a previous sip response. As you can see in the sip debug (labled in red). <--- Transmitting (NAT) to

Hello gentlemen, I am new here. I have a D-Link DPH-80S SIP phone (it's a non-US model), and I am trying to make it work with Asterisk. I tried versions 1.0.7 and yesterday's CVS and the behavior is the same. The phone registers with no problem, and can accept calls. But when I try to make outgoing call, there is a series of invite requests from the phone, to which asterisk responds

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8 my caller ID is not working WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515 -- Joseph

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

What does it mean? Jun 19 20:18:39 NOTICE[17484]: chan_sip.c:5472 check_auth: stale nonce received from '<sip:8862270@vpbx.elmit.com>' bye Ronald

Hello everyone, I am just starting a project for school in RoR and I am a complete n00b ^^ Here''s my problem : I need to get the user''s login stored in the session but for some reason I cannot. Here''s my login method code : def login if request.post? @user = User.find_by_username(params[:login]) if @user and @user.password_is? params[:password]

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

The use of the nonce looks right to me. Can somebody point out what is going wrong here? Jul 4 14:37:04 VERBOSE[12919]: Sip read: REGISTER sip:voip.victoria.tc.ca SIP/2.0 Via: SIP/2.0/UDP 199.60.222.229:5060;branch=z9hG4bKUF2CCx Max-Forwards: 70 To: pcnavideo <sip:pcnavideo@voip.victoria.tc.ca> From: pcnavideo <sip:pcnavideo@voip.victoria.tc.ca>;tag=w23oUOnv7kR Call-ID: