similar to: What is this error message? (check_auth: stale nonce received from ...)

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:

I'm trying to get an IAX trunk to failover to a local trunk it the trunk is down. This is what I've been working on: [macro-forward1]; exten => s,1,Dial(IAX2/192.168.1.1/${ARG1},20) exten => s,2,Goto(call-${DIALSTATUS},1) exten => s-CONGESTION,1,Dial(LOCAL/${ARG2},20) exten => s-CHANUNAVAIL,1,Dial(LOCAL/${ARG2},20 ;end macro-forward1 exten =>

I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID <sip:5551212@192.168.1.X>' Does anyone know what "stale nonce" is? Thanks! Paul Conn -------------- 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

I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

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

Here's a flipside of this subject: what is the absolute cheapest Linux device that can be connected to a PC's USB port? That has just enough power for a minimal Asterisk server running on it. The Asterisk just maintains a CDR database on its Flash memory, which it periodically submits over the PC's network connection with an HTTP hit on a remote full-service Asterisk server? No call

Sorry for disturbing, but I sent some messages today and I am not seeing them on this list. Can sombody tell me, in case this message appear on the list. Thank you

I've installed Sven Slezak's Notify module. He gives the follow as an example line to put into extensions.conf exten => s,1000,Notify(${CALLERIDNUM}|${CALLERIDNAME}|${EXTEN}/ sunnybook) I understand what is going on with this line but I don't know where in the extensions.conf file to put it? Thanks, Chris Griffin cgriffin@33keys.com

Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '45629314783bd11604363618632f07b9@201.x.x.x' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

Has anyone ever been party to an integration of ACT CRM platform with Asterisk? Thanks Cory Andrews

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..

I have a 2 port T1 card doing PRI passthrough, Span 1 answers from Telco, Span 2 sends to my existing phone system(Nortel). My Span1 gets sent to the context from-pri, detailed here: [from-pri] exten => _49XX,1,Set(CALLERID(all)=${CALLERID(all)}) exten => _49XX,2,Dial(Zap/g2/${EXTEN},,twk) exten => _49XX,3,Congestion() exten => _49XX,4,Set(CALLERID(all)="") exten =>

Originally posted by: mailto: Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I

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

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

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