similar to: stale nonce?

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:

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:

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 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

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

I have this error Aug 27 13:31:46 NOTICE[2863] chan_sip.c: stale nonce received from '656720189<sip:6562720189@69.12.30.109;user=phone>' generated for two equipment hardwired in asterisk. Some friend can help me? Thank's Jo?o Carlos Moura

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 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

Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with a dialplan that essentially routes any call starting with an "8" to asterisk. All other US 7 and 10 digit calls, 911, etc, route via the spa3k's fxo port. Is there a way in extensions.conf to: - inspect the dialed exten number, - if first digit is "8", drop the 8, - continue through each

Thanks Rich, I have an SPA-3000 laying around, so I will attempt to set it up in a little more conventional manner (although your method looks like a winner for a home test PBX). Would you mind posting or PM your current config to me, maybe screenshots if you PM. If I start with that it will take less time to get to the point where the SPA-3000 is a true FXO-FXS gateway for *. I will be happy to

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

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

Asterisk Project Security Advisory – AST-2020-002 Product Asterisk Summary Outbound INVITE loop on challenge with different nonce. Nature of Advisory Denial of Service Susceptibility Remote

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..

Author: Darren Moffat <darrenm at opensolaris.org> Repository: /hg/zfs-crypto/gate Latest revision: d22dab8b3cf8927274becf145e811389891f4fb7 Total changesets: 1 Log message: lint nits and CCM IV/nonce size fix Files: update: usr/src/common/zfs/zcrypt_common.c update: usr/src/uts/common/fs/zfs/zio_crypt.c

I need to place a call using a "pin code". To access an external line, the host PBX (a Ericsson MD-110) will require that I dial *72*pincode#phone_number to complete any (trunk) call. When I send the number, my Sipura 3000 will reject the call with "Forbidden - wrong password on authentication for INVITE" (see below). All other calls sent to the Sipura box without the

freebsd quite current ports tree 1.01 asterisk spa3k at 2.0.11(GWg) for calls in from the pstn side of an spa3k to asterisk, i get sporadic short beeps. they are not related to sip re-reg time, which is all that has occurred to me so far. calls in from the fxs side of the spa3k and out through nufone do not exhibit the beeps. calls from the fxs side of the spa3k out the fxo side do have the

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?