search for: nonces

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'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 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

Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use

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..

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:

OpenSSH Security Advisory: legacy-certs.adv This document may be found at: http://www.openssh.com/txt/legacy-cert.adv 1. Vulnerability Legacy certificates generated by OpenSSH might contain data from the stack thus leaking confidential information. 2. Affected configurations OpenSSH 5.6 and OpenSSH 5.7 only when generating legacy certificates. These must be

OpenSSH Security Advisory: legacy-certs.adv This document may be found at: http://www.openssh.com/txt/legacy-cert.adv 1. Vulnerability Legacy certificates generated by OpenSSH might contain data from the stack thus leaking confidential information. 2. Affected configurations OpenSSH 5.6 and OpenSSH 5.7 only when generating legacy certificates. These must be

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

06.10.2015 1:22, Joshua Colp ?????: > On 15-10-05 05:58 PM, Dmitriy Serov wrote: >> 05.10.2015 23:24, Joshua Colp ?????: >>> On 15-10-05 05:22 PM, Dmitriy Serov wrote: >>>> Hello. Do I understand correctly that the current implementation >>>> res_pjsip does not support ZRTP? >>>>

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

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

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

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

Hi, my dovecot installation works since months and clients authenticate using CRAM-MD5. But today I got the first chance to test an client that supports DIGEST-MD5 - and it doesn't work. Because of lack of other supporting clients and servers I'm now at the point I don't know which side is to blame. The error I get after the client answers the servers challenge is "-ERR

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

I'm trying to debug a new spa3k to * config issue and am receiving hundred's of the following: Sep 3 09:38:51 NOTICE[23092] chan_sip.c: stale nonce received from '784-7103 <s ip:7847103@208.2.2.2>' What's the typical issue behing this message?

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: