similar to: "timeout on non-critical invite" spamming log

On 12/30/2017 08:18 PM, Dovid Bender wrote: > Script kiddies trying to find vulnerable systems that they can make > calls on. Lock down the box with iptables and use fail2ban to block > them. The via is probably bogus unless a box at the DoD was comprimised. > > > > On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at

I've been getting a lot of timeouts on non-critical invite transactions. I turned on sip debug. They were the result of SIP invites like this: Retransmitting #10 (NAT) to 185.107.94.10:13057: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057 From:

Hi All I'm getting tens of thousands of these messages ever hour in the Asterisk CLI for Asterisk 13.22.0: [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 1924200000-502043860-301870737 on non-critical invite transaction. [Apr 20 15:59:46] WARNING[45462]: chan_sip.c:4127 retrans_pkt: Timeout on 301794058-652332923-1834701069 on non-critical invite transaction. [Apr

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a

I'm getting invites to very high ports every 30 seconds from a particular ip address: Retransmitting #10 (NAT) to 5.199.133.128:52734: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972 To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748

On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at

I wonder if I could have that patch, maybe I could add it to my fail2ban regexp and if you have the correct regexp, I would apperciate that as well. Thanks. On Wed, 29 Aug 2018 19:18:29 -0400, Telium Support Group wrote: > > Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened

OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote:

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group <support at telium.ca> wrote: > Depending on log trolling (Asterisk security log) misses a lot, and also > depends on the SIP/PJSIP folks to not change message structure (which has > already happened numerous time). If you are comfortable hacking > chan_sip.c you may prefer to get the same messages from the AMI. It still

I'm experiencing one-way voice paths, followed by a hangup on one softphoine and not the other. Also, caller ID has odd outputs -- and I wonder if the problems are related. My configuration has Asterisk and a Linphone softphone running on the same box. I have a PC, and on that PC I use X-Lite or SJPhone to connect to the Linphone instance. When I call from the PC to Linphone: * I call

Hello list, I am confronted with the following problem : making a call only leasts for about 30 seconds, then the call is ended. The CLI shows : [May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 954539948-5060-2 at 192.168.1.100 for seqno 11 (Critical Response) -- See doc/sip-retransmit.txt. [May 21 14:31:50] WARNING[25345]:

Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all

Hello All, Got one customer and he is getting disconnection within 15 seconds when he tries to make outbound calls. Initially, it was working fine without any glitches... Other customers on the same system are working fine, its just with this customer only. This is the error message thrown by Asterisk on the CLI: - Jan 18 12:23:30 WARNING[30532]: chan_sip.c:1228 retrans_pkt: Maximum retries

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

Hi all, I am using * as a PBX for a Broadvoice VoIP account. It had been working well since about last November, although not perfectly (similar disconnection problems, although I am pretty sure it had to do with my PPPoE setup, but I think these issues were resolved). As of a few weeks ago, though, I started having serious problems. Basically, I can start up * and connect to Broadvoice and

We're trying to get a SIP peer going between our asterisk box and our provider. It should then ring our phone. The call does come in and it does execute the extension in the dial plan. But the provider says they never get a 200 OK back and therefore they send another INVITE and then after a few seconds drop the call. Here's our setup: sip.conf [ngt-trunk] type=peer qualify=yes port=5060

HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17

Hi, How would I go about troubleshooting this: [Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. [Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno 101

Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and only in voicemail) after about 20 seconds with the error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt).". I