similar to: No subject

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

On Fri, Jan 9, 2015 at 5:24 PM, Michelle Dupuis <mdupuis at ocg.ca> wrote: > I'd suggest taking a look at the free edition of SecAst ( > www.generationd.com). It handles these messages perfectly (and can also > use AMI security events) - so you don't need to constantly be updating > fail2ban rules. It's a drop in replacement for fail2ban. > > -M- > >

> On 27Jun, 2015, at 15:34, Michelle Dupuis <mdupuis at ocg.ca> wrote: > > Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? Do you mean large number of calls or how loud the call is? -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News)

When a caller calls my google voice phone number, I must answer, wait and press one to accept. Sometimes even that does not work. I have tried a few different things to get asterisk to place the call in an answered state and send the DTMF 1 with the Dial macro. I found Malcom Davenports wiki page regarding Google calling which has been very helpful in troubleshooting the issue.

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

Hi guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

And after reload ALL your phones are unreachable for 2 minutes! Imagine you have several thousands devices unreachable for 2 minutes. How much calls will fail during that time? Regards, Mindaugas Kezys Kolmisoft UAB=20 VoIP Billing Solutions e-mail: info at kolmisoft.com URL: http://www.kolmisoft.com -----Original Message----- From: asterisk-users-bounces at lists.digium.com =

Maybe someone could point in the right direction. I have a small facility that's running around 40 Polycom 301/501 phones, Asterisk 1.4.18 running under Mandriva 2007.1. The phones were assigned a DHCP address in the 10.10.10.x range. Today, the DHCP server failed and to get them back online, I loaded the dhcp-server onto another system (Also running Mandriva) and copied the dhcpd.conf

I saw the following lines in the log this morning. From my router logs I see that the connection went down as my ISP was doing maintenance for a few minutes last night. I can understand the external registrations timing out, but why do the phones become unreachable. They are on the internal lan within the same subnet as the Asterisk server. Internal DHCP and DNS was functional. If I had a PRI card

I've got a number (10) Cisco 7960's connected to my network. All the phones work perfectly except one. The asterisk console keeps throwing up: Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 sip_poke_noanswer: Peer '4001' is now UNREACHABLE! Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 handle_response: Peer '4001' is now REACHABLE! Jun 22 15:42:08

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Hello, I am logging directly into file and also to syslog. Here is snippet from my /etc/asterisk/logger.conf: messages => notice,warning,error,verbose syslog.local0 => notice,warning,error,verbose But the logs look different: VERBOSE[7609][C-00000013] pbx.c: NOTICE[3042] chan_sip.c: Peer '1111' is now UNREACHABLE! vs. VERBOSE[7609][C-00000013]: pbx.c:2925 in

Hi All, I'm wondering if anyone can share any info on why I frequently get peer timeouts like below, and receive 489 messages from another A*k server on the same LAN. For the peers, we've one L2 switch. ICMP is <1ms. The CPU of the main A*k server is usually < 2%. So I can't see why we'd get such large delays. The phones are all Cisco 7940s (SIP 2xx) The 489 originate

[Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer '1000' is now UNREACHABLE! Last qualify: 110 [Mar 21 23:24:18] NOTICE[9931]: chan_sip.c:26242 sip_poke_noanswer: Peer 'patton' is now UNREACHABLE! Last qualify: 20 I also get errors for connections to SIP servers for which I have "register" entries in the [general] section of sip.conf. The

> On 2019-11-16 03:29, Fourhundred Thecat wrote: > Hello, > > I am logging directly into file and also to syslog. > Here is snippet from my /etc/asterisk/logger.conf: > > messages => notice,warning,error,verbose > syslog.local0 => notice,warning,error,verbose > > But the logs look different: > > VERBOSE[7609][C-00000013] pbx.c: >

hi folks. we recently deployed 10 Cisco 7960G w/ SIP firmware 7.3 on our network and we start having problems of dropping calls (actually the calls wasn't dropped it just the sound was muted for about 5-10 seconds, but most users will think the call dropped and hangup/redial). i've check the console output. there was a lot of messages like the following: Sep 28 15:00:49 NOTICE[8182]:

I'm having trouble setting up tls/srtp secure communications on my Asterisk server- I'm still rather new at working with Asterisk. I have enabled tls and encryption and I have csipsimple with tls build on the phone. I'm currently only testing one phone with this capability so far, and the rest still work in the current state. My logging looks like this with verbose turned up:

Roman: I created two bash scripts called Mail2Fax and Fax2Mail for use with the asterisk sever. They leverage the app_txfax and app_rxfax scripts, along with ast_fax. They make using these apps a lot easier, including being able to mail to fax@domain.ca for outgoing faxes and then extracting phone numbers from the subject line! (Makes it easy to use with Sendmail without complex rules /

Ok, the script is attached... I'll post it on www.generationd.com when I have a chance. If you have any updates & improvements please email them to me! (The command line parameter handle is pretty stupid - just grew from testing to production without cleanup). MD _____ From: Craig Huff [mailto:huffcslists at gmail.com] Sent: Saturday, September 08, 2007 11:34 AM To:

I see repeated over and over the following messages: NOTICE[1142106560]: chan_sip.c:4988 handle_response: Peer '1001' is now REACHABLE then 5 minutes later: NOTICE[1142106560]: chan_sip.c:5958 sip_poke_noanswer: Peer '1001' is now UNREACHABLE both messages repeated over and over Any clue what I can do to fix this? Is there any where I can look up these Notices to find