I looked closer at the Mikrotik settings and found out someone applied a NAT rule on traffic going out the LAN interface.
That was causing the source IP to get re-written.
I deleted it and now incoming calls are working.
Thanks for the second pair of eyes.
Ok so 13.0.6.6 has been released to "stable" smae issue still there, No save option in advanced and wont load packages manually or via the package manager. this has been tested on 3 systems and all the same.
If its being tested and working before stable can it be confirmed what packages are required. as in teh Freepbx distro there is something obviously missing.
We dont even see any activity or errors in teh httpd logs.
Ian
the oss endpoint manager was (a possibly still is) and interesting product but when Andrew joined schmooze some years ago we switched to the paid endpoint manager. the paid version is not without bugs but it does provide an easier mechanism for updating the basefiles and it does support a wider variety of phones. Yes you do have to pay for it, but given the amount of time i suspect some of you have invested so far and getting the oss epm to work, i think you will find the paid emp pretty cheap.
There is no save option because everything saves when you change it. Just confirmed this is working.
Can't confirm this. Working fine here.
Every time the developer (not me) submits the work I check it on the freepbx distro.
P=51 is the vlan tag for all grand stream phones. I have over 200 phones Deployed and never have had any issues.
Anybody had any luck with the live provisioning option with the gxp2160 without rebooting with freepbx. I heard that freepbx supports this or will support this.
i am updating my voip lab from a freepbx server v2.9.0.9 -->v12.0.76.2 (about time, right?).
any chance that i can save some time copying over the 2.9 config to the new 12.0?
or do i need to do it all manually?
tnx
ams
I would love to, but for some reason, every time I try to register for an account or log in, it doesn't let me.
The login is the same as you use to login here
I see 7 failures so you have the username right... the password again is the same as you use to login to the forums.
I would, but it seems I can't get my SPA3102 to register the trunk with the chan_sip driver, for resons which I haven't figured out yet :).
I really like the approach of the responsive firewall and think it is a great idea. But I can see the same behavior with the firewall as jst68. It is successfully blocking IPs, so it is working and I have it rightly configured.
But browsing through the logs I see attacks from various IPs with a certain amount of time in between each try not beeing blocked at all even though it is a continues attack going on for a long period of time.
In between these timestamps there are attempts from other IPS, also with a certain amount of time between each try. This way they avoid beeing blocked.
Is there any setting that can be used to prevent these kind of attacks? It would be nice to have an IP blocked after lets say 5 failures. Or do I need to also use fail2ban to prevent this?
Just extracting one IP it looks like this
[2016-03-07 16:10:31] NOTICE[30385] res_pjsip/pjsip_distributor.c: Request from '"12345" ' failed for '89.163.242.32:5082' (callid: f8593ed356d031570a20b2c24cc014ab) - No matching endpoint found
[2016-03-07 16:23:15] NOTICE[31889] res_pjsip/pjsip_distributor.c: Request from '"12345" ' failed for '89.163.242.32:5083' (callid: ff7fe6c764eccef01d3970c745bcbb66) - No matching endpoint found
[2016-03-07 16:35:54] NOTICE[31889] res_pjsip/pjsip_distributor.c: Request from '"12345" ' failed for '89.163.242.32:5079' (callid: a5cb2b1229e563eb4ffbe19561fbace0) - No matching endpoint found
[2016-03-07 16:48:32] NOTICE[30385] res_pjsip/pjsip_distributor.c: Request from '"12345" ' failed for '89.163.242.32:5076' (callid: 27888250307bb52b208642c02a7efb5d) - No matching endpoint found
[2016-03-07 18:04:28] NOTICE[31889] res_pjsip/pjsip_distributor.c: Request from '"12345" ' failed for '89.163.242.32:5081' (callid: 102cf1fbe164a8f5432725096dbfafe9) - No matching endpoint found
But after the first entry the log records also these other IPs
[> 2016-03-07 16:10:55] NOTICE[31889] res_pjsip/pjsip_distributor.c: Request from '"2212" ' failed for '209.239.127.96:5070' (callid: cfb5f293c87e62515c88ea2ac26b9723) - No matching endpoint found
[2016-03-07 16:11:00] NOTICE[30385] res_pjsip/pjsip_distributor.c: Request from '"1001" ' failed for '89.163.242.64:5071' (callid: 7729504046e6e3c2e5e2f456ffbfe05b) - No matching endpoint found
When a user hits the voicemail button on the phone, the *97 number is displayed, of course. I'd like to make it just a little bit more user friendly, and have it display "Voice Mail" also.
This can be done with the 'connectedline()' command in the dialplan, and this is what I've added to extensions_custom.conf
exten => _*97,1,Set(CONNECTEDLINE(number,i)=*97)
same => next,Set(CONNECTEDLINE(name,i)=Voice Mail)
same => next,Set(MAILBOX=${SIPPEER(${CHANNEL(peername)},mailbox)})
same => next,VoiceMailMain(${MAILBOX},s)
same => next,Hangup(normal_clearing)But that's quite a lot for a small thing, especially as I'd like to add more 'friendly names' for system features. There has to be a better way.
Any suggestions?
also, I've come to realise that I don't really understand what the underscore is for before the matched number, and it's quite hard to google for! Would love an explanation for that one 
Gotcha.
Hope that's the "letsencrypt'' stuff. That's going to be sooooo useful!
Good luck with it.
[edit: typo and subsequent grammar change]
Agreed. I don't need to know when a backup process does its job properly, I need to know when it failed.
For now, your best bet would be a well crafted email filter to just delete the 'success' messages.
You can see some of our work here: https://github.com/analogic/lescript/pull/9
Hi Team,
After enable User & Devices Mode cpu utilization above 50% and some time calls will drops
Asterisk (Ver. 11.21.0): Peers
PBX Firmware: 6.12.65-32
PBX Service Pack: 1.0.0.0
[XXX@phones ~]$ top -cd1
top - 00:52:11 up 4:30, 1 user, load average: 5.04, 5.02, 5.00
Tasks: 242 total, 6 running, 236 sleeping, 0 stopped, 0 zombie
Cpu(s): 64.2%us, 1.0%sy, 0.0%ni, 34.7%id, 0.0%wa, 0.0%hi, 0.1%si, 0.0%st
Mem: 264484516k total, 9527664k used, 254956852k free, 221704k buffers
Swap: 4194296k total, 0k used, 4194296k free, 4992556k cached
PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND
4181 asterisk 20 0 293m 24m 7776 R 99.7 0.0 255:47.30 /usr/bin/php -q /var/lib/asterisk/agi-bin/user_login_out.agi login 21016 2255
4339 asterisk 20 0 293m 24m 7776 R 99.7 0.0 255:17.94 /usr/bin/php -q /var/lib/asterisk/agi-bin/user_login_out.agi login 21017 2256
5178 asterisk 20 0 293m 24m 7776 R 99.7 0.0 251:07.21 /usr/bin/php -q /var/lib/asterisk/agi-bin/user_login_out.agi login 21015 2254
17044 asterisk 20 0 293m 24m 7772 R 99.7 0.0 75:35.30 /usr/bin/php -q /var/lib/asterisk/agi-bin/user_login_out.agi login 21008 2247
17461 asterisk 20 0 293m 24m 7776 R 99.7 0.0 73:46.71 /usr/bin/php -q /var/lib/asterisk/agi-bin/user_login_out.agi login 21007 2246
17478 asterisk 20 0 4858m 69m 14m S 19.7 0.0 10:30.62 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
Voicemail messages can be replayed via the handset using the UCP, which is perfect because the office computers do not have sound capability.
Recorded calls show only with a download link, no 'play on phone' like the voicemails have. Is this a fault in my system or behaviour by design? Is it possible to change?
Ah, The underscore prefix allows the use of pattern matching. So the particular example above does not need it. I'll remove it.
Thanks for the tip.
Extension names are not limited to single specific extension "numbers". A single extension can also match patterns. In the extensions.conf file, an extension name is a pattern if it starts with the underscore symbol (_).