Copy /var/www/html/admin/modules/firewall (or whatever it’s called) from another system. FreePBX should then see it as not installed but locally available.
↧
Unable to Parse XML response from Mirror
↧
Is it possible to use your 'away' status to redirect calls?
Thanks Bill - I haven’t found this setting actually does anything! Have you? 
Dave - thanks - it’s not so much the getting the status that I am struggling with, it’s getting the call flow to change its behaviour DEPENDING on the status.
↧
↧
FreePBX 15 GUI does not configure Asterisk properly!
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
↧
Unable to Parse XML response from Mirror
Thanks I fixed it by manually downloading the module from here: https://github.com/FreePBX/firewall/releases
Then I uploaded it and install it.
↧
BLF light green even if device is offline
I’m trying to configure some of our GxP 2160 phones to correctly display BLF states for other devices/extensions.
While it generally works (that is, when an extension is busy, the BLF light turns red), I’d like the BLF light to either be turned off or red, when the monitored extension is in “State:Unvavailable”. Instead, it stays green even when the device responsible for the extension has been physically turned off.
CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
100@ext-local : &Custom:DND100,Custo State:Idle Presence:available Watchers 0
207@ext-local : PJSIP/207&Custom:DND State:Unavailable Presence:not_set Watchers 1
Any ideas appreciated 
Yes, and this is happening on our up2date FreePBX 15.x installation.
↧
↧
Unable to Parse XML response from Mirror
Spoke too soon. When I installed the firewall this way it shows up as unsigned. I cannot sign it because it tries to go online then I get the same Parse error that is in the title. I cannot update any of my PBX’s right now. Are the repos down?
↧
External IP authentication behind NAT
IP authentication doesn’t require you to send a registration string. Further more you set the External IP in FreePBX under Settings -> Asterisk SIP settings -> NAT Settings.
Hope that helps.
↧
Unable to Parse XML response from Mirror
↧
Unable to Parse XML response from Mirror
Thanks miken32 this worked. The module that I copied over was already signed.
↧
↧
When putting someone on hold get an error "call transaction doesnt exist"
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
↧
Https://mirror.freepbx.org is not working
I’m having same issue with Freepbx 13 and 14
↧
Is it realistic to target removing all ERRORS during startups and reloads?
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
↧
No audio between some extensions (strange)
That sounds like a good plan Jameel. Usually, I don’t see these types of audio issues for systems where all the phones and the pbx are on the same subnet with no NAT between them, especially when it’s not consistent, and assuming there are no IP collisions going on. As for the packet tracing, there are different ways to go about it. I think many like to use tcpdump, but for a simple check of the rtp flow, before digging deeper, I like to use tshark just because I’m used to it.
From the PBX command line, I’d do the following:
[root@freepbx]# tshark host 10.12.21.65
In this example, my phone is at 10.12.21.65, and my PBX is at 10.12.4.11. While tshark is running, I will place a call from my test phone. The output that flies by will look something like:
1 0.000000000 10.12.21.65 -> 10.12.4.11 SIP/SDP 1111 Request: INVITE sip:103@10.12.4.11 |
2 0.018268165 10.12.4.11 -> 10.12.21.65 SIP 592 Status: 401 Unauthorized |
3 0.055583814 10.12.21.65 -> 10.12.4.11 SIP 407 Request: ACK sip:103@10.12.4.11 |
4 0.068091626 10.12.21.65 -> 10.12.4.11 SIP/SDP 1399 Request: INVITE sip:103@10.12.4.11 |
5 0.104914332 10.12.4.11 -> 10.12.21.65 SIP 394 Status: 100 Trying |
6 1.090579212 10.12.4.11 -> 10.12.21.65 SIP 659 Status: 180 Ringing |
7 1.115429589 10.12.4.11 -> 10.12.21.65 SIP/XML 1218 Request: NOTIFY sip:101@10.12.21.65:5060;ob |
8 1.147347429 10.12.4.11 -> 10.12.21.65 SIP 659 Status: 180 Ringing |
9 1.157864282 10.12.21.65 -> 10.12.4.11 SIP 590 Status: 200 OK |
10 1.975603976 10.12.21.65 -> 10.12.4.11 SIP 566 Request: REGISTER sip:10.12.4.11:5060 |
11 1.977003543 10.12.4.11 -> 10.12.21.65 SIP 600 Status: 401 Unauthorized (0 bindings) |
12 1.980277889 10.12.21.65 -> 10.12.4.11 SIP 855 Request: REGISTER sip:10.12.4.11:5060 |
13 1.984652942 10.12.4.11 -> 10.12.21.65 SIP 546 Status: 200 OK (1 bindings) |
14 2.004116904 10.12.4.11 -> 10.12.21.65 SIP 672 Request: NOTIFY sip:101@10.12.21.65:5060;ob |
15 2.018600766 10.12.21.65 -> 10.12.4.11 SIP 414 Status: 200 OK |
16 2.701966854 10.12.4.11 -> 10.12.21.65 SIP/SDP 1096 Status: 200 OK |
17 2.846513984 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18456, Time=1976200720
18 2.861299491 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18457, Time=1976200880
19 2.870240161 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62174, Time=2068152745
20 2.875086836 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18458, Time=1976201040
21 2.889239405 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62175, Time=2068152905
22 2.894463921 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18459, Time=1976201200
23 2.909481741 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62176, Time=2068153065
24 2.918681238 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18460, Time=1976201360
25 2.929525192 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62177, Time=2068153225
26 2.945471823 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18461, Time=1976201520
27 2.949280087 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62178, Time=2068153385
28 2.956528507 10.12.4.11 -> 10.12.21.65 RTP 214 PT=ITU-T G.722, SSRC=0x64CADBD9, Seq=18462, Time=1976201680
…a lot more of this…
200 4.669428890 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62264, Time=2068167145
201 4.689204523 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62265, Time=2068167305
202 4.709188690 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62266, Time=2068167465
203 4.729021260 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62267, Time=2068167625
204 4.748711515 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62268, Time=2068167785
205 4.768801404 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62269, Time=2068167945
206 4.788601085 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62270, Time=2068168105
207 4.808806558 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62271, Time=2068168265
208 4.828580204 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62272, Time=2068168425
209 4.848547744 10.12.21.65 -> 10.12.4.11 RTP 214 PT=ITU-T G.722, SSRC=0xC06C8722, Seq=62273, Time=2068168585
210 4.866766218 10.12.4.11 -> 10.12.21.65 SIP 476 Request: BYE sip:101@10.12.21.65:5060;ob |
In this example, I can see RTP packets going back and forth between the PBX and phone. In some cases with audio issues, you’d just see the flow going in one direction which can lead to clues as for what to look at next.
↧
↧
Billing Module
a2billing wants to take control which defeats a lot of features.
What exactly does it take control of? We’ve been using it for 10 years now to track billing on outbound calls. Currently installed on about 75 hosted PBXs. The only time it gets in the way is the extra CDR generated for each call.
↧
Login into queue using Phone Apps
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
↧
A Community Update
In late January , we’re releasing updates for FreePBX to support the new, upcoming requirements for emergency calling.
I assume you mean “the new, upcoming American requirements” here. I trust this won’t get in the way for the rest of us? Any details on what changes this will involve?
In February , we’re doing work to improve pieces of infrastructure used by our open source projects.
LOL considering the module servers have been down for at least 24 hours, you may want to bump that one up a few weeks.
In the March/April timeframe, I’ll update you on the significant engineering time we’re investing to modernize some of the internal “plumbing” of FreePBX, to help it keep up with recent technology changes.
Dear god, please don’t tell us you’re moving more code away from PHP to NodeJS. Your short-sighted decision to require a whole new infrastructure for basic functionality means a lot of servers stuck at 13 over here. If you really wanted to move from PHP because of performance concerns, I would have much preferred to see some kind of compiled code like Python, Go, or Rust that didn’t require a whole new daemon.
more participation from me and from others at Sangoma in the forums
Just make sure it’s good participation. Ignore or respectfully engage with the trolls, and don’t immediately dismiss everyone’s problems as an error on their part. I haven’t been around the forums much, but these were definitely problems I saw in the past.
We are working on a series of short videos
I’d suggest making sure there are written versions of these, for the visually impaired (and people who just don’t like videos!)
↧
Is it possible to use your 'away' status to redirect calls?
My mistake for taking documentation at its word apparently. I tried it, and agree it doesn’t work.
https://issues.freepbx.org/browse/FREEPBX-18552 seems to be relevant. I don’t get what the REST Apps module has to do with it.
↧
↧
Announcements not working any more
I will do some A-B testing… My FreePBX is running on my local LAN, so ports are probably not the issue.
↧
One extension - two phones. Both work but not showing registered in endpoint
I have an executive that asked to have his Polycom and Mitel phone in his office on the same extension. I used PJSIP and changed max contacts to 2 and then put in the mac address for the second phone. I deleted the configs in endpoint manager and rebuilt them. They show as 77227-1 and 77227-2. I wiped both phones and rebooted them. Both phones came up with the right config and work just fine, however, they don’t show as connected in endpoint manager. Because they don’t show up in endpoint manager the polycom will not update the date and time so it’s always blinking the incorrect date/time. Other issues like jolt click to dial only calls the polycom and not the mitel.
↧
Is it possible to use your 'away' status to redirect calls?
No I don’t know what the REST apps module has to do with it either. AFAIK REST apps are just “to control PBX functions and call settings directly from the screen of their phone.” So quite how/why it’s relevant to FreePBX call flow logic doesn’t make any sense to me. Zulu is already successfully changing the user’s presence state. It’s the bit AFTER THAT that’s not working.
btw, I have a ticket open about this issue here too
https://issues.freepbx.org/browse/FREEPBX-20830
↧