What further information can I provide to aid in getting help analyzing this problem?
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Problem with delay in call setup
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Q xact reports
It can only know about data it has received since it was licensed. The data is not stored anywhere without the module
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Q xact reports
so does that mean that the queues and agents it finds are the only ones that have received calls since the module was installed and that as the days progress more queues and agents will start showing up?
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Installation Problems
I would have to agree with your comments on PJSIP. I have really struggled to get trunks working on PJSIP that easily work on Chan_Sip.
The extensions seem OK but general reliability appears very poor, the channel has stop quite a few time for me with all the extensions becoming unavailable. I think the problem is PJSIP does not support dynamic ip addresses which sadly many of us have to use.
It really feels like a massive step backwards, I hope chan_sip is supported for quite a while yet, I don't think I could rely on a PBX that only had PJSIP.
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Failure on PBX to PBX Call over a trunk
With a little investigation I think I have worked this out. If you change type=peer to type=friend the calls seem OK being transferred back again.
I am just a little worried I may have made myself a security problem, all seems OK at the moment but I am still checking.
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Failure on PBX to PBX Call over a trunk
Maybe I was a bit ahead of myself, it only works when the calling extension is on PJSIP not chan-sip.
Back to the drawing board I think.
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IVR with Enable Direct Dial slow response
I went ahead and created another IVR with direct dial enabled which can be called from the main IVR by pressing 1. No latency doing it that way,
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FreePBX 12, Asterisk 13 and PJSIP trunk config
We are having a similar issue. Looking for an update also.
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Networking issues
OK a little more about my setup. I had Comcast Business come in and install internet here just a few days ago now. I have one device plugged into the gateway, a windows PC. Eventually there will be a few more but at the moment I only want that gateway (cable modem) providing service to dedicated hardware. I purchased a block of 5 usable static IP's this in all reality gives me 6 IP's to use. The sixth being assigned to the gateway and that gateway providing DHCP to a LAN. So my Windows PC has an internal IP range address. Now I have verified that the Static IP's do work by requesting a proper IP from the gateway using the IPv4 properties in windows. I had another hard drive with a LAMP stack running Debian that is able to use all the IP's.
I do not want to post my external facing IP's in open forum. This would be a recipe for disaster! I am quite certain that I will get plenty of attempts at my setup regardless of what I do but hey why make it worse 
ping google.com > unknown host google.com
ping 96.xx.xxx.xx > (my gateway address) Destination Host Unreachable (Works from Windows Computer)
mii-tool eth0 > eth0: negotiated 100baseTx-FD flow-control, link on
I redirected the output of the route -n to a txt file and moved it to a usb for this one.Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
xx.xx.xxx.80 0.0.0.0 255.255.255.248 U 0 0 0 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 eth0
0.0.0.0 xx.xx.xxx.86 0.0.0.0 UG 0 0 0 eth0
So if you will notice the first entry in that output shows the last octet as .80 I do not know where that is coming from that is nothing I have programmed in any of my config files. The last entry for the Gateway is correct and coincides with what was provided by Comcast. I assume its like your last octet of .0
I am all ears and willing to do just about anything. If you need a non redacted copy of an output I can send via PM
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Networking issues
Forgot the ping from Gateway to FreePBX box...
Pinging xxx.xxx.xxx.xxx:
Request timed out.
Request timed out.
Request timed out.
Ping statistics:
Pings sent: 3 (0 per second); Replies received: 0 (0 per second)
Bytes sent: 192 (48 per second); Bytes received: 0 (0 per second)
0 replies passed verification (0 failed)
Min time: 0 ms; Max time: 0 ms; Avg time: 0 ms; Total time: 4994 msBTW I am literally assigning an external facing IP to my FreePBX box in an identical setup to what I have being hosted on a VPS currently.
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Directory not using VM Greeting, spells name only
Looks OK, here's the log, it starts playing the letters right after:
2016-01-27 19:26:47] VERBOSE[41396][C-0000003e] pbx.c: Goto (directory,1,1)
[2016-01-27 19:26:47] VERBOSE[41396][C-0000003e] pbx.c: Executing [1@directory:1] Answer("SIP/voipms_ny4_mabos-00000037", "") in new stack
[2016-01-27 19:26:47] VERBOSE[41396][C-0000003e] pbx.c: Executing [1@directory:2] Wait("SIP/voipms_ny4_mabos-00000037", "1") in new stack
[2016-01-27 19:26:48] VERBOSE[41396][C-0000003e] pbx.c: Executing [1@directory:3] AGI("SIP/voipms_ny4_mabos-00000037", "directory.agi,dir=1,retivr=true") in new stack
[2016-01-27 19:26:48] VERBOSE[41396][C-0000003e] res_agi.c: Launched AGI Script /var/lib/asterisk/agi-bin/directory.agi
[2016-01-27 19:26:48] VERBOSE[41396][C-0000003e] file.c: Playing 'cdir-welcome.slin' (language 'en')
[2016-01-27 19:26:51] VERBOSE[41396][C-0000003e] res_agi.c: Playing 'silence/1.ulaw' (escape_digits=1234567890#) (sample_offset 0) (language 'en')
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Debian8 fresh install Error Class Userman not found
/usr/src/freepbx# php -i|head
phpinfo()
PHP Version => 5.6.17-0+deb8u1
System => Linux asteriskdebian 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) x86_64
Build Date => Jan 13 2016 09:09:23
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Debian8 fresh install Error Class Userman not found
I been trying to find some sort of link since you mentioned it on version of php i can see in the install.php file a refrence to php version . maybe I should go to a lower version of php ?
/usr/src/freepbx# cat install.php |grep PHP
if (version_compare(PHP_VERSION, '5.3.3', '<')) {
out(sprintf(_("FreePBX Requires PHP Version 5.3.3 or Higher, you have: %s"),PHP_VERSION));
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WebRTC browser calls FreePBX 13 (Stable)
Unaware that WSS is not supposed to work, I got it working today on FreePBX 13 and Asterisk 11.13 through no special effort other than to enable Asterisk's HTTPS server on 8089 and supply a valid certificate. I was already using HTTPS in Apache for admin and UCP. What is it that makes this unsupported--or was it fixed in the last few weeks?
I can make outgoing calls but cannot receive calls. The WebRTC client immediately appears unregistered to Asterisk by failing to respond to a qualify:
[2016-01-27 20:57:32] NOTICE[12903]: chan_sip.c:29479 sip_poke_noanswer: Peer '991101' is now UNREACHABLE! Last qualify: 0
I can't tell whether this is a WebRTC issue or a typical network (NAT, firewall) issue, though my SIP phones on the same network are experiencing no difficulties with registration.
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WebRTC browser calls FreePBX 13 (Stable)
The JsSIP client was registering with a contact containing "transport=ws" even though it is using a WSS connection. Thus outgoing calls from WebRTC worked fine but incoming calls were going to the wrong transport.
I hacked in a little patch to the js to force it to register as transport=wss (thanks to @artfulhacker for the tip) and now the client registers fine and receives signaling but dumps incoming calls with a SIP 488 "Not acceptable here" -- some different protocol (codec) mismatch this time.
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WebRTC browser calls FreePBX 13 (Stable)
As I already said. The module does not support wss at this time. By module I mean the module in freepbx. I never said a thing about asterisk. In fact behind the scenes rob and I are working on let encrypt with webrtc and the freepbx module along with tls support among other things.
We've both made wss calls and tls calls. Through freepbx.
The code for webrtc in freepbx is open source. Anyone can contribute. Go for it.
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Fax Pro not e-mailing incoming faxes **resolved**
Two considerations. The button only shows up when you're within 90 days of support expiring. Also, you need to check online for updates for it to tell you that. If you're within that timeframe AND the button isn't there when you check for updates, we'd be interested in having you contact support to have a look
As far as other places to renew it though, you can always go to the portal and update it there. That doesn't have the 90 day restriction, you can update it at any time, and for multiple years if you want.
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Endpoint manager for Snom showing time 30 min behind
Using the latest FreePBX 13 distro we have installed EndPoint manager and after reseting the NPT server and the correct timezone the phones using the Commercial EndPointManager are 30 min. behind the date on the PBX and the rest of the phones on the network.
System is using:
FreePBX 13.0.54
Asterisk 11.20.0
SHMZ release 6.6 (Final)
cpe:/o:schmooze:linux:6:GA
Phones are Snom S-300 firmware version 7
Any suggestions?
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WebRTC browser calls FreePBX 13 (Stable)
Actually, it does look like Asterisk is mostly to blame. https://issues.asterisk.org/jira/browse/ASTERISK-24330
If not for that issue, I believe the JsSip would be working fine.
So I do not believe it is correct for FreePBX or JsSip to be making hacky workarounds to deal with Asterisk. Asterisk needs to fix their bug.
Thanks for your work on LE/TLS @tm1000! Glad to see it.
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Multiple Extensions Answering A Call?
I am trying to figure out if there is anyway to meet this goal and cannot for the life of me figure out if this function exists!
I need any of 20 extensions to be able to initiate a call (to a misc destination or some code lets call it 12345). This would make all the other extensions (19) ring. However, when 1 person picks up the call it continues to ring other extensions that can continue to answer the call and be added to said call.
Basically I want to be able to send a request (ring) to extensions and upon answering they are added to a conference. If this functionality doesn't exist or a work around to get a similar functionality I would gladly fund a developer to make an extension that would do this.
Thanks!
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