@bajramia wrote:
Hi All,
We have made a dessision switching our phone system from Avaya to Sangoma PBXact UC 2000
with all commercial modules and 600 sangoma 705 phones and 18 Vega 3050 gateways, I want to ask and see which protocol should i use to build all the extensions Chan_Sip or PJSIP.Thank you
Posts: 1
Participants: 1
@sentinelace wrote:
I can’t remember how I did this before. We have a user setup to follow me to their cell phone, if they don’t answer, it used to fall back to the PBX voicemail side. Now it is not and it goes to the voicemail on the cell phone. Is there a setting to make it always fall back to the pbx voicemail if not answered by the follow me number?
Posts: 3
Participants: 2
@Matthew99 wrote:
Hi All
i have a 2 Queues set up like so
Queue 1 if unanswered fails over to Queue 2 then on to voicemail
i am running Asternic call stats,
when i run a test call that hits Queue 1 is unanswered then goes to Queue 2 then Voicemail the call stats show the call as unanswered in Queue 2 and not queue 1
so i don’t get a correct report on missed calls from Queue one.
anyone had this issue or suggest a way round it to get more accurate stats
Posts: 1
Participants: 1
@steve_pbuk wrote:
I’m happy to raise this as a support ticket but wanted to check first to see if this is a known issue. Or maybe even something that has been done by design.
We use a FreePBX server with Sangoma S500 VoIP phones. All of the S500s connect to the PBX via the VPN feature build into Endpoint Manager.
This works great and is very reliable most of the time.
The only time this setup seems to be unreliable is if I want to make changes. Below are a few examples:
Adding a new extension
If I add a new extension every phone that is connected via VPN will become “unavailable” under the “pjsip show contacts” command. At this point all calls drop and the phones can not make any calls. This situation happens as soon as I add the extension - before I click “Apply” in the top right.Phone template changes
If I make a basic change such as renaming a button in the phone template and then save and update the phones, it will cause all phones using this template to reboot. This only happens when the phones are connected via the VPN. This is a nightmare because some computers are daisy chained to the phones for network access.Firmware Updates
Not 100% certain this is related yet but the few times I have pushed out phone firmware updates I seem to run into issues such as phones constantly showing a messaging saying they are updating firmware. I did some testing and updates seems to work OK if I did a couple of phones at a time. Seem to run into problems when I did 8+ phones at a time. Not sure if this is related to VPNs or not though!Can anyone from Sangoma advise if points 1 and 2 are to be expected? I don’t remember seeing anything related to this in the Wiki when I first setup the VPNs.
Server is almost fully up to date. Last ran updates a few weeks ago. Below are the current module versions I have.
Phone firmware is: 2.0.4.55
+---------------------+-------------+----------------------------------------+------------+ | Module | Version | Status | License | +---------------------+-------------+----------------------------------------+------------+ | accountcodepreserve | 13.0.2.2 | Enabled | GPLv2 | | amd | 13.0.2 | Enabled | GPLv3+ | | announcement | 13.0.7.3 | Enabled | GPLv3+ | | areminder | 14.0.4.2 | Enabled | Commercial | | arimanager | 13.0.4 | Enabled | GPLv3+ | | asterisk-cli | 14.0.1 | Enabled | GPLv3+ | | asteriskinfo | 13.0.7.1 | Enabled | GPLv3+ | | backup | 14.0.10.1 | Enabled | GPLv3+ | | blacklist | 14.0.1 | Enabled | GPLv3+ | | broadcast | 14.0.1.9 | Enabled | Commercial | | builtin | | Enabled | | | bulkhandler | 13.0.14.7 | Enabled | GPLv3+ | | calendar | 14.0.2.6 | Enabled | GPLv3+ | | callback | 13.0.5.2 | Enabled | GPLv3+ | | callerid | 13.0.8.13 | Enabled | Commercial | | callforward | 14.0.1.3 | Enabled | AGPLv3+ | | calllimit | 13.0.5.5 | Enabled | Commercial | | callrecording | 14.0.3 | Enabled | AGPLv3+ | | callwaiting | 14.0.1.1 | Enabled | GPLv3+ | | campon | 13.0.4.1 | Enabled | GPLv3+ | | cdr | 14.0.5.14 | Enabled | GPLv3+ | | cel | 14.0.2.8 | Enabled | GPLv3+ | | certman | 14.0.3.1 | Enabled | AGPLv3+ | | cidlookup | 14.0.1.7 | Enabled | GPLv3+ | | conferences | 13.0.23.12 | Enabled | GPLv3+ | | conferencespro | 14.0.2.5 | Enabled | Commercial | | configedit | 13.0.7.1 | Enabled | AGPLv3+ | | contactmanager | 14.0.4.9 | Enabled | GPLv3+ | | core | 14.0.18.36 | Enabled | GPLv3+ | | cos | 13.0.12.2 | Enabled | Commercial | | customappsreg | 13.0.5.4 | Enabled | GPLv3+ | | cxpanel | 14.0.1 | Enabled | GPLv3 | | dahdiconfig | 14.0.1.2 | Enabled | GPLv3+ | | dashboard | 14.0.3.3 | Enabled | AGPLv3+ | | daynight | 14.0.1 | Enabled | GPLv3+ | | dictate | 13.0.5 | Enabled | GPLv3+ | | digium_phones | 13.0.7.4 | Enabled | GPLv2 | | directory | 13.0.19.5 | Enabled | GPLv3+ | | disa | 13.0.6.6 | Enabled | AGPLv3+ | | donotdisturb | 14.0.1.1 | Enabled | GPLv3+ | | endpoint | 14.0.2.145 | Enabled | Commercial | | extensionroutes | 13.0.10.7 | Enabled | Commercial | | fax | 14.0.2.5 | Enabled | GPLv3+ | | faxpro | 14.0.3 | Enabled | Commercial | | featurecodeadmin | 13.0.6.4 | Enabled | GPLv3+ | | findmefollow | 14.0.1.20 | Enabled | GPLv3+ | | firewall | 13.0.57.1 | Enabled | AGPLv3+ | | framework | 14.0.3.18 | Enabled | GPLv2+ | | fw_langpacks | 14.0.1 | Enabled | GPLv3+ | | hotelwakeup | 14.0.1.4 | Enabled | GPLv2 | | iaxsettings | 14.0.1.4 | Enabled | AGPLv3 | | infoservices | 13.0.1.3 | Enabled | GPLv2+ | | irc | 2.11.0.7 | Enabled | GPLv3+ | | ivr | 14.0.3 | Enabled | GPLv3+ | | languages | 14.0.1.2 | Enabled | GPLv3+ | | logfiles | 13.0.10.5 | Enabled | GPLv3+ | | manager | 13.0.2.5 | Enabled | GPLv2+ | | miscapps | 13.0.3.1 | Enabled | GPLv3+ | | miscdests | 13.0.5 | Enabled | GPLv3+ | | music | 13.0.22.3 | Enabled | GPLv3+ | | outroutemsg | 13.0.2.1 | Enabled | GPLv3+ | | paging | 14.0.4 | Enabled | GPLv3+ | | pagingpro | 14.0.2.12 | Enabled | Commercial | | parking | 13.0.19.8 | Enabled | GPLv3+ | | parkpro | 14.0.2 | Enabled | Commercial | | pbdirectory | 2.11.0.6 | Enabled | GPLv3+ | | phonebook | 13.0.6.1 | Enabled | GPLv3+ | | phpinfo | 13.0.2 | Enabled | GPLv2+ | | pinsets | 13.0.9 | Enabled | GPLv3+ | | pinsetspro | 13.0.9.12 | Enabled | Commercial | | pm2 | 13.0.5 | Enabled | AGPLv3+ | | pms | 14.0.2.22 | Enabled | Commercial | | presencestate | 14.0.1.7 | Enabled | GPLv3+ | | printextensions | 13.0.3.1 | Enabled | GPLv3+ | | queueprio | 13.0.2 | Enabled | GPLv3+ | | queues | 14.0.2.22 | Enabled | GPLv2+ | | qxact_reports | 14.0.6 | Enabled | Commercial | | recording_report | 14.0.1.15 | Enabled | Commercial | | recordings | 13.0.30.12 | Enabled | GPLv3+ | | restapi | 13.0.21.1 | Enabled | AGPLv3 | | restapps | 13.0.92.19 | Enabled | Commercial | | ringgroups | 14.0.1.5 | Enabled | GPLv3+ | | sangomacrm | 13.0.4.32 | Disabled; Pending upgrade to 14.0.1.10 | Commercial | | setcid | 13.0.6.2 | Enabled | GPLv3+ | | sipsettings | 14.0.27.5 | Enabled | AGPLv3+ | | sipstation | 14.0.1.8 | Enabled | Commercial | | sms | 14.0.4.5 | Enabled | Commercial | | soundlang | 14.0.5 | Enabled | GPLv3+ | | speeddial | 2.11.0.4 | Enabled | GPLv3+ | | superfecta | 14.0.7 | Enabled | GPLv2+ | | sysadmin | 14.0.16 | Enabled | Commercial | | timeconditions | 14.0.2.15 | Enabled | GPLv3+ | | tts | 13.0.10 | Enabled | GPLv3+ | | ttsengines | 13.0.7.3 | Enabled | AGPLv3 | | ucp | 14.0.2.10 | Enabled | AGPLv3+ | | userman | 14.0.3.43 | Enabled | AGPLv3+ | | vmblast | 13.0.8 | Enabled | GPLv3+ | | vmnotify | 14.0.1.1 | Enabled | Commercial | | voicemail | 14.0.2 | Enabled | GPLv3+ | | voicemail_report | 13.0.13.3 | Enabled | Commercial | | vqplus | 14.0.1.9 | Enabled | Commercial | | weakpasswords | 13.0.2 | Enabled | GPLv3+ | | webcallback | 13.0.11.2 | Enabled | Commercial | | webrtc | 14.0.3.7 | Enabled | GPLv3+ | | xmpp | 14.0.1.15 | Enabled | AGPLv3 | | zulu | 14.0.3.31.3 | Disabled; Pending upgrade to 14.0.4.6 | Commercial | +---------------------+-------------+----------------------------------------+------------+
Posts: 2
Participants: 1
@sattva wrote:
Good day,
when trying to add a new certificate from the mobile operator through the new menu of the certificate - the certificate was uploaded using the previously generated CSR Reference - an error appears:
“There was no error importing the certificate: openssl_x509_read (): cannot be coerced into an X509 certificate!”
Posts: 1
Participants: 1
@mvogel4949 wrote:
Has anyone found a reliable pull chord phone that could be used in a nursing home setting? Either SIP or Analog. I’ve found some options but they all tie to a proprietary system.
Posts: 1
Participants: 1
@bpatterson117 wrote:
We got a Grandstream GXV-3275 I made a template for it and assigned it a firmware version that said that it was applicable for the GXV3275. Once it loaded the firmware and rebooted, it wont boot passed the initial Grandstream Logo.
Anyone else have this issue?Has anyone been able to successfully load a firmware from SD card on this model of phone?
Posts: 1
Participants: 1
@cgallery wrote:
I’m running a fairly old version of FreePBX/Asterisk right now. All my incoming calls come into a ring group, and picking-up on any phone in that ring group causes the other phones to show the call as missed.
From speaking with people, it appears newer versions have changed this behavior so that a call answered at any extension causes the other extensions to receive a message that the call was received elsewhere, so they won’t show me a missed call message.
I got a real good deal on some GXP2160 phones and was thinking of upgrading to a newer version of FreePBX, but I wouldn’t want to change the way missed calls are reported, is there a way to adjust FreePBX/Asterisk to behave like it used to?
Posts: 1
Participants: 1
@SpringfieldCA wrote:
Hello everyone
I am new into the PBX system.
We run our server in Elastix and from there I managed all the extensions. However I am not able to find how to set up one extension as the main one that can receive the calls. I am sorry if am not using the right words for describing what I need but I am really lost with this.
Thanks in advance for your help!
Posts: 2
Participants: 2
@ctarbet wrote:
I currently have a FreePBX 2 server that needs to be replaced. T1 / PRI working great with Digium TE110P card.
The new replacement FreePBX 14 system is ready to go, except that the T1 / PRI connection doesn’t work. It shows a yellow alarm when connected. I’m using a different, but same model card (TE110P).
I have the card jumper set to T1 (not E1) and I’ve set other settings according to the PRI provider.
ESF/B8ZS
pri_cpe
National ISDN 2
Clock Source 0
Line Build Out 0db(<133 ft)
23 used channels
Posts: 1
Participants: 1
@kristiandg wrote:
Good morning, all.
I may have noticed something odd with the QUEUES module, but need to run it by some folks.
We want to make a queue that basically rings the longest-waiting agent, but if they’re on a call, the calls sit in queue for another idle agent. We also have a Join Announcement set to only play if no agents are available.
Sounds easy enough, but…
I set the queue to SKIP BUSY AGENTS, and for testing we put just 1 agent in the queue (static member). The first call goes straight to the agent (as expected - no announcement, just rings), but a second caller calls in, they hear the Join Announcement, then they get routed to the Failover Destination (in this case, voicemail). The second caller never gets placed in queue to wait for the agent to become available. We’re trying to implement Callback Queueing, but it never makes it to that step because the agent is on a call and the queue is essentially “closed” rather than waiting in queue.
On top of that, we’ve told it to speak the queue position and hold time (once). Neither happened. The Periodic Announcements were also set to use a QCB module every 1m30s, also never happened.
Skip Busy Agents: Yes
Music on Hold Class: Inherit (Agent Ringing)
Join Announcement: (When No Free Agents)
Max Wait: 3m
Max Wait: (tried both Strict and Loose)
Agent Timeout: 15s
Retry: None (also tried 15s and 60s) - see note below about the oddities on this setting
Auto Pause: No (also tried Yes)
Auto Pause on Busy: No (also tried Yes)
Auto Pause on Unavailable: No (also tried Yes)
Max Callers: 0 (also tried 2)
Join Empty: Yes
Leave Empty: No
Announcement Frequency: 15s (for testing)
Min Announcement Interval: 30s (for testing)
Announce Position: Yes
Announce Hold Time: Once
Breakout Type: Queue Callback
Queue Callback Module: QCB-Test
Repeat Frequency: 1m30sHere’s the really odd part about RETRY - when retry is set, the music on hold resets at that time, almost as if the call as left and re-entered the queue completely.
Thoughts? I’m perplexed…
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Participants: 1
@sentinelace wrote:
I need follow me to go to a cell phone even if the user is on the phone. It just rings forever. I enabled all the optional destinations to force follow me with the same result. If the phone is idle, it works perfect. What setting am I missing?
Posts: 2
Participants: 2
@ofir29200 wrote:
i just installed freepbx 12 on ubuntu 14.04.
i tried to install freepbx 13 on ubuntu 16.04 and centos 7, but idk why, the installation will always fail in the end with “cannot connect with asterisk” or something like that.
now that i have it running i got to a new issue, when i try to apply the config to add the extensions, i get this error:Reload failed because retrieve_conf encountered an error: 126
exit: 126
sh: 1: /var/lib/asterisk/bin/retrieve_conf: Permission deniedcan you help me with it?
thank you!
Posts: 3
Participants: 3
@stormforever1 wrote:
Hello All,
New to FreePBX only about 3 weeks in…came from a Cisco Call Manager and UCCX environment… FreePBX firmware: 10.13.66-22 service pack: 1.0.0.0… if I dial our conference line ext it plays a message then ask for you to enter the Conference ID once you hit any number it plays: we have not received a valid response and it disconnects the call…I am not sure if I am looking into the correct area: Applications - IVR - ConfBridge
Enable Direct Dial: Disabled
Timeout: 10
Alert Info: None
Ringer Volume Override: None
Invalid Retries: 0
Append Announcement to Invalid: No
Return on Invalid: No
Invalid Recording: Default
Invalid Destination: None
Timeout Retries: 0
Append Announcement on Timeout: No
Return on Timeout: No
Timeout Recording: Default
Timeout Destination: None
Return to IVR after VM: No
Posts: 1
Participants: 1
@Adalis wrote:
The customer service manager at one of our clients has two DID’s terminating to her extension. One originating in California and one in Colorado. FMFM is setup on the extension to go to her cell phone if she is away from her desk.
The Colorado DID and any call coming through their system works as configured. “Confirm Calls” is turned on. The California DID rings the FMFM, the deskphone rings and if unanswered the cell phone rings once and immediately goes to the cell phone voicemail.
The inbound routes are setup exactly the same, no privacy or other settings turned on in one or the other differently.
FMFM is configured to ring the deskphone 7 seconds and then the FMFM list (the extension and the cell number with the #) for 30 seconds. “No Answer” is setup for normal behavior.
This extension is in a ring group, but the issue does not occur when coming though the system only when the DID is dialed directly.
Not sure why this would matter, but the “Max Contacts” is set to 3 in advanced on the extension.
Appreciate any suggestions or questions…
Posts: 2
Participants: 2
@gitim wrote:
HI
I am following this instruction to apply cisco call manager path
and
Freepbx version 14, asterisk version 14 and running
rpmbuild -ba asterisk14.spec
I get this error, while libxml2 development is installed already, any idea how I can fix this?
Posts: 1
Participants: 1
@christchurch wrote:
TLDR:
Seems like FreePBX/Asterisk box is intermittently not relaying the RTP stream from a Snom PA1 door intercom to Snom 710 extension phone.Situation:
Installed FreePBX (on dedicated on site Dell Server) and Snom phones last year, all has been working fine since. Recently tried installing a Snom PA1 as a door intercom. Basically just a Snom phone in a box with a loudspeaker and relays for the door. Person at door presses a button, the PA1 dials a ringgroup, the ringgroup dials some extensions, somebody answers, lets the person in by pressing 1# (this activates relay 1 for several seconds). I have 7 buttons on the door and created 7 ring groups (one for each button). Some buttons just call one phone and some call multiple using the ‘ringall’ strategy. Other benefit of using a ringgroup it can be set to just terminate the call after 20 seconds rather than go to answerphone which we dont want.So I get the sytem in place and all seems to be working fine, then I start getting complaints that sometimes it isn’t working, the person picking up the phone cannot hear the person at the door (but the person at the door can hear them). So I set call recording up on all the ring groups, ensure Asterisk is logging, and I also setup a monitor port on our switch so I can capture all packets going to and from the FreePBX box using Wireshark on another server. The results are below and this is where I get stuck… the only thing I can spot is that the FreePBX box is not relaying the RTP stream from the intercom to the phone, but is relaying the stream from the phone to the intercom just fine. I have also tried using a dummy extension and FM/FM instead of a ringgroup and still have the same problem. The data below is using a dummy extension 364 and FM/FM.
Some information:
PA1 Intercom : ext 401 Name: KC Door IP: 192.168.0.33 FWver: snomPA1-SIP 8.7.5.75
710 Phone : ext 416 Name: John Dixon IP: 192.168.0.38 FWver: snom710-SIP 8.9.3.80
DummyExt : ext 364 FM/FM to extension 416
FreePBX : IP: 192.168.0.200 FWver: 14.0.3.19 / 12.7.5-1807-1.sng7Allowed CODECs on Snom devices (Snom defaults):
g722,pcmu,pcma,gsm,g726-32,aal2-g726-32,g723,g729,telephone-eventAllowed CODECs on FreePBX:
ulaw,alaw,gsm,g726Don’t think it’s a CODEC problem as all other devices have been working fine and according to the Wireshark logs the PA1 ia correctly sending g711U to the correct FreePBX port. The RTP audio stream from the intercom decodes fine in Wireshark (you can hear it) but the call recording in FreePBX does not have this audio, it only records the audio from the phone (ext416) not from the intercom (ext401).
All are connected to the same LAN switch in the same building.
Please see attached logs… remember this is an intermittent problem, sometimes it works OK, sometimes it doesn’t. It always seems to work fine on an evening when I am in on my own testing, but not so much during the day when people are in the offices! Not sure if that is scientific!
The files with “working” are from a time when it worked just fine, the files with “not working” are from a time when it didn’t work, but all other things being equal.
Any help greatly appreciated as it’s driving me nuts! Cheers!
Posts: 1
Participants: 1
@aerodes wrote:
Hi everyone,
I want to install FOP2 for FreePBX 14.0.3.17. I read from FOP2 website that can be done with this bash script. Does it really works?
wget -O - http://download.fop2.com/install_fop2.sh | bash.I want some feedback.
Regards.
Posts: 1
Participants: 1
@raymonvdm wrote:
I`m using multiple trunk for outbound calls and i have setup 3 outbound routes.
Route1 if the user calls 0xxxxxxxxxx the calls are routed trunk 1
Route2 if the user calls 9xxxxxxxxxx the calls are routed trunk 2
Route3 if the user calls 99xxxxxxxxxx the calls are router trunk 3Under the Trunk Dial Number Manipulation Rules
i want to strip of the additional 0 , 9 and 99 and add 31 to it, but i don`t understand the logic of the prepend - prefix - match patern
Posts: 1
Participants: 1
@nortelvoip wrote:
I have a PBX setup at my house and I keep getting calls that when answered are showing my own caller id. When I answer it is a Microsoft licensing scam. What is the best way to block these calls? I’m on vitelity sip trunks.
Posts: 1
Participants: 1
