Teamviewer is not a better solution than any of the three Stewart listed…in fact it’s far worse because SNG7 is not intended to support an X environment and will likely break again in the future even if you get it working now. I trust the FreePBX web interface, VPN and ssh far more than teamviewer. Sounds like maybe you need a cloud installation of FreePBX - if you really expect a router is going to get reset or switched out out of your control, hosting a PBX at that office is going to be a problematic because you will lose all access and FreePBX will almost certainly stop working at some point. SIP servers are just sensitive to firewall and routing issues.
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Install XFCE on FreePBX OS
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Remote Extensions drop registration after a while and never recover
All right, here is a fresh Astersk log relating to extensions 250 251…now it is on now it is off
[2018-05-13 09:40:39] NOTICE[3012] chan_sip.c: Peer ‘251’ is now Reachable. (21ms / 2000ms)
[2018-05-13 09:41:01] NOTICE[2932] chan_iax2.c: Restricting registration for peer ‘329’ to 60 seconds (requested 300)
[2018-05-13 09:41:01] NOTICE[2938] chan_iax2.c: Restricting registration for peer ‘477’ to 60 seconds (requested 300)
[2018-05-13 09:41:24] NOTICE[3012] chan_sip.c: Peer ‘250’ is now Reachable. (18ms / 2000ms)
[2018-05-13 09:41:43] NOTICE[3012] chan_sip.c: Peer ‘251’ is now UNREACHABLE! Last qualify: 21
[2018-05-13 09:41:56] NOTICE[2941] chan_iax2.c: Restricting registration for peer ‘329’ to 60 seconds (requested 300)
[2018-05-13 09:41:56] NOTICE[2937] chan_iax2.c: Restricting registration for peer ‘477’ to 60 seconds (requested 300)
[2018-05-13 09:42:28] NOTICE[3012] chan_sip.c: Peer ‘250’ is now UNREACHABLE! Last qualify: 18
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Can’t get FreePBX to fully work with ht503
Hi Ariel can you please check my previous unswer I have posted also some log files
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Install XFCE on FreePBX OS
Thanks for your advice. Problem is that I will receive a physical antenna which I need to attach to hardware myself. So I need to actually have physical hardware in order to make the setup work.
Maybe I could connect the freepbx box that’s attached to the antenna to a cloud installation but I’m afraid it will add extra latency to the calls. And server prices in the (Asian) country I’m deploying are expensive so it would also add unnecessary cost.
You guys have inspired me to dive into dynamic DNS as a potential solution though, I will try to make it work without GUI and teamviewer.
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Announcement Won't Play
More testing is interesting.
I’ve added another new trunk using pjsip.
Connectivity, Inbound Route, (for this DID), Set Destination = Announcements, Select Exterminate Call.
When called, asterisk plays the exterminate message as expected.
So the Announcement seems to work with chan_pjsip trunk, but not with chan_dongle.
Could this be correct ?
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Install XFCE on FreePBX OS
Put in a raspberry pi with Anyedsk or Teamview and control it that way. That will give you the access you need and you dont need to mess with the pbx
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Moving from SIP to PJSIP
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Can’t get FreePBX to fully work with ht503
You have probably defined your extension and trunk incorrectly, specifically the UDP port. For the extension, use port 5060, for the trunk use port 5062
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Can’t get FreePBX to fully work with ht503
pjsip and these kinds of FXO gateways don’t play well together. In pjsip, endpoints are normally identified by IP address (won’t work if you are using both FXS and FXO on the same device) or ‘User ID’ in the From header (won’t work if you are sending caller ID on incoming FXO calls). See No incoming calls (this is about a Linksys SPA3102, but it basically does the same thing as an HT503 and has the same issues).
IMO the easy solution is to use a chan_sip trunk for the FXO. Though not tested, there are two solutions that should work with pjsip. One is the identify_by=auth_username parameter as suggested by @avayax . The other is the ‘line’ parameter, normally used to distinguish multiple trunks with the same provider and discussed in various threads.
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Can’t get FreePBX to fully work with ht503
This is apparently fixed in:
core version 14.0.18.1
core version 13.0.122.21
I haven’t tried the fix yet and I don’t know how it’s been implemented, but my ticket on this issue has been addressed:
https://issues.freepbx.org/browse/FREEPBX-17158?filter=-2
So PJSIP should work fine with those gateways on those core versions.
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How to setup a Gamma PJSIP trunk
I have got my FreePBX 14 system installed and received my IP details from Gamma, but I am struggling to setup the PJSIP trunk. This is the first time I have used FreePBX so sorry if I am just being stupid.
So far all I have is the Gamma IP in ‘SIP server IP’, port 5060 in SIP server port, and set authentication to none as Gamma uses IP authentication. With these settings I can receive calls but cannot make calls.
Any help would be appreciated.
Justin
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How to setup a Gamma PJSIP trunk
I know nothing about Gamma but in general, this is easy to debug. At the Asterisk command prompt, do
pjsip set logger on
then attempt an outgoing call. Your outbound INVITE will appear in the Asterisk log, as well as any reply.
If there is a reply (403, 404, 503, etc.) that should be a clue as to what is wrong. There may also be a Warning or other header in the reply with additional information about the error.
If there is no reply at all, check that you have correctly set your external IP address in your list of authorized addresses in the Gamma portal. Also check that the Via header in your INVITE contains your external IP address (if not and your PBX is behind a NAT, check that the NAT settings in Asterisk SIP settings are correct and restart Asterisk if you change them).
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Manager, task processor
You mean in manager_custom.conf? Like this?
eventfilter=Event: AgentCalled
eventfilter=Event: NewState
eventfilter=Event: AgentConnect
eventfilter=Event: QueueMember
eventfilter=Event: QueueStatusComplete
eventfilter=Event: QueueMemberPaused
eventfilter=Event: QueueMemberAdded
eventfilter=Event: QueueMemberRemoved
eventfilter=Event: QueueMemberStatus
eventfilter=!Event: RTCP*
eventfilter=!Event: VarSet
eventfilter=!Event: Cdr
eventfilter=!Event: DTMF
eventfilter=!Event: AGIExec
eventfilter=!Event: ExtensionStatus
eventfilter=!Event: ChannelUpdate
eventfilter=!Event: ChallengeSent
eventfilter=!Event: SuccessfulAuth
eventfilter=!Event: HangupRequest
eventfilter=!Event: SoftHangupRequest
eventfilter=!Event: NewAccountCode
eventfilter=!Event: MusicOnHold
eventfilter=!Event: LocalBridge
We are already doing this.
Any other ideas / suggestions are very welcome.
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Add Stasis application in dialplan (to use Asterisk ARI with FreePBX)
I want to use Asterisk REST Interface (ARI) with FreePBX :
wiki.asterisk.org/wiki/display/AST/Getting+Started+with+ARI
In this tutorial, we have to do this :
3) Create a dialplan extension for your Stasis application. Here, we’re choosing extension 1000 in context default - if your SIP phone is configured for a different context, adjust accordingly.
extensions.conf
[default]
exten => 1000,1,NoOp()
same => n,Answer()
same => n,Stasis(hello-world)
same => n,Hangup()
In FreePBX, we can’t edit the file extensions.conf directly (“Do NOT edit this file as it is auto-generated by FreePBX.” in the header of this file).
So, I have to add my dialplan in extensions_custom.conf
But I don’t know, what I have to write, to add Statis application for all incoming calls like in this tutorial.
When i try to connect to Asterisk using wscat:
$ wscat -c “ws://localhost:8088/ari/events?api_key=asterisk:asterisk&app=hello-world”
connected (press CTRL+C to quit)
I CAN’T see this message:
== WebSocket connection from ‘127.0.0.1:37872’ for protocol ‘’ accepted using version ‘13’
Creating Stasis app ‘hello-world’
How I can do this ?
Someone here has used ARI (Asterisk REST Interface) with FreePBX ?
Thank you.
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Add Stasis application in dialplan (to use Asterisk ARI with FreePBX)
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Polycom Color Expansion Module not populating
This seems to be a recurring subject, but I haven’t found a solution in the forums or issue tracker so I’ll post again…
We got a Polycom VVX501 phone and color expansion module for testing with a different phone system and seeing as it’s a very nicely designed device and Polycom devices tend to be extremely cross-platform friendly, we’d take a shot at getting it to work with FreePBX and the Commercial Endpoint Manager. The phone itself seems fine, but the expansion module isn’t populating. I think we’ve got the settings correct (added EXP-COLOR-1, set some speed dials and a BLM key in the EXP section of the device configuration, and … they show up on the phone itself rather than the expansion module. The configuration files generated appear to be correct, but I don’t have a huge amount of experience with that level of detail with provisioning Polycom devices.
Has anyone run into this and figured out a solution they’re willing to share?
One thing I did find is that Polycom’s UC software gets cranky if you have duplicate destinations set for speed dials - it logs an error and discards them. Not sure if this can be sanely sanity-checked by Endpoint Manager (it may be a big corner case that requires excessive effort into for one vendor), but it’s something to be aware of.
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Can’t get FreePBX to fully work with ht503
I had the fxs 5060 fxo 5062 , uncaditional forward to VoIP setting 5062 and trunk settings in fpbx 5062 but with this configuration I never got any incoming calls to FreePBX , when I changed the unconditional forward to 5060 I got incoming calls but the fxs can’t get registered but I got calls in the other internals . Do I have to configure the fxo port as an extension to fpbx in order to work and if yes do I have to I configure it ?
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Multiple Fail2Ban notifications.... where should I be looking?
This appears to be fixed in firewall 13.0.55.1 for me
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Can’t get FreePBX to fully work with ht503
You are confusing the gateway ports with the pbx ports. The unconditional forward must be directed to the port where the pbx is listening and the trunk must be directed to the port where the gateway is listening.
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Inbound calls from PSTN to FreePBX through Grandstream GXW410X FXO Gateway fail
I am trying to configure a Grandstream GXW410x as a trunk on my FreePBX server.
While I am able to make outgoing calls through the GXW410 without an issue. When I receive an incoming call, the extensions never ring and the following error is displayed in the FreePBX log chan_sip.c: username mismatch, have , digest has <>.
I followed these instructions to configure the GXW410 to work with FreePBX.
I also tried adding match_auth_username=yes from some forum posts I found with similar issues with other gateways, but this did not resolve the error.
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