You have to also add ##* into the dialplan on the Sangoma phone.
Account -> Advanced -> Dial Plan (on a S500)
Also you can setup Sangoma phones so you can press transfer and long press a BLF button for a phone and it will transfer the call to voicemail.
I have not programmed a Polycom phone yet so sorry I dont know much about them.
Yes. Use a ring group.
You can’t remove that. It’s part of the token check
I don’t understand your last post.
For example, assume that your house/office subnet is 192.168.1.0/24 and the garage subnet is 192.168.2.0/24.
Please explain any differences from these assumptions:
Also, please post whether the SPA122 is using pjsip or chan_sip, and the make and model of relevant networking devices.
This question has many answers, but more importantly would be what requirements do you have.
An example, if you have a few locations with 30+ users and you loose internet, you may still want the users inside the office to be able to communicate internally, that would then require a PBX at each location.
Now you would use IAX to communicate between offices, this can be done via the web, or a more secure answer is that vpn you mentioned.
If you a smaller offices, maybe the above isn’t required., so you use 1 PBX for all users, then from the firewalls at each location configure your VPN and configure the multiple networks into your single pbx.
I would check SIP IP address settings.
Please tell us the rest of the story. If you have some administrative control over said NAT and can forward the required ports, it doesn’t matter what kind of NAT you have.
If all your extensions are internal (on the same LAN as the PBX) and all your trunks are analog, PRI and/or GSM gateways, your server doesn’t need to be connected to the internet so it also doesn’t matter what kind of NAT you have.
After comparing the guide on the wiki and what I have done here.
in my guide make samples generates /etc/asterisk/asterisk.conf (as billsimon mentioned) which I modify to set the runuser and rungroup
and next I ran asterisk using init.d, I thought that is the reason that make samples was keeping all the errors away for my setup.
The wiki guide on the other hand does everything as the root user, and so I thought I had a solution to make my install setup work.
I tried replacing:
sudo make samples
with:
sudo cp configs/samples/asterisk.conf.sample /etc/asterisk/asterisk.conf
This way I don’t actually have to run make samples, and can still have asterisk.conf for asterisk to run without root.
however, same result, and that is the only line that I changed
I do not know why using the make samples command would prevent the errors, but it certainly seems to, I have installed start to finish 10+ times with various configurations.
I am stumped for the moment, but I did do some more cleanup on the guide, and borrowed a few things from the guide on the wiki.
I took a look at your original post and the no audio.pcapng file. Though the PBX <-> MacBook SIP is there, I unfortunately couldn’t see the SIP from/to the originating extension (on China Mobile?), I’m guessing that frpc is running on the Pi and the internally looped traffic is not captured by Wireshark.
My suspicion is that Asterisk views the external extension as local so doesn’t substitute the public IP address in the SDP.
Could you run FreePBX in your Google Cloud server instead of on the Pi? If not, please explain your constraints. I documented a ‘cookbook’ solution for that; see https://www.dslreports.com/forum/r31813676-Almost-free-PBX-in-the-Google-Cloud .
Or, set up an OpenVPN server on the Pi and forward its UDP port via FRP. The mobile should then be able to get a VPN connection to the Pi and appear as a local extension.
Asterisk thinks 522 is busy, and doesn’t even attempt to dial.
to start with… i am trying to install with centos7… so i know its not the dist your using…
for some reason i cant seem to get PJsip to install… i have used modified version of your intrustctions and i have used the intructions just to install centos 7 freepbx 14 with no pathc. when ever i log in to asterisk and try a command like pjsip show endpoints it says no such command… however when i run the same command in my production freepbx the commands seem to work.
Chech that the pjsip modules are not ‘noload’ in
/etc/asterisk/modules.conf
I’ve never encountered someone having so many issues with not running make samples.
The distro, which is installed hundreds of times a day, uses the same methodology to install freepbx as is in the wiki. We never run make samples and we never add asterisk.conf. We never have as many issues as you are having.
Again. As Dicko pointed out. When you run make samples you expose your system. Even the asterisk wiki says this.
I can endorse @tm1000 's last post, personally I have been installing asterisk from source for various platforms for many (many) years and literally hundreds of time.
I discovered very early on that “make samples” NEVER was useful, all the files as asterisk puts it “for reference” are in
/usr/src/asterisk-14.6.0/configs/samples
/usr/src/asterisk-14.6.0/configs/basic-pbx
is a working PBX
If you really MUST run make samples, I would caution you to look in the Makefile and hold very close to your head the ominous:-
# Overwite config files on "make samples" or other config installation targets
OVERWRITE=y
JM2CWAE
i am working on trying to get GVSIP going… but i am stuck having issues even getting PJSIP installed even as as part of a standard build. using the guild located here… https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7
i am not seeing any errors displayed as i walk thru tall the steps.
howerver when ever i log in to asterisk and try a command like pjsip show endpoints it says no such command… however when i run the same command in my production freepbx the commands seem to work.
there is nothing about PjSIP in /etc/asterisk/modules.conf
i used CentOS-7-x86_64-Minimal-1804.iso that i just downloaded the other day.
i have probably attempted building this 20 times…
any help would be great… probably somethings simple i am over looking.
looks like it installed Asterisk 14.5.0 the last time i tried
nothing in there… i have started ta new thread…
side note Xekon… why are you installing asterisk with just sudo ./configure… as this is 13.21.1 it really needs --with-pjproject-bundled unless i am missing somthing.
Actually the 13.21.1 is an artificial artifact to play happy with FreePBX, the branch is Master and if you cared to
./configure --help
you would maybe notice your “missing” thing
--with-pjproject-bundled
Use bundled pjproject libraries (default)
but there is no harm adding that switch if you want
When you “make menuselect” check that all the res_pj* are checked in “Resource Modules” and pjsip in the “channel drivers”.
Even if the modules are built they wont be loaded if there is an unfulfilled precondition, from asterisk
module load chan_pjsip
and take it from there (Personally I tripped up on aggressively using statsd (stasis) )
Which is why I keep trying to find different ways of removing make samples, but every time I remove that line I have errors after Freepbx install, it really makes no sense at all. I am going to keep trying to figure it out.
I think the next step instead of guessing, is to look at everything it does, actually find that section of the code in the asterisk install files so that I can closely examine it line for line. Maybe then something would click.
after veifiy makemenu had everything selected i tried running module load chan_pjsip and i am now getting this
2018-07-07 13:24:00] WARNING[163829]: loader.c:526 load_dynamic_module: Error loading module ‘chan_pjsip’: /usr/lib64/asterisk/modules/chan_pjsip.so: undefined symbol: ast_sip_cli_traverse_objects
That behavior is, as ever, all defined in “Makefile” and conveniently in the “samples” section 