Anybody can recommend good wholesale DID provider (s) for US and International DID’s
Thanks
Himala
Anybody can recommend good wholesale DID provider (s) for US and International DID’s
Thanks
Himala
As @dicko said, you can set up multiple trunks, one for each IP address from which the provider can send calls.
Or, you can use a single trunk and add two lines to sip_additional_custom.conf for each address; see example for provider Callcentric https://www.callcentric.com/support/device/asterisk/14 .
Or, with a single trunk, you can add some custom dial plan to distinguish your provider’s calls based on the From domain; see https://www.callcentric.com/support/device/did_trixbox .
Or, you can use pjsip. If the provider lists all his hosts in his DNS, it’s automatic. Otherwise, you can put a list of addresses or networks in the Match parameter.
Flowroute works well.
For that last bit, you just need to go through all you flows more carefully to ensure there is no way into the queue after hours.
For the first bit, no idea off the top of my head.
For US DIDs… lots of providers to pick from.
For Intl DIDs… I use Twilio for some and DIDWW for others.
tcp 0 0 securesipwilder.copoc:mysql securesipwilder.copoc:47278 ESTABLISHED
I don’t have one in front of me, but is comes down to a setting in the phone and the PBX.
There is an option in both the phone and FreePBX on how to handle MWI subscriptions.
I would suggest changing the phone and leaving FreePBX default.
netstat -ntpa|grep 3306
@Stewart1 I never tested the PJSIP driver for trunk. It supposed to work like a chan_sip trunk ? Or my provider has to be also in PJSIP ?
(post withdrawn by author, will be automatically deleted in 24 hours unless flagged)
That is as suspected the ucp process, disable it temporarily and see what happens
Take a look at
http://www.anveodirect.com/
https://www.voxbeam.com/dids
https://www.alcazarnetworks.com/origination.php (US and Canada only)
https://starcompartners.com/origination#wholesale (US and Canada only)
I agree with @adolfoc on Twilio and DIDWW.
For better advice, please estimate your requirements:
Quantity of DIDs?
Total monthly minutes?
Number of channels?
Need toll-free?
Need SMS?
Failover handling?
Will numbers be ported from another provider?
Countries needed?
unfortunately this doesn’t works even putting in the server name( i tried everythink). About messagenet I
just tried with same results, only a little better with g722 (ehiweb doesn’t support g722) but I think that at this point it’s clear that is fanvil poor quality firmware related problem, I just tried with an old cisco SPA514G and there is no delay in the echo test after 30 seconds and voice quality is really better, unfortunately I have only one of these cisco so I can’t test internal calls. anyway I want to really thank you for your support and explanations.
regards
There is no problem with pjsip at one end and chan_sip on the other. Unless your provider is quite small, he likely isn’t even using Asterisk.
If you are still in the evaluation / testing stages, I recommend that you give pjsip a try. If it works right away or with reasonable effort, stick with it. For handling calls coming from addresses other than the one to which you registered, IMO pjsip does a much better job.
The system is playing your call, but you aren’t hearing it. Make sure the file referenced actually exists and make sure the ‘.slin’ version is available to the server.
Out on a limb: maybe a codec problem? Don’t you have to transcode everything on your dongle to GSM?
This is just the craziest thing. I disabled ucp and ucp node server. I restarted mysqld and pretty soon its back
Thanks for trying to help. I feel it has something to do with the HTML pages as I can get to User Control Panel and all the menus show up for selecting… just nothing displays on the dashboard.
Disable fop2
service fop2 stop
New to FreePBX, and am trying to set it up with Cisco SPA514G phones. Installed FreePBX 14.0.3.6 from the ISO on it’s own dedicated machine. Don’t have any trunk lines - just wanted to set this up internally for testing. I’ve gotten to the point where I’ve created extensions, and enabled voicemail. Went to go setup voicemail and get the prompt to “enter your name and press #”. I press the # button but the system does not recognize it. Played around with other menus as well, i.e. accessing voice mail where you can press keys 2, 0, *, # etc and pressing the keys does not register with the system. Not sure at this point why the keys aren’t working - any pointers would be appreciated!