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By @UKVoIPForums
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#5825
With Asterisk and FreePBX moving closer to the removal of chan_sip I decided to make the switch myself. Below is a copy of my Voipfone PJSIP settings that I configured a few days ago with FreePBX 13.0.197.22 and so far so good. :nerd:

First thing you will need to do is enable the "SIP Channel Driver" to use both chan_sip and chan_pjsip. You can do that by navigating to the Settings > Advanced Settings configuration page and scrolling down until you see the "SIP Channel Driver" setting. Make sure that "both" is selected in the dropdown box. If it isn't, make the change and click "Submit" to save it and then "Apply Changes" to implement it.

Now that's out of the way, let's create a Voipfone PJSIP Trunk...
  1. Navigate to Connectivity > Trunks > + Add Trunk > Add SIP (chan_pjsip) Trunk page.
    You should be at the following screen:
    Image
  2. Under the "General" tab section make the following changes:
    • Trunk Name = Voipfone-(ACCOUNT_NUMBER)
    • Outbound CallerID = (PHONE_NUMBER)
    It should look something like this:
    Image
  3. Click on the "pjsip Settings" tab and in the "General" tab section make the following changes:
    • Username = (ACCOUNT_NUMBER)
    • Secret = (ACCOUNT_PASSWORD)
    • Authentication = Outbound
    • Registration = Send
    • Language Code = English - United Kingdom
    • SIP Server = sip.voipfone.net
    • SIP Server Port = 5060
    • Context = from-pstn
    • Transport = 0.0.0.0-udp
    Again, it should look like this:
    Image
  4. While still in the "pjsip Settings" tab click on the "Advanced" tab and make the following changes:
    • General Retry Interval = 20
    • Expiration = 60
    • Contact User = (ACCOUNT_NUMBER)
    • From Domain = sip.voipfone.net
    • From User = (ACCOUNT_NUMBER)
    • Client URI = sip:(ACCOUNT_NUMBER)@(YOUR_IP):5060
    • Server URI = sip:(ACCOUNT_NUMBER)@46.31.225.185:5060
    • AOR Contact = sip:(ACCOUNT_NUMBER)@46.31.225.185:5060
    • Match (Permit) = 46.31.225.185,(YOUR_IP)
    You can leave the rest at their defaults.
    Image
    Image
  5. Finally click on the "Submit" button to save your changes and then "Apply Changes" to implement them.

Additional information on how to convert extensions from chan_sip to chan_pjsip can be read here: WelshPaul @ FREEPBX - New tool to assist converting from SIP to PJSIP


NOTE: Replace the following (in the above configuration) with your real Voipfone credentials:
  1. (ACCOUNT_NUMBER) = Your Voipfone account number (the 8-digit number starting with 3) followed by star and your 3-digit extension number.
  2. (ACCOUNT_PASSWORD) = Your Voipfone extension account password. Passwords for all your extensions can be found in your Control Panel.
  3. (YOUR_IP) = Your IP Address (Find it here: https://www.whatsmyip.org)
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By WelshPaul
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#5980
Hi @nasarz the PJSIP trunk settings for a Sipgate Basic account in UK is as follows:

Tab General:
TrunkName: SipgatePJSIP (or whatever you want to call it)
Outbound caller ID: <YourPhoneNumber> e.g. “<01792123456>”

Tab pjsip Settings - General
Username: sipgate acount ID, e.g. “1234567a0”
Secret: sipgate account password
SIP Server: sipgate.co.uk
SIP Server Port: 5060
Context: from-pstn

Tab pjsip Settings - Advanced
Expiration: 600
From User: sipgate acount ID, e.g. “1234567a0” (if you get this wrong, you’ll hear sth like “all lines are busy, pls try again later” on outgoing calls)
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