- Wed 6th Jan 2021, 00:05
#6075
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Before we start, you should understand that A&A’s offer two different methods we can use to register a trunk against:
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 an Andrews & Arnold PJSIP Trunk...
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 Andrews & Arnold credentials:
- SIP Phone
- To your server via SIP
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 an Andrews & Arnold PJSIP Trunk...
- Navigate to Connectivity > Trunks > + Add Trunk > Add SIP (chan_pjsip) Trunk page.
You should be at the following screen:
- Under the "General" tab section make the following changes:
- Trunk Name = AAISP-(441234567890)
- Outbound CallerID = (PHONE_NUMBER)
- Click on the "pjsip Settings" tab and in the "General" tab section make the following changes:
- Username = (ACCOUNT)
- Secret = (PASSWORD)
- Authentication = Outbound
- Registration = Send
- Language Code = English - United Kingdom
- SIP Server = voiceless.aa.net.uk
- SIP Server Port = 5060
- Context = from-trunk
- Transport = 0.0.0.0-udp
- While still in the "pjsip Settings" tab click on the "Advanced" tab and make the following changes:
- DTMF Mode = RFC 4733
- Forbidden Retry Interval = 20
- Fatal Retry Interval = 20
- General Retry Interval = 20
- Expiration = 60
- Qualify Frequency = 20
- Contact User = (ACCOUNT)
- From Domain = (YOUR_IP)
- From User = (ACCOUNT)
- Client URI = sip:(ACCOUNT)@voiceless.aa.net.uk:5060
- Server URI = sip:voiceless.aa.net.uk:5060
- AOR Contact = sip:(ACCOUNT)@voiceless.aa.net.uk:5060
- Match (Permit) = voiceless.aa.net.uk
- Click on the "Codec" tab and select the following codecs:
- alaw
- ulaw
- 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 Andrews & Arnold credentials:
- (ACCOUNT) = Your Andrews & Arnold phone number in international format (+441234567890).
- (PASSWORD) = Your Andrews & Arnold SIP Password. Your SIP Password can be found in the Andrews & Arnold Control Pages under the "Outgoing" tab.
- (YOUR_IP) = Your IP Address (Find it here: Settings > Asterisk SIP Settings > NAT Settings > External Address)
- Appreciate my support? Feel free to buy me a coffee.
- Voipfone are offering you the chance to trial their VoIP service for free for 30 days. Sign me up!
- Sign up for a Dropbox account using my referral link and get an extra 500 MB of bonus space. Sign me up!
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