VoIP Phones - Asterisk: Difference between revisions

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Don't feel comfortable with live number in example. No point in allowing ulaw because AAISP don't use it

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Revision as of 08:57, 14 September 2025 (view source) Adsb (talk \| contribs) m (Make the outbound context clear) ← Older edit		Latest revision as of 15:33, 17 September 2025 (view source) Adsb (talk \| contribs) m (Don't feel comfortable with live number in example. No point in allowing ulaw because AAISP don't use it)
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==PJSIP: Trunk registration== Here is an example of a working pjsip.conf setup where Asterisk will register with A&A ~~to receive calls~~ in the same way as a SIP phone does in order to make outgoing calls. Incoming calls are sent to all registered SIP "phones".<br /> It is recommended you read the "PJSIP: NAT Issues: Keep-Alive / Anti-Idle" section below as you may wish to comment out or drastically increase the qualify_frequency line(s) if your Asterisk is not behind NAT. In pjsip.conf: [~~reg_442082881111~~reg_441234567890] type = registration retry_interval = 20 contact_user = maininbound expiration = 120 outbound_auth = ~~auth_reg_442082881111~~auth_reg_441234567890 client_uri = sip:+~~442082881111~~441234567890@voiceless.aa.net.uk server_uri = sip:voiceless.aa.net.uk [~~auth_reg_442082881111~~auth_reg_441234567890] type = auth password = ~~NotRealPasswordHere~~SecretPasswordGoesHere username = +~~442082881111~~441234567890 [aaisptrunk] type = aor contact = sip:+~~442082881111~~441234567890@voiceless.aa.net.uk qualify_frequency=20 [aaisptrunk_servera] type = aor contact = sip:+~~442082881111~~441234567890@a.voiceless.aa.net.uk qualify_frequency=20 [aaisptrunk_serverb] type = aor contact = sip:+~~442082881111~~441234567890@b.voiceless.aa.net.uk qualify_frequency=20 disallow = all allow = alaw ~~allow = ulaw~~ direct_media = no rtp_symmetric = yes aors = aaisptrunk,aaisptrunk_servera,aaisptrunk_serverb outbound_auth=~~auth_reg_442082881111~~auth_reg_441234567890 ~~Calls~~The "contact_user" option in the registration section sets the context for incoming calls to Asterisk, in this example calls come into the context "maininbound" in extensions.conf ===extensions.conf=== exten = maininbound,n,Voicemail(222@default,us) You can dial out via the trunk with (probably in a context like "from-internal"): ~~[mainoutbound]~~ exten => _X.,1,Dial(PJSIP/${EXTEN}@aaisptrunk,,) ==PJSIP: Trunk without registration== AsIf anyou ~~alternative~~don't ~~to registering~~need Asterisk ~~like~~to itmake ~~were~~outgoing ~~a SIP phone~~calls, you can have A&A send incoming calls directly to Asterisk. Use the above example but do not include the top section for "[reg_442082881111]" and the 'outbound_auth' item in the aaisptrunk endpoint. Then set the AAISP control panel to point to your server by hostname or IP address:<br /> Outgoing calls require registration, and you'll automatically receive incoming calls to registered "phones". If you register Asterisk and have calls sent directly to Asterisk you'll receive 2 copies of each call. [[File:Asterisk pjsip noregistration.png\|border]] exten => _X.,1,Dial(SIP/voiceless-out/${EXTEN}) </syntaxhighlight> For inbound calls (assuming you're routing ~~call~~calls to a ~~registered~~context ~~Snom~~named "snom"): <syntaxhighlight lang="ini"> [voiceless-in]