VoIP Phones - Asterisk: Difference between revisions
Content deleted Content added
m Change the password shown - don't know if the account details were of a live service |
m sip.conf removed as of Asterisk 21 |
||
| Line 7: | Line 7: | ||
= Configuration = |
= Configuration = |
||
Asterisk |
Asterisk historically had two methods to configure SIP connections: the legacy "sip.conf" (SIP) and the more modern "pjsip.conf" (PJSIP). As of Asterisk 21 "sip.conf" has been officially removed so there's no longer a choice (other than running an older Asterisk version). |
||
Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues, '''however''' it's been reported that PJSIP doesn't support in-band DTMF detection properly. You may need to switch back to legacy sip.conf if this affects you. The official recommendation on the [https://trac.pjsip.org/repos/wiki/FAQ#dtmf PJSIP FAQ] seems to be to write your own plugin if you need it. In-band DTMF support seems like an important thing to have, so we suggest raising a bug to report a missing feature in PJSIP if this affects you! |
Newer installations of Asterisk should be configured to use PJSIP as it will be more supported as Asterisk development continues, '''however''' it's been reported that PJSIP doesn't support in-band DTMF detection properly. You may need to switch back to legacy sip.conf if this affects you. The official recommendation on the [https://trac.pjsip.org/repos/wiki/FAQ#dtmf PJSIP FAQ] seems to be to write your own plugin if you need it. In-band DTMF support seems like an important thing to have, so we suggest raising a bug to report a missing feature in PJSIP if this affects you! |
||