>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to that person) >> >> B) issue credentials per device and keep them all separate. Use >> extensions.conf to ring them all >> >> Having written the question out carefully, it seems obvious that A is >> the way to do this, but it's sort of contrary to the advice in the book >> so I thought I would ask.
Ben Ford <[email protected]> writes: > I'm no expert on the user side of things, but I would prefer option A. Of > course, this is completely your preference. Asterisk will allow either > option, so you have some flexibility there. One of the advantages of option > A is that you can have multiple devices (like you mentioned) that can all > be rung at once simply if the user has a desk phone, mobile work phone, etc. Thanks. It seems that A allows the user to choose devices without changing the PBX config, and that the real issue is if one wishes to allow calling the various devices separately, or if they are really just one logical device. In this case I intend them to be one logical device as in "dial the set of softphones Alice has configured and currently has connected". I have looked further and turned up two wrinkles, one of which is easy and obvious in hindsight: one has to set max_contacts high for the user's softphone aor: https://blogs.asterisk.org/2017/11/29/pjsip-mis-configuration-can-cause-loss-sip-registrations/ and the second appears to be an artifact of syntax processing and not trivial to deal with. PJSIP doesn't dial all contacts when dialing an aor, so one needs to use PJSIP_DIAL_CONTACTS. However, that can return the empty string and thus lead to a syntax error, leading to the need to write code to fix formatting: https://asteriskfaqs.org/2019/06/09/asterisk-users/dialpjsip_dial_contactsalice-pjsip_dial_contactsbob-how-not-to-fail-if-one-endpoint-has-no-registered-aor.html For the second issue, it would be nice if Dial just discarded empty destinations, as in Dial(PJSIP/foo&) Dial(PJSIP/foo&&PJSIP/baz) as would result from the following if there were no bar registrations Dial(PJSIP/foo&${PJSIP_DIAL_CONTACTS/bar}) Dial(PJSIP/foo&${PJSIP_DIAL_CONTACTS/bar}&PJSIP/baz) which seems to be how almost everyone (perhaps actually everyone) wants to use PJSIP_DIAL_CONTACTS. It is starting to seem that configuring multiple endpoints is easier than maintaining code to remove a trailing or double &. Greg -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
