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| 00:19.48 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.22.0 (2018/07/12) 16.0.0-rc1 (2018/08/08), Standard: 15.5.0 (2018/07/12); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 03:56.41 | livinginside | what's the most straightforward way to Playback() an audio file to the callee _and_ the caller? Or, just to the callee rather than the caller? |
| 03:58.18 | livinginside | e.g. not via Dial() aA; after the call's in progress, i.e. via a feature calling Gosub()? |
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| 11:28.47 | allizom | Hi. I'm learning asterisk, and I'm using pjsip. I've got some phones connected, but not at all times, and I've got multiple endpoints which can have multiple contacts. But if I try to Dial(${PJSIP_DIAL_CONTACTS(first)}&${PJSIP_DIAL_CONTACTS(second)}) when second has no contacts, the call will fail. How can I ring each contact for multiple endpoints even in the case something is not connected? |
| 11:48.06 | Samot | You cant' ring a contact that doesn't exist. |
| 11:49.25 | allizom | Samot: in that case I want to ring just what's currently connected |
| 11:50.03 | allizom | but currently the call just fails without ringing anything at all |
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| 12:09.07 | allizom | Samot: the issue seems to be that ${PJSIP_DIAL_CONTACTS(second)} is evaluated to empty and so the string that is dialed ends with an & which brings this error: Dial argument takes format (technology/resource) |
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| 12:14.16 | Samot | Then you will need to look for states of contacts. |
| 12:16.44 | allizom | Samot: I'm searching for it but "states of contacts asterisk" is not bringing me relevant results |
| 12:17.15 | Samot | Doesn't ${PJSIP_DIAL_CONTACTS($EXTEN)} do it? |
| 12:17.31 | Samot | So if it's 100 PJSIP_DIAL_CONTACTS(100) |
| 12:17.44 | Samot | That should attempt to dial all the contacts for endpoint 100 |
| 12:18.28 | allizom | ok. but I need to call contacts for more than one endpoint |
| 12:19.13 | Samot | I get that |
| 12:19.27 | Samot | But that looks for all the registered contacts and dials them for an endpoint |
| 12:19.33 | Samot | So you would need to do that for each endpoint. |
| 12:20.57 | allizom | yes, my issue is that if I dial both of them, separating with an &, what happens when one endpoint has no contacts is to have a dialed string which ends in an ampersand, which makes it invalid |
| 12:23.23 | Samot | Then you need to check the endpoint for contacts before dialing I guess. |
| 12:23.37 | Samot | I really don't use PJSIP for a lot of stuff right now so I haven't messed with this. |
| 12:24.18 | allizom | I see. Most guides/books I've seen are concerned with SIP rather than PJSIP in asterisk |
| 12:24.47 | allizom | but then I also saw pjsip is recommended for new setups, and I'm just starting out |
| 12:32.21 | sibiria | Samot: what do you use instead? opensips or kamailio or something? |
| 12:32.42 | Samot | I use Kamailio+Asterisk |
| 12:33.01 | Samot | Have for over a decade. |
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| 22:04.31 | tonybaqain | hello, can anyone please let me know if there is any other App other than Telephone for mac to test asterisk ? with numpad and so on |
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| 22:09.55 | craigify | tonybaqain, zoiper has a free edition |
| 22:10.51 | Reinhilde | if you have an android phone or androidx86 on your mac, CSipSimple |
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| 22:39.10 | sibiria | tonybaqain: Linphone |
| 22:39.48 | sibiria | but Telephone has DTMF capabilites, if that was your concern |
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