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| 02:28.28 | Janos | Is there a way to manually store things in sorcery ? I would like to cache an expensive call in my dialplan made with the CURL function, I was thinking to cache it in the asterisk DB but if I could use sorcery with automatic expire it would be awesome, using asterisk 16 by the way |
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| 14:02.25 | muAdmDev | Is it possible, that if I call a number and the callee blocked me, I get "unallocated number" as a result? |
| 14:16.17 | Samot | No. |
| 14:16.51 | Samot | Unallocated means the terminating carrier (other side) doesn't have the number. |
| 14:17.06 | Samot | Or the number doesn't exist and your carrier can't find where it routes to. |
| 14:18.02 | muAdmDev | thx, that's what I expected. second case could be possible. number exists, but one/some carriers can't route to it |
| 14:18.49 | Samot | That's not how that works. |
| 14:19.07 | Samot | If it is a valid number then it should have some sort of LRN to tell the carriers who owns it. |
| 14:19.14 | Samot | And which network is should be routed to. |
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| 14:20.43 | muAdmDev | what if carriers or the LRNs are faulty. We had cases before, when numbers were reachable via one carrier but not via another |
| 14:24.05 | Samot | Then it would be a different error. |
| 14:24.33 | Samot | Unallocated = Unknown Number/Does Not Exist. |
| 14:25.17 | Samot | Knowing where the location of something is not the same as knowing how to _get to it_ |
| 14:26.35 | muAdmDev | ok, thx |
| 14:26.42 | Samot | So you can dial this number from your mobile or another carrier service and it works? |
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| 17:00.18 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.31.0 (2020/02/04) 16.8.0 (2020/02/04) Standard: 17.2.0 (2020/02/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 18:01.30 | zxpereaxz | i guys |
| 18:05.19 | zxpereaxz | somebody can help me with a problem ?I have asterisk and one queue with ring-all strategy with three users or extensions when the three extensions is OK all work, but when an extension is disconected not rinning the others two extensions |
| 18:07.09 | zxpereaxz | any idea of this or where i can find help for this |
| 18:16.27 | igcewieling | If you are using FreePBX, ask on #FreePBX. If you are not using FreePBX then put a pastebin.com of the CLI output showing the problem happening. |
| 18:16.39 | igcewieling | someone might be able to help. |
| 18:18.07 | zxpereaxz | i use issabel PBX.... |
| 18:19.20 | igcewieling | never heard of it. |
| 18:19.53 | Samot | It's a project from AsterNIC. |
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| 18:23.30 | sahmed | Hello, One thing to know, if A-side client don't offer video codec in sdp then asterisk doesn't offer any video codec in the B-side ? even though if we set PJSIP_MEDIA_OFFER(video)=vp8 |
| 18:26.55 | Samot | igcewieling: It was less than stellar. I installed the ISO and within 24 hours deleted the VM. |
| 18:27.36 | [TK]D-Fender | I see a reference to Exliastix and 3CX. Is that running a Pre 3CX fork that actually is based on *? |
| 18:27.43 | Samot | As of Nov 2019, it still was on Asterisk 13 and ZERO PJSIP support. |
| 18:28.04 | Samot | Elastix sold to 3CX. |
| 18:28.09 | Samot | There is no more Elastix. |
| 18:28.30 | Samot | The guys from AsterNIC are in the same country as Elastix was. This fills the gap. |
| 18:28.39 | zxpereaxz | this is my pastebin https://pastebin.com/9W8qCzK2 |
| 18:28.51 | Samot | However, they just go Asterisk 16 in their new 2020 ISO |
| 18:29.07 | Samot | And they aren't supporting PJSIP due to lack of support/help from the community. |
| 18:30.12 | Samot | From what I've seen from them, PJSIP *might be* a paid channel solution. So for your Issabel PBX to support PJSIP you'd have to pay them for it. |
| 18:30.32 | Samot | So yeah, totally unimpressed. Such a shame too, their other stuff is nice. |
| 18:31.07 | [TK]D-Fender | zxpereaxz, pastebin shows nothing useful. You should show the queue status before the caller is evens ent in |
| 18:31.10 | [TK]D-Fender | sent in* |
| 18:33.16 | [TK]D-Fender | "queue show" |
| 18:33.25 | [TK]D-Fender | Also "sip set debug off" |
| 18:33.30 | [TK]D-Fender | We don't need that |
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| 18:41.13 | zxpereaxz | new pastebin with queue status: https://pastebin.com/St9U5DZP |
| 18:42.35 | zxpereaxz | in this case the extension 137 is disconnected, if i remove this extension from queue the queue work it |
| 18:43.08 | [TK]D-Fender | [2020-02-04 13:40:02] WARNING[31669][C-00000122]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 18:43.47 | [TK]D-Fender | Whatever this thing is using for state devices clearly isn't working because its trying |
| 18:43.53 | [TK]D-Fender | And we can't support this GUI... |
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| 23:32.37 | sahmed | It looks like, in asterisk-16 with pjsip, there have nothing like "videosupport=always", am I correct? |
| 23:34.02 | sahmed | so, asterisk doesn't offer any video if A-Side has no video, is this correct? anyone have any idea, please... |