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| 00:12.00 | UncleKiwi | ok the gs is the issue |
| 00:12.05 | UncleKiwi | cisco was sweet |
| 00:21.08 | *** join/#asterisk infobot (ibot@rikers.org) |
| 00:21.08 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.15.0 (2017/04/07), 11.25.1 (2016/12/08), Standard: 14.4.0 (2017/04/07); 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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| 00:42.23 | UncleKiwi | Samot:ess there is firmware or setting i need correct in the gs to get it to work |
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| 01:12.41 | Samot | I don't use Grandstream. |
| 01:12.49 | Samot | So I couldn't tell you. |
| 01:31.31 | drmessano | Heh |
| 01:31.51 | drmessano | Is the GS being used as the router? |
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| 02:45.17 | igcewieling1 | ~grandstream |
| 02:45.17 | infobot | [grandstream] the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 02:45.26 | igcewieling1 | I've heard they have improved. |
| 03:37.34 | carrar | their Grandstream GXP2000 was certainly crap |
| 03:38.16 | carrar | 2100 series was noticable improovement |
| 03:44.53 | lorsungcu | carrar: that's not really saying much |
| 03:45.00 | lorsungcu | 2100 series was miserable |
| 03:45.15 | carrar | worked pretty well for me |
| 03:45.21 | carrar | using encryption etc... |
| 03:45.31 | lorsungcu | inherited a system with ~60 2124 where BLF would just stop once in a while |
| 03:45.39 | carrar | ah |
| 03:45.47 | carrar | I started from scratch |
| 03:45.50 | lorsungcu | support finally told us to re-add the first one, save the settings, then reboot and add the rest |
| 03:45.52 | lorsungcu | and it would work |
| 03:46.06 | lorsungcu | and it did |
| 03:46.17 | carrar | inheriting a system can always be the suck :) |
| 03:46.52 | lorsungcu | always is |
| 03:47.02 | lorsungcu | we just rebuilt it with shit that works |
| 03:47.04 | lorsungcu | no worries |
| 03:49.26 | drmessano | Grandstream are big in Japan |
| 03:49.33 | carrar | heh |
| 03:49.37 | carrar | Cisco is big in Japan |
| 03:49.48 | carrar | all the companies I've been to here |
| 03:49.53 | drmessano | So is Hasselhoff |
| 03:49.57 | carrar | haha |
| 03:50.00 | carrar | so true |
| 03:50.03 | carrar | and always will be |
| 03:50.16 | drmessano | He's hooked on a feeling |
| 03:50.38 | carrar | https://www.osburn.com/stream/stream_00003097.php |
| 03:50.59 | drmessano | <3 |
| 03:51.15 | drmessano | Yeah I would love to come to work every day dressed as Sonic The Hedgehog |
| 03:51.19 | drmessano | and nobody say a thing |
| 03:51.24 | carrar | haha |
| 03:51.31 | drmessano | Because, Japan |
| 03:51.38 | carrar | not when it gets fricken hot and humid here |
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| 06:51.02 | UncleKiwi | some of the grandstream stuff is great |
| 06:51.22 | UncleKiwi | better than cisco in many cases |
| 06:51.37 | UncleKiwi | with regard to use with asterisk |
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| 15:44.45 | proute | Hi everybody |
| 15:46.45 | proute | I use asterisk 1.8. In this Asterisk, I have 2 trunk SIP (from the same provider). In each trunk , I have did number. So, my problem is when i received a call, each incoming call go to the the peer (same trunk)... |
| 15:47.03 | proute | Is there a solution to route the did numbers in the good trunk? |
| 15:47.11 | proute | thankk for your help |
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| 17:09.05 | Samot | proute: Trunks tells Asterisk where to accept calls from and where to send them. |
| 17:09.35 | Samot | If you have two SIP trunks that have the same host= setting... |
| 17:09.51 | Samot | It's just going to match on the first peer |
| 17:10.19 | Samot | The trunk sends calls to the context listed |
| 17:20.00 | igcewieling1 | When you register your Asterisk to the provider you can provide a destination. |
| 17:21.03 | igcewieling1 | egister => username:secret@host/callbackextension |
| 17:21.16 | igcewieling1 | callbackextension is what I am thinking of. |
| 17:36.29 | nathani | Can someone recommend SIP termination provider for quality calls to Canada / US . I currently use Voip.ms but the quality is lacking |
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| 18:11.13 | drmessano | flowroute |
| 18:11.21 | drmessano | nathani: |
| 18:11.42 | nathani | Thanks |
| 18:15.22 | proute | igcewieling1: I have several did numbers for each trunk |
| 18:16.47 | Samot | proute: Do you register both trunks? |
| 18:16.49 | igcewieling1 | proute: do you register for each one? |
| 18:16.57 | drmessano | proute: Then route with dialplan, not with contexts |
| 18:17.04 | Samot | ^^^^ |
| 18:18.04 | proute | Yes, I register both trunks. |
| 18:18.44 | igcewieling1 | Then the only option is to do what Samot and drmessano said. |
| 18:18.53 | Samot | Do you have insecure=port? |
| 18:19.05 | Samot | -? that was just to show it was a question. |
| 18:19.17 | Samot | Or insecure=invite,port |
| 18:19.22 | drmessano | Just dump them all into one context and route them |
| 18:19.26 | drmessano | Not a big deal |
| 18:19.32 | Samot | Yeah.. |
| 18:19.36 | drmessano | Stop using Contexts to seperate calls from peers |
| 18:19.40 | proute | Samot: yes, I tried with insecure=port, invite and invite,port |
| 18:19.42 | drmessano | When they're all the same peer |
| 18:19.44 | drmessano | Basically |
| 18:19.48 | drmessano | Because they are.. |
| 18:19.58 | drmessano | There is no authentication on incoming |
| 18:20.03 | drmessano | So you're not matching anything |
| 18:20.07 | drmessano | Except IP |
| 18:20.09 | proute | all did numbers are in the same context |
| 18:20.16 | drmessano | ok? |
| 18:20.17 | Samot | ??? |
| 18:20.20 | drmessano | So whats the issue? |
| 18:20.21 | Samot | So what's the issue? |
| 18:20.28 | Samot | hah |
| 18:20.37 | drmessano | I know |
| 18:20.39 | drmessano | Dude |
| 18:20.44 | Samot | BRUH |
| 18:20.49 | drmessano | Who cares which peer it matches |
| 18:20.58 | drmessano | They all have the same IP |
| 18:20.59 | Samot | Yeah. |
| 18:21.02 | drmessano | It doesnt matter |
| 18:21.29 | proute | my issue is that all incoming call use the same channel (same trunk) |
| 18:21.35 | drmessano | So? |
| 18:21.55 | Samot | proute: I said this earlier, trunks are there for AUTHing |
| 18:22.06 | Samot | It's a peer like anything else. |
| 18:22.32 | Samot | Are you saying you want to control how many calls each of those trunks can have? |
| 18:22.41 | Samot | And thus which incoming calls route to them? |
| 18:22.50 | proute | Samot: yes, excatly |
| 18:22.54 | Samot | OK |
| 18:22.58 | Samot | Remote port |
| 18:23.00 | Samot | re |
| 18:23.03 | Samot | Remove port |
| 18:23.03 | drmessano | Only took us like an hour to get to that lol |
| 18:23.26 | Samot | insecure= <-- Don't use this for auth what is set here |
| 18:23.37 | Samot | So insecure=invite <-- Don't auth INVITES |
| 18:23.50 | Samot | insecure=port <-- Don't match port |
| 18:24.07 | Samot | You want it to match the port |
| 18:29.46 | igcewieling1 | (02:23:03 PM) drmessano: Only took us like an hour to get to that lol <-- why I rarely help anymore. |
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| 18:42.07 | proute | Samot: I tried with insecure=invite, and the problem is still here |
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| 19:13.13 | drmessano | Yeah |
| 19:14.07 | drmessano | I dont see this working |
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| 19:38.59 | Samot | I would just do context |
| 19:39.05 | Samot | in the dialplan... |
| 19:40.30 | Samot | There are two ways I see this being done. |
| 19:41.10 | Samot | 1) Use two IPs on the PBX, have them send to each IP |
| 19:41.50 | Samot | 2) Use PJSIP for the second trunk so it's on a different port. |
| 19:43.52 | Samot | Well I guess three.. |
| 19:44.14 | Samot | Track the "call count" by the DID |
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| 22:17.43 | dan_j | Hi. Is it possible to create a featurecode which, when dialled, transfers both sides of the call into a meeting room? |
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| 23:31.41 | nathani | I am using x-lite, and flowroute not sure how to get outbound callerid working |
| 23:33.38 | nathani | nvm -they had it in their documentation |
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| 23:39.43 | cmendes0101 | I'm trying to use mixmonitor with the r and t settings to get both in and out files but these audio files are difference lengths like the silence has been removed. Is there a settings to have it record the full length? The audio file that is mixed seems fine |