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| 01:40.56 | snadge | how does silence supression work in asterisk? |
| 01:41.29 | snadge | tldr.. customer has a voice message system, and our upstream hangs up due to a 5 minute media timeout.. looks like they're using silence supression, ie, we're not getting any RTP packets from them after you press 9 to record your message |
| 01:44.39 | snadge | hmm.. asterisk doesn't support silence suppression for codec g711a |
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| 02:11.52 | snadge | im trying rtpkeepalive=60 .. wish me luck ;) |
| 02:18.58 | snadge | that worked.. thanks for all your help again guys.. dunno what i would do without yas :P |
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| 02:52.31 | igcewieling | It is good to design your dialplan with your IVRs hanging up eventually. Also good to have a max message length on voicemail |
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| 06:51.33 | snadge | From: "0466XXXXXX" <sip:095XXXXX@202.43.X.X> |
| 06:52.02 | snadge | asterisk 13.10.0 .. its showing the username as the src in the cdrs |
| 06:52.14 | snadge | instead of the number |
| 06:52.32 | snadge | i really should know the answer to how to fix this.. but apparently i don't :D |
| 06:52.49 | snadge | the username matches the user context |
| 06:55.20 | snadge | i think this behaviour has changed since asterisk 12 |
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| 11:02.12 | bylzz | Hi, I'm not getting any dialtones when its ringing but when call is established sound is fine, any hints as to what might be wrong? |
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| 11:45.08 | kpjim | does anyone know if it is safe to destroy peers during <cli> sip reload? |
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| 12:25.26 | Crack92 | Hi everyone. Iâm following this wiki to install asterisk and freepbx on my Centos 7 minimal http://wiki.freepbx.org/display/FOP/Installing+FreePBX+13+on+CentOS+7 |
| 12:26.20 | Crack92 | however when starting asterisk i get this error: No ethernet interface found for seeding global EID. You will have to set it manually. |
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| 12:27.24 | Crack92 | even though asterisk and freepbx start iâm not able to connect any phones to my extensions using both pjsip and chan_sip |
| 12:28.05 | Kunsi | Crack92: for FreePBX support, see #freepbx |
| 12:28.43 | Crack92 | Kunsi: yes i wrote there, as i tought this is an asterisk issue i wrote here also |
| 12:29.32 | Kunsi | We will not support anything freepbx-related |
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| 12:58.24 | saltsa | http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerPres <-- what does presentation and screened mean in this context? |
| 12:58.44 | saltsa | or what's the difference between them |
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| 13:03.32 | Kunsi | i'd say presentation means "presendet (transmitted) to phone", and screened means "displayed to user" |
| 13:03.39 | Kunsi | but don't really know |
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| 17:18.41 | WIMPy | saltsa, Kunsi: Partially. The PI determines if the ID should be dislayed, yes, but the SI tells you the level of trust. |
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| 18:16.19 | rrittgarn | when using realtime, does the built in realtime config have the capabilities of trying one ODBC connection first, if that times out/ fails, etc. move on to another connection? |
| 18:16.45 | rrittgarn | example trying to pull voicemessages out of a DB if the DB doesn't get back quick enough, can it try a secondary server? |
| 18:41.12 | [TK]D-Fender | nope |
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| 19:28.17 | Spengler | does anyone in here use vitelity sbc service from behind a NAT router |
| 19:28.45 | seiggy | anyone know if there's a shortcut in ARI for dealing with Currency playback? |
| 19:31.21 | igcewieling | not sure what you mean by their sbc service, but I use their standard sip service and there are not NAT issues. Remember to set the endpoint with NAT when creating it in the Vitelity gui |
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| 20:28.21 | rrittgarn | in app_voicemail, when running in realtime, the system puts the whole message in your messages table, including recording blob. Is it possible to put just the envelope in the DB, and the binary data in the filesystem? or am I going to have to look into using ARI Voicemail to accomplish this? |
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| 20:38.09 | [TK]D-Fender | rrittgarn, Nope. |
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| 21:37.26 | SpaceInvaders | -- Incorrect password '' for user '6003' (context = default) |
| 21:37.35 | SpaceInvaders | does anyone know why voicemail would suddenly stop seeing dtmf? |
| 21:40.09 | DivideBy0 | seiggy: currency as in money? |
| 21:40.22 | seiggy | yep |
| 21:41.02 | DivideBy0 | maybe something like this: http://lists.digium.com/pipermail/asterisk-users/2010-June/249333.html |
| 21:42.39 | seiggy | yea, but I need it in ARI, and I also need it to respect the proper english too |
| 21:42.52 | seiggy | ie: "One Dollar" vs "Five Dollars" |
| 21:43.24 | seiggy | and "Five Dollars and One Cent" or "One dollar and Fifteen Cents" |
| 21:44.05 | seiggy | I just wrote a quick C# class that does all the rules and converts a decimal number to a collection of prompts to hand off. Should work, was just hoping I could pass a decimal number and a currency type and have it play it with all the correct rules |
| 21:46.16 | SpaceInvaders | ok it's not a dtmf issue. still troubleshooting |
| 21:47.38 | DivideBy0 | seiggy: sorry, I read "not in ARI" |
| 21:47.53 | seiggy | haha, all good |
| 21:48.36 | DivideBy0 | you can play a sequence in v14, so it could be all one play request to make it easier, but the grammar and all that is up to you in ARI |
| 21:49.44 | seiggy | ok, that's cool. Just wanted to make sure I wasn't duplicating functionality that already existed |
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