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| 01:33.58 | radicaldev | can someone help me figure out why asterisk stop sending one leg of RTP after a reinvite? |
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| 02:19.43 | radicaldev | arg, it was a codec change. |
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| 12:38.05 | ntz | servus |
| 12:40.06 | ntz | I have one meta-Q: ... please does anybody has a menuselect.makeopts file for asterisk 11.x LTS with some minimalistic configuration ? |
| 12:41.02 | ntz | we use asterisk just as fax gateway (t38 over sip) with ael ... I'm tired of traversing through whole menuselect categories thinking of each feature while unchecking it |
| 12:42.33 | SeiGGy | so what's the hot open source Asterisk Load Balancer / Clustering solution these days? |
| 12:44.17 | Tim_Toady | ntz: it would be easier to maintain a modules.conf with just the things you need. |
| 12:45.10 | ntz | Tim_Toady: well, yeah - we had a discussion about exactly this matter but the side who is receiving asterisk finally got into preffering minimalistic asterisk - but you're true, it was also my proposal |
| 12:45.37 | Tim_Toady | much more easier and more flexible in case you forgot something or need some added functionality |
| 12:45.39 | [TK]D-Fender | Minimum to do WHAT? |
| 12:45.55 | ntz | just sip+ael+t38 support |
| 12:46.04 | [TK]D-Fender | AEL? ew |
| 12:46.16 | [TK]D-Fender | So you expect no dialplan function really? |
| 12:46.26 | [TK]D-Fender | NO DB integration? |
| 12:46.29 | [TK]D-Fender | transcoding? |
| 12:46.32 | ntz | but it was meta-Q: ... I have no real problem with that other than it's just labour-intensive to do it |
| 12:46.32 | [TK]D-Fender | those are ALL optional... |
| 12:46.59 | [TK]D-Fender | Don't take a meta-Q and then be uselessly vague about it :) |
| 12:47.05 | [TK]D-Fender | Get specific |
| 12:47.05 | ntz | [TK]D-Fender: you're true in this .... we will ofc need some of those |
| 12:47.29 | ntz | but at least I'd wanted to kick out modules and functions that are rarely used (or we won't use them for sure) |
| 12:47.42 | [TK]D-Fender | Sooo ... "I want stuff, but as little stuff as possible, but the stuff I need, but can't list for you" |
| 12:48.10 | [TK]D-Fender | You know those .so's are pretty easy to figure as to which do what... |
| 12:48.26 | ntz | [TK]D-Fender: my biggest problem is that I am not asterisk expert at all ... I managed to build it, make it working, to create a configuration for our specific case and to pack it to packages |
| 12:48.51 | [TK]D-Fender | Codecs are obvious, channel dirvers are obvious, dialplan apps mostly as well |
| 12:49.22 | ntz | ok, thanks guys, I'll try to convince receiving side that minimalistic asterisk will be just worse option that full asterisk with proper configuration |
| 12:49.35 | ntz | s/that full/than full/ |
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| 13:06.46 | flashdel | hi folks, i would like to receive SMS via asterisk (over one sim card only) and after the sim card is received, the content should be send to an email-adress. Background is, that i need to use a mobile tan for a credit card which is in use by many co-workers and we dont want to exchange a mobile phone all the time. Is that possible and if yes, what device would you recommend for a HP server? Or are there other ideas which would fit better? |
| 13:12.12 | WIMPy | I'd look in to the various SMS applications out there. |
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| 13:30.42 | flashdel | you mean, buying a smartphone and using it? Never thought of that^^ |
| 13:34.58 | WIMPy | No I was thinking about the PC things. Don't know what's in now. Back when I did SMS stuff, I used gnokii. |
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| 15:11.14 | jjrh | For asterisk systems that will have a lot of active channels - CPU is the primary concern right? |
| 15:11.46 | jjrh | Talking ~200-400 active channels |
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| 15:35.26 | DivideBy0 | I think call setup and tear down is more of it, so if they're really short, you'll have a lot more cpu usage than if they're all long calls |
| 15:35.58 | DivideBy0 | there was an astricon presentation about it last year, I think it was called something about benchmarks jjrh, it's on youtube |
| 15:48.58 | jjrh | Cool thanks - but my main concern when procuring hardware should be CPU speed correct? |
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| 15:59.05 | jjrh | https://www.youtube.com/watch?v=tYKWE8EYD9Q |
| 15:59.20 | jjrh | I'm guessing that's the video DivideBy0 |
| 15:59.45 | somepoortech | jjrh: I was testing recently for CPU impact in asterisk, I found sipp to be helpful for testing load |
| 16:00.29 | jjrh | somepoortech: well i'm trying to the procurement guy what hardware to buy. |
| 16:01.12 | somepoortech | jjrh: I was only testing ~100 or so channels on a single core of a Xeon E5-2697 |
| 16:01.40 | jjrh | Yeah i'm testing against a Intel(R) Xeon(R) CPU X3470 @ 2.93GHz |
| 16:02.38 | jjrh | and that's doing alright - it's a older machine. But I'm thinking i'll just tell the guy to get like 16gb of ram (or more) and whatever is a decent xeon right now |
| 16:02.48 | somepoortech | jjrh: the entire chip? I imagine you will run out of packets per second then cpu... still would be a good idea to test it if possible |
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| 16:03.55 | jjrh | This is just my dev machine - some repurposed dell 1U machine |
| 16:04.30 | jjrh | yeah this CPU is a little old - launched in Q3 09 |
| 16:04.58 | somepoortech | unless you are transcoding you should be fine cpu wise... even on ancient cpu |
| 16:08.16 | jjrh | Yeah - we run ~30-40 phones off a P3 equivalent box at my office. |
| 16:08.30 | jjrh | with uh 512mb of ram I think... |
| 16:08.32 | jjrh | maybe 1gb |
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| 17:42.05 | st2 | hello. i've got problem with webrtc |
| 17:42.19 | st2 | in browser logs: Called with SDP without ice-ufrag and ice-pwd. |
| 17:43.54 | st2 | but there is [*] res_rtp_asterisk in menuselect |
| 17:44.07 | st2 | and uuid-dev is installed in my system |
| 17:44.50 | st2 | it is strange but ldd not show uuid library |
| 17:44.51 | st2 | ldd /usr/lib/asterisk/modules/res_rtp_asterisk.so |
| 17:44.52 | st2 | linux-vdso.so.1 => (0x00007fff5377d000) |
| 17:44.52 | st2 | libpthread.so.0 => /lib/x86_64-linux-gnu/libpthread.so.0 (0x00007f4715372000) |
| 17:44.52 | st2 | libc.so.6 => /lib/x86_64-linux-gnu/libc.so.6 (0x00007f4714fad000) |
| 17:44.52 | st2 | /lib64/ld-linux-x86-64.so.2 (0x00007f47157b6000) |
| 17:45.04 | st2 | what i missed |
| 17:45.28 | st2 | could you please help me? |
| 17:46.20 | st2 | asterisk version is 11.20.0 |
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| 18:55.40 | st__ | excuse me. am i missed smth about ice-pwf issue? |
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| 19:28.18 | puzzled | st2: for webrtc better use latest asterisk version. either latest 13 or even 14. webrtc is a fast moving target and some of the webrtc required stuff/fixes are only in the very latest versions |
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| 19:52.46 | cpugenius | any of you guys do any significant ip faxing? i have a box that is running an old version of callweaver and was wondering what else out there is worth knowing about |
| 19:54.49 | st__ | puzzled: thank you, for your answer. but i had already working asterisk with webrtc support on 11.20 version. but today i had to update openssl. and now i can not compile even previous version( |
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| 21:06.36 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.10.0 (2016/07/21), 11.23.0 (2016/07/21), Standard: 14.0.0-beta1 (2016/07/27); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
| 21:10.46 | scv | interesting crash https://paste.ee/r/DEN9M |
| 21:10.53 | scv | i wonder how endpt could be null there |
| 21:11.20 | scv | i have a sneaking suspicion that there was memory pressure but dunno for sure :/ |
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| 21:17.27 | file | scv, what version of PJSIP? |
| 21:25.18 | scv | whatever's bundled with 13.10.0 |
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| 22:58.51 | fbnts | Hi, I wondered if anyone could help - I've been battling with an issue with transfer calls that have originated on Grandstream IP Phones. Incoming calls can be blind transferred find but if the phone made the call I get transfer failed on the phone. Asterisk console logs: |
| 22:58.55 | fbnts | <PROTECTED> |
| 22:59.36 | fbnts | If I enable sip debug I also see: SIP/2.0 503 Service Unavailable (can't handle one-legged xfers) |
| 23:01.41 | file | I'd suggest providing a pastebin of the complete console output |
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| 23:07.15 | fbnts | file: ok, is there a way to redirect the console output to a file (including sip debug) |
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| 23:14.59 | fbnts | file: This is an extract of when the transfer fails. Let me know if you need any further lines above or below (the capture was about 5000 lines in total!) http://pastebin.com/4EQVzy9T |
| 23:15.45 | file | a decent chunk before would be needed |
| 23:15.59 | file | the pastebin is what happened, not how it got to that point |
| 23:21.46 | fbnts | file: I think this covers from the initial sip invite for the call. My extension I was testing from was 2011 and was testing with calling Voicemail at 123 and then trying to transfer to 123 (I know stupid but it still gives the same error as transferring to another extension) |
| 23:21.46 | fbnts | http://pastebin.com/JS5auXgg |
| 23:22.18 | file | you can't blind transfer apps |
| 23:23.28 | fbnts | ah I will try again - So I can't call voicemail on 123 and then blind xfer it to my mobile? |
| 23:23.36 | file | nope |
| 23:24.32 | fbnts | ah ok, I will check again with a real call (I will call my mobile and blind xfer it to another extension) |
| 23:25.36 | fbnts | I get the same console error: chan_sip.c:24051 handle_response_notify: Got OK on REFER Notify message - I will turn on debugging and capture that as well |
| 23:35.11 | fbnts | file: I'm just trying to extract the relevent data from the debug log. I also did a tcpdump of the sip traffic here: http://pastebin.com/AAYPGYkV |
| 23:38.52 | fbnts | file: This is the verbose log (sorry its huge!): http://pastebin.com/LR6BJwjS |
| 23:40.35 | file | that REFER had a different NOTIFY... |
| 23:40.38 | file | 404 Not Found |
| 23:41.40 | file | Failed SIP Transfer to non-existing extension 2021 in context outbound |
| 23:43.19 | fbnts | ah, I think I understand now. Its looking for the destination in the current context. The handsets usually use context internal which would match 2021 |
| 23:44.01 | *** join/#asterisk raspberrypifan (~raspberry@73.106.72.78) |
| 23:48.14 | fbnts | file: Yep I've sorted that! Was me having a stupid moment some time ago dropping all outbound calls into another context (can't even remember why!) I have fixed that and its all working now |
| 23:48.36 | file | kaboom |
| 23:49.17 | fbnts | I was then testing with transferring VM to my mobile not realising that you can't transfer apps |
| 23:55.18 | raspberrypifan | hello yall so whats a good recommdned gsm gateway |
| 23:59.51 | fbnts | Thanks for your help file - night |