| 00:04.46 | *** join/#asterisk zaidzagha (~zaidzagha@46.185.138.33) |
| 00:05.20 | zaidzagha | hello , Would it be possible to restrict some extensions from calling a certain extension |
| 00:05.35 | zaidzagha | like restrict extension number 101 from calling 102 |
| 00:07.32 | [TK]D-Fender | Answered in #freepbx |
| 00:07.37 | [TK]D-Fender | (which is where this belongs) |
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| 01:53.53 | WangDang_ | Hi, I'm working through the O'reilly asterisk book as I set up a server. It notes that the default ports for the rtp portion of a sip call are 10,000-20,000. If I look at /etc/services, I see that a handful of ports in that range already have assigned services. Should I be worried about this apparent conflict? |
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| 02:03.40 | [TK]D-Fender | Not unless you're actually running them on those exact ports |
| 02:03.44 | [TK]D-Fender | that's just for naming. |
| 02:06.56 | [TK]D-Fender | You can also just change the range * uses on its side any time you like |
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| 02:26.54 | malaphus | Anyone have any hints on how I might be able to stream audio over the internet to asterisk's MoH, when the original audio is in AAC format? I got mpg123 streaming working for mpeg1 and mpeg2, but the stream I want to use is only AAC |
| 02:33.18 | WangDang_ | [TK]D-Fender: thanks for the reply. I've decided to limit the range * uses since I don't need anywhere that many ports for SIP. Thanks |
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| 16:03.15 | leffelad | Hi all, having issues with routing traffic back to a VPN SIP client, get a good RTP stream for 10-11 seconds, then it hits the retransmission limit, tried NAT and other settings, asterisk box is 10.0.0.x, vpnclient is 10.0.10.x, this a routing issue or asterisk not NAT'ing correctly? |
| 16:03.33 | leffelad | anyone got any ideas on this one? |
| 16:57.35 | [TK]D-Fender | * doesn't NAT |
| 16:57.45 | [TK]D-Fender | your routing does. |
| 16:57.57 | [TK]D-Fender | As for what's happening.. we'd have to actually see what's going on. |
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| 17:29.51 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 17:30.56 | EmleyMoor | Voicemail (and other things) won't respond to keypresses on my desk phone (Aastra 6731i) - is this likely a setting in sip.conf that's amiss, or on the phone? |
| 17:33.40 | [TK]D-Fender | either |
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| 17:42.21 | EmleyMoor | Ah, got it... |
| 17:43.20 | EmleyMoor | Phone's setting needs to be "SIP INFO" |
| 17:45.32 | [TK]D-Fender | needs to MATCH |
| 17:45.42 | [TK]D-Fender | I'm pretty sure that isn't the only option on either side. |
| 17:47.50 | EmleyMoor | I think it's a recently changed default that's being followed... |
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| 19:21.59 | kraptv | oh man. I am completely annoyed at Cisco SIP XML - trying to enable ssh for the latest firmware. any chance a pair of eyes would take a look at my config?? |
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| 21:11.01 | bsdice | Have I mentioned that Digium working with Jira is a real breeze |
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| 21:32.19 | *** join/#asterisk jvwjgames (~jvwjgames@2601:7:6880:81b0:d93e:382f:7d84:1e54) |
| 21:32.39 | jvwjgames | Can anyone help me with speech recognision |
| 21:32.48 | jvwjgames | I need speech recognision |
| 21:32.56 | bsdice | recognition |
| 21:33.05 | jvwjgames | yes |
| 21:33.37 | jvwjgames | can anyone help me with that? |
| 21:39.11 | jvwjgames | bsdice: can you help me |
| 21:39.17 | bsdice | no, sorry |
| 21:39.25 | bsdice | this sounds like a lot of work |
| 21:39.31 | bsdice | 3rd party modules |
| 21:39.39 | bsdice | go with DTMF options |
| 21:39.45 | jvwjgames | can't |
| 21:39.51 | bsdice | Press 1 for operator |
| 21:39.53 | bsdice | why? |
| 21:40.02 | jvwjgames | my phone system can't correctly identify DTMF tones |
| 21:40.16 | jvwjgames | google voice strips all DTMF tones |
| 21:40.22 | [TK]D-Fender | If you can't recognize THOSE cleanly what makes you think you stand a chance at VOICE? |
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| 21:41.03 | [TK]D-Fender | jvwjgames, http://zaf.github.io/asterisk-speech-recog/ |
| 21:42.10 | bsdice | heh, Google down -> you are down |
| 21:42.18 | bsdice | I don't like such constructs |
| 21:43.12 | jvwjgames | true |
| 21:43.32 | [TK]D-Fender | Because yes, Google going down is the riskiest part of this equation.... |
| 21:43.35 | jvwjgames | also don't like the speech api 50/day requests |
| 21:43.48 | [TK]D-Fender | Oh wait... how are those calls arriving in the first place? |
| 21:43.50 | [TK]D-Fender | GOOGLE VOICE |
| 21:43.59 | jvwjgames | yes |
| 21:44.13 | jvwjgames | cause sip trunks are expensive |
| 21:44.29 | [TK]D-Fender | Go google up "CMU sphinx asterisk" and go try some alternatives |
| 21:44.37 | jvwjgames | ok |
| 21:45.02 | [TK]D-Fender | <jvwjgames> cause sip trunks are expensive <- Hardly. |
| 21:46.11 | skrusty | evening |
| 21:46.36 | jvwjgames | they are $25 a month for a two way sip trunk |
| 21:47.18 | skrusty | shop around |
| 21:47.27 | jvwjgames | what about freeswitch |
| 21:47.30 | jvwjgames | would that work |
| 21:47.49 | [TK]D-Fender | As what?> |
| 21:47.52 | skrusty | for what? |
| 21:48.03 | [TK]D-Fender | FreeSWITCH is a PBX platform... not a voice recognition engine |
| 21:48.32 | [TK]D-Fender | <jvwjgames> they are $25 a month for a two way sip trunk <- Maybe that one service you found. There are tons of others |
| 21:48.55 | jvwjgames | oh cause that link that someone gave me said freeswitch can be used with speechrec |
| 21:49.08 | skrusty | so can asterisk |
| 21:49.16 | skrusty | but you need a speech recog engine |
| 21:49.19 | [TK]D-Fender | use WITH, not used AS |
| 21:49.38 | skrusty | there are lots out there, some free some not |
| 21:50.03 | skrusty | http://www.voip-info.org/wiki/view/Voice+recognition |
| 21:50.24 | [TK]D-Fender | skrusty, Trust me... from what we see he'll only be loking at "free" |
| 21:50.33 | skrusty | :) |
| 21:50.56 | [TK]D-Fender | jvwjgames, jvwjgames What level of input are you expecting? |
| 21:51.25 | skrusty | tbh, if you want free and simple, use the zaf agi [TK]D-Fender mentioned earlier |
| 21:51.35 | skrusty | almost zero setup |
| 21:51.56 | skrusty | and yeah, if you're worried about google being down, cough up some cash! |
| 21:52.42 | jvwjgames | ok |
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| 23:12.31 | pcAngel | Hi guys, we're having bandwidth issues from all of the SDP ringing (or sometimes blank audio) we're receiving during the call initiation phase (after the invite, before 200 OK / 503 / etc). Is there a way already built in to asterisk that we can avoid this? I was thinking it would be optimal if we could avoid sending SDP with the 200 OK, and then reinvite after we receive the 200 OK. I've |
| 23:12.31 | pcAngel | heard that PRACK is supposed to solve this somehow, but I haven't been able to figure it out when researching it in the past. |
| 23:15.32 | pcAngel | Contacting our VOIP providers has resulted in responses saying that there's either too many ULCs involved and they aren't handling media, or that the PSTN equipment forces this behavior |
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