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| 00:56.54 | snadge | supyo |
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| 04:54.19 | tcpdump | Is it normal for a VOIP client to spam your stun server over and over with binding requests about once per second? |
| 04:55.59 | snadge | im going to say probably not |
| 04:58.57 | tcpdump | Hmm looks like its sending an empty "change request" |
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| 06:00.57 | drmessano | Well |
| 06:01.06 | drmessano | it's not normal to run a Stun server |
| 06:01.08 | drmessano | So there's that |
| 06:02.17 | tcpdump | drmessano: define "normal"? Apple uses a STUN server for facetime. |
| 06:02.40 | tcpdump | So does Google for hangouts and Duo |
| 06:02.53 | drmessano | Sure.. because for them, one size fits all |
| 06:03.26 | drmessano | I don't know anyone using a Stun server in production for Asterisk clients.. It's extremely rarely ever needed, if ever |
| 06:04.03 | tcpdump | Im reading a doc that says to put configuration changes in "chan_sip" . Is that referencing the main sip.conf file, or is that located elsewhere? |
| 06:04.42 | drmessano | sip.conf |
| 06:05.10 | tcpdump | So just anywhere in the body thats not a peer or client? |
| 06:05.15 | tcpdump | Thanjs drmessano |
| 06:05.26 | drmessano | No, there are sections |
| 06:05.30 | drmessano | What are you trying to do? |
| 06:05.43 | lorsungcu | > put configuration changes |
| 06:06.02 | drmessano | lol |
| 06:06.12 | drmessano | lorsungcu: Useless straight guy |
| 06:06.18 | tcpdump | hmmm, i did a find, and I don't see any sections titled "chan_sip" in the sip.conf, thats why I ask. |
| 06:06.24 | drmessano | There isn't |
| 06:06.26 | drmessano | So again |
| 06:06.28 | drmessano | Specifically |
| 06:06.31 | drmessano | What are you trying to do |
| 06:08.33 | tcpdump | configure ICE/TURN/STUN support: https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+%28ICE%29+in+Asterisk |
| 06:08.41 | tcpdump | Following that page. |
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| 06:09.23 | drmessano | [general] section of sip.conf |
| 06:09.39 | drmessano | or per peer, as it states |
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| 06:10.36 | drmessano | and the stunaddr option goes in rtp.conf |
| 06:10.59 | tcpdump | "and can be enabled inside chan_sip both globally" - thats what confused me. |
| 06:11.02 | tcpdump | Got it, so general. |
| 06:11.03 | tcpdump | thx |
| 06:11.05 | tcpdump | let me fix that |
| 06:11.13 | lorsungcu | np |
| 06:15.47 | tcpdump | Hmm, any idea how to troubleshoot why it's not including it in the INVITE packets? I made sure all the config matched that doc. |
| 06:17.29 | drmessano | Did you reload chan_sip and rtp? |
| 06:19.27 | tcpdump | yes, and when that didn't work I just restarted asterisk completely. |
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| 08:01.59 | dnit | Hi |
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| 08:05.29 | dnit | I am bridging my agents channel with outgoing call to give access of outgoing call's early media to my agent. |
| 08:07.31 | dnit | I am using a hangup handler while bridging my agents channel and outgoing calls channel , so that when the outgoing call hangs up my agent lands up into the holding bridge ( using BridgeWait ). |
| 08:08.17 | dnit | But just before the hangup handler is invoked my agent logs out. |
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| 13:38.50 | darkunderlord | morning |
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| 14:22.16 | darkunderlord | anyone know a quick way to test all phones on a system to see if they have DND set on the phone itself? I'm using SPA525G2's from Cisco FYI |
| 14:23.24 | Samot | No. |
| 14:23.34 | Samot | Because it can be set both at the PBX level and the phone level. |
| 14:23.35 | lorsungcu | Could script that with sipp |
| 14:23.37 | Samot | Depending on the phone. |
| 14:23.40 | Samot | Yes. |
| 14:23.43 | Samot | You can call them all |
| 14:23.52 | lorsungcu | Basically the only way |
| 14:23.58 | Samot | But |
| 14:24.07 | Samot | That will just return a 480 Busy |
| 14:24.28 | [TK]D-Fender | And should to it pretty instantly |
| 14:25.16 | Samot | But just looking for 480 Busy's could return a false positive. |
| 14:25.46 | [TK]D-Fender | if there are no calls going on for the devices that's pretty much solid |
| 14:25.54 | Samot | True |
| 14:26.08 | Samot | This would need to be done at night |
| 14:26.17 | Samot | Or when there are no users at their phones to guarantee this. |
| 14:26.25 | Samot | But then again... |
| 14:26.31 | Samot | They could be on DND at night. |
| 14:26.45 | Samot | But if I'm not on a call and I see a weird call come in.. |
| 14:26.53 | Samot | I could just hit "Reject" on my phone.. |
| 14:26.59 | Samot | Which will send back a BUSY |
| 14:27.16 | [TK]D-Fender | that's the "instant" part |
| 14:27.34 | [TK]D-Fender | if it takes 2 seconds to ACK that you don't want it then that should be able to be judged |
| 14:27.44 | [TK]D-Fender | Also many phones won't send a ringing back in that case |
| 14:27.50 | [TK]D-Fender | You'd have to confirm with your model |
| 14:28.36 | Samot | Bottom line there is no "easy" way |
| 14:28.48 | Samot | Or "quick" |
| 14:29.49 | darkunderlord | ok thanks |
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| 14:30.22 | darkunderlord | guess I could script to hit all phones and look at XML. Just hoped it was easier and less brute force :) |
| 14:31.29 | Samot | Problem with DND is, phones have that feature. |
| 14:31.43 | Samot | Directly on the phone which DND's the _entire_ phone |
| 14:31.53 | Samot | And there are feature codes that do DND at the PBX level. |
| 14:32.18 | Samot | So unless you've programmed your phones to remove the default DND option and set a softkey DND to use feature codes... |
| 14:32.28 | Samot | You will have to call the phone. |
| 14:32.36 | Samot | Otherwise, you could just look at the hints. |
| 14:33.12 | Samot | Because the feature codes for DND will change the state of the device in the hints. |
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| 15:02.54 | _8bits | hello, what does this line same => n,GoToIf($[ "${CALLERID(num):0:1}" = "1" ]?dial:hangup) |
| 15:03.02 | _8bits | I can't call from 200 to 114 extension |
| 15:03.25 | igcewieling | That matches the first digit of the callerid. |
| 15:03.53 | igcewieling | 200 doesn't start with 1 so calls from 200 would not match |
| 15:03.59 | _8bits | How can I check two options |
| 15:04.04 | _8bits | i mean if it matches 1 or 2 |
| 15:04.11 | igcewieling | huh? |
| 15:04.23 | _8bits | <PROTECTED> |
| 15:04.29 | _8bits | I mean how to do it like that :D |
| 15:05.02 | igcewieling | put in another line to match the 2nd one. |
| 15:05.10 | igcewieling | why not stop matching on CallerID? |
| 15:05.45 | _8bits | But if first like will be true it will go to hangup instantly |
| 15:05.51 | _8bits | line |
| 15:06.36 | igcewieling | <PROTECTED> |
| 15:06.54 | igcewieling | then set the 3rd line to a hangup. |
| 15:21.53 | lvlinux | Anybody have an idea why I would be consistently getting one way audio? I can hear the remote party but they can't hear me. My firewall/router (pfSense) is showing the ports are connecting fine (and I captured the RTP stream on it and my voice was there in the stream). Pastebin of SIP debug here: http://paste.debian.net/978661/ Asterisk is behind NAT but is setup for external media addresses etc. as it |
| 15:21.59 | lvlinux | should be. |
| 15:22.21 | igcewieling | disable direct medial |
| 15:22.26 | lvlinux | Multiple providers give the same result. |
| 15:22.42 | lvlinux | I believe it is disabled but I'll make sure. |
| 15:23.12 | igcewieling | https://support.onsip.com/hc/en-us/articles/204029430-PFSense-Firewall-Settings-for-VoIP |
| 15:23.46 | lvlinux | yup it's disabled |
| 15:24.51 | lvlinux | And I've done all the standard pfSense modifications (static NAT ports, etc, forwarded RTP ports to Asterisk box, etc.) |
| 15:25.11 | lvlinux | (which is what the article says to do) |
| 15:25.24 | Samot | So your IP phone and ASterisk are on the same network? |
| 15:25.30 | lvlinux | yes |
| 15:25.49 | Samot | And when do you have audio issues? |
| 15:25.52 | Samot | On inbound calls? |
| 15:26.25 | lvlinux | outbound---i haven't tested inbound as I don't receive them, just outbound. |
| 15:26.35 | Samot | OK |
| 15:26.39 | Samot | So you can hear the callee |
| 15:26.46 | Samot | But the callee cannot hear you? |
| 15:26.51 | lvlinux | yes |
| 15:26.55 | Samot | OK |
| 15:26.56 | Samot | So.. |
| 15:27.03 | Samot | Buy a real router |
| 15:27.10 | Samot | Get rid of pfSense |
| 15:27.16 | Samot | And your problem will go away |
| 15:27.23 | Samot | Because this is OUTBOUND |
| 15:27.37 | Samot | This is your PBX sending the audio to the PSTN |
| 15:28.10 | Samot | And pfSense's "outbound NAT" is pure garbage. |
| 15:28.15 | lvlinux | ??? I don't see how a different router would make any difference unless pfSense just isn't forwarding outbound RTP (which it says it is). |
| 15:28.24 | Samot | What forwarding? |
| 15:28.27 | lvlinux | pfSense outbound NAT is fine when you mess with it and get it off the defaults. |
| 15:28.39 | Samot | PBX sends packet to Router, router sends packet to PSTN |
| 15:28.45 | Samot | Sigh. |
| 15:28.47 | Samot | OK. |
| 15:28.47 | lvlinux | yes |
| 15:29.07 | Samot | No outbound audio means your RTP isn't either leaving your network |
| 15:29.15 | lvlinux | The state tables show the proper ports being used to (that match the SDP). |
| 15:29.16 | Samot | Or not making to the destination |
| 15:29.30 | Samot | or making it to the destination with information is doesn't understand. |
| 15:29.54 | Samot | You need to call your provider then. |
| 15:30.05 | Samot | You need to confirm they are getting your audio |
| 15:31.38 | igcewieling | lvlinux: if you are using chan_sip, pastebin the [general] section. |
| 15:31.46 | lvlinux | It's pjsip |
| 15:32.04 | Samot | OK. |
| 15:32.08 | Samot | pjsip set logger on |
| 15:32.11 | lvlinux | When I make a call with my phone registered to an Asterisk box that is out on the internet, everything works fine. |
| 15:32.17 | igcewieling | I'd try with chan_sip. Most of the docs out there will assume chan_sip. |
| 15:32.21 | lvlinux | Yes i did the pjsip logger on---that's what the pastebin is. |
| 15:34.30 | lvlinux | Here it is again: http://paste.debian.net/978661/ |
| 15:34.38 | Samot | OK |
| 15:34.47 | Samot | Now do: rtp set debug on |
| 15:34.49 | Samot | Make a call |
| 15:34.58 | Samot | Lets see if RTP is being sent to 46.165.225.157 like it should be. |
| 15:35.13 | lvlinux | it is |
| 15:35.13 | lvlinux | i did that and saw the RTP both directions |
| 15:35.19 | Samot | oK |
| 15:35.19 | Samot | So |
| 15:35.23 | Samot | Call your provider. |
| 15:35.30 | Samot | You need to confirm they are setting the audio |
| 15:35.44 | Samot | s/setting/seeing/ |
| 15:35.55 | lvlinux | I also did a packet capture on the firewall, took the capture file, opened it in wireshark, and heard my voice out of it. |
| 15:36.15 | Samot | OK |
| 15:36.16 | lvlinux | But this happens with two separate providers. |
| 15:36.17 | Samot | So.. |
| 15:36.32 | igcewieling | With chan_sip, I'd expect to see messages like <--- Transmitting (NAT) to 208.88.56.93:5060 ---> when NAT is involved. |
| 15:36.50 | Samot | What is the direct_media setting on the trunk? |
| 15:37.03 | lvlinux | direct_media=no |
| 15:37.08 | Samot | OK. |
| 15:37.14 | lvlinux | It's also on the endpoint configs too. |
| 15:37.42 | Samot | Show a debug of the call going to the other provider. |
| 15:37.46 | Samot | That has the same issue. |
| 15:38.27 | Samot | Providers like DIDLogic, etc are generally "media agnostic" |
| 15:38.34 | lvlinux | I'll have to wait till I get back to the PBX and phone to do that. |
| 15:38.42 | Samot | And send it straight through to the upstream carrier they are using. |
| 15:38.52 | Samot | And sometimes those providers have the same upstream carriers. |
| 15:39.15 | lvlinux | Yes, and the sip debug showed everything sending properly as far as I could tell. |
| 15:39.16 | Samot | You could be in a situation where both your providers are sending you to the same upstream. |
| 15:39.29 | Samot | So you need to look at the SDP in both calls. |
| 15:39.45 | lvlinux | The other provider that I've tried is voip.ms, which proxies the media though. |
| 15:39.54 | Samot | proxies.... |
| 15:39.56 | Samot | Doesn't handle. |
| 15:40.08 | lvlinux | no? I thought they did. |
| 15:40.22 | Samot | No. |
| 15:40.30 | Samot | Proxy does not mean handle. |
| 15:40.42 | Samot | It means I proxy it through so it all looks like it comes from the same place. |
| 15:40.55 | Samot | So instead of you have 15 media IPs because of all my carriers... |
| 15:40.57 | Samot | You have 1 |
| 15:41.01 | Samot | My proxy IP |
| 15:41.21 | lvlinux | ah, k. |
| 15:41.35 | Samot | Now.. |
| 15:41.41 | Samot | That being said, if their proxy is setup right |
| 15:41.53 | Samot | They should have an rtp log |
| 15:42.07 | Samot | That shows the RTP/RCTP packets being sent. |
| 15:42.12 | Samot | What ports, etc were in use. |
| 15:42.18 | Samot | If there were drops, etc |
| 15:42.21 | Samot | So that is a bonus. |
| 15:43.48 | lvlinux | k I guess when I get back I'll try a checking the debug on a voip.ms call and see if there's any difference. And if not, I'll have to get in touch with one of the providers. I hate contacting the providers like pulling teeth though... |
| 15:44.16 | Samot | Well with issues likes this, yes. |
| 15:44.27 | Samot | Because they basically become MiTM for you. |
| 15:44.46 | Samot | The issue could be with the upstream |
| 15:44.54 | Samot | But the upstream won't work directly with you |
| 15:45.01 | Samot | And you can't send stuff directly to the upstream |
| 15:45.09 | Samot | So they have to play middle man |
| 15:45.40 | lvlinux | I may try some other providers too and see if I can get any of them to work. |
| 15:46.04 | Samot | But since this has pfsense involved. |
| 15:46.09 | lvlinux | Might be a good idea to test with an Asterisk box I have on a VPS too, and see if it gets RTP from me. |
| 15:46.27 | Samot | Your packets could be sent with bad data and the upstream is just ignoring them. |
| 15:46.59 | lvlinux | what do you mean by "bad data"? What could pfSense be doing that buggers up RTP?? |
| 15:47.00 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 15:47.10 | Samot | Not sending the packet correctly |
| 15:47.40 | Samot | So the upstream may look at it and go "oh this doesn't belong here" and drop it. |
| 15:47.44 | Samot | or who knows |
| 15:48.10 | Samot | In my experience, pfSense is not the router to use for SIP solutions. |
| 15:49.46 | lvlinux | Hmm, I've had great success with it elsewhere. I hope that's not the case, as I really like some of the features it offers. |
| 15:51.06 | Samot | Well at this point, any issues on Asterisk have been ruled out. |
| 15:51.17 | Samot | You say that at the router you hear audio |
| 15:51.22 | Samot | But that's at the router |
| 15:51.25 | Samot | internally still |
| 15:51.53 | Samot | Not after it has been handled for outbound traffic, setting the packet to show it is behind NAT, etc, etc. |
| 15:52.16 | Samot | You need to see what is happening after the packet is routed to the Internet |
| 15:52.33 | Samot | So yeah maybe setting up a VPS and seeing if you have the same issue... |
| 15:52.55 | lvlinux | The capture I did was on the WAN interface of the router, so it _should_ have been what was actually sent out, but who knows. |
| 15:54.15 | lvlinux | But yeah, I already have Asterisk on a VPS that I'm using for something else, so I'll see if it gets audio from me too. |
| 15:55.04 | lvlinux | If it was the router though, you'd think that doing a direct call from the SIP phone directly registered to the provider (without going through mky Asterisk box) would have the same problem. |
| 15:55.10 | lvlinux | idk |
| 15:55.40 | lvlinux | I guess I'll pick it back up tomorrow. Thanks for the suggestions. |
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| 16:10.03 | lorsungcu | lvlinux: station to staiton calls are ok |
| 16:10.04 | lorsungcu | ? |
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| 16:32.45 | Asterisco | hi |
| 16:33.01 | Asterisco | someone could help to understand the queues |
| 16:33.02 | Asterisco | ? |
| 16:33.16 | Samot | <PROTECTED> |
| 16:33.41 | Asterisco | is necessary use it? |
| 16:33.42 | Samot | Hire someone |
| 16:34.03 | Samot | No they are not |
| 16:34.13 | Asterisco | exten => 0951111111,n,Dial(SIP/11&SIP/12&SIP/13&SIP/14,60,m(musica-attesa)) |
| 16:34.24 | Asterisco | if i use this |
| 16:34.42 | Asterisco | all phones ring.. so the first operator could answer to the phone? |
| 16:34.43 | Samot | ??? |
| 16:34.59 | Asterisco | right? |
| 16:35.29 | Samot | Have you tried it? |
| 16:35.37 | Asterisco | but if i recive 5 calls? |
| 16:36.10 | Samot | All five calls get sent to all the devices |
| 16:36.29 | Samot | They either answer or dont |
| 16:37.27 | Asterisco | if the first 4 operator are speaking to the phone... |
| 16:37.41 | Asterisco | ....and 5 person call nothing could respond |
| 16:37.53 | [TK]D-Fender | <Asterisco> is necessary use it? <- of course not |
| 16:38.05 | Asterisco | so, after about 30 seconds when the operator #3 close his call... |
| 16:38.18 | [TK]D-Fender | no |
| 16:38.20 | Asterisco | could take phone #5 |
| 16:38.22 | [TK]D-Fender | no |
| 16:38.22 | Asterisco | right? |
| 16:38.24 | [TK]D-Fender | no |
| 16:38.24 | Samot | It will ring them until timeout or answer |
| 16:38.33 | [TK]D-Fender | they ring IMMEDIATELY |
| 16:38.38 | [TK]D-Fender | it will NOT retry |
| 16:38.47 | Samot | So now |
| 16:38.57 | Asterisco | so nothing could respond to the call? |
| 16:39.08 | [TK]D-Fender | Whoever CAN wring will ring |
| 16:39.14 | [TK]D-Fender | whoever DOES answer answers |
| 16:39.40 | nibbier | hi. So somehow I don't have the numer a caller dialed natively, so I need to extract it form some sip header (Set(DN=${SIP_HEADER(TO)})). This is very sad, as I can't work with all these nice dialplan features like patternmatching any more, as this does not match on ${DN} but the dialed number (${EXTEN}?). can i work around this? |
| 16:40.28 | [TK]D-Fender | Goto(context,${DN},1) |
| 16:40.32 | [TK]D-Fender | after breaking it off |
| 16:40.39 | [TK]D-Fender | then your normal extens can continue |
| 16:40.53 | nibbier | but this wont work with pattern matching? |
| 16:40.58 | Samot | nibbier: thats going to get the full header |
| 16:40.59 | [TK]D-Fender | it will |
| 16:41.10 | Samot | You need to strip the unneeded stuff |
| 16:41.18 | [TK]D-Fender | indeed |
| 16:41.31 | nibbier | yes, I do that already, just didnt paste :) |
| 16:41.40 | Samot | The <sip: |
| 16:41.50 | *** join/#asterisk atotclic (~Atotclic@unaffiliated/atotclic) |
| 16:41.50 | Samot | And everything after @ |
| 16:41.55 | nibbier | [TK]D-Fender: every time I ask a question here, you immediately give me the answer. awesome, thanks :) |
| 16:42.03 | [TK]D-Fender | You're welcome |
| 16:43.11 | nibbier | Samot: i cut at the @ and am only interested in the last 3 digets... so all set. thanks also! |
| 16:43.49 | RovingWriter | [TK]D-Fender is da real MVP |
| 16:47.17 | igcewieling | Major Vicious Personality? |
| 16:57.05 | *** join/#asterisk aljone (~androirc@2.55.175.122) |
| 16:57.39 | drmessano | igcewieling: Do you ever get tired of trolling? |
| 16:57.47 | drmessano | Asking for a friend |
| 17:02.50 | tcpdump | What's the most common audio codec for SIP in asterisk? |
| 17:03.06 | drmessano | g711 |
| 17:07.54 | *** join/#asterisk gusto (~gusto@2a01:c844:1046:820:efb2:5e13:74f1:8c37) |
| 17:08.49 | tcpdump | 1. Of course I have to make sure my client has the g711 codec, I'd assume? 2. If Im editing my peer config, and I add allow=g711, and allow=ulaw, it will prefer them in the order I put them in the config, if I understand that right? |
| 17:09.10 | drmessano | uh |
| 17:09.18 | drmessano | There is no "g711" |
| 17:09.29 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-170.cust.bezeqint.net) |
| 17:09.45 | drmessano | ulaw and alaw are g711u and g711a |
| 17:10.03 | drmessano | so allow ulaw and allow alaw |
| 17:10.33 | drmessano | and "preferred order" is simplifying it greatly |
| 17:10.46 | drmessano | Are your clients foreign? |
| 17:13.30 | tcpdump | no, so just ulaw. |
| 17:13.32 | tcpdump | In the US |
| 17:16.21 | *** join/#asterisk LunaLovegood (~alice@75.98.139.193) |
| 17:17.03 | drmessano | Dont even bother with alaw then |
| 17:17.57 | LunaLovegood | How do I make Asterisk forget a pjsip contact? I've tried setting max_contact=0 in the aor and it doesn't remove the old one. |
| 17:18.23 | *** join/#asterisk Oatmeal (~Suzeanne@2600:1700:d0a1:85a0:e8e2:8316:4796:6cbe) |
| 17:23.06 | lorsungcu | LunaLovegood: reload the module? |
| 17:23.44 | LunaLovegood | module reload res_pjsip.so |
| 17:23.48 | LunaLovegood | doesn't work |
| 17:23.56 | lorsungcu | doesnt work or doesnt remove contact |
| 17:23.57 | LunaLovegood | unless you mean a full unload, then load? |
| 17:24.04 | LunaLovegood | doesn't remove contact |
| 17:24.13 | lorsungcu | i kind of expected that |
| 17:24.19 | lorsungcu | i think you need to wait for it to drop off |
| 17:24.26 | LunaLovegood | yeah i guess |
| 17:24.30 | lorsungcu | or maybe you can qualify it? |
| 17:24.50 | LunaLovegood | tried that, even with qualify_frequency=2 |
| 17:24.55 | *** join/#asterisk karelk (~karel@31.10.154.117) |
| 17:25.08 | LunaLovegood | it just stays Unavailable |
| 17:25.13 | LunaLovegood | but it doesnt remove it |
| 17:25.45 | LunaLovegood | oh well, I'll just create a new endpoint/aor for it |
| 17:34.15 | *** join/#asterisk atotclic (~Atotclic@unaffiliated/atotclic) |
| 18:08.43 | igcewieling | when using chan_sip, the command is "sip unregister X" where X is the peer name. pjsip should have something similar. |
| 18:14.19 | Samot | LunaLovegood: pjsip show endpoints |
| 18:14.32 | Samot | This will show all the endpoints and their AORs |
| 18:14.48 | Samot | pjsip show aors <-- will show aors |
| 18:15.09 | Samot | pjsip show contacts <-- will show the contacts of the aors |
| 18:36.23 | *** join/#asterisk Oatmeal (~Suzeanne@2600:1700:d0a1:85a0:e8e2:8316:4796:6cbe) |
| 18:56.30 | *** join/#asterisk jpsharp (~jsharp@linode.fivecats.org) |
| 18:58.27 | jpsharp | I've got a pair of grandstream phones talking to Asterisk over TLS. They seem to be getting into a registration pissing contest. Whichever phone has registered last can make calls, the other one gets "username mismatch. have X, digest has Y" where X is the SIP ID of the other phone. What config am I missing somehwere? Both phones are set to "nat=force_rport,comedia". |
| 19:04.13 | *** join/#asterisk atotclic (~Atotclic@unaffiliated/atotclic) |
| 19:09.21 | atotclic | hello |
| 19:09.42 | atotclic | snmp asterisk not work |
| 19:11.28 | *** join/#asterisk stefanauss (~stefanaus@95.239.117.111) |
| 19:16.05 | darkunderlord | ok I have a weird one.....I do fax detection and it takes about 6 seconds to successfully detect on the PRI I'm using. Here is my part of the dialplan I use. https://pastebin.com/Unz9AFzd I have alot of customers that apparently are getting a fast busy signal at 7 and less seconds. I'm not specifically answering the call right away, and for some reason think that I tested doing that and it worked better not doing the answer. Anyone run |
| 19:16.06 | darkunderlord | into this kind of thing? |
| 19:26.53 | *** join/#asterisk atotclic (~Atotclic@unaffiliated/atotclic) |
| 19:28.46 | igcewieling | I thought fax detection required the channel be answered first. |
| 19:29.05 | *** join/#asterisk salviadud (~ralfalfa@189-211-190-134.static.axtel.net) |
| 19:29.21 | darkunderlord | apparently no. :) |
| 19:29.25 | darkunderlord | lol |
| 19:30.49 | darkunderlord | I think if I answer first and then wait 6 seconds, it'll make them hear silence. That's usually worse |
| 19:31.01 | *** join/#asterisk sekil (~sekil@cable-89-216-192-72.dynamic.sbb.rs) |
| 19:31.15 | igcewieling | answer, run the ringing app |
| 19:31.47 | darkunderlord | I would, but I've had people complain and not stay on the phone after 3 rings, fake or not. People suck, but when they are customers..... |
| 19:32.43 | darkunderlord | If they would let me put an IVR in the beginning, then It would start playing that and eventually detect the fax. |
| 19:32.53 | darkunderlord | but they are having meetings on whether to let me do that or not. LOL |
| 19:33.07 | darkunderlord | it's an old scrap and steel business that is oldschool and wants a human answering |
| 19:41.49 | jpsharp | I think I figured it out. I didin't have a "type" entry in my realtime database. |
| 19:41.51 | igcewieling | a better solution is to use a dedicated fax number. |
| 19:42.56 | *** part/#asterisk jpsharp (~jsharp@linode.fivecats.org) |
| 19:46.11 | darkunderlord | well Ringing and then wait for 7 seconds only takes 2 rings. I'll try that I think. Thanks |
| 19:46.51 | RovingWriter | i wonder how many people use realtime from database |
| 19:47.04 | RovingWriter | its not documented very well, but, i have made it work before. |
| 19:47.12 | darkunderlord | I use realtime |
| 19:47.19 | darkunderlord | with mysql |
| 19:47.30 | RovingWriter | yeah thats what i did it with before too |
| 19:47.46 | RovingWriter | do u put your dialplan in there too? |
| 19:47.48 | darkunderlord | why are you wondering then? :) |
| 19:48.00 | darkunderlord | I'd think more people use realtime than not. Config files suck |
| 19:48.15 | darkunderlord | although, I don't use config realtime, just my peers, etc |
| 19:48.32 | RovingWriter | i would think that too, but its documented so poorly that I imagine more people use config files |
| 19:55.33 | darkunderlord | yeah the config files are just easier, but when it comes to the shear number of peers, it's easier for that. |
| 19:57.21 | RovingWriter | yeah, for my realtime stuff i still do dialplan in config file |
| 21:06.14 | shanth | i have install festival on ubuntu trusty, but there is no festival_server binary. how do i get festival to run as a server? |
| 21:06.17 | shanth | installed* |
| 21:09.15 | shanth | ah nvm it was in man page :) |
| 21:10.17 | shanth | festival --server |
| 21:15.45 | drmessano | People still use Festival? |
| 21:21.15 | RovingWriter | whats better, drathir? |
| 21:21.20 | RovingWriter | er, drmessano |
| 21:21.27 | RovingWriter | espeak? |
| 21:25.47 | drmessano | Google TTS |
| 21:31.37 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 21:37.34 | shanth | just for testing stuff without making recordings drmessano |
| 21:37.46 | shanth | but i'll check out google tts for sure |
| 21:38.13 | drmessano | Festival is shit |
| 21:38.18 | shanth | yeah it is haha |
| 21:38.32 | drmessano | So just skip it |
| 21:38.33 | shanth | works for what im doing plus it's already installed |
| 21:38.56 | shanth | $cust just wants a demo of something and $boss said use festival |
| 21:39.21 | shanth | i will install google tts on my dev box though |
| 21:39.58 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 21:40.10 | shanth | what application do i use if i want to give the user time to enter a key after hearing some options? |
| 21:45.00 | RovingWriter | google TTS never really worked well for me. as in, when I try to use it, I just get long dead air |
| 21:45.17 | RovingWriter | shanth, WaitExten(x) |
| 21:46.17 | shanth | thanks RovingWriter |
| 21:46.35 | *** join/#asterisk [J]oules (uid217910@gateway/web/irccloud.com/x-ubsvkhwfavdltdje) |
| 21:47.57 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |
| 21:52.30 | RovingWriter | no problemo |
| 21:53.09 | drmessano | I never had a problem with it |
| 21:53.17 | drmessano | When did you try Google TTS? |
| 21:54.03 | RovingWriter | oh like 2 years ago haha |
| 21:54.30 | drmessano | Yeah, that's not such a great reference.. Come on |
| 21:54.44 | RovingWriter | haha |
| 21:54.58 | RovingWriter | got a "howto" you use to set it up? |
| 21:55.11 | drmessano | http://zaf.github.io/asterisk-googletts/ |
| 21:57.57 | RovingWriter | yeah, thats the one i used. |
| 21:58.36 | RovingWriter | when i use it in the dialplan, its just dead air. |
| 21:58.44 | RovingWriter | and no debug output anywhere |
| 21:59.53 | drmessano | Oh hang on |
| 21:59.56 | drmessano | https://zaf.github.io/asterisk-speech-recog/ |
| 22:00.10 | drmessano | You need that.. uses the Speech API |
| 22:00.43 | RovingWriter | what? i dont need to recognize speech :P |
| 22:00.47 | *** join/#asterisk iheartlinux (~jwpierce3@mail.trunkmasters.com) |
| 22:01.01 | RovingWriter | I need to go the other way |
| 22:01.11 | RovingWriter | text to speech :") |
| 22:04.00 | drmessano | I thought there was a new Google TTS that uses the newer public API |
| 22:04.08 | drmessano | That's apparently the other way |
| 22:08.12 | Samot | I'm testing Watson in the next couple of days |
| 22:09.27 | Samot | Because that will do both TTS and STT |
| 22:10.07 | Samot | And it appears to send back pcm 8k format streams so perfect. |
| 22:10.53 | *** join/#asterisk ttaylor (~ttaylor@vpn.duh.net) |
| 22:24.33 | shanth | the more asterisk i learn i think this really isn't so bad |
| 22:30.38 | drmessano | shanth: In a few weeks you'll be ready to spin up your own broken-as-hell distro |
| 22:30.44 | drmessano | Then you know you've made it |
| 22:30.53 | drmessano | Sorry, I shouldn't say a couple weeks |
| 22:31.02 | drmessano | You seem like a fast learner |
| 22:31.05 | drmessano | Maybe a week then |
| 22:32.22 | RovingWriter | been playing with this watson thing now since Samot mentioned it. I like it. |
| 22:35.16 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:38.50 | shanth | any reason to use NoOp over Verbose? |
| 22:40.05 | RovingWriter | how can I get OGG to be within 'core show file formats' ? |
| 23:27.59 | *** join/#asterisk mmlj4 (~mmlj4@47-48-196-90.static.gwnt.ga.charter.com) |