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| 04:24.16 | overyander | i'm trying to test out calendaring. my username needs to have a \ in it and my password needs to contain special characters &@%; etc. Should I wrap those in single quotes, double quotes or is it not necessary? |
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| 05:36.28 | jkister | Just upgrading from meetme to confbridge on asterisk 11 -- only piece left over is converting Page(SIP/foo&SIP/bar&...,diqA(myannounce) - is there any better way to convert than to make lots of spool/outbound files (~25) that ring into the same confbridge ? Or even using the Originate API -- either way I'm worried about timing of all the phones answering (auto-answer intercom) |
| 05:57.35 | jkister | -- and then ending the conference/making all the auto-answered phones hang up would get all messy too |
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| 05:59.05 | Lope | how can I set the language inside my context in extensions.conf? |
| 06:00.16 | Lope | When I googled, I found an application called SetLanguage. http://www.asteriskguru.com/tutorials/setlanguage.html but it's not in my /usr/lib/asterisk/modules/ |
| 06:04.44 | Lope | okay, got it. Set(CHANNEL(language)=silent-lang); |
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| 06:27.07 | Lope | I've got an issue where my voicemail is not recording properly anymore. It used to... now I just hear about 1 second from whatever sound asterisk played before starting the recording, as well as a beep, and then it ends. |
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| 06:28.43 | Lope | The recorded wav is only 1 second. even if I talk for 10sec while leaving the msg. |
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| 06:36.37 | Lope | okay the problem only occurs if I play a very short greeting then go straight to voicemail. If I run a menu etc, then go to voicemail, it's fine. |
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| 12:31.58 | Lope | any idea why my music on hold might sound all jittery with a lot of popping and clicking sometimes? |
| 12:32.12 | Lope | Am I supposed to be running a realtime kernel or something like that? |
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| 13:58.03 | guest9930_ | Howdy, do you have any ideas of how to translate numbers to E.164 in the dialplan? Are there modules for this or is this all about reinventing the wheel in the dialplan? |
| 13:59.22 | WIMPy | Up to 4 lines of Goto or switch. Doesn't seem that complicated. |
| 14:00.45 | guest9930_ | 4 lines? Sounds amazing. I tried and I ended up with more like 20-30. Our numbers come in and all kinds of formats. 00CC, CC, NATIONAL |
| 14:01.22 | WIMPy | How do you get more formats? |
| 14:03.41 | guest9930_ | My logic looks like this that I fist look if the number starts with +, if so yes it must be E.164 already. If not check if it starts with 00, if so remove 00 and add +. If it starts with 0 it must be a national, add +45 |
| 14:03.46 | guest9930_ | and so on, it ends up with some lines |
| 14:04.51 | WIMPy | Sounds like you're dealing with the usual 4 formats then, which would be one line per format. |
| 14:05.17 | WIMPy | Or for Denmark only 3, as there are no area codes. |
| 14:05.55 | guest9930_ | WIMPy: hang on i'll show you a code snippet |
| 14:07.40 | guest9930_ | http://pastebin.com/FVr6ZZq0 |
| 14:07.48 | guest9930_ | This is my dialplan for changing CallerID only, not request URI |
| 14:09.29 | WIMPy | Replace those GotoIfs int the reformatting part with ExecIfs. That will remove half the lines for that part. |
| 14:09.40 | guest9930_ | Ok, I will start with that |
| 14:10.12 | WIMPy | For the dialled part it's even easier as you can use the patterns instead of having to do comparisons. |
| 14:11.22 | WIMPy | Yeah, would be nice if you could do that kind of thing globally for any call. |
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| 14:21.49 | WIMPy | Oh, and if you get yur comparisons right, you don't need to take extra care of all those words. |
| 14:22.43 | Lope | any idea why my audio is super jittery? |
| 14:24.30 | Lope | it sounds like I'm losing 50ms of audio every 100-1500ms (random) |
| 14:25.01 | WIMPy | Bad network connection? |
| 14:25.13 | Lope | is there any way to test for that? |
| 14:25.34 | WIMPy | You probably just did it :-) |
| 14:25.55 | Lope | My asterisk server is taking incoming calls from a CallCentric DID. O' |
| 14:26.06 | WIMPy | And yes, there's RTCP. But I'm not sure how reliable that information is. |
| 14:26.28 | Lope | I've got a CallCentric SIP account for testing. When I call the CC directory service the audio is crystal clear. When I call my asterisk server it's super jittery. |
| 14:26.54 | WIMPy | Where do you call from? |
| 14:27.03 | Lope | My Asterisk server is running on a DigitalOcean VPS. Could it be the CPU-time-sharing that happens on a VPS that's causing it? |
| 14:27.18 | Lope | I'm in Malaysia, my VPS is in Singapore, CallCentric is in the USA. |
| 14:27.51 | WIMPy | Possible, but unless the load is really high, less likely. |
| 14:27.53 | Lope | I've tried adding this option to my dialplan but it didn't make any difference: Set(JITTERBUFFER(adaptive)=2000,,600) |
| 14:28.19 | Lope | WIMPy: what is RTCP? |
| 14:28.39 | WIMPy | Adding jitterbuffers midway is not really a good idea. They should be used on the endpoints only. |
| 14:29.16 | Lope | I see, well my Asterisk server is sort of acting as an endpoint while it's playing Music on hold. My music on hold is what sounds the most jittery. |
| 14:29.20 | WIMPy | RTCP is used to collect RTP statistics. You can find several variables named RTP* in Asterisk. |
| 14:30.29 | Lope | What would you do? |
| 14:31.18 | WIMPy | I think the recognized answer is: Get used to it. |
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| 14:32.20 | Lope | But it's unacceptable. The jitter is horrible? |
| 14:33.18 | WIMPy | Well, to use VOIP you need an absolutely perfect network connection. the kind that cannot even been taken for granted on a LAN. |
| 14:33.47 | WIMPy | For some odd reason it seems to work most of the time, however. |
| 14:34.39 | Lope | Maybe jitterbuffer will help. "Function JITTERBUFFER not registered" I'll try adding it. |
| 14:36.01 | WIMPy | Jitterbuffers will only work for received data. So wehn you're listening to MOH from your Server, a JB on there won't make any difference. |
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| 14:44.24 | Lope | WIMPy: I see, my current test confirmed what you said :/ |
| 14:44.41 | Lope | I set a jitter buffer of 5000 and resync of 10000. it did nothing :p |
| 14:46.15 | Lope | Previously I noticed the bass in my MOH caused the audio to skip. So I ran a high pass filter on it, which reduced the skipping. I wonder if it's still an issue. |
| 14:46.28 | Lope | Because the skipping does seem to occur on the beat more than off the beat. |
| 14:47.18 | WIMPy | The network doesn't care about the type of music. |
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| 14:54.55 | Lope | WIMPy: right again. I saved a sound with a constant tone as my MOH, it skips all the same. |
| 14:58.59 | Lope | I think it's my router :) |
| 15:00.03 | Lope | I noticed when I had 2 SIP phones connected and I called the one through the other the MOH was jittery. But if I just play the MOH and take the other phone offline, it's much more steady, still with the odd skip/crackle. |
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| 15:01.52 | Lope | Haha I enabled QoS on my router, seems MUCH better. |
| 15:05.23 | Lope | night night |
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| 15:13.53 | tompaw | Guys, what could be the reason for a one-way audio in this case: http://prntscr.com/acp4kn ? |
| 15:14.15 | tompaw | (10.57.12.210 is my head set, it does not hear remote end) |
| 15:14.31 | tompaw | There's no nat involved, SDP IPs seem to be ok, direct_media is set to no |
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| 15:14.53 | tompaw | 207... is the PSTN (flowroute) |
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| 17:05.49 | litn | Hello, what's the best way to keep, for example, the last 24 hours of raw sip calls for debugging with my provider if anything comes up? the rtp streams etc |
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| 17:10.13 | DrDamnit | @litn probably tshark |
| 17:10.25 | DrDamnit | litn, probably tshark |
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| 18:13.30 | nerdcore | I'm having some issues with my Asterisk setup; I use IAX2 to communicat from asterisk to my upstream provider and this appear to be working fine, but my phone extensions which reside on the LAN with the Asterisk server use SIP and this appears to be disconnecting when it shouldn't. I'm reading about sip.conf options and timeouts. Any thoughts on what could cause these timeouts? http://nerdcore.net/mike/asterisk-timeouts-1457460756.log |
| 18:14.52 | nerdcore | I've read the linked wiki article but I'm pretty much a VOIP newb and haven't gotten to reading through RFC 3261 yet :) |
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| 18:16.52 | nerdcore | also reading about the "nat=" option in sip.conf and I'm not clear on its use. As my SIP phones are on a LAN with my Asterisk box, which then connects upstream through a different interface to my provider using IAX2, I'm thinking I should set "nat=no". Is this correct? |
| 18:17.55 | nerdcore | I appear to be using Asterisk 11.13.1 (Debian 8) |
| 18:20.50 | *** join/#asterisk snmp (~q@mx.yito.ru) |
| 18:21.26 | newtonr | If your phones are not behind NAT then yeah you probably want nat=no for their peer/friend entries |
| 18:23.51 | nerdcore | newtonr: Someone else set this up for me, but it's seriously problematic. :( I understand that since SIP is not used going out to the Internet (the phones register with Asterisk directly on a LAN), then I am "not using NAT"? |
| 18:24.03 | *** join/#asterisk Synthase_ (uid63346@gateway/web/irccloud.com/x-wpeptwgsoxmutdsr) |
| 18:24.15 | nerdcore | it was set nat=yes in sip.conf, which seems wrong now that I'm looking it over. Not sure why they did that :/ |
| 18:25.04 | Kevin` | if I use page to create a one-way sip call with asterisk, will it still request/send data for the incoming sound? how can I control this for bandwidth reason? |
| 18:26.07 | newtonr | nerdcore, you need to tell Asterisk that the peers are behind NAT if they are behind NAT. if they are not behind NAT then you want nat=no. |
| 18:26.27 | [TK]D-Fender | Kevin`: No |
| 18:28.28 | newtonr | nerdcore, if Asterisk and the phones are on the same LAN there should not be NAT between them.. |
| 18:29.22 | nerdcore | newtonr: thanks. I think some very poor assumptions were made about this config; Getting it fixed up now... |
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| 18:31.31 | nerdcore | Is 60s a very low value for "maxexpiry" in sip.conf? I see the default is 1h. Someone set this to 60s and there are an aweful lot of registrations happening :/ |
| 18:34.25 | Synthase_ | Environment details? |
| 18:34.25 | Kevin` | can someone help with this? trying to get webrtc working "[Mar 8 12:31:59] WARNING[29713][C-00000000]: chan_sip.c:10541 process_sdp: Can't provide secure audio requested in SDP offer" |
| 18:36.04 | [TK]D-Fender | Kevin`: You didn't set up SRTP and the other side expects it |
| 18:36.17 | [TK]D-Fender | Kevin`: So go set it up or have them sttop trying to use it |
| 18:36.40 | Synthase_ | @nerdcore If this is a LAN, you can likely afford the overhead, although less than five minutes is unnecessary. |
| 18:37.39 | Kevin` | [TK]D-Fender: how do I set up srtp? |
| 18:38.00 | [TK]D-Fender | Kevin`: There are docs & the sample configs |
| 18:38.05 | [TK]D-Fender | You'll ahve to go read them. |
| 18:38.31 | Kevin` | [TK]D-Fender: in theory sure, but I already have the srtp module loaded and "encryption=yes". the examples don't have any more |
| 18:38.39 | nerdcore | Synthase_: thank I've bumped it up to 10m |
| 18:39.10 | [TK]D-Fender | Kevin`: Can't jsut say "encrption without setting up certificates, etc |
| 18:39.28 | Kevin` | [TK]D-Fender: again, the examples for srtp don't have certificates |
| 18:39.44 | Kevin` | and yes, you can use encryption without certificates |
| 18:39.51 | Kevin` | the docs don't say srtp requires pki either |
| 18:42.04 | [TK]D-Fender | https://www.google.ca/#q=asterisk+srtp+howto |
| 18:42.19 | [TK]D-Fender | Ton of guides out there for this |
| 18:42.41 | Kevin` | well except the first one doesn't have any of what you just said |
| 18:42.55 | Kevin` | what do you think I was looking at? =p |
| 18:43.18 | [TK]D-Fender | I wouldn't waste time with even a guess |
| 18:43.47 | Kevin` | so you go and link me to the same guide missing the same info |
| 18:43.54 | [TK]D-Fender | And that first one is in the long dead voi-info Wiki which shouldn't have been used for years now |
| 18:44.08 | [TK]D-Fender | I linked to a GOOGLE search with a whole page worth of guides |
| 18:44.24 | [TK]D-Fender | If you're going to take me as having suggested the FIRST link then you have a reading problem. |
| 18:44.44 | [TK]D-Fender | [13:42][TK]D-FenderTon of guides out there for this <- Didn't say "I eman only that first link!" |
| 18:44.56 | [TK]D-Fender | But I see where this is going... |
| 18:45.14 | [TK]D-Fender | gets back to work on other matters |
| 18:45.45 | Kevin` | right now i'm looking for how to convert a java key store to a whateverasterisklikesbutprobablyisn'tdocument key store |
| 18:54.49 | *** join/#asterisk gdmlnx (~Gdm@ns1.puan.com.ar) |
| 18:55.35 | gdmlnx | Hi guys, any help with ss7 signanlling ? .. I got the following error : dahdi_ss7_error: [1] No siganlling link available sending message! |
| 19:08.33 | *** part/#asterisk AndreasLutro (~andreas@lutro.me) |
| 19:09.55 | nerdcore | Could I ask for a quick review of my sip.conf to ensure nothing is set to a really stupid value (like maxexpiry=60)? http://nerdcore.net/mike/asterisk-sip-2016-03-08.conf |
| 19:10.07 | nerdcore | I took the passwords out (it isn't actually "password ;) ) |
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| 20:20.20 | davlefou | Hi, i have put an Asterisk behind some nat, what port i have to open for sip users? |
| 20:27.12 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
| 20:27.15 | newtonr | davlefou, https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT |
| 20:27.52 | newtonr | "TCP/UDP 5060 and UDP 10000-20000" |
| 20:55.37 | newtonr | davlefou, you can alter the RTP ports in rtp.conf |
| 20:56.33 | davlefou | newtonr, thanks |
| 20:57.11 | davlefou | But i don't undestand, all seems works but i have no sound. |
| 21:09.15 | newtonr | davlefou, best to wireshark it out, check the SDP in the SIP dialog and see where RTP is supposed to be going, then look at the RTP streams and see where they are going |
| 21:09.50 | newtonr | davlefou, you can google a variety of guides on how to troubleshoot issues with RTP |
| 21:10.06 | newtonr | usually comes down to misconfiguration, Firewalls or SIP ALGs |
| 21:14.02 | davlefou | newtonr, My asterisk was in data serveur where i install an virtual machine. It was an goautodial 3.3 os. |
| 21:15.03 | newtonr | Okay :) |
| 21:16.29 | davlefou | All seems okay but no sound. |
| 21:20.28 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 21:20.34 | newtonr | davlefou, Sounds like you had better get to wiresharking. :) |
| 21:21.11 | davlefou | I have try tcpdump. |
| 21:30.12 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 21:32.53 | davlefou | WARNING[3611]: chan_sip.c:3983 retrans_pkt: Retransmission timeout reached on transmission |
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| 21:46.01 | drab | hi, I just added this "exten => 4360,1,Dial(SIP/4200)" expecting that if nobody at 4200 picked up it'd gone to 4200's voicemail, but it doesn't, which I guess makes sense, but it's not what I need |
| 21:46.55 | drab | I could assign that extension to the same phone 4200 is assigned to, but they don't need to make calls through that, just picking up if anybody calls 4360 |
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| 21:59.10 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:59.34 | [TK]D-Fender | drab, There is no voicmail unless you call it in the dialplan. |
| 21:59.44 | [TK]D-Fender | And there is no such thing as "assigned" |
| 22:00.55 | voip | add the otehr extension to 4200's voicemail box |
| 22:00.57 | voip | try that :) |
| 22:01.31 | voip | 4360 |
| 22:09.31 | drab | [TK]D-Fender: I have a _4NNN extension in the dialplan that has voicemail and calls ${EXTEN}, but I thought asterisk was supposed to use the closest match, which it must be doing because it does call 4200 |
| 22:09.54 | drab | so there must be something else I don't understand in the way it continues to process the dialplan after no answer from 4200 |
| 22:10.52 | [TK]D-Fender | If Dial doesn't get answered then it goes to the next priority |
| 22:11.14 | drab | right, but there's no other priority in 4360 definition |
| 22:11.24 | drab | there is an broader match, but I thought that didn't count |
| 22:11.29 | drab | after the first match |
| 22:11.59 | drab | so it matches exten => 4360,1,Dial(SIP/4200)" and if 4200 doesn't pick up that's it, it never looks at the other block for _4NNN |
| 22:12.17 | [TK]D-Fender | You should show us actual full dialplan and the call |
| 22:12.26 | [TK]D-Fender | Because these single lines are not the big picture |
| 22:12.32 | [TK]D-Fender | PB it all up |
| 22:12.48 | drab | right, sec |
| 22:15.15 | *** join/#asterisk Cory (~C@unaffiliated/cory) |
| 22:19.09 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 22:19.49 | drab | [TK]D-Fender: the config http://pastebin.com/UyvnVJs1 |
| 22:21.09 | [TK]D-Fender | "dialplan show" |
| 22:21.10 | [TK]D-Fender | and the call |
| 22:23.17 | [TK]D-Fender | that pattern is also a conceptual mistake |
| 22:33.19 | drab | I think I figured what my problem was. From https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 22:33.26 | drab | Please be aware that because of the way auto-fallthrough works, if Asterisk can't find the next priority number for the current extension or pattern match, it will also look for that same priority in a less specific pattern match. |
| 22:34.16 | drab | so priority 2 for 4360 wasn't found in 4360, but was in _4XXX, which led to voicemail etc. adding a priority 2 Hangup() worked as expected |
| 22:34.24 | drab | (ie it stopped there, no voicemail etc) |
| 22:34.35 | drab | [TK]D-Fender: what's the conceptual mistake in that pattern? |
| 22:36.00 | [TK]D-Fender | PB IT ALL |
| 22:36.09 | [TK]D-Fender | exten => _[4][0-9]XX,1,Dial(SIP/${EXTEN},15,t) |
| 22:36.13 | [TK]D-Fender | bad idea |
| 22:36.15 | [TK]D-Fender | th 4 is FIXED.. |
| 22:36.20 | [TK]D-Fender | 0-9 is all digits |
| 22:36.26 | [TK]D-Fender | so that's a waste as there is a LETTER for it |
| 22:36.35 | [TK]D-Fender | _4XXX |
| 22:36.36 | [TK]D-Fender | ^ |
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| 23:45.20 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-zwautitesqqdsied) |
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| 23:51.57 | *** join/#asterisk DrivenMad (~DMz@24.18.107.229) |
| 23:53.31 | DrivenMad | Can you use asterisk to provide a data connection to a device, lets say a smartphone by dialing into it?? even though i know it might be slow.. but its a data connection for stuff like email. |
| 23:55.14 | DrivenMad | i was thinking about old school dialup modems... the picture in my mind is like this.... server running at home connected to internet running asterisk/freepbx/ etc. taht will anwser google voice number. then provide tones like a modem maybe... |
| 23:55.50 | DrivenMad | basically i ahve a moto-e phone that has unlimited calls.. but no data (wifi data only) |
| 23:56.09 | WIMPy | Well, there is that IAXmodem thing, but that's VERY limited. |
| 23:56.38 | WIMPy | And you still need a data connection to call a modem. |
| 23:56.42 | DrivenMad | but if i could call my server and get a slow data connection from it though the free call .. well bobs your uncle :) |
| 23:57.04 | DrivenMad | IAXmodem.. i will look at that for sure... |
| 23:57.06 | WIMPy | That might not be included in your free calls. |
| 23:57.26 | DrivenMad | maybe.. |
| 23:58.02 | WIMPy | And if you want to do CSD, some rate adaption makes more sense than a modem, I.e. V.110 / V.120. |
| 23:58.13 | DrivenMad | I just love the idea that i might be able to get a slow data connection if i still ahve cell service :) |
| 23:58.31 | WIMPy | Hook up a modem. |
| 23:58.32 | DrivenMad | very cool thx WIMPy :) |