| 00:00.09 | robert_ | I didn't say it was an error. |
| 00:00.23 | igcewieling | though for best results I would set both peers to the same codec when possible |
| 00:00.29 | robert_ | I'm trying. |
| 00:00.50 | robert_ | I don't know what RudyValencia has set for his codecs, though. |
| 00:01.04 | *** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
| 00:01.11 | igcewieling | do a "sip show peers" and make sure the two softphone peers show the public IP, not the internal private IP. |
| 00:01.40 | igcewieling | robert_: asterisk can transcode between those two codecs. |
| 00:01.45 | RudyValencia | my codecs are speex 32000/speex 16000/speex 8000/pcmu 8000/pcma 8000/gsm 8000 |
| 00:02.13 | robert_ | oh |
| 00:02.22 | robert_ | let's settle on 32kz speex. |
| 00:02.51 | robert_ | and it's saying "Incorrect password" for our mailboxes. |
| 00:03.02 | igcewieling | Stick to ulaw until you get everything working |
| 00:03.14 | igcewieling | aka pcmu |
| 00:04.14 | RudyValencia | I have set the codec to ulaw only |
| 00:04.55 | Spengler1 | i just installed a new digium d50 phone and dtmf doesn't work |
| 00:05.05 | RudyValencia | so now I just need to figure out how to make voicemail work |
| 00:05.18 | Spengler1 | any tips for getting it to work? dtmf works just fine on soft phone just not digium hard phone |
| 00:06.03 | Spengler1 | i'm thinking it may have something to with the digitmap |
| 00:06.06 | Spengler1 | not sure though |
| 00:06.06 | igcewieling | do you have working audio now? |
| 00:06.16 | RudyValencia | robert_: now try your VM box |
| 00:06.26 | igcewieling | Spengler1: digium phones are not as popular as other brands and fewer people are around to help. |
| 00:07.39 | robert_ | RudyValencia: Okay. |
| 00:08.02 | robert_ | RudyValencia: "19:07:29 Not acceptable here" |
| 00:09.07 | robert_ | RudyValencia: MicroSIP supports ULAW? |
| 00:09.10 | RudyValencia | yes |
| 00:09.14 | RudyValencia | it's called pcmu |
| 00:09.51 | robert_ | Oh, okay. |
| 00:10.15 | robert_ | No audio. |
| 00:11.15 | RudyValencia | I had yet to start asterisk back up |
| 00:11.20 | robert_ | Oh, okay. |
| 00:13.29 | RudyValencia | The voicemail seems to work. |
| 00:14.35 | RudyValencia | we have it working now, with ulaw only |
| 00:15.13 | RudyValencia | we're trying 32kbit speex now |
| 00:15.25 | RudyValencia | is there a way to make the music on hold sound better btw? |
| 00:17.06 | RudyValencia | mp3 hold music is still failing |
| 00:18.54 | TriJetScud | RudyValencia: limit the bandwidth from 400-3.4K |
| 00:19.14 | TriJetScud | and then downsample it to 4K |
| 00:19.17 | TriJetScud | erm... 8K |
| 00:19.46 | robert_ | so TriJetScud we fixed it without needing any forwarding. :D |
| 00:20.04 | TriJetScud | heh, I guessed stun was more than enough? |
| 00:20.22 | TriJetScud | works with consumer grade crap really |
| 00:20.25 | RudyValencia | stun and port forwarding weren't even needed. |
| 00:20.30 | robert_ | oh, we apparently don't need stun |
| 00:21.02 | RudyValencia | all we needed was nat=yes and directmedia=no |
| 00:22.50 | robert_ | igcewieling: so can we use hold music sampled at >8khz (wav or mp3)? |
| 00:22.57 | TriJetScud | heh asterisk's probably doing some NAT magic |
| 00:23.08 | robert_ | heh, yeah. |
| 00:23.37 | TriJetScud | and robert_, just filter the music down from the 400-3.4K range and downsample it to 8K |
| 00:24.03 | TriJetScud | from what I can say about freeswitch, downsampling mp3's is done dynamically, not sure about asterisk |
| 00:24.24 | robert_ | format_wav.c:112 check_header_fmt: Unexpected frequency mismatch 32000 (expecting 8000) |
| 00:24.42 | TriJetScud | yeah, robert_ use audacity to change it to 8K first |
| 00:24.52 | TriJetScud | after you've done the post processing |
| 00:25.04 | TriJetScud | and you may want to decrease the volume of the moh by -20dB or so |
| 00:25.33 | robert_ | hm |
| 00:25.40 | TriJetScud | given if the audio is coming in too hot |
| 00:27.58 | *** join/#asterisk dr0ck (~dr0ck@c-67-172-153-201.hsd1.co.comcast.net) |
| 00:28.54 | robert_ | can I change it to support different moh formats? |
| 00:30.51 | TriJetScud | and oh, after looking up, source based coders like speex doesn't compress music all that well now come to think of it |
| 00:44.59 | igcewieling | robert_: RudyValencia I'm sure the Asterisk book talks about what is needed for clients behind nat. |
| 00:45.03 | igcewieling | ~book |
| 00:45.04 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 00:45.36 | igcewieling | read the book and avoid another wild goose chase on some other issue you find |
| 00:46.49 | *** part/#asterisk KNERD (~KNERD@24.175.249.177) |
| 00:49.47 | robert_ | so now i'm trying to get my phone hooked up. |
| 00:50.30 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
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| 00:54.50 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 01:02.45 | jmetro | What would be the best codec to run my auto attendant voice files in? Right now ive got them in 16000hz 16bit pcm signed wav named as .sln16 |
| 01:03.00 | robert_ | igcewieling/TriJetScud: can I change it to support different moh formats? |
| 01:14.01 | *** join/#asterisk deo (~deo@222.127.13.226) |
| 01:26.03 | carrar | jmetro, whats the highest quality codec phone that you have connected? |
| 01:26.18 | robert_ | oh wait, no. I'm sorry, I want a different bitrate. |
| 01:26.54 | carrar | vs what is the codec your carrier is using |
| 01:26.55 | *** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe) |
| 01:27.17 | jmetro | Carrar g722 |
| 01:27.19 | jmetro | 64kb |
| 01:27.38 | carrar | if you want it to sound great to your phones you can use that |
| 01:27.39 | beefcafe | hi, if I wanted to have asterisk check my emails and call me when there's something new, what should I do? |
| 01:27.53 | carrar | but outside callers probably won't connect as g722 |
| 01:28.08 | carrar | so you probably want a codec for outside calls too to use |
| 01:28.11 | jmetro | they will on our demo =) |
| 01:28.24 | jmetro | i meant more for the audio file, i dont care past that |
| 01:28.45 | jmetro | g711 or 722 etc, doesnt matter. i mean what file type/rate/etc should i be using. |
| 01:29.33 | jmetro | trying to get the best sounding auto attendant. The voice sounds beautiful as an mp3 on my PC but kinda awful going to my g722 polycom through the system |
| 01:35.14 | *** join/#asterisk BludSuckingFiend (~pi@cpe-24-160-205-248.insight.res.rr.com) |
| 01:44.20 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 01:46.48 | robert_ | eesh that book is expensive. |
| 02:04.01 | *** join/#asterisk dfighter (~dfighter@arcemu/staff/dfighter) |
| 02:17.39 | igcewieling | jmetro: sln16 should sound the best, but I've been unable to figure out exactly how to generate those files. I bet if you play one of your .sln16 sound files included with Asterisk it will sound fine |
| 02:17.52 | igcewieling | robert_: what book is expensive? |
| 02:18.35 | igcewieling | jmetro: you never should use mp3 format with asterisk. |
| 02:20.22 | *** join/#asterisk cbdev (~fnord@hieristdas.internetzuen.de) |
| 02:24.25 | robert_ | igcewieling: that Asterisk book, lol. |
| 02:24.48 | igcewieling | robert_: if you can't afford it, read the online version |
| 02:27.33 | robert_ | we NEED to change the bitrate to 16khz. it sounds absolutely awful. |
| 02:28.40 | igcewieling | I recommend testing it with ulaw as the phone codec to eliminate and specific issues with transcoding from sln16 to the more exotic (or at least newer) codecs. |
| 02:29.45 | robert_ | hm |
| 02:29.52 | robert_ | so we should transcode it to ulaw? |
| 02:31.03 | igcewieling | Keep your "master high quality" sounds in sln16. When possible use the asterisk "file" CLI command to convert the sln16 files to all the formats you will allow on your PBX. This avoids having to transcode on the fly |
| 02:31.13 | *** join/#asterisk haryv (~Netvergen@d216-232-130-90.bchsia.telus.net) |
| 02:32.00 | igcewieling | We keep ulaw, g722, g729, and sln16 versions of our soundfiles. |
| 02:32.21 | robert_ | hm |
| 02:32.27 | robert_ | g729? |
| 02:32.43 | igcewieling | g729 is a commercial codec |
| 02:33.02 | igcewieling | you should be asking fewer questions and reading more chapters in the Asteisk book. |
| 02:33.32 | haryv | been years since using asterisk |
| 02:34.20 | haryv | downloaded ubuntu to install elastix that was a waste of time. not built for that distribution |
| 02:35.17 | igcewieling | we use centos |
| 02:38.45 | haryv | yup guess so |
| 02:40.18 | *** join/#asterisk chang33 (~jimmy@114-32-7-232.HINET-IP.hinet.net) |
| 02:40.20 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 02:42.30 | chang33 | hi all |
| 02:43.04 | chang33 | does asterisk support exchange text messages during a call? |
| 02:44.36 | igcewieling | what protocol does exchange use for text messages? |
| 02:45.46 | chang33 | SIP MESSAGE? |
| 02:45.55 | igcewieling | do you mean microsoft exchange? |
| 02:46.26 | chang33 | NO...send/receive message |
| 02:47.07 | igcewieling | chang33: I believe it does. Check info on Asterisk 11 |
| 02:48.23 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 02:51.25 | Spengler1 | what do you guys think about an asterisk system in a small business environment for 6 phones? |
| 02:51.34 | jmetro | asterisk a+ |
| 02:51.52 | igcewieling | Spengler1: how dependent are you on phones? |
| 02:51.59 | Spengler1 | very |
| 02:52.18 | igcewieling | how reliable is your internet? |
| 02:52.21 | Spengler1 | i built a system out with an hp proliant ml110 ; 6 digium d50s and a poe switch |
| 02:52.27 | Spengler1 | around 3k |
| 02:52.37 | Spengler1 | internet is reliable but i was going to stick with pots |
| 02:53.06 | Spengler1 | voip is something to consider for the future ; looking at about 50Mb/s internet |
| 02:53.11 | igcewieling | Get yourself an ITSP at least for when the pots go down. |
| 02:53.30 | igcewieling | you can also use it for outgoing (easier to failover to POTS) |
| 02:53.50 | jmetro | i only ever hear about problem with pots, going over internet has been so stress free |
| 02:53.51 | Spengler1 | i considered that also |
| 02:54.08 | Spengler1 | what ITSP do you recommend? |
| 02:54.21 | igcewieling | Us. 8-) |
| 02:54.27 | igcewieling | where are you located? |
| 02:54.34 | Spengler1 | Maryland |
| 02:54.42 | igcewieling | Vitelity works well. |
| 02:55.18 | Spengler1 | SIP? |
| 02:55.56 | igcewieling | Of course. |
| 02:56.10 | Spengler1 | I only ask because I have seen IAX based providers |
| 02:56.26 | igcewieling | Vitelity also supports IAX, but their core busines is SIP |
| 02:56.28 | Spengler1 | Do you think 3k is reasonable for a small business system? |
| 02:56.37 | igcewieling | I have no idea. |
| 02:56.46 | igcewieling | for the hardware or the installation too? |
| 02:56.55 | pabelanger | anybody ever seen this before: [2013-03-06 17:23:41.437] WARNING[15448] chan_sip.c: just did sched_add waitid(10688) for sip_reinvite_retry for dialog rOmNCuI.7lX7piJUaS1J9Op4hxzVfzWg in handle_response_invite |
| 02:57.22 | Spengler1 | just for hardware |
| 02:57.43 | igcewieling | pabelanger: yes. according to file it is harmless. |
| 02:57.53 | igcewieling | Spengler1: is the POTS cards included? |
| 02:58.00 | Spengler1 | yes |
| 02:58.10 | Spengler1 | digium card |
| 02:58.17 | pabelanger | igcewieling: interesting, when did he mention that? |
| 02:58.27 | igcewieling | I don't think it is unreasonable, but I have NO idea what others (including the company I work for) charge. |
| 02:58.39 | igcewieling | pabelanger: a few days ago when I asked the exact same question. |
| 02:59.02 | igcewieling | I can't stand sales people. |
| 02:59.15 | jmetro | spengler1: depending on the server you are buying. asterisk doesnt need much to run on |
| 02:59.19 | Spengler1 | I'm going to give Vitelity a try too |
| 02:59.32 | Spengler1 | i currently am using broadvoice |
| 02:59.42 | Spengler1 | for my test system ; running it inside a vm |
| 03:00.19 | pabelanger | igcewieling: okay, thanks. Will ping him in the morning |
| 03:00.38 | pabelanger | if the warning is unless, then see about change the level of it, can my log files are getting spammed |
| 03:00.45 | igcewieling | pabelanger: if you find out differently, let me know. |
| 03:01.11 | pabelanger | igcewieling: are you using a sip proxy with asterisk? |
| 03:01.24 | Spengler1 | igcewieling : what do you recommend for hardware ? |
| 03:01.28 | Spengler1 | i was thinking proliant ml110 |
| 03:01.31 | igcewieling | pabelanger: yes and no. It is complicated. |
| 03:01.46 | file | It happens if the remote side has not fully processed the existing invite |
| 03:01.50 | igcewieling | Spengler1: I prefer Sangoma, but Digium cards are supposed to be food. |
| 03:01.59 | igcewieling | and good too! |
| 03:02.39 | file | It sends back a request pending, and then chan_sip retries after a random period of time |
| 03:02.45 | Spengler1 | Do they provide linux based server appliances? |
| 03:02.56 | file | Reinvite race conditions are fun |
| 03:04.01 | igcewieling | file: any way to make asterisk delay for a slightly longer amount of time to reduce glare? |
| 03:05.08 | igcewieling | Spengler1: I was referring to the POTS cards |
| 03:05.38 | Spengler1 | igcewieling : do you have any recommendations for servers? |
| 03:05.41 | file | Don't believe so, at least not without code mods |
| 03:06.02 | igcewieling | Spengler1: none whatsoever. I hate hardware, it always lets you down. |
| 03:06.31 | Spengler1 | what do you run your asterisk box on? |
| 03:06.39 | file | And now I sleep |
| 03:06.54 | haryv | file, what does digium charge these days for support? |
| 03:07.28 | Spengler1 | I looked it up and I think it was $595 / year for 1 system |
| 03:07.37 | igcewieling | Spengler1: I have installed about 40 asterisk servers in the past 3 years and I have never actually physically seen any of them. I have no idea what we buy, but I know our vendor puts them together themselves |
| 03:09.28 | Spengler1 | igcewieling : thats cool , whats the largest call center you have setup in terms of phone count? |
| 03:09.50 | igcewieling | Spengler1: never set up a call center on asterisk. |
| 03:10.26 | Spengler1 | igcewieling : what about office |
| 03:10.58 | igcewieling | I'm pretty sure we have a customer with 150+ phones. |
| 03:12.00 | Spengler1 | cool ; scales nicely i assume. I was talking with digium support and he told me once i got to a certain size not to use ulaw |
| 03:12.10 | Spengler1 | too much bandwidth |
| 03:13.18 | Spengler1 | he said buy the G.729 codec |
| 03:14.07 | igcewieling | sales convinces some poor sod to sign up with our service, accounting orders the hardware, the office tech boots my ISO and it builds and sets up everything. Techs bring the server, router, switch, and phones to the customer. Techs plug in an IP KVM and cradlepoint router with ECVO service, I VNC into the KVM and set up networking and the like. Techs collect extension/name info and sets up the phones on the desks and set up extensi |
| 03:15.00 | igcewieling | Spengler1: we have asterisk servers which have around 150+ calls at peak times, but those are not used as "PBXs" |
| 03:15.26 | Spengler1 | are you using cradlepoint with a cellular connection? |
| 03:16.06 | igcewieling | yes. REALLY SUCKS, but since half the time the networking info changes at some point we don't set up the IP info until the server is on-site |
| 03:16.52 | Spengler1 | i have a small business that uses it as their primary internet with Verizon LTE and it works about 10 times better than their old DSL |
| 03:16.55 | igcewieling | once we get the router installed and the server on our internet connection we don't need the cradlepoint anymore. |
| 03:17.10 | Spengler1 | its costly though but they need it because one of the employees works from home. it works pretty good |
| 03:18.13 | Spengler1 | sounds like a cool job igcewieling ; what states does your company operate out of? |
| 03:19.12 | igcewieling | Spengler1: We use XO and other carriers for the "last mile" so we have a "nation wide" service area, but we only have staff on Long Island and dispatch out to NYC/NJ and Philly |
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| 03:46.24 | haryv | jet something was one of the original sip provider I used |
| 03:51.07 | *** part/#asterisk studybot_ (~studybot_@gateway/tor-sasl/studybot/x-68286794) |
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| 04:23.32 | joako | join #suse |
| 04:33.47 | joako | Is AsteriskNOW depreciated? There are recent security updates to the base OS but DAHDI is not update for the new kernel version |
| 04:34.39 | WIMPy | What kernel version? |
| 04:34.54 | WIMPy | I'm not sure the latest is supported, yet. |
| 04:43.08 | *** join/#asterisk FireAndIce (~FireAndIc@219.91.132.37) |
| 04:43.11 | joako | WIMPy, Kernel 2.6.18-348.2.1.el5. In the repository the newest DADHI is for kernel 2.6.18.308 which appears to be from 2012-02-22 |
| 04:44.42 | WIMPy | Last week it didn't work with 3.8.0, but 3.4.0 definitely works. |
| 04:45.23 | WIMPy | That's from http://downloads.asterisk.org/pub/ however. |
| 04:46.13 | joako | You´re talking about building from source? |
| 04:46.30 | WIMPy | yes |
| 04:48.50 | joako | Im talking about the RPM from the repository for AsteriskNOW and CentOS: http://packages.digium.com/centos/5/current/x86_64/RPMS/?C=M;O=D |
| 04:49.04 | joako | I don´t understand why constantly DAHDI is not updating for the kernel security patches |
| 04:51.23 | WIMPy | The fine difference between hardware supported by Linux and hardware with support for Linux. |
| 04:51.42 | robert_ | I still want 16khz audio for hold music and voices. :/ |
| 04:55.44 | robert_ | ooh |
| 04:55.49 | robert_ | ADPCM, I think is what I want. |
| 04:57.50 | robert_ | wait |
| 04:58.10 | robert_ | does Asterisk ONLY read 8khz wav files? |
| 04:59.30 | robert_ | or is it up to the codec reading the file? |
| 05:07.19 | joako | robert_, Well technically a non-wav file is not an 8khz wav file so the codec determines what codec you can use |
| 05:07.43 | *** join/#asterisk Adam0928 (~Adam0928@c-71-224-47-45.hsd1.pa.comcast.net) |
| 05:07.46 | joako | But ideally you will manually convert the sound files to the native codec(s) you use to avoid transcoding |
| 05:09.01 | robert_ | so can I only use 8khz anything? |
| 05:09.18 | robert_ | (mp3, ogg, etc.) |
| 05:09.53 | robert_ | we transcode it beforehand and upload it to our asterisk server. |
| 05:10.55 | joako | PSTN is 8khz so what is the point exactly? |
| 05:11.02 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-vmibwpogtnczagco) |
| 05:11.26 | robert_ | I want 16khz internally, though. So that's not possible? |
| 05:12.34 | Adam0928 | no it is |
| 05:13.16 | WIMPy | I think slin16 is the format you want. |
| 05:13.21 | joako | robert_, Yes it does seem possible: https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats |
| 05:13.59 | WIMPy | Or 'core show file formats'. |
| 05:15.22 | Adam0928 | http://ftp.tiaonline.org/tr-41/tr411inactive/Public/0-Archive/2005-08-Halifax/TR41.1-05-08-004-TSB-116A-Ballot-Version.pdf |
| 05:15.29 | robert_ | oh, sweet |
| 05:16.18 | Adam0928 | ^SUMMARY OF VOICE QUALITY RECOMMENDATIONS FOR IP TELEPHONY |
| 05:16.35 | robert_ | oh so I just 'format = (format_name)", then? |
| 05:16.44 | robert_ | and yes, I shall read that. thank you. |
| 05:17.30 | Adam0928 | it does not pertain directly to asterisk |
| 05:20.32 | robert_ | hm |
| 05:20.51 | robert_ | oh geez |
| 05:21.07 | robert_ | this sounds like porn music it's so slow.. <_< |
| 05:21.44 | ChannelZ | you've done it wrong |
| 05:23.30 | Adam0928 | do you plan on playing a lot of music thru your machine? |
| 05:24.10 | robert_ | no, this is our server. |
| 05:24.40 | robert_ | ChannelZ: hm, okay. oh wow. it's still 48khz |
| 05:24.41 | Adam0928 | server = machine |
| 05:24.42 | robert_ | fixes that. |
| 05:25.10 | robert_ | oh, you mean do we plan to have a lot of people on hold? |
| 05:25.15 | Adam0928 | i was going to ask why not use a higher sampling rate then |
| 05:25.49 | Adam0928 | no voices dont step outside the 8 khz range, which is why Ma Bell uses it |
| 05:25.52 | robert_ | we would if Asterisk isn't normally this slow and asterisk complain about it. |
| 05:26.38 | Adam0928 | why not cd quality rates |
| 05:26.41 | Adam0928 | like 96 |
| 05:27.02 | WIMPy | That's a lot more than CD. |
| 05:27.47 | Adam0928 | sorry 44.1 |
| 05:28.00 | robert_ | yeeeah |
| 05:28.04 | Adam0928 | i forgot it was that low |
| 05:28.05 | robert_ | it's reeeeeeeeeeeeeeeeeeeeeally slow. |
| 05:28.25 | robert_ | wrong codec, perhaps? |
| 05:28.30 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-147.red.bezeqint.net) |
| 05:28.48 | Adam0928 | slow as in choppy |
| 05:28.55 | Adam0928 | or slow as in speed |
| 05:29.09 | robert_ | as in speed. |
| 05:29.21 | robert_ | somebody could slow-dance to this song. |
| 05:29.27 | robert_ | and usually it's up-tempo. |
| 05:29.31 | Adam0928 | i was having the opposite problem with my voicemails |
| 05:29.50 | Adam0928 | it sounded like alvin leaving messages |
| 05:30.32 | robert_ | oh |
| 05:30.33 | robert_ | hm |
| 05:31.13 | Adam0928 | no one but me uses that machine tho so i just rebooted it and it was fine |
| 05:31.34 | robert_ | oh |
| 05:31.44 | robert_ | I did it wrong, it seems. (as noted earlier.) |
| 05:31.46 | robert_ | Codec mismatch on channel SIP/100-0000000d setting write format to slin from slin16 native formats (g722) |
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| 05:36.17 | robert_ | oh |
| 05:36.22 | robert_ | my channel, oops. |
| 05:37.33 | robert_ | ugh |
| 05:48.25 | robert_ | oh |
| 05:48.31 | robert_ | my client just needs to be fixed, |
| 05:48.43 | robert_ | shouldn't be that hard, I don't think. |
| 05:49.20 | robert_ | (I don't want too a too-high bitrate; this is going to eventually interface with the public.) |
| 05:55.56 | *** join/#asterisk gerhard7 (~gerhard7@195-241-233-43.ip.telfort.nl) |
| 05:59.22 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 06:03.29 | ChannelZ | via what? |
| 06:03.52 | robert_ | some SIP trunk; I haven't decided which yet. |
| 06:07.16 | ChannelZ | Well chances are it's going to wind up on a T1 at 8khz anyway. I'm not sure if any ITSPs using wideband have any peering with mobile providers, a few of who are doing some wideband |
| 06:08.32 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-195-147.red.bezeqint.net) |
| 06:09.13 | beefcafe | is moh sent as unicast or multicast? |
| 06:10.36 | joako | ChannelZ, That´s basically what I said an hour ago |
| 06:10.53 | ChannelZ | I'm not sure how multicast MOH would work/make sense but it's all unicast |
| 06:13.02 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 06:13.26 | robert_ | yeah |
| 06:14.12 | robert_ | I know. |
| 06:14.19 | ChannelZ | I guess if you're going to put each other in the office on hold a lot you can delight in higher quality MOH |
| 06:15.14 | robert_ | lol yeah; I'd rather higher quality internally. |
| 06:15.59 | robert_ | are there any speex codecs? |
| 06:16.14 | robert_ | oh nevermind |
| 06:16.21 | ChannelZ | yes |
| 06:16.25 | robert_ | I need to make sure audacity can write to speex |
| 06:19.16 | robert_ | hm |
| 06:20.51 | robert_ | where are the speex codecs? |
| 06:22.11 | robert_ | oh |
| 06:22.21 | robert_ | speex is OGG VORBIS, isn't it? |
| 06:23.47 | ChannelZ | more or less but with very specific properties |
| 06:23.54 | robert_ | ah. |
| 06:24.27 | ChannelZ | actually that's not true. |
| 06:25.05 | ChannelZ | Speex is based on CELP. |
| 06:25.39 | ChannelZ | It's "made" by the same group as Vorbis and Theora |
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| 06:28.00 | robert_ | ah. |
| 06:28.13 | robert_ | grrr |
| 06:28.28 | ChannelZ | RAWR! |
| 06:28.31 | robert_ | what ISN'T 8khz that IS supported by both the clients and server? |
| 06:28.42 | ChannelZ | that depends on your clients |
| 06:28.56 | robert_ | sigh. |
| 06:29.35 | ChannelZ | Softphones are more likely to have wideband codecs but hard devices, it's an extra feature you pay more for. |
| 06:29.51 | ChannelZ | Although I think Digium's phones do some wideband maybe |
| 06:30.46 | robert_ | ah. |
| 06:30.53 | ChannelZ | g722 it looks like |
| 06:31.14 | *** part/#asterisk tonsit (~FtC@tonsit.com) |
| 06:32.01 | robert_ | hm |
| 06:34.35 | joako | I think even Grandstream claims their ¨products¨ support G722 |
| 06:34.48 | robert_ | so g722 supports 16kbps? |
| 06:35.34 | ChannelZ | kind of |
| 06:35.50 | beefcafe | From: "asterisk" <sip:asterisk@10. |
| 06:35.58 | beefcafe | how do I change the 'asterisk' here? |
| 06:37.03 | robert_ | ChannelZ: "kind of"? |
| 06:37.06 | ChannelZ | that's a default sort of caller ID if the channel doesn't have any |
| 06:37.35 | ChannelZ | robert_: well it's 16khz at 14 bit. |
| 06:38.00 | beefcafe | ChannelZ: how do I set it to something else? |
| 06:38.19 | ChannelZ | and it's of course lossy so it's not exactly what you'd think of 16khz quality, whatever that might be. |
| 06:38.30 | robert_ | Oh, cool. 11025 is actually sufficient. |
| 06:38.52 | ChannelZ | beefcafe: set the caller ID either in your dialplan prior to Dial()ing or on the channel, like if it's a sip peer in sip.conf |
| 06:39.43 | ChannelZ | exten => whatever,n,Set(CALLERID(all)=Boobs <12345>) |
| 06:39.54 | beefcafe | ChannelZ: it's an OPTIONS sip:10. message |
| 06:40.01 | robert_ | haha boobs |
| 06:40.06 | ChannelZ | oh.. then who cares? |
| 06:40.14 | beefcafe | I do :P |
| 06:40.18 | beefcafe | is there a way to change it? |
| 06:40.29 | robert_ | I would too if I had to stare at it all the time. :p |
| 06:40.37 | robert_ | And I probably will at some point. |
| 06:40.45 | robert_ | so I feel you, beefcafe :p |
| 06:40.55 | beefcafe | thanks robert_ |
| 06:41.08 | ChannelZ | you stare at SIP debugs all the time to see OPTIONS messages? |
| 06:42.00 | beefcafe | http://forums.asterisk.org/viewtopic.php?f=1&t=80340 |
| 06:42.06 | beefcafe | someone who cared |
| 06:43.04 | ChannelZ | interesting. Didn't know you could put callerid in the general section though I suppose it makes sense |
| 06:43.09 | *** join/#asterisk apurvtwr (b64722fb@gateway/web/freenode/ip.182.71.34.251) |
| 06:43.44 | beefcafe | thanks ChannelZ, I'll give it a try |
| 06:47.43 | *** join/#asterisk Vince-0 (~vincent@105-237-1-58.access.mtnbusiness.co.za) |
| 06:48.55 | beefcafe | ChannelZ: it works. thanks again :) |
| 06:50.12 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-mtpdkkylqalwokiq) |
| 06:50.44 | ChannelZ | well you found the answer... but you're welcome :) |
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| 06:57.49 | *** join/#asterisk schmidts (~schmidts@vie-91-186-159-076.dsl.sil.at) |
| 06:57.51 | schmidts | good morning |
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| 07:20.00 | robert_ | ChannelZ: channel.c:5075 ast_write: Codec mismatch on channel SIP/100-0000002e setting write format to slin from slin16 native formats (g722) |
| 07:20.21 | robert_ | only I don't know what slin16 is |
| 07:25.16 | schmidts | robert_ slin16 is a wideband variation of slin. if you use g722 which is also a wideband codec, if asterisk needs to convert or playback sound or moh it converts it to slin16 internal |
| 07:25.40 | apurvtwr | hi everyone. is it possible to write to socket from asterisk context ? |
| 07:26.18 | robert_ | hm |
| 07:26.50 | apurvtwr | I want to send events like ringing, answered or hangup to a remote client |
| 07:27.23 | *** join/#asterisk Bradada (~Bradada@223-143-46-143.dynamic.hinet.net) |
| 07:27.35 | schmidts | apurvtwr if you mean the dialplan with asterisk context, the answer is no. to get something like this you would have to write your own resource module which listen on the asterisk event dispatcher and then send this to your client |
| 07:27.57 | robert_ | hm |
| 07:28.14 | schmidts | apurvtwr but there is allready a solution for this, why not using asterisk manager interface (AMI) ? |
| 07:28.31 | schmidts | robert_ when do you see this message? |
| 07:28.48 | robert_ | schmidts: when I dial our internal "Play hold music" extension. |
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| 07:29.11 | robert_ | schmidts: I don't want it downsampled. |
| 07:29.44 | schmidts | robert_ then you need a sound file which is either g722 converted or slin16 compatible |
| 07:30.25 | robert_ | I'm using a 16-bit signed PCM |
| 07:30.31 | robert_ | named .wav16 |
| 07:31.01 | robert_ | (default_hold.wav16) |
| 07:31.19 | apurvtwr | schmidts: thanks. I actually tried asterisk java. I have a class that implements Asteriskserverlistener and propertychangelisterner. I add this listener to every asteriskchannel. Now on a call event, It shows mulitple events, because of intermediate channels. |
| 07:32.47 | schmidts | apurvtwr AMI is only an interface where every event gets posted, you dont need a listener on every channel, only one should be enough but your client has to find the right channels. I am not sure but imho you can set an unique ID for ami for each channel |
| 07:33.05 | schmidts | robert_ with 24kHz? |
| 07:33.19 | schmidts | sorry 16kHz of course ;) |
| 07:33.55 | robert_ | schmidts: yes. |
| 07:34.00 | robert_ | :p |
| 07:34.16 | robert_ | I figured out the HARD way the khz doubled. |
| 07:35.06 | schmidts | robert_ what does a core show channel .... shows you on the three codecs? read format, write format and native format |
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| 07:38.11 | robert_ | it says it's not transcoding. |
| 07:38.32 | robert_ | all are g722 |
| 07:38.39 | robert_ | but it still sounds different than our source. |
| 07:40.26 | *** join/#asterisk work (work@2002:5289:c108:d:a0e2:7c81:c437:39af) |
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| 07:48.46 | schmidts | robert_ is your source mono? |
| 07:50.12 | robert_ | yes. |
| 07:50.29 | robert_ | we saw somewhere that we had to downmix it to mono. |
| 07:50.43 | robert_ | orr will Asterisk do that on it's own? |
| 07:50.57 | schmidts | ofcourse you have, phone speakers are mostly only mono ;) |
| 07:51.15 | robert_ | or* :p |
| 07:51.19 | robert_ | yeah. |
| 07:51.35 | schmidts | it will but not the same way as an audio prog would do |
| 07:51.42 | schmidts | or not at the same quailty ;) |
| 07:52.09 | schmidts | what do you mean with sounds different? in what way does it sound different? |
| 07:53.02 | *** join/#asterisk Bradada (~Bradada@223-141-190-60.dynamic.hinet.net) |
| 07:53.06 | robert_ | it sounds cleaner in OUR copy of the mono version than in the copy we're hearing over Asterisk. |
| 07:53.35 | robert_ | it sounds like it's AM radio in the copy coming from Asterisk. Even on softphones. |
| 07:53.54 | schmidts | have you tried to record the output using monitor and listen to this soundfile or even take a tcpdump and get the raw sound stream out there? |
| 07:55.12 | robert_ | whoa |
| 07:55.12 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 07:55.18 | robert_ | that sounds actually really good. |
| 07:57.39 | schmidts | the monitor output or raw stream? |
| 07:57.55 | robert_ | I think I'm getting it, slowly. |
| 07:58.47 | work | hello I want to run TE405P in EDSS1 enviremont what packages should install I have installed AsteriskNOW |
| 07:58.49 | *** join/#asterisk resist0r (~resist0r@unaffiliated/mrresist0r) |
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| 08:05.03 | robert_ | hah |
| 08:05.05 | robert_ | there we go. |
| 08:05.12 | robert_ | I love you, sox. |
| 08:06.30 | robert_ | NOW we just need to un-rape Allison Smith's voice. |
| 08:09.19 | robert_ | I could convert this to g722 |
| 08:09.25 | robert_ | now that I have the desired quality |
| 08:09.43 | schmidts | :D |
| 08:09.49 | robert_ | :D |
| 08:10.03 | robert_ | it sounds exactly like the master now. Oh god, this is heaven. :D |
| 08:10.32 | robert_ | hm |
| 08:10.36 | robert_ | there's a pop at the end |
| 08:11.05 | robert_ | oh |
| 08:11.16 | robert_ | there's one in the source. Okay. |
| 08:11.57 | beefcafe | robert_: which tool do you use for conversion? |
| 08:12.47 | robert_ | beefcafe: I was using audacity. We'll keep editing with it, but I just switched to sox for conversion to sln16. :D |
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| 08:13.41 | robert_ | beefcafe: this is awesome! it sounds JUST LIKE PLAYING MUSIC OVER A SKYPE CALL. :D |
| 08:14.00 | robert_ | screw skype. |
| 08:14.38 | beefcafe | I feel skype sounds better than g722 |
| 08:14.57 | robert_ | but not by much, though. |
| 08:15.05 | robert_ | it does sound better |
| 08:15.18 | robert_ | you can hear the room around you |
| 08:15.32 | beefcafe | yeah |
| 08:15.47 | beefcafe | I wonder if there's a codec for CD or DVD quality :D |
| 08:15.56 | robert_ | that would rock. :D |
| 08:16.28 | robert_ | so now we just need to get voice prompts done |
| 08:16.37 | robert_ | and power through voicemail |
| 08:16.41 | robert_ | and we'll be set. |
| 08:16.44 | beefcafe | hates doing prompts |
| 08:16.56 | beefcafe | uses google tts for that |
| 08:17.01 | robert_ | haha |
| 08:17.18 | beefcafe | google speaks way more languages than me |
| 08:17.19 | schmidts | :D +1 beefcafe |
| 08:17.24 | robert_ | yeah but you aren't us. we have a person with a nice deep voice for our prompts. :D |
| 08:17.35 | robert_ | yeah, they do. |
| 08:17.46 | beefcafe | may I borrow your prompts? :P |
| 08:17.46 | robert_ | and we'll use them for spanish. |
| 08:17.48 | schmidts | beefcafe imho there is slin192 |
| 08:17.49 | robert_ | :P |
| 08:18.22 | schmidts | robert_ take a look at the pitchshift function before recording something, thats really awesome if you want to get a deep voice |
| 08:18.35 | beefcafe | my phones don't support slin192 |
| 08:18.57 | robert_ | schmidts: ooh, nice. my colleague has a deep voice already. :D |
| 08:19.08 | robert_ | deep voice + pitchshift = epic win probably. |
| 08:19.40 | schmidts | beefcafe i dont know any phone which supports slin192 by now |
| 08:20.22 | beefcafe | any soft client for that? |
| 08:20.36 | robert_ | beefcafe: I like MicroSIP. |
| 08:20.47 | robert_ | It's Windows and has nice aero glass theming. |
| 08:21.17 | beefcafe | cool |
| 08:21.22 | robert_ | yeah. :D |
| 08:26.15 | beefcafe | the best codec ever: gooverthere |
| 08:26.26 | robert_ | lolwut |
| 08:26.44 | beefcafe | go over there |
| 08:26.53 | robert_ | I figured that was what you meant :p |
| 08:27.02 | robert_ | I was just making sure, lol |
| 08:27.09 | beefcafe | better sound than bluray :D |
| 08:27.25 | robert_ | in case I was hasing a case of the stupids. :p |
| 08:27.31 | robert_ | teh stupids, rather. xD |
| 08:27.34 | robert_ | and yeah. :D |
| 08:28.11 | robert_ | links beefcafe with -lgooverthere |
| 08:28.13 | robert_ | :p |
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| 08:28.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:28.40 | beefcafe | what's your sip address? :D |
| 08:28.54 | robert_ | PM's it to you. :D |
| 08:30.29 | schmidts | robert_ i guess codecs will be enhanced until ears 1.0 even cant find any difference |
| 08:30.36 | *** join/#asterisk Neptu (~Hej@mail.avtech.aero) |
| 08:30.53 | robert_ | schmidts: indeed. :p |
| 08:31.14 | robert_ | schmidts: then you get ears 2.0, linked directly with -lbrain :p |
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| 08:33.15 | schmidts | :D |
| 08:33.18 | robert_ | :d |
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| 08:35.26 | *** join/#asterisk tokozedg (5cf1517e@gateway/web/freenode/ip.92.241.81.126) |
| 08:36.07 | tokozedg | Hi, in asterisk 1.4 what's best way to append certain info to file? |
| 08:36.20 | tokozedg | I don't have FILE function. |
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| 08:40.10 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 08:40.14 | kaldemar | tokozedg: probably app System or func SHELL (if the latter even exists). |
| 08:42.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:42.37 | tokozedg | kaldemar: I thought system too, is it normal to use System in production with heavy call traffic? |
| 08:45.25 | kaldemar | normal and normal. it may not work as well as you'd like. test it properly. |
| 08:45.52 | tokozedg | ok, thanks kaldemar |
| 08:46.24 | ChannelZ | one has to do what one has to do if you're gonna run a 7 year old PBX :) |
| 08:50.18 | schmidts | ChannelZ pff 1.2 is still running fine for me |
| 08:51.53 | ChannelZ | features smeatures! heh |
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| 09:54.07 | as001 | hello during configuring my asterisk 1.11 box i saw following line: checking for SSL_CTX_set_tlsext_use_srtp in -lssl... no Is it something which will prevent me from using srtp ? It is on Debian squeeze |
| 09:55.24 | as001 | I am trying to configure asterisk for webrtc and to test it with jssip library but I get error 1106. Has someone some experience with this topic ? |
| 09:57.26 | as001 | ws close code 1006 is the error from demo web site |
| 09:59.41 | as001 | Asterisk version 11.2.1 |
| 10:04.50 | as001 | I followed this link: https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support with change in nat settings which Asterisk wanted because there is depracted nat settings on page. |
| 10:06.44 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 10:13.00 | as001 | I think this error is somehow connected with asterisk http server. I have enabled=yes,bindaddr 0.0.0.0,bindport 8088 in http.conf but when I type mydomain:8088/ws in chrome browser i get message "upgrade required". with mydomadin:8088 I get 'The requested URL was not found on this server.' |
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| 11:36.13 | ruben231 | hi guys any idea on this..? -------------------> http://pastebin.com/S184fKQn |
| 11:39.29 | ruben231 | any idea guys how to completely remove dahdi, i compile it through source..? |
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| 11:43.21 | beefcafe | how do I get webrtc? the wiki page is inadequate |
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| 11:46.19 | ruben231 | any idea guys |
| 11:49.48 | kaldemar | ruben231: you don't have dahdi, therefore no need to remove it. either don't compile chan_dahdi.so at all in asterisk or don't load it (noload => chan_dahdi.so in /etc/asterisk/modules.conf). |
| 11:50.59 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.72.150) |
| 11:53.50 | beefcafe | why does it say Upgrade Required for webrtc? |
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| 11:54.17 | ruben231 | <PROTECTED> |
| 11:54.54 | Greenlight | Asterisk totally deadlocked, dropped phones and wouldn't let them register, CLI unresponive. In the log I see logs of "Exceptionally long voice queue length queuing to"; are they likely cause or effect of the deadlock? |
| 11:55.43 | ruben231 | <PROTECTED> |
| 11:59.37 | *** join/#asterisk disposable (disposable@shell.websupport.sk) |
| 12:12.33 | WIMPy | ruben231: 4% is about as far away from "it works" as I've ever seen. |
| 12:20.02 | disposable | can somebody share their experience of running a busy asterisk in KVM? I'm interested in how it handled the absence of real clock source. how did it affect calls, MoH, meetme, confbridge? |
| 12:21.06 | disposable | by busy i mean 50+ sip peers. conferences of 10+ participants. |
| 12:23.19 | *** join/#asterisk dgv (~dgv@201.21.191.198) |
| 12:32.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 12:34.22 | file | beefcafe, it says that because the HTTP server is trying to upgrade the connection from HTTP to websockets which is used to transport the signaling - going to it in your browser directly is useless |
| 12:38.53 | beefcafe | file: how should I use a web interface? |
| 12:39.47 | file | I don't understand the question. |
| 12:45.55 | *** join/#asterisk FireAndIce (~FireAndIc@219.91.142.242) |
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| 13:05.53 | kaldemar | ruben231: you first asked how to remove DAHDI. your dahdi_test gives results that suggests that DAHDI is completely unusable. what do you need DAHDI for? |
| 13:12.43 | beefcafe | Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 13:12.52 | beefcafe | I keep getting this |
| 13:13.20 | beefcafe | file: like a web phone with dialpad? |
| 13:13.21 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-fqwpxqzellwotzkp) |
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| 13:44.16 | R1ck | I've written an AGI script with python and starpy, when the script finishes (it does everything it's supposed to do), asterisk logs: "-- <SIP/167060001-0000001c>AGI Script agi://127.0.0.1:4573 completed, returning 0 |
| 13:44.20 | R1ck | [Mar 7 14:40:50] ERROR[9275]: utils.c:1164 ast_carefulwrite: write() returned error: Broken pipe" |
| 13:44.25 | R1ck | how do I prevent that? |
| 13:44.46 | *** join/#asterisk hbonath (~hbonath@rrcs-98-102-153-237.central.biz.rr.com) |
| 13:45.43 | *** join/#asterisk jnicola (~jnicola@host2.186-124-123.telecom.net.ar) |
| 13:45.57 | jnicola | Hello! |
| 13:46.04 | jnicola | could somebody help me? i have problems configuring a digium card |
| 13:46.21 | ruben231 | kaldemar: does this not indicae dahdi is working..? ----------------------> http://pastebin.com/XhtxQ1Wd<-------------if i may ask |
| 13:46.29 | jnicola | channels change to blocked state, one by one. I have to restart dahdi service to unblock channels. |
| 13:46.49 | jnicola | Im from argentina and use r2 signaling |
| 13:47.29 | jnicola | dahdi is working, but it works bad. |
| 13:47.50 | jnicola | some days i dont have blocked channels, |
| 13:48.23 | jnicola | but some days, i have to restart dahdi service 2 or 3 times |
| 13:49.57 | beefcafe | Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 13:50.06 | beefcafe | what's wrong with ssl? |
| 13:50.42 | acidfoo | beefcafe, before it was working? |
| 13:51.16 | beefcafe | I didn't notice it wasn't. I think it's working now if it's related to tls |
| 13:52.24 | kaldemar | ruben231: depends on how you define "working". i wouldn't say that anything is working based on that. |
| 13:52.42 | *** join/#asterisk davlefouAMD (~david@197.15.50.19) |
| 13:53.23 | beefcafe | I'm getting the ssl problem every minute or so |
| 13:54.08 | kaldemar | R1ck: does your script read response to every command it sends to asterisk? |
| 13:54.11 | *** join/#asterisk gerhard7 (~gerhard7@195-241-233-43.ip.telfort.nl) |
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| 13:55.34 | R1ck | kaldemar: I'm guessing starpy should take care of that |
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| 14:21.25 | webguy2 | we are running asterisk (obviously) and have digium phones connected. our it guy has left, so i am working things. one of the phones has trouble dialing out, but all other phones are solid. what is the location of the conf file that would be used for one off changes? |
| 14:21.25 | *** part/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz) |
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| 14:32.36 | ghost75 | get it guy |
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| 14:34.07 | webguy2 | ghost75: long story. i am looking for individual digitmap configuration |
| 14:34.14 | webguy2 | at an extension level |
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| 15:22.15 | eXcAliBuR | when using digium conf manager, how can I force the phone to get it's settings based on its mac address, right now it shows a list and I can select the extension, I have mac=<mac address> in my res_digium_config file but it seems to do nothing. |
| 15:24.20 | eXcAliBuR | i seem to be missing config_auth |
| 15:28.52 | *** join/#asterisk camerin (hoax@newelite.bshellz.net) |
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| 15:51.31 | Kyosh | what is an acceptable jitter value for RTP/voip? |
| 15:54.05 | igcewieling | Kyosh: usually 20 - 40 ms depending on the size of the jitterbuffer on the phone |
| 15:54.31 | Kyosh | so jitter values of about 1.67 to 3 ms is fine? hmm |
| 15:54.35 | Kyosh | thanks |
| 15:55.00 | igcewieling | if the jitter is smaller than the jitterbuffer there wont be audio issues |
| 15:55.05 | Kyosh | that seems like a strong difference in latency |
| 15:55.16 | Kyosh | understood |
| 15:56.05 | Kyosh | so asterisk does not use a jitter buffer? |
| 15:56.21 | Greenlight | Kyosh: It can, but usually the endpoints dejitter |
| 15:57.28 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 15:57.59 | Kyosh | got it |
| 15:58.18 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:58.19 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:59.00 | Kyosh | now to find he jitter buffer on the aastra 6757i :p |
| 15:59.11 | igcewieling | Kyosh: check the docs? |
| 15:59.55 | Greenlight | You say you're having issues with <5ms jitter? |
| 16:01.12 | Kyosh | not having issues. just reading up on it. |
| 16:01.27 | Kyosh | igcewieling: the docs are not stating anything about the JB |
| 16:02.00 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
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| 16:38.20 | lodac | would it be possible to use IVR to dial an extention that would be on CCME (through a sip trunk). ex. User calls main number, they enter extention 20510 (which is on CCME) it will then connect it to that ext, upon no anwser it would return them back to Asterisk for voicemail (which voicemail is working). |
| 16:43.02 | navaismo | yep,possible |
| 16:43.35 | lodac | would I have to use asterisk_custom and create an ext for each one? |
| 16:43.51 | lodac | or could I do it from Extentions |
| 16:44.46 | lodac | sip_additional... rather |
| 16:46.47 | navaismo | uh? So you are using freepbx |
| 16:47.21 | navaismo | join # freepbx |
| 16:48.33 | lodac | I am doing most by cmd line, but I am also checking it on gui for other techs and provide documentation. sorry for the confusion |
| 16:49.49 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::1) |
| 16:49.51 | cusco | hi... |
| 16:50.05 | cusco | app_queue.c:9938 load_module: No entries were found for ringinuse/ignorebusy in sippeers table |
| 16:50.16 | cusco | do I need a colum named ringinuse? |
| 16:50.22 | cusco | or a colum state ..? |
| 16:50.25 | cusco | I can't find it in the docs |
| 16:51.19 | lodac | Thanks, I figured it out by trial and error. |
| 16:56.08 | *** join/#asterisk coreyf1513 (~cfarrell@ws2.cfware.com) |
| 17:00.40 | pabelanger | Best way to troubleshoot echo on a PRI? |
| 17:01.12 | pabelanger | Have a site that seems to only be having problems with outbound calls to the CO |
| 17:01.15 | pabelanger | inbound seems good |
| 17:01.53 | leifmadsen | if you hear echo, it's the far end |
| 17:02.08 | leifmadsen | likely some analog path partway in the path |
| 17:02.30 | leifmadsen | that's typically the reason in my experience, as echo can't be generated by the PRI |
| 17:03.41 | pabelanger | leifmadsen, Hmm... Ya, that is what I am trying to trace down |
| 17:03.45 | pabelanger | where it is actually happening |
| 17:06.35 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
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| 17:49.48 | igcewieling | pabelanger: what PRI card? |
| 17:50.03 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
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| 17:56.14 | pabelanger | 03:02.0 Communication controller: Digium, Inc. Wildcard TE210P/TE212P dual-span T1/E1/J1 card 3.3V (rev 02) |
| 17:56.36 | igcewieling | I assume it has a built in echo canceler? |
| 17:57.09 | pabelanger | Ya, HWEC |
| 17:58.19 | igcewieling | you have the EC addon module? |
| 17:58.45 | pabelanger | what do you mean? |
| 17:59.06 | igcewieling | From the docs "The TE200 Series cards, unless otherwise equipped, utilize Asterisk to perform software-based echo cancellation" |
| 17:59.21 | igcewieling | does it show a HW EC in dmesg when the card drivers load? |
| 17:59.23 | pabelanger | Ya, have hardware echo |
| 17:59.25 | pabelanger | yes |
| 17:59.42 | pabelanger | Going to upgrade the firware for it later this evening |
| 17:59.46 | pabelanger | I think it is old |
| 17:59.50 | pabelanger | see if that helps |
| 17:59.50 | igcewieling | pabelanger: contact Digium support. I believe echo on a card with onboard EC is a supportable issue. |
| 18:00.22 | igcewieling | When I was unemployed I made my living for 2 months selling Tellabs EC units to people with Digium cards which had HW EC. |
| 18:01.02 | igcewieling | pabelanger: you can try lowering your txgain and see if it helps |
| 18:01.18 | pabelanger | igcewieling, Ya, that is what we are doing this afternoon |
| 18:01.24 | pabelanger | waiting for client to get back from lunch |
| 18:09.29 | *** join/#asterisk wltjr_ (~wltjr@unaffiliated/wltjr) |
| 18:10.39 | wltjr_ | greetings, is sip qualification a good thing? since I started using that, seeing remote sip peers like the providers trunk becoming unreachable, at times it has been due to network issues, but not sure if that is always the case |
| 18:11.13 | wltjr_ | read stuff about keepalive being better, and qualify interrupting calls and such, as I have seen, when peers become unreachable, if they really are |
| 18:13.15 | igcewieling | wltjr_: I like qualify=10000 and qualifyfreq=120 unless you have NAT involved, then I make qualifyfreq=30 |
| 18:17.58 | wltjr_ | igcewieling: ok, I can try increasing that, it seemed to help some when I did before, presently at 5000/30, so guess I should keep the 30 since I am natted for the main peer I am having issues with, and increase qualify time |
| 18:18.42 | wltjr_ | pretty sure its issues with comcast, they seem to have brief routing issues at random times, I see it on a few diff * boxes conneced to same peer in diff parts of the city, on comcast network |
| 18:19.07 | leifmadsen | a lot of the time the firewall/NAT equipment configuration makes a big difference |
| 18:19.20 | leifmadsen | we'll see peers time out for practically no reason, almost independent of the qualify settings |
| 18:19.37 | leifmadsen | always ends up being something on the remote endpoint, like ALG or whatever other settings |
| 18:19.51 | *** join/#asterisk CunningPike (~CunningPi@d206-116-73-41.bchsia.telus.net) |
| 18:19.54 | wltjr_ | leifmadsen: I will have to look into the nat devices, just linux routers, couple buffalos and one linksys, running openwrt |
| 18:20.46 | wltjr_ | though I can do pings and see the same inconsistent times, thus prretty sure its routing issues with Comcast, a few times I know for a fact it was, their techs even admitted it, other times when they were on sight diagnosing noise issues |
| 18:21.15 | wltjr_ | leifmadsen: just not sure if they are doing it for no reason, or a real issue |
| 18:21.49 | *** join/#asterisk kikohnl (~keith@udp278022uds.hawaiiantel.net) |
| 18:22.41 | wltjr_ | but as long as others are using qualify I feel a bit better, some stuff I was reading was saying its better to use keepalive and not qualify, because qualify might interrupt calls and that I have seen |
| 18:31.02 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 18:37.32 | pabelanger | anybody else with Bria have echo problems on speaker phone? |
| 18:37.53 | Qwell | Bria makes hardphones now? |
| 18:38.02 | pabelanger | soft |
| 18:38.08 | pabelanger | andriod |
| 18:45.47 | *** join/#asterisk teff (~teff@client-80-1-160-118.bsh-bng-011.adsl.virginmedia.net) |
| 18:45.48 | Qwell | top |
| 18:45.50 | cusco | pabelanger: bria works flawlessly for me in iphone |
| 18:45.57 | Qwell | This isn't my console. |
| 18:46.22 | hal-j | I need to pick up Bria |
| 18:46.30 | pabelanger | cusco, okay, thanks. Guess I need to find somebody running on andriod |
| 18:47.16 | hal-j | pabelanger: their support folks are technically capable folks, if you wanted to go that route |
| 18:47.49 | pabelanger | hal-j, Ya, haven't reached out to them yet |
| 18:47.58 | pabelanger | was just checking others XP first |
| 18:48.11 | hal-j | I know the support guys there |
| 18:48.13 | cusco | also it may work on a handset and not on another one |
| 18:48.42 | pabelanger | hal-j, where should I contract then at? |
| 18:48.44 | pabelanger | contact* |
| 18:51.13 | hal-j | check the website, follow supprot links, I'd bet |
| 18:53.22 | AkkerKid | SIP to Asterisk to FXS to Fax Machine Is T38 possible? |
| 19:00.11 | igcewieling | Addisk: only with Asterisk 11 |
| 19:00.16 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 19:00.55 | igcewieling | ...er.. AkkerKid only with Asterisk 11 |
| 19:03.42 | AkkerKid | hmmm... |
| 19:04.10 | AkkerKid | is there anything else I can do to improve on what i'm running curently? |
| 19:04.19 | AkkerKid | which is 1.6.2 i believe... |
| 19:05.14 | igcewieling | AkkerKid: define "improve". Asterisk versions before v11 do not in any way shape or form support T.38 gateway features (SIP to FXS). |
| 19:05.27 | *** join/#asterisk LoKoMurdoK (~soporte@fedora/LoKoMurdoK) |
| 19:05.31 | *** part/#asterisk LoKoMurdoK (~soporte@fedora/LoKoMurdoK) |
| 19:05.33 | igcewieling | Correction, Asterisk 10 might, but there is no reason to use 10 .vs. 11 |
| 19:05.51 | AkkerKid | as it is now i have somewhat flaky support for sip to fax through FXS |
| 19:05.53 | igcewieling | If you want to "improve" the situation then upgrade. |
| 19:06.04 | igcewieling | AkkerKid: that would be with ulaw, not T.38 |
| 19:06.20 | igcewieling | i,e, FaxOverVoiceOverIP using ulaw. |
| 19:06.27 | igcewieling | that will never be reliable. |
| 19:06.50 | AkkerKid | ok just making sure |
| 19:07.49 | AkkerKid | i'm using ulaw now. i've turned off echo cancelation. I'm just wondering if there are any other tweaks I could employ to further improve transmission |
| 19:07.50 | *** join/#asterisk _methods (~minotaur@192.81.210.43) |
| 19:09.47 | igcewieling | several. upgrade to Asterisk 11 and use T.38 or use POTS. |
| 19:10.09 | igcewieling | other than that you are wasting your time tying to solve a known limitation of SIP. |
| 19:10.15 | AkkerKid | understood. |
| 19:10.20 | AkkerKid | thanks for your help. |
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| 19:29.14 | *** join/#asterisk linlin (will@173.243.115.75) |
| 19:30.19 | pabelanger | So, -10 tx / rx seems to be the happy spot for echo |
| 19:30.49 | linlin | hey everyone. im deciding to finally make the move for my home phone to an asterisk system... currently have dialtone through comcast digital voice. how hard is it to port a residential number to a sip provider and who can you reccomend for this service? I've used vitelity.net in the past for toll free did's etc and they were very good. any other succestions for a service that will have |
| 19:30.50 | linlin | minimal usage? |
| 19:31.13 | lorsungcu | vitelity is ok, flowroute is better |
| 19:31.23 | lorsungcu | it'll cost you somewhere between 10-25$ |
| 19:31.55 | linlin | per month ? |
| 19:32.48 | lorsungcu | no |
| 19:32.50 | lorsungcu | to port |
| 19:33.01 | *** join/#asterisk jkroon (~jkroon@105.246.173.47) |
| 19:33.09 | jkroon | hi guys, it seems under some circumstances asterisk (11.2.1) will keep indefinitely postponing the destruction of dialog (Autodestruct on dialog 'BW094317320070313483758522@196.35.130.5' with owner SIP/IS-00000e3e in place (Method: BYE). Rescheduling destruction for 10000 ms) - given enough of these they can (and in my case does) eventually take the system down. |
| 19:33.10 | lorsungcu | their rates are posted on their site, check it there |
| 19:33.19 | jkroon | any suggestions on what to look for and how to fix it? |
| 19:33.50 | linlin | alright cool ill look into both |
| 19:35.09 | linlin | back some years ago...the last time i messed with asterisk at home...i was using sipura boxes for FXS devices... are sipura boxes still good for FXS or is there something newer and better out? my system will not have PCI card slots so it has to be an external box to interface to the phones until I eventually replace with ip phones |
| 19:35.38 | lorsungcu | they are fine, although you'll probably have better luck finding them as cisco/linksys now |
| 19:36.19 | *** join/#asterisk Neptu (~Hej@c213-89-2-159.bredband.comhem.se) |
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| 19:38.31 | linlin | are these chinese pci-e fxs/fxo cards any good? |
| 19:38.50 | igcewieling | jkroon: I experienced a similar problem and never figured out it, but it isn't happening anymore. You might want to double check and make sure ALL NAT'd clients are indeed working and you don't see any retransmit messages when you enable sip debug |
| 19:39.21 | jkroon | mostly happens on the one peer that I do NOT have NAT to/from. |
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| 20:02.17 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 20:02.20 | saint_ | hi all |
| 20:02.53 | lorsungcu | sup yo |
| 20:04.10 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 20:04.24 | *** join/#asterisk k610 (~K610@cable-78.29.241.186.coditel.net) |
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| 20:09.48 | *** join/#asterisk zoidberg- (~zoidberg@irc2.labs.cybergangsta.net) |
| 20:10.13 | zoidberg- | Hey guys, whats the easiest way to compile asterisk from source with debugging symbols? |
| 20:30.02 | *** join/#asterisk polysics (~Adium@host22-73-dynamic.16-79-r.retail.telecomitalia.it) |
| 20:30.11 | polysics | hi there |
| 20:30.22 | polysics | can I dial an user on another SIP machine using Dial()? |
| 20:30.41 | polysics | what's the syntax for dialing, say, 500@othermachine.com, please? |
| 20:31.35 | *** part/#asterisk zoidberg- (~zoidberg@irc2.labs.cybergangsta.net) |
| 20:35.36 | ChannelZ | Dial(SIP/somewhere.com/500) |
| 20:36.05 | pabelanger | Aside from SIPVicious, what other voip scanners are there? |
| 20:36.10 | polysics | not SIP/500@whatever ? |
| 20:36.10 | pabelanger | I want to do a pen test on my box |
| 20:36.17 | polysics | ok, so that's what was wrong |
| 20:36.18 | pabelanger | since, somebody made it past my proxy |
| 20:37.13 | polysics | would that be something that scans for users? trying 100, 101 etc? |
| 20:37.25 | ChannelZ | Dial(SIP/500@somewhere.com) pretty much means the same thing AFAIK |
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| 20:43.08 | *** part/#asterisk ghost75 (~trechber@dslb-188-105-026-174.pools.arcor-ip.net) |
| 20:45.21 | igcewieling | pabelanger: are you running freepbx? |
| 20:46.59 | leifmadsen | no he is not |
| 20:47.06 | leifmadsen | paul does things correctly |
| 20:47.20 | leifmadsen | when he says proxy, he really means proxy (kamailio) |
| 20:47.37 | igcewieling | ah. |
| 20:48.17 | polysics | paul and leif are amongst the enlightened few :-) |
| 20:50.02 | pabelanger | igcewieling, what leifmadsen said |
| 20:50.40 | wdoekes | pabelanger: how do you know someone made it past the proxy? did they get through or past it? |
| 20:50.47 | igcewieling | now if I could only find a consultant who is a T.38 god. |
| 20:52.25 | pabelanger | wdoekes, because I can see traffic in asterisk attempting to dialout |
| 20:52.32 | pabelanger | all registrations are handled in kamailio |
| 20:52.38 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-026-174.pools.arcor-ip.net) |
| 20:52.47 | pabelanger | so, either they know the username / password. Which is not likely. only 2 accounts |
| 20:52.56 | pabelanger | or they somehow found a way past my auth |
| 20:53.23 | pabelanger | So, I just enabled ANTI_FLOOD in kamailio, so stop the flood and works |
| 20:53.32 | pabelanger | but really curious how they made it into asterisk |
| 20:54.32 | wdoekes | or rather, how they made it through the proxy.. assuming you do the auth there? |
| 20:54.42 | pabelanger | correct |
| 20:55.02 | wdoekes | pedantic=yes ? |
| 20:55.42 | pabelanger | it is |
| 20:56.08 | wdoekes | ok.. so it wasn't in-dialog requests starting new calls |
| 20:56.36 | pabelanger | Ya, I didn't have any debug or traces going |
| 20:56.38 | pabelanger | but I do now |
| 20:56.45 | wdoekes | rotating pcaps |
| 20:56.48 | pabelanger | so, I'll have to wait until another attack |
| 20:56.53 | wdoekes | never leave home without ', |
| 20:56.56 | wdoekes | 'm |
| 20:56.58 | pabelanger | or find the tool to attack my self |
| 20:58.09 | wdoekes | I'm assuming you don't have useful xlogs in the proxy that help |
| 20:59.17 | pabelanger | wdoekes, correct |
| 20:59.24 | pabelanger | for the future I do |
| 21:01.24 | wdoekes | bad preloaded route handling? |
| 21:01.53 | pabelanger | possible |
| 21:02.00 | pabelanger | mostly the default configuration file |
| 21:02.18 | pabelanger | if so, want to fix it and submit it back upstream |
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| 21:22.46 | thegoat | quick question. are the sql scripts for creating the mysql cdr DB in the source files anywhere? |
| 21:34.32 | kaldemar | thegoat: see under contrib |
| 21:34.45 | thegoat | cool, tks |
| 21:37.50 | *** join/#asterisk ccrnp (~ccr@208-90-186-2.warp2biz.com) |
| 21:39.30 | ccrnp | Hi room |
| 21:40.55 | ccrnp | I have an issue where IAX trunk won't send 180 ringing sip message to sip extensions. |
| 21:41.11 | ccrnp | has anyone encountered ringback issues with IAX trunk >? |
| 21:42.07 | pabelanger | wdoekes, wow, figured it out. INVITEs are not authenticated |
| 21:45.45 | Qwell | well there's your problem |
| 21:46.03 | pabelanger | indeed |
| 21:47.12 | igcewieling | this is why I don't try to configure Kamailio myself. LOL~ |
| 21:47.41 | igcewieling | Does anyone happen to know a T.38 guru who does consulting? |
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| 21:56.26 | *** join/#asterisk _abc_ (~user@unaffiliated/ccbbaa) |
| 21:57.04 | _abc_ | Hello. I have trouble getting sccp by git. Can you confirm that it fails with an unexpected EOF error? Here: git clone git://git.xivo.fr/official/xivo-libsccp.git |
| 21:57.08 | _abc_ | thanks |
| 21:58.04 | _abc_ | (aside: git is getting my gut) |
| 21:58.08 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 22:04.44 | navaismo | _abc_, same error here |
| 22:04.57 | _abc_ | thanks |
| 22:04.58 | navaismo | Cloning into 'xivo-libsccp'... |
| 22:04.58 | navaismo | fatal: The remote end hung up unexpectedly |
| 22:05.17 | _abc_ | is that skinny version merged into current 1.8 asterisk or not? |
| 22:05.25 | _abc_ | It is not clear what version is in that repo |
| 22:07.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:08.46 | _abc_ | anyway, thanks |
| 22:08.52 | _abc_ | I will be back some day heh ;) |
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| 22:24.52 | menti | Hi guys, i'm trying to connect Asterisk with Asterisk.NET library, i have an exception and i discovered that it's because the socket return an exception...... Asterisk is refusing the connection, i googled it, but i can't find the solution |
| 22:28.42 | menti | have you ever had this problem? or have any idea? Thank you so much in advance guys |
| 22:33.28 | [TK]D-Fender | what does ASTERISK say? |
| 22:34.01 | menti | Asterisk Nothing, the problem is with the socket connection |
| 22:34.10 | menti | and i made a tcpdump (i have traffic) |
| 22:34.33 | [TK]D-Fender | care to show us? |
| 22:34.53 | [TK]D-Fender | Also to prove that you have a proper config to respond to it with ... |
| 22:37.52 | menti | Show you how? i mean... what config do you want to see? |
| 22:38.12 | [TK]D-Fender | ... |
| 22:38.22 | [TK]D-Fender | the one your your CONNECTION requires you to have set up |
| 22:43.34 | [TK]D-Fender | And times up... I''m out for the night.... |
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| 23:34.31 | jsgoecke | Hello all |
| 23:34.49 | jsgoecke | So, I am on 64-bit Ubuntu and did do: ./configure --libdir=/usr/lib64 |
| 23:35.07 | jsgoecke | But still get this error when I try to start Asterisk 11: error while loading shared libraries: libasteriskssl.so.1 |
| 23:35.26 | jsgoecke | Searching the interwebs I can't find a solution beyond the config option or disabling |
| 23:35.28 | jsgoecke | Any other ideas? |
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| 23:55.02 | igcewieling | jsgoecke: libasteriskssl.so.1 is likely installed somewhere, do a find and tell us where it actually is |
| 23:55.12 | jsgoecke | it is in /usr/lib64 |
| 23:55.21 | jsgoecke | along with the rest of the Asterisk modules |
| 23:55.46 | jsgoecke | this is ubuntu 12.10 btw |
| 23:56.01 | igcewieling | if you have no other ideas, try copying it to /usr/lib64/asterisk/modules and/or /usr/lib/asterisk/modules |
| 23:56.50 | jsgoecke | yeah, I figured it might just be that |
| 23:56.53 | jsgoecke | ty |
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| 23:58.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 23:59.02 | robert_ | asterisk keeps complaining that it can't ast_db_put |
| 23:59.05 | jsgoecke | did the trick of course |