| 00:04.42 | LemensTS | Anyone know of a milliwatt test number? |
| 00:06.48 | hardwire | provide your own through an aux DID route :) |
| 00:09.38 | LemensTS | On another server with a DID, just send it into 1004? |
| 00:15.36 | leifmadsen | LemensTS: I can set that up real quick |
| 00:15.39 | leifmadsen | can you dial a SIP URI? |
| 00:15.43 | leifmadsen | or does it have to be a DID? |
| 00:16.32 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 00:17.20 | leifmadsen | LemensTS: sip:milliwatt@leifmadsen.com |
| 00:17.22 | LemensTS | leifmadsen: yea that would be cool |
| 00:17.26 | leifmadsen | LemensTS: or 416-479-0259 x4 |
| 00:17.35 | LemensTS | leifmadsen: appreciate it! |
| 00:17.41 | leifmadsen | np was easy :) |
| 00:17.50 | leifmadsen | and now I'm off for a night of board games and drinking |
| 00:18.05 | LemensTS | good luck cheaters win |
| 00:18.07 | LemensTS | ;) |
| 00:20.59 | *** join/#asterisk Godfather_ (~estanteri@223.Red-88-19-153.staticIP.rima-tde.net) |
| 00:31.46 | LemensTS | im getting like 1433 on my RX, is that even possible when it should be 14844 |
| 00:32.58 | LemensTS | dahdi_monitor 3 -vv |
| 00:34.16 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 00:45.52 | LemensTS | Anyone? Ill pay. |
| 00:48.01 | KNERD | Are there VMWare images for AsteriskNOW? |
| 01:17.42 | *** join/#asterisk shadowapex (~William@adsl-99-107-163-194.dsl.lsan03.sbcglobal.net) |
| 01:18.43 | shadowapex | Does anyone know of a method of handling faxes with Asterisk where you can store the actual fax files within a database? |
| 01:22.06 | shadowapex | Trying to create a realtime faxing solution where we can store faxes in a database and all the fax connection sip/iax details in the database; kind of like the cc_sip_buddies table |
| 01:24.35 | pabelanger | sql blob? |
| 01:24.39 | shadowapex | Yeah |
| 01:25.47 | *** join/#asterisk LemensTS (~matthew@adsl-70-130-149-3.dsl.stlsmo.swbell.net) |
| 01:27.50 | shadowapex | We'd like to have faxing sort of like how you can use Asterisk with the voicemail-odbc-storage addon that lets you store voicemail messages in the database. |
| 01:37.33 | shadowapex | :< |
| 01:47.49 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |
| 01:56.56 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
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| 02:23.29 | *** join/#asterisk manji (~manjiki@94-193-165-46.zone7.bethere.co.uk) |
| 02:26.47 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-cyayhuenwugweope) |
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| 02:30.53 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 02:38.30 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 02:46.51 | saxa | hi ppl, I have a problem with SIP connection, behind a nat, sometimes it works, sometimes not. I can hear the music if I put the IP number in the phone, and if I put the domain name , I can make the call but I don't hear anything |
| 02:47.12 | saxa | do I need to enable some special option in sip.conf ? |
| 02:48.14 | saxa | for example if I change the codec from gsm to ulaw it doesnt work anymore |
| 02:48.41 | iprouteth0 | saxa: is the CLI telling you anything? |
| 02:51.07 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-wodtfgstoowqfvwo) |
| 02:53.07 | saxa | iprouteth0: yeah, it shows everything as if it functions normal |
| 02:53.28 | saxa | so I don't get anything useful from there |
| 02:53.31 | iprouteth0 | have you used rtp debug? |
| 02:53.54 | saxa | i set up externhost= |
| 02:54.02 | saxa | have not tried rtp debug |
| 02:54.05 | saxa | let me see |
| 02:54.31 | saxa | for example, now I tried and it works |
| 02:54.58 | saxa | I have changed from gsm to ulaw and in the first dial attempt i didnt hear anything |
| 02:55.21 | saxa | after 5 minutes i retried and it worked |
| 02:55.24 | saxa | huh |
| 02:55.27 | saxa | strange |
| 02:55.40 | saxa | it needs so long to update ? |
| 02:56.18 | iprouteth0 | rtp debug showing you anything? |
| 02:56.39 | saxa | let me try to start rtp debug |
| 02:57.28 | iprouteth0 | is the call inbound or outbound? |
| 02:58.58 | saxa | http://pastebin.com/F87qjzAS |
| 02:59.24 | saxa | iprouteth0: i have a sip phone connected in my home via internet to my office, where is the asterisk box |
| 02:59.51 | saxa | so i try to call from my 1005 extension to my asterisk box |
| 03:00.58 | iprouteth0 | so it's an on-net call, but the destination endpoint is remote? |
| 03:01.06 | saxa | the ip you see in the pastebin is my internal ip at home |
| 03:02.03 | saxa | iprouteth0: yes, 1005 is my ext, in my home, from where I connect via internet (sip) to my office |
| 03:02.19 | saxa | my office is about 14km from my home |
| 03:02.22 | iprouteth0 | hmmm... |
| 03:02.37 | iprouteth0 | what is your canreinvite set to? |
| 03:02.55 | saxa | i have not set this option |
| 03:03.10 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |
| 03:03.52 | saxa | ok, seems that my bandwith is not enogh |
| 03:04.06 | iprouteth0 | that might explain why it is intermittent |
| 03:04.30 | saxa | when i ssh into the asterisk box in my office from my home, and try to call , I do not hear the music |
| 03:04.32 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 03:04.39 | saxa | if I disconnect |
| 03:04.40 | iprouteth0 | Though it would seem to me that the RTP stream should be going to your public IP at home |
| 03:04.43 | saxa | it works |
| 03:04.52 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:05.11 | iprouteth0 | works everytime if SSH session is not open? |
| 03:05.15 | saxa | iprouteth0: thats also true, i find it strange that it lists my internal ip |
| 03:06.26 | iprouteth0 | if the phone is registered successfully though, and you're getting audio sometimes, it may not be an issue that it's displaying private IP for that extension |
| 03:06.50 | saxa | iprouteth0: i tried to call many times and it worked all of the times |
| 03:07.02 | iprouteth0 | audio stream worked too? |
| 03:07.09 | saxa | yeah |
| 03:07.17 | saxa | i also parked myself |
| 03:07.26 | iprouteth0 | what is your connection bandwidth? |
| 03:07.27 | saxa | and i heard music |
| 03:07.29 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 03:07.53 | saxa | i have 10MB/1MB at home and 1MB/256kb at office |
| 03:08.19 | saxa | let me try with ssh open now |
| 03:09.10 | iprouteth0 | i imagine the office upload bandwidth to be the issue |
| 03:09.29 | drmessano | ~sipnat |
| 03:09.30 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 03:09.41 | drmessano | Follow that, if you havent already |
| 03:09.42 | *** join/#asterisk sourcode (~code@ppp-58-11-75-225.revip2.asianet.co.th) |
| 03:09.50 | iprouteth0 | if it fails while SSH is connected, see if there is a different when using ssh -C |
| 03:11.31 | saxa | http://pastebin.com/b3ENJtuw |
| 03:11.49 | saxa | as you can see from that, the first time it worked, and I could hear the music |
| 03:12.14 | saxa | but then I try to call the voicemail and it apeared in the console, but I have not heard anything |
| 03:12.28 | saxa | thats with the ssh connected |
| 03:12.57 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:13.19 | *** join/#asterisk Aut0Exec (~root@65.75.65.130) |
| 03:13.50 | saxa | iprouteth0: i retired to call and on the first attempt it worked, and i could hear voicemail |
| 03:14.07 | saxa | on seccond i called moh and it worked also |
| 03:14.16 | Aut0Exec | hi guys... i'm just getting into asterisk(asterisk nub here) how to i utilize a phone adapter to use more than 1 phone... like say 4 phones? |
| 03:14.17 | saxa | so most probably is the bandwith |
| 03:15.00 | saxa | drmessano: thx, will try to check that guide tomorrow |
| 03:15.54 | iprouteth0 | I'm not sure you even have enough pipe to effectively set a QoS scheme |
| 03:16.21 | iprouteth0 | I only have 384K myself, and it can sometimes struggle |
| 03:16.29 | iprouteth0 | and my extensions are local |
| 03:16.38 | *** join/#asterisk DrDigi (~mmurphy@108-69-211-16.lightspeed.frokca.sbcglobal.net) |
| 03:17.05 | iprouteth0 | because your extension is not local it may utilize additional bandwidth though i'm not sure |
| 03:17.06 | saxa | yup, probably the problem is the pipe :( , unfortunately i cant upgrade that |
| 03:17.20 | Aut0Exec | would I need a bigger phone adapter? |
| 03:17.25 | saxa | will try to leave gsm as codec |
| 03:17.33 | iprouteth0 | may help |
| 03:17.40 | iprouteth0 | not sure what the gsm compression is |
| 03:17.49 | iprouteth0 | if you can use g729, might be good too |
| 03:17.56 | saxa | gsm should be the lowest one iirc |
| 03:18.15 | iprouteth0 | I can only utilize g711u due to my provider limitation |
| 03:18.16 | iprouteth0 | s |
| 03:18.23 | Aut0Exec | :| |
| 03:18.28 | saxa | will try g729 , need to check which one is the less consumable one |
| 03:18.47 | saxa | Aut0Exec: reform your question |
| 03:19.02 | saxa | Aut0Exec: which adapter are you talking to _ |
| 03:19.25 | saxa | s/to/about |
| 03:20.04 | saxa | iprouteth0: thx for your help in any case, I will try to look at that deeper tomorrow |
| 03:20.19 | Aut0Exec | saxa, soz i just finished watching the systm vid on the asterisk site... and i'm looking at the sipura phone adapter they are using but in the demo they areonly using 1 phone... my question is what if i wanted like a little office setup.. what kind of equipment would I need? |
| 03:20.33 | saxa | so probably using iax and sip together on that connection wont work at all. |
| 03:20.40 | iprouteth0 | saxa: happy to help. I've found helping others assists my learning alot as well |
| 03:21.03 | iprouteth0 | [I have an linksys ATA with two FXS ports |
| 03:21.07 | iprouteth0 | gets you two extensions |
| 03:21.27 | saxa | Aut0Exec: sipura is a soft phone ? |
| 03:21.46 | Aut0Exec | lol no phone adapter.... fxs, fxo |
| 03:21.47 | saxa | if so, you just need 4 different computers connected together |
| 03:21.49 | Aut0Exec | analog to voip |
| 03:21.55 | saxa | oh |
| 03:21.56 | iprouteth0 | it may be possible if you were to disconnect test jack in PSTN NID you might be able to plug your ATA into existing POTS home wiring |
| 03:22.00 | saxa | i have a tdm410 |
| 03:22.23 | iprouteth0 | I got my linksys ATA for $20 US... works great! |
| 03:22.44 | saxa | this card have 4 in or 4 out depends on what you put in for fxs or fxo |
| 03:22.45 | Aut0Exec | oh really |
| 03:22.52 | Aut0Exec | ok |
| 03:22.59 | Aut0Exec | so that would only give me 4 phones? |
| 03:23.12 | Aut0Exec | if i wanted like 8 then get 2 cards? |
| 03:23.13 | Aut0Exec | etc? |
| 03:23.24 | saxa | Aut0Exec: that would give you 4 analog lines |
| 03:23.26 | iprouteth0 | ATAs might be cheaper |
| 03:23.46 | saxa | Aut0Exec: digium.com has many solutions |
| 03:23.51 | drmessano | I love using a Linksys ATA to feed a house |
| 03:24.00 | Aut0Exec | oh really |
| 03:24.03 | Aut0Exec | ata ok |
| 03:24.05 | Aut0Exec | i'll check that |
| 03:24.07 | saxa | you have cards with up to 48 or 50 channels iirc |
| 03:24.08 | iprouteth0 | drmessano: I take it you've done it that way? :) |
| 03:24.15 | drmessano | Quite a bit |
| 03:24.27 | iprouteth0 | drmessano: it's more or less what the cable ISPs do for their VOIP setups |
| 03:24.46 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 03:24.52 | iprouteth0 | i've always wanted to try it, but my FXS at home is in use for my ADSL |
| 03:24.59 | drmessano | Find a phone jack near an ethernet connection, make sure ma bell is unplugged at the NID, and plug the ATA into the jack.. Done |
| 03:25.25 | iprouteth0 | figured it would be that easy. As long is NID is disco'ed |
| 03:26.08 | drmessano | I've also fed both lines into a jack.. Just need to cut a cord and fan the red/green to the outer pair, use a simple 2 line splitter |
| 03:26.14 | p3nguin | Everyone does it this way because it's the easiest way. |
| 03:26.30 | drmessano | or get one of those that spits out line 2 on one side of the adapter on green/red |
| 03:26.38 | iprouteth0 | We have markets where we've run FTTP/FTTH, and do much the same with existing wiring |
| 03:26.44 | iprouteth0 | but we feed it into the ONT |
| 03:26.54 | Aut0Exec | iprouteth0, where did u get ur ATA? |
| 03:27.42 | iprouteth0 | Aut0Exec: http://www.google.com/search?q=tdm410&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a#sclient=psy&hl=en&client=firefox-a&rls=org.mozilla:en-US%3Aofficial&biw=1280&bih=832&tbs=shop:1&q=linksys+pap2-na&aq=0p&aqi=p-p1g4&aql=f&oq=&pbx=1&bav=on.2,or.r_gc.r_pw.&fp=d57722a22572de1d |
| 03:27.47 | Aut0Exec | k thanks |
| 03:27.56 | iprouteth0 | hope the link works |
| 03:28.08 | iprouteth0 | anyway I just searched for linksys PAP2 on google shopping |
| 03:28.12 | drmessano | I miss the days of modifying PAP2s from Vonage |
| 03:28.18 | drmessano | Got them as cheap as $5 |
| 03:28.21 | iprouteth0 | My father actually got it for me for my birthday |
| 03:28.33 | iprouteth0 | Mine came unlocked |
| 03:28.48 | drmessano | I unlocked like 50+ of those things |
| 03:28.49 | drmessano | lol |
| 03:28.53 | Aut0Exec | iprouteth0, how many phone would that allow you to connect? |
| 03:29.13 | iprouteth0 | Aut0Exec: two, unless you utilize the method drmessano and I were speaking of |
| 03:29.27 | iprouteth0 | I shouldnt say two phones |
| 03:29.36 | iprouteth0 | cause you can use splitters and the like |
| 03:29.40 | drmessano | If you split the jack out you can put quite a few phones on it.. It's 2 EXTENSIONS though |
| 03:29.42 | iprouteth0 | think of it more like two analogue circuits |
| 03:29.57 | drmessano | ^^^ |
| 03:30.06 | drmessano | It's two LINES.. |
| 03:30.16 | Aut0Exec | ok |
| 03:30.39 | drmessano | Just dont put some old phone with a bell ringer on it on there |
| 03:30.51 | iprouteth0 | lol |
| 03:30.53 | drmessano | it will sound like a dying phone when ringing |
| 03:31.19 | iprouteth0 | yeah, definitely use more modern analogue phones |
| 03:31.33 | Aut0Exec | lol ok |
| 03:31.56 | iprouteth0 | man. I really want to mess around with asterisk and a PRI |
| 03:32.21 | iprouteth0 | maybe I'll have to bother someone at work for that... |
| 03:32.28 | Aut0Exec | if i use splitters thats like maybe 4 phones |
| 03:32.49 | Aut0Exec | i'm real nub here |
| 03:33.29 | KNERD | nubs in tubs |
| 03:33.31 | iprouteth0 | Aut0Exec: If your deployment is just for at home, and you are not using xDSL, then you can disconnect the telco from your NID(main phone junction, usually outside) |
| 03:33.45 | iprouteth0 | and then connect your ATA to one of your phone jacks |
| 03:33.56 | iprouteth0 | then use regular phones in all of your existing phone jacks |
| 03:34.33 | Aut0Exec | ohhhhh |
| 03:34.35 | Aut0Exec | i gatcha |
| 03:34.48 | Aut0Exec | :) |
| 03:34.54 | iprouteth0 | if you don't disconnect the Telco from the NID you're likely to damage your ATA |
| 03:35.22 | iprouteth0 | again if you are using any kind of a DSL connection, that won't work |
| 03:35.30 | Aut0Exec | ok |
| 03:35.37 | Aut0Exec | sounds good |
| 03:35.41 | iprouteth0 | unless you run ONE phone jack that connects to telco, and the rest of the jacks in home are disconnected from telco |
| 03:35.58 | iprouteth0 | of course on the ONE ran to telco is where your DSL modem would connect |
| 03:36.13 | iprouteth0 | you haven't mentioned what type of ISP service you have |
| 03:36.22 | Aut0Exec | lol dude i dont live in USA |
| 03:36.27 | Aut0Exec | i use cable |
| 03:36.39 | Aut0Exec | i have like 3gig down |
| 03:36.46 | Aut0Exec | and like 1 up |
| 03:37.06 | Aut0Exec | no xDSL |
| 03:38.00 | Aut0Exec | i guess in the mean time until I get a phone adapter I can use a soft phone right? |
| 03:38.09 | Aut0Exec | soft sip phone? |
| 03:41.22 | iprouteth0 | yes |
| 03:41.32 | iprouteth0 | what OS are you using? |
| 03:47.16 | saxa | fromdomain = yourdomain.com |
| 03:47.28 | saxa | does this works with asterisk 18 ? |
| 03:47.38 | saxa | 1.8 |
| 03:48.08 | saxa | or is externhost= better to use ? |
| 03:48.17 | saxa | all of those in sip.conf . |
| 03:52.27 | iprouteth0 | I use externhost and externrefresh |
| 03:52.34 | iprouteth0 | along with no-ip dynamic dns |
| 03:53.33 | iprouteth0 | I think fromdomain is like an ACL for where * will accept calls from |
| 03:53.36 | saxa | ok, thats also what i used, since it was already in sip.conf |
| 03:53.55 | iprouteth0 | oh nm |
| 03:54.07 | saxa | i saw that fromdomain option in voip-info.org |
| 03:54.09 | iprouteth0 | fromdomain is to identify yourself when making outbound to another domain |
| 03:54.35 | iprouteth0 | One thing I love about the Asterisk GUI is that there are info flags for just about every option |
| 03:54.45 | iprouteth0 | really helps point you in the right direction sometimes |
| 03:55.16 | saxa | heh, i need to try it once |
| 03:55.31 | saxa | i use just a plain selfcompilled * on slackware |
| 03:55.32 | iprouteth0 | it was pretty easy for me to install on my setu |
| 03:55.32 | iprouteth0 | p |
| 03:55.40 | iprouteth0 | should be easy on yours too |
| 03:55.47 | saxa | probably yes :) |
| 03:55.50 | iprouteth0 | i'm running on gentoo, but * is out of the portage tree |
| 03:56.05 | p3nguin | externhost is a general sip parameter used for when your Asterisk is behind a NAT. It has no relation to fromdomain, which is a peer parameter. |
| 03:56.11 | iprouteth0 | I just downloaded from SVN and built. there is a checkconfig option that helps you set it up also |
| 03:56.23 | saxa | but for me was straightforward to compile it. The code is very very mature |
| 03:56.39 | saxa | hey p3nguin , hello. |
| 03:57.35 | saxa | p3nguin: its listed on voip-info.org under the tips section of sip behind nat |
| 03:58.07 | saxa | so that is why i asked if its better to use externhost or fromdomain |
| 03:58.49 | saxa | ok, time to sleep over here. thx iprouteth0 , c u ppl. |
| 03:58.54 | iprouteth0 | fromdomain is for identifying when making a call to a non-peer, as some destinations require this identification |
| 03:59.20 | iprouteth0 | saxa: np, ttyl if I see ya around the channel |
| 04:00.16 | saxa | ok, night ;) |
| 04:00.32 | saxa | tomorrow I will try more stuff with this setup |
| 04:00.38 | saxa | thx again and bye |
| 04:07.18 | *** join/#asterisk CRCinAU_ (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 04:07.24 | CRCinAU_ | p3nguin: ping? |
| 04:08.14 | p3nguin | Yeah? |
| 04:08.36 | CRCinAU_ | your sccp phones. |
| 04:08.48 | p3nguin | Okay. |
| 04:08.48 | CRCinAU_ | do you have the conference buttons/functions enabled? |
| 04:09.05 | CRCinAU_ | I've defined a soft button key set in sccp.conf |
| 04:09.07 | p3nguin | I have MeetMe. |
| 04:09.21 | p3nguin | That creates a conference. |
| 04:09.24 | CRCinAU_ | and whenever I have confrn in there, it comes up with ??? |
| 04:09.32 | CRCinAU_ | ie ??? on the actual button |
| 04:10.32 | CRCinAU_ | so meetme lets you dial someone else too? |
| 04:10.37 | CRCinAU_ | then join them to the group? |
| 04:14.37 | p3nguin | I have confrn in both onhold and conntrans. |
| 04:14.49 | CRCinAU_ | does it display on the actual phones button? |
| 04:15.20 | CRCinAU_ | ie on my display, I see: [ hold ] [ EndCall ] [ Transfer ] [ ??? ] |
| 04:15.52 | p3nguin | ???? should be "more" |
| 04:16.06 | p3nguin | It takes you to a second page of buttons. |
| 04:16.33 | CRCinAU_ | nah - I trimmed it down to 4 buttons on the conntrans |
| 04:16.39 | p3nguin | When I use the conference soft key, it internally uses MeetMe. |
| 04:16.46 | CRCinAU_ | hmmmm |
| 04:17.46 | CRCinAU_ | I think I might have to research the button maps a little more |
| 04:20.21 | p3nguin | It's possible I'm mistaken. I'm heavily drugged right now and not able to think clearly about this. |
| 04:20.26 | CRCinAU_ | as for some reason, I can't transfer either. |
| 04:20.40 | p3nguin | I'm trying to recall exactly what's going on, looking at my configs and my phone. |
| 04:21.15 | CRCinAU_ | I'll do the same after this F15 install has finished :p |
| 04:21.36 | CRCinAU_ | netinstalls are *damn* slow - but at least I can get to a shell ;) |
| 04:22.40 | p3nguin | To transfer, you need to be on a call. Press the transfer key, and it will put the call on hold and give you a new dial tone. Dial a new number, and after you get an answer, press the transfer key again to transfer this new call to the original call which is on hold. |
| 04:23.00 | p3nguin | I know for a fact that one is working because I use it often. |
| 04:23.41 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 04:23.52 | *** join/#asterisk CRCinAU_ (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
| 04:23.55 | CRCinAU_ | sighs |
| 04:24.23 | p3nguin | As for the confrn and meetme, I need to look into that again. I don't use it much, so I'm not all that familiar with it. I do remember testing it and saw chan_sccp create a meetme conference. |
| 04:24.23 | CRCinAU_ | I tried calling my MOH extension, hit transfer, then dialled another phone, then hit transfer again and I got a "Unable to complete transfer" |
| 04:24.57 | p3nguin | Call the other phone and establish the connection between the two phones. |
| 04:25.12 | p3nguin | After you get that done, press the transfer key. |
| 04:25.16 | CRCinAU_ | then I run out of phones to transfer it to ;) |
| 04:25.32 | CRCinAU_ | wait - I can transfer it to a fax machine I guess :p |
| 04:25.43 | p3nguin | Just pay attention and follow along. |
| 04:25.46 | p3nguin | Call the other phone and establish the connection between the two phones. |
| 04:25.54 | p3nguin | After you get that done, press the transfer key. |
| 04:26.01 | p3nguin | Then you'll hear a new dial tone and your call will instantly be on hold. |
| 04:26.34 | CRCinAU_ | ahhh - that seemed to work. |
| 04:26.37 | p3nguin | Now dial your MoH extension. It should answer almost right away. Press the transfer key again, and your phone should have no remaining calls on it. |
| 04:26.52 | p3nguin | You've now transferred your call that was on hold to the moh. |
| 04:27.51 | CRCinAU_ | hmmm ok |
| 04:28.04 | CRCinAU_ | it just seems that if the moh is the first call made, I can't transfer it. |
| 04:32.14 | p3nguin | There's something about chan_sccp that knows shit. |
| 04:32.35 | p3nguin | It's like it knows it's only half of a call or something. |
| 04:33.06 | CRCinAU_ | maybe.... |
| 04:33.22 | LemensTS | Anyone ever see channels go bad on a digium card? I have an fxo module on channel 1 and 3, on channel 3 I have echo. I swapped the fxo modules, still the same on channel 3. |
| 04:33.35 | CRCinAU_ | I'm wondering why the confrn comes up as ??? though... that seems not really correct. |
| 04:33.45 | LemensTS | /etc/fxotune.conf is same for channels 1 & 2 |
| 04:33.47 | p3nguin | Okay, I'm really messed up... I don't know what I'm doing, but my confrn key says UNDEFINED and my MeetMe key is asking for a number. |
| 04:33.48 | LemensTS | I mean 1 & 3 |
| 04:33.57 | leifmadsen | it comes up as ??? literally? |
| 04:34.05 | p3nguin | It probably does. |
| 04:34.22 | p3nguin | Mine says UNDEFINED, so ??? would be equivalent. |
| 04:34.33 | p3nguin | What does Google say about chan_sccp and confrn? |
| 04:37.10 | CRCinAU_ | p3nguin: http://chan-sccp-b.sourceforge.net/doc/new_features.html#nf_sccp_softkeys |
| 04:37.15 | CRCinAU_ | kinda makes me wonder |
| 04:41.47 | p3nguin | That page basically says what I started to try to tell you... meetme key is used to conference two calls and yourself. |
| 04:42.11 | p3nguin | But I don't understand the reason there is a separate conference key, which does not work for either of us. |
| 04:42.52 | CRCinAU_ | me either. |
| 04:42.55 | p3nguin | I used to use the Confrn key when I ran SIP on my phone, but when I went to SCCP I have to use the MeetMe key. |
| 04:42.58 | CRCinAU_ | thats where I was getting confused... |
| 04:43.07 | CRCinAU_ | to me, Conference is a conference :P |
| 04:43.18 | CRCinAU_ | meetme is - well, not a conference lol |
| 04:43.21 | p3nguin | MeetMe is a conference, in Asterisk. |
| 04:43.26 | p3nguin | It's a conference in Asterisk. |
| 04:43.39 | p3nguin | That's the only thing MeetMe does, actually. |
| 04:43.52 | CRCinAU_ | hmmmm |
| 04:44.27 | p3nguin | Confrn when I used SIP was actually not a conference, but was 3-way calling. |
| 04:45.16 | p3nguin | I had a serious issue with the cnf_join_enable SIP setting. |
| 04:45.45 | p3nguin | By default, it bridges the two other calls together when you hang up your phone. |
| 04:45.49 | p3nguin | by default |
| 04:45.57 | p3nguin | That sucks. |
| 04:46.34 | CRCinAU_ | indeed. |
| 04:46.41 | p3nguin | Once I found out there was a fix, I just defined the setting with a value of 0. Then when I hang up, it disconnects the other two lines. |
| 04:47.31 | CRCinAU_ | See, I just want the default behaviour as described in that page for a conference :p |
| 04:48.02 | p3nguin | But that was back in the day. Now MeetMe is what gets used, and there are other options for making sure the other calls disconnect. |
| 04:48.30 | CRCinAU_ | so realisticly, if I replace Confrn with meetme, it should work the same? |
| 04:49.40 | p3nguin | Give it a try. You might have to enable it for the line(s) on the phone and you'll want to define good meetmeopts. |
| 04:51.28 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 04:52.18 | CRCinAU_ | Hmmmm |
| 05:15.33 | *** join/#asterisk Aut0Exec (~root@65.75.65.130) |
| 05:16.14 | CRCinAU_ | ok - so it seems MeetMe has been replaced with ConfBridge |
| 05:16.19 | CRCinAU_ | however I'll be buggered if I can get it gonig |
| 05:16.21 | Aut0Exec | question... if i'm using a dd-wrt or openwrt router .. do I still need a phone adapter? |
| 05:17.13 | Aut0Exec | i'm a nub here |
| 05:17.13 | p3nguin | aut0exec: That depends if you need both a router AND a phone adapter. |
| 05:17.34 | p3nguin | It's sort of like trying to decide if you need both a router and an alarm clock. |
| 05:17.49 | Aut0Exec | huh |
| 05:18.08 | Aut0Exec | i was thinking to plug a phone in the back of a cat5 jack |
| 05:18.33 | p3nguin | If you have an IP phone, you don't need a phone adapter. |
| 05:18.36 | Aut0Exec | i want to just install asterisk on my router |
| 05:18.42 | Aut0Exec | oh ok |
| 05:18.48 | p3nguin | By phone adapter, I assume you mean ATa. |
| 05:18.50 | p3nguin | ATA |
| 05:18.52 | Aut0Exec | yes |
| 05:19.00 | Aut0Exec | what does ATA stand for anyways? |
| 05:19.14 | brainiac | Analog Telephony Adapter |
| 05:19.16 | p3nguin | ATAs are for connecting analog phones to Ethernet. |
| 05:19.18 | Aut0Exec | k |
| 05:19.22 | p3nguin | ~ata |
| 05:19.22 | infobot | ata is probably Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 05:19.22 | Aut0Exec | yeah bingo |
| 05:19.49 | Aut0Exec | so if its already an ip phone then no need |
| 05:19.50 | Aut0Exec | i gatcha |
| 05:19.57 | p3nguin | Will you use an IP phone or a bell phone? |
| 05:20.11 | Aut0Exec | well looks like ip phone |
| 05:20.21 | p3nguin | Then you don't need an ATA for it. |
| 05:20.26 | Aut0Exec | ok |
| 05:20.31 | Aut0Exec | they are expensive huh? |
| 05:20.47 | p3nguin | If $30 is expensive, yes, they are expensive. |
| 05:20.51 | Aut0Exec | lol |
| 05:20.53 | Aut0Exec | amazon? |
| 05:20.57 | p3nguin | ebay |
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| 05:21.31 | Aut0Exec | k |
| 05:21.41 | Aut0Exec | cisco yes? |
| 05:22.54 | p3nguin | Yes, Linksys/Cisco |
| 05:23.05 | p3nguin | PAP2T |
| 05:23.15 | Aut0Exec | ok thanks bud. :) |
| 05:23.45 | p3nguin | If you need to connect an analog phone to it AND also connect your phone line from the wall jack to it, consider the SPA-3102. |
| 05:24.06 | Aut0Exec | ok |
| 05:24.13 | Aut0Exec | thanks |
| 05:24.30 | Aut0Exec | /quit |
| 05:33.31 | CRCinAU_ | p3nguin: I'm smacking my head against a wall here... |
| 05:33.37 | CRCinAU_ | I have the meetmenum = 700 |
| 05:33.47 | CRCinAU_ | if I dial 700 from all phones, they all come into the conference |
| 05:33.57 | CRCinAU_ | but it just doesn't work when using the MeetMe button on the phone |
| 05:35.05 | *** join/#asterisk AB3LS (Geoff@174.122.9.52) |
| 05:38.03 | p3nguin | I'm seeing recent "work on conference" on this channel driver. Maybe they know it isn't working. Have you researched it to see if they know it doesn't work? If they think it works, you need to let them know it doesn't. |
| 05:39.01 | CRCinAU_ | I've hunted around what I can find - but theres not a great deal of info... |
| 05:39.22 | CRCinAU_ | from looking at the output, I think it creates a new MeetMe room for every single call instead of the same room |
| 05:39.34 | CRCinAU_ | See, I have a call established |
| 05:39.45 | CRCinAU_ | then I hit "MeetMe", the call goes on hold and I get a dialtone |
| 05:40.15 | CRCinAU_ | I dial in another number, then it connects me to the meetme (ConfBridge) but leaves the first call on hold and doesn't dial the second number |
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| 05:44.25 | CRCinAU_ | thinks about pulling down the latest svn to test |
| 05:48.04 | p3nguin | I'm searching all the revs in the past two months to see what is going on. I can't use the latest because of the MWI light bug, but my last good rev is kind of old. |
| 05:49.37 | CRCinAU_ | I noticed that theres a configure option: --enable-conference... |
| 05:49.40 | CRCinAU_ | so I'm tinkering ;) |
| 05:53.06 | CRCinAU_ | hmm - now the Confrn button comes up |
| 05:53.10 | CRCinAU_ | but says "Key not active" |
| 05:54.14 | CRCinAU_ | yay |
| 05:54.22 | CRCinAU_ | now sccp reload crashes asterisk :p |
| 05:54.50 | p3nguin | I'm checking out an older revision to see what's going on in it. |
| 05:55.52 | CRCinAU_ | meetme still fails the same though |
| 05:56.40 | CRCinAU_ | anyhow - I'm going to get some munchies... bbl |
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| 07:02.03 | CRCinAU_ | sooo |
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| 07:03.15 | CRCinAU_ | p3nguin: any luck finding anything out? |
| 07:03.45 | jmordica | problem. When i initiate a backup from my second or failover server through ssh to the primary server, the primary server creates a backup locally and doesn't restore the second server |
| 07:03.54 | jmordica | asterisknow. freepbx 2.8 |
| 07:08.27 | CRCinAU_ | p3nguin: this is somethign: http://sourceforge.net/tracker/?func=detail&aid=3131059&group_id=186378&atid=917045 |
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| 07:14.34 | jkroon | jmordica, wrong # |
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| 07:15.06 | jkroon | your chances of being helped goes up exponentially if you try using the asterisknow support channels. |
| 07:15.31 | jmordica | yea sorry.. but those guys aren't helping with the issue because i feel like it's something asterisk is doing.. |
| 07:15.38 | *** join/#asterisk Pranav_rcmas (~prp_rcmas@117.193.203.21) |
| 07:15.55 | jkroon | asterisk has nothing to do with backups or ssh. the two keywords you mentioned. |
| 07:17.32 | CRCinAU_ | p3nguin: lodged this: http://sourceforge.net/tracker/?func=detail&aid=3207447&group_id=186378&atid=917045 |
| 07:20.04 | p3nguin | The first one had a comment about changing confrn to conf... I tried that hours ago and the soft key was completely gone. |
| 07:20.33 | p3nguin | The next comment says it's fixed in rc 3.1 ... but I used rc 3.1, and it's not fixed. |
| 07:21.15 | CRCinAU_ | yeah - I lodged a bug |
| 07:21.21 | CRCinAU_ | I think its the way things are called |
| 07:21.31 | CRCinAU_ | the first call never makes it to the meetme / confbridge room |
| 07:21.39 | CRCinAU_ | the second number is never dialed |
| 07:22.10 | CRCinAU_ | but the phone doing all the work does get put into the meetme / confbridge room - even though the second number is never dialed. |
| 07:22.20 | CRCinAU_ | p3nguin: did you say you have the meetme stuff working? |
| 07:22.38 | p3nguin | Are you using MeetMe or ConfBridge? I have 1.4, so I can't compile chan_sccp with conference support. |
| 07:22.43 | CRCinAU_ | ahhh |
| 07:22.52 | CRCinAU_ | I'm on 1.6 - which gives me confbridge |
| 07:23.11 | CRCinAU_ | although even using --enable-conference doesn't seem to work |
| 07:23.17 | p3nguin | I can only use MeetMe, so maybe that's why I lost the conf button completely. |
| 07:23.22 | CRCinAU_ | however, I haven't used conf instead of confrn |
| 07:23.37 | p3nguin | I did and the key disappeared completely. |
| 07:24.46 | CRCinAU_ | nah - I just get: Key not active |
| 07:25.16 | p3nguin | When I have conf, it's gone. When I use confrn, it's UNDEFINED. |
| 07:25.31 | CRCinAU_ | yeah - confrn gives me ??? |
| 07:25.33 | p3nguin | But I think it's because I can't use conference support at all. |
| 07:25.39 | CRCinAU_ | however conf brings up the button ok |
| 07:25.52 | CRCinAU_ | not that it works ;) |
| 07:25.57 | p3nguin | Okay, so you at least get half of the way there. |
| 07:26.25 | CRCinAU_ | yeah - I get a button that doesn't work ;) LOL |
| 07:26.35 | CRCinAU_ | just waiting for the phone to come back up..... |
| 07:26.38 | p3nguin | That's more than before. I've read a couple revisions mentioning conference work, so I don't know if conference works in the current revision or not. |
| 07:27.05 | CRCinAU_ | it doesn't work in 3.1rc |
| 07:27.25 | p3nguin | That's what I figured. I don't know why that comment indicates that it does. |
| 07:27.38 | CRCinAU_ | that being said, meetme is broken in 1.6 |
| 07:27.46 | p3nguin | I doubt that. |
| 07:28.04 | CRCinAU_ | well, how chan-sccp-b works with it is |
| 07:28.14 | p3nguin | chan_sccp is broken. |
| 07:28.22 | p3nguin | MeetMe works fine. |
| 07:28.56 | CRCinAU_ | does my bug make sense to you? |
| 07:29.02 | CRCinAU_ | https://sourceforge.net/tracker/?func=detail&aid=3207447&group_id=186378&atid=917045 <-- |
| 07:29.08 | p3nguin | Did you check chan_sccp-b v2 at all? I can't remember what features it had and didn't have. |
| 07:29.21 | CRCinAU_ | nah - haven't touched it |
| 07:29.35 | p3nguin | What you filed in the bug report makes sense, but we've also discussed the problem quite a bit. |
| 07:30.38 | CRCinAU_ | nods |
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| 07:31.22 | p3nguin | And... I've got to run again. Keep me updated on your bug report. I'm interested in the fix. |
| 07:31.55 | CRCinAU_ | I reakon its making different confbridge rooms for each call |
| 07:31.58 | CRCinAU_ | mayb |
| 07:32.27 | CRCinAU_ | in the CLI, where can I find the ConfBridge rooms etc? :\ |
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| 07:42.11 | *** join/#asterisk Docfxit (~ATUUser5@netblock-208-127-208-174.dslextreme.com) |
| 07:42.38 | Docfxit | Hi, |
| 07:43.00 | Docfxit | Is anybody awake? |
| 07:46.19 | carrar | shhh |
| 07:47.03 | Docfxit | Sorry. I didn't mean to wake you. |
| 07:47.10 | wdoekes2 | *yawns* |
| 07:47.29 | Docfxit | Could I get a quick question in? |
| 07:47.40 | wdoekes2 | ~ask |
| 07:47.40 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:49.18 | Docfxit | I need to change the subnet of my server from 192.168.1.xxx to 192.168.2.xxx. I have changed the phone in sip.conf. I have changed the IP and the gateway in the phone. What else do I need to change? |
| 07:49.48 | Docfxit | I have also changed the IP of the Ubuntu network. |
| 07:49.58 | carrar | change the IP of the server? |
| 07:50.08 | wdoekes2 | +10 points for carrar |
| 07:50.26 | carrar | change the IP's in your DHCP scope |
| 07:50.37 | Docfxit | Is that different than the network address of Ubuntu? |
| 07:50.51 | Docfxit | I'm using static IP's. |
| 07:51.09 | carrar | yuck |
| 07:51.21 | Docfxit | Where else would I change the ip of the server? |
| 07:51.30 | carrar | everywhere you set IP's |
| 07:51.59 | Docfxit | That's great. Where else are IP's set? |
| 07:52.04 | wdoekes2 | that allow list of your smtpd! |
| 07:52.09 | carrar | It's your server |
| 07:52.12 | wdoekes2 | s/that/the/ |
| 07:52.36 | Docfxit | What file would that be in? |
| 07:52.44 | carrar | heh |
| 07:53.55 | wdoekes2 | Docfxit: the short answer.. we don't know your setup, you should. and if you don't, just fix it when you notice that something doesn't work |
| 07:54.28 | benngard | i receive a call in context "b", would like to transfer (by hittting #) the call to an extension i context a, whats the smartest way to do that? |
| 07:54.40 | Docfxit | The phone isn't registering. |
| 07:54.41 | carrar | "and if you don't, maybe you should take the time to learn your system" :) |
| 07:55.00 | carrar | registering to what? |
| 07:55.07 | Docfxit | I can ping the phone and I can ping the server with the new subnet. |
| 07:55.09 | carrar | the old IP? |
| 07:55.20 | Docfxit | Registering to the server. |
| 07:55.21 | benngard | can i change the context to "a" before excecute dial |
| 07:55.30 | carrar | yes |
| 07:55.43 | benngard | how? |
| 07:56.10 | carrar | goto(a,${EXTEN},1) |
| 07:57.48 | benngard | yes, that will work, but i would like to stay in "b" context, got more stuff to do after the dial |
| 07:58.11 | carrar | they wny change to a? |
| 07:58.14 | carrar | then |
| 07:58.37 | carrar | stay in b |
| 07:58.38 | benngard | all my extensions are in "a" dont want to create them in "b" |
| 07:58.45 | carrar | include a into b |
| 07:58.46 | benngard | to |
| 07:59.02 | benngard | thats what i think about |
| 07:59.13 | carrar | stop thinkin, and start doin! |
| 07:59.38 | benngard | thinking of createing a "c" context where all internal numbers are, include "c" in "a" and "b" |
| 08:00.03 | carrar | Sounds HOT++ |
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| 08:02.33 | wdoekes2 | Docfxit: you need to find out if the issue is in the phone or the asterisk. sip set debug on in asterisk. if you see no traffic, then either the phone setup is bad, or the network setup is bad |
| 08:04.20 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 08:05.31 | Docfxit | wdoekes2, Thanks. I'll try that. |
| 08:07.00 | *** join/#asterisk manji (~manjiki@94-193-165-46.zone7.bethere.co.uk) |
| 08:11.17 | wdoekes2 | ($10 says your phone doesn't know how to reach the new asterisk-subnet) |
| 08:13.28 | wdoekes2 | gone |
| 08:27.15 | Docfxit | I changed the both the IP address and the gateway to 192.168.2.xxx |
| 08:27.52 | Docfxit | I'd be happy to give you the $10 to figure out the problem. |
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| 08:38.03 | Docfxit | When I try sudo asterisk -vvvvvvr I get an error saying Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) The file is there with 0 bytes. |
| 08:41.16 | koffu | hi, could somebody provide help to troubleshoot my asterisk problem? Asterisk 1.6.2.9-2, debian squeeze i686. Asterisk is randomly stop to accept new calls, take 100% CPU. log is show nothing, strace show a wait loop, lsof is show a lot of opened asterisk.ctl pipes. after a kill - asterisk is using just one or two. |
| 08:50.20 | *** join/#asterisk prometheanfire (~mthode@mx1.mthode.org) |
| 08:50.46 | prometheanfire | is it normal for asterisk to have over 10k sockets open for only 33 calls? |
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| 08:52.05 | *** join/#asterisk nighty^ (~nighty@x114242.dynamic.ppp.asahi-net.or.jp) |
| 08:54.23 | prometheanfire | seems to be rtp ports not being closed |
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| 08:59.58 | prometheanfire | https://issues.asterisk.org/view.php?id=17255 |
| 09:00.00 | *** part/#asterisk prometheanfire (~mthode@mx1.mthode.org) |
| 09:14.25 | benngard | hmm, strange, i am doing some tests here, adding a member to a queue, call the queue, se that the call is "landing" on the member, the phone is ringing, if the member aint asnwering within 15 seconds, the member will pauesd and dailplan execution continues, but: "queue show" claims that the member "has taken 1 calls", how come, the member didnt anser |
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| 10:36.20 | koffu | strace show me a loop |
| 10:36.20 | koffu | futex(0x81a0468, FUTEX_WAKE_PRIVATE, 1) = 0 |
| 10:36.20 | koffu | nanosleep({0, 1000}, NULL) = 0 |
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| 11:58.46 | *** join/#asterisk JD__ (~raws20@p5DF6372A.dip.t-dialin.net) |
| 11:59.41 | JD__ | Hello, try to reinstall my asterisk, because init script (/usr/sbin/asterisk) is broken brings up error: |
| 11:59.43 | JD__ | ./configure: line 7122: ACX_PTHREAD: command not found |
| 11:59.43 | JD__ | ./configure: line 7147: syntax error near unexpected token `ALSA,' |
| 11:59.43 | JD__ | ./configure: line 7147: `AST_EXT_LIB_SETUP(ALSA, Advanced Linux Sound Architecture, asound)' |
| 11:59.52 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 12:00.16 | JD__ | any suggestions? |
| 12:17.07 | JD__ | okay, sorry broke source files -> error |
| 12:17.11 | JD__ | *n |
| 12:17.16 | JD__ | thanks anyway |
| 12:22.55 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
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| 13:40.47 | jan_bangna | i'm trying the Amazon EC2 AMI from voxilla to play around with asterisk. now when i launch the instance i can only pick from a list of a few ports to open the firewall for. can i open more ports later on? |
| 13:46.20 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:49.58 | jan_bangna | oh found the custom firewall settings |
| 13:55.24 | jkroon | hi guys, when configuring dahdi PRI links, is there any way to test whether the remote endpoint is providing timing or not or doesn't this matter? |
| 13:56.17 | jkroon | on BRI links it's simple, dahdi_scan will report what the jumper is set to and this makes a physical wiring change, PRI doesn't seem to be that easy to deal with. |
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| 14:09.13 | phix | hi gang |
| 14:23.27 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 14:26.09 | LemensTS | after 'fxotune -i 4', i just run 'fxotune -s' to apply the changes right, I don't have to do 'service dahdi restart' or anything between them? |
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| 15:15.50 | dlynes | Is there a certain minimum gcc version for asterisk 1.6.2.17? I seem to be getting a bunch of errors being generated because various headers are using gnu'isms that probably don't exist in my version of the gcc compiler |
| 15:16.14 | dlynes | Fwiw, I'm using gcc 4.1.2 that comes with Centos 5 |
| 15:17.13 | dlynes | Also, fwiw, I'm compiling module code against the asterisk headers |
| 15:47.51 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-yflflwciwdrrhzax) |
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| 16:39.23 | volga629 | i have licence for 5 channels g792a but not all my extension have ability to use this codec. How I can enable to use different codec in different extensions |
| 16:40.15 | *** join/#asterisk sourcode (~code@ppp-58-11-75-225.revip2.asianet.co.th) |
| 16:40.50 | dlynes | volga629, allow=g729 for each of those peers, or allow=g729 in your general section, and don't override your allow/disallow settings in each of your peers |
| 16:40.57 | volga629 | I tried use deny all and allow ulaw&g729 for all extensions but it dropping call when not all extension has the same allowed codecs |
| 16:41.50 | volga629 | hmm |
| 16:41.54 | dlynes | volga629, of course |
| 16:42.12 | dlynes | volga629, because it can't transcode |
| 16:42.23 | dlynes | volga629, which probably means your g729 codec is not installed properly |
| 16:43.01 | dlynes | volga629, do a 'g729 show licenses' from the command line |
| 16:43.07 | dlynes | volga629, what does it spit back at you? |
| 16:45.34 | dlynes | Is there a certain minimum gcc version for asterisk 1.6.2.17? I seem to be getting a bunch of errors being generated because various headers are using gnu'isms that probably don't exist in my version of the gcc compiler |
| 16:45.59 | volga629 | I checked it show licence ok |
| 16:46.16 | volga629 | I can paste in pastebin |
| 16:46.22 | dlynes | volga629, can you paste the single line on here? |
| 16:46.42 | dlynes | volga629, no need to pastebin; it's only one line |
| 16:47.01 | dlynes | volga629, pastebin is if you have more than 2 lines |
| 16:47.14 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-136-43.dsl.stlsmo.sbcglobal.net) |
| 16:48.20 | dlynes | volga629, also how many simultaneous calls are you trying to transcode at once? |
| 16:50.11 | volga629 | right now it 2 extensions with g729a |
| 16:50.22 | dlynes | volga629, where? |
| 16:51.05 | volga629 | 3 calls |
| 16:51.35 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-fbytqnfdqrmpluyg) |
| 16:51.50 | dlynes | volga629, if you're trying to do three calls that are from one sip endpoint to another sip endpoint with differing codecs on both ends, it will not work, because you only have licenses for 5 channels |
| 16:51.56 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 16:52.06 | dlynes | volga629, consider a channel to be a call leg for purposes of licensing |
| 16:52.22 | volga629 | this remote users which using tunnel, and when I usng just ulaw or alaw it working fine |
| 16:52.33 | volga629 | so we need more licence |
| 16:52.50 | dlynes | volga629, if you have a call coming on g729 and you want to pass it off onto a phone using ulaw, that's two licenses |
| 16:53.18 | WIMPy | That doesn't seem to make sense. |
| 16:53.21 | dlynes | volga629, if you have a call coming in on g729 and you want to dump it into ulaw voicemail, that's one license |
| 16:53.40 | WIMPy | Isn't one licence for one encoder and one decoder? |
| 16:53.46 | volga629 | hmm |
| 16:54.03 | dlynes | WIMPy, one license is for one encoder or one decoder |
| 16:54.04 | volga629 | that interesting math |
| 16:54.18 | dlynes | WIMPy, each encoder and decoder takes up a licnese |
| 16:54.44 | dlynes | WIMPy, it seems to trip up every new user of the digium codecs |
| 16:55.26 | WIMPy | Hmm. bad. |
| 16:55.35 | dlynes | volga629, while your calls are happening, you can type 'g729 show licenses' to see how many licenses are actually being used for the calls that are happening |
| 16:55.57 | dlynes | WIMPy, i'm guessing you've never used the digium g729 codecs? |
| 16:56.10 | WIMPy | Correct |
| 16:56.49 | volga629 | so how I need allow in correct way use g729 for remote users ? |
| 16:57.34 | dlynes | volga629, disallow=all, allow=g729 |
| 16:57.47 | dlynes | volga629, and the system will transcode to g729 if you have enough channels |
| 16:58.05 | volga629 | let me try |
| 16:58.08 | dlynes | volga629, if you're using a newer version of asterisk, you can also do disallow=all ; allow=g729 ; allow=ulaw |
| 16:58.20 | dlynes | volga629, and that will allow it to fail over to ulaw if it can't negotiate g729 |
| 16:59.06 | dlynes | So, typically you'd purchase twice the number of channels as you expect to have simultaneous calls that need transcoding |
| 17:00.02 | volga629 | that for each extension ? |
| 17:00.15 | dlynes | so, if you have an 8 line analogue card, and you want to serve up SIP lines to your customers that are offsite, you'd typically buy 16 licenses |
| 17:00.42 | dlynes | so that you can transcode to g729 to send it across ADSL or whatever it's going across efficiently |
| 17:01.41 | dlynes | volga629, it all depends on how you have your extensions set up |
| 17:02.16 | dlynes | volga629, if your extensions can handle ulaw (and they're local), i would leave them on ulaw, so that you don't need to do transcoding |
| 17:02.50 | dlynes | volga629, it's all a question of how much bandwidth your users have, how much bandwidth you have, whether the sip extension supports ulaw, gsm, g729, ... and so on and so forth |
| 17:03.15 | dlynes | Generally if the sip device supports gsm, i'll transcode to gsm instead of g729, so that i'm not using up precious licenses |
| 17:03.43 | dlynes | gsm's almost as good as g729, and it's free |
| 17:04.18 | volga629 | for local no problem of on bandwidth and for remote we should use g729 because we using it through vpn tunnel |
| 17:04.37 | dlynes | volga629, so there you go, then...you already know your answer |
| 17:05.53 | volga629 | yes, I will allow for remote g729 and for local ulaw |
| 17:07.00 | dlynes | volga629, just remember if you have two call legs, and you need to transcode, you require two licenses |
| 17:07.06 | *** join/#asterisk Shnootz (~Hanan@62.145.76.186) |
| 17:07.21 | dlynes | volga629, and if you one call leg and the other call leg is voicemail or something similar, you only need one license |
| 17:07.40 | Shnootz | hi i need asistance with xorcom asteribank |
| 17:07.59 | Shnootz | my asteribank is not working properly |
| 17:07.59 | volga629 | let me try test |
| 17:08.16 | Shnootz | it is stuck at e4e4:1161 on the firmware |
| 17:08.34 | dlynes | Shnootz, probably the only people that can help you is Xorcom |
| 17:08.43 | dlynes | Shnootz, have you tried emailing them or calling them? |
| 17:09.02 | Shnootz | not yet |
| 17:09.19 | Shnootz | i thought i will get quicker reply here |
| 17:09.23 | dlynes | Shnootz, yeah...if it's firmware related on the asteribank, they're probably going to be the only folks that can help you |
| 17:09.39 | jMyles | Hello everyone. I'm new to asterisk and enjoying it. Thanks so much for your efforts. |
| 17:09.54 | Shnootz | ok thanks dlynes |
| 17:09.55 | dlynes | Shnootz, you're actually the first person I've seen in ten years or so even mention asteribank on this channel |
| 17:10.08 | Shnootz | :) |
| 17:10.17 | dlynes | Shnootz, but fwiw, how is the product? |
| 17:10.27 | Shnootz | very good |
| 17:10.33 | dlynes | Shnootz, we were considering it at one point, and might still consider it in the future |
| 17:10.41 | dlynes | Shnootz, call quality's still good? |
| 17:10.46 | Shnootz | <PROTECTED> |
| 17:10.52 | Shnootz | yeah |
| 17:11.04 | dlynes | Shnootz, cool...you use it for door entry systems as well? |
| 17:11.14 | Shnootz | no |
| 17:11.24 | Shnootz | just as a hybrid pbx |
| 17:11.28 | dlynes | Shnootz, so you don't use the external control, then? |
| 17:11.56 | dlynes | Shnootz, i.e. the relay ports? |
| 17:12.08 | Shnootz | yes i do |
| 17:12.15 | Shnootz | but did not use them |
| 17:12.22 | dlynes | ah |
| 17:12.33 | dlynes | Shnootz, that's the main reason we were looking at the asteribanks |
| 17:12.46 | dlynes | Shnootz, so that we could have one box that did both phone system and door entry system |
| 17:12.55 | Shnootz | i am guessing in the future we will integrate them |
| 17:13.16 | dlynes | Shnootz, like for integrating into a Mircom or something similar |
| 17:13.18 | Shnootz | until now my customers didnt have that option in the analog system |
| 17:13.39 | dlynes | yeah...we don't like leaving the thinking up to our customers |
| 17:13.51 | dlynes | if we did, we'd be missing out on a lot of potential sales |
| 17:16.06 | volga629 | dlynes: you the man thank you again |
| 17:22.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:27.52 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 17:35.58 | capitan | what web server does asterisk gui use? |
| 17:37.15 | dlynes | capitan, apache/freepbx |
| 17:39.16 | capitan | dlynes, really? i thought i read somewhere that it was separate |
| 17:40.16 | dlynes | capitan, the original version was different from freepbx; newer versions use freepbx |
| 17:40.35 | dlynes | capitan, well...actually...let me clarify |
| 17:40.58 | dlynes | capitan, asterisk live uses freepbx and apache; asterisk gui (afaik) is a dead product |
| 17:41.36 | dlynes | capitan, that being said, according to the topic in #asterisk-gui, they're looking for a new maintainer |
| 17:49.46 | capitan | wow... you're right. thanks for the info :) |
| 17:56.00 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:56.50 | p3nguin | AsteriskNOW offers a choice of Asterisk GUI (in addition to a choice of FreePBX) at install time. |
| 17:57.01 | p3nguin | I've never heard of "asterisk live." |
| 17:59.31 | p3nguin | And if the Asterisk GUI doesn't have its own built-in httpd, then it can use any httpd you feel like configuring. |
| 18:00.52 | *** join/#asterisk eugeneoden (~goden@63.133.138.10) |
| 18:05.00 | atan2 | Perhaps a long shot here, but when a sip adapter registers I see "Registered SIP '44' at 192.168.1.101:5060" |
| 18:05.09 | atan2 | Why does Asterisk show me that users local IP? |
| 18:05.14 | atan2 | The server is not on the same network. |
| 18:07.15 | *** join/#asterisk zkn (~zkn@82.131.70.89.cable.starman.ee) |
| 18:13.05 | atan2 | Missed the nat=yes bit. There we go. |
| 18:13.07 | atan2 | :) |
| 18:13.39 | zkn | Hello, does anyone know of a way how to use regexten generated extension in the dialplan to dial that respective user that this regexten extension belongs to? |
| 18:14.30 | dlynes | p3nguin, sorry....asterisk now, not asterisk live :O |
| 18:17.05 | *** join/#asterisk eject_ck (~eject_ck@83-218-247-119.dynamic.vega-ua.net) |
| 18:19.01 | eject_ck | hi all |
| 18:19.13 | eject_ck | I'm trying to get g729 working under openbsd |
| 18:19.16 | eject_ck | 4.8 |
| 18:19.23 | eject_ck | [Mar 12 20:15:33] WARNING[28363]: loader.c:387 load_dynamic_module: Error loading module 'codec_g729.so': File not an ELF object |
| 18:19.23 | eject_ck | [Mar 12 20:15:33] WARNING[28363]: loader.c:795 load_resource: Module 'codec_g729.so' could not be loaded. |
| 18:20.03 | eject_ck | I'm under amd64 |
| 18:23.55 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-qvorzyicbnsbbqhi) |
| 18:25.56 | p3nguin | Huge difference between live and NOW. AsteriskNOW is a complete OS that you install. "Live" indicates that it would run from the removable media. |
| 18:27.41 | Diffen | Evning, if i use the r option (ring signal) in a queue setting, am I not allowed to play a periodic announcement to the caller if the caller are queued? |
| 18:33.48 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 18:41.33 | *** join/#asterisk tris (~tristan@173-164-188-122-SFBA.hfc.comcastbusiness.net) |
| 18:52.27 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 18:54.04 | dlynes | Is there a certain minimum gcc version for asterisk 1.6.2.17? I seem to be getting a bunch of errors being generated because various headers are using gnu'isms that probably don't exist in my version of the gcc compiler |
| 18:55.15 | dlynes | eject_ck, did you grab the openbsd version of the codec module? |
| 18:55.39 | dlynes | eject_ck, erm...actually...is there even a version for openbsd? |
| 18:55.51 | dlynes | eject_ck, much less 64-bit openbsd? |
| 19:09.37 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 19:12.40 | p3nguin | The codec module from Digium is codec_g729a.so, which indicates to me that he's using a stolen module and using it illegally. I wouldn't offer him help, because you will likely tarnish your own name. |
| 19:13.15 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 19:13.44 | Juggie | p3nguin, there are some countries where pattents do not apply. |
| 19:14.05 | Juggie | that does make the alternative version legal, it is not stolen it is a free implementation that due to patent laws may be illegal in your country. |
| 19:14.22 | leifmadsen | regardless, it can't be supported here :) |
| 19:14.31 | Juggie | agreed :) |
| 19:14.37 | zkn | i thought there existed also an open source version of g729 |
| 19:14.41 | Juggie | but its not inherantly 'illegal |
| 19:14.43 | Juggie | ' |
| 19:15.07 | Juggie | obviously digium is based in a country where patents apply and as such that is why the licensed version is offered. |
| 19:15.21 | Juggie | most (all) of the money goes to pay off the patent pool. |
| 19:17.36 | leifmadsen | zkn: the patents on g729 make it so regardless of code accessibility or not, it is still patent encumbered |
| 19:17.47 | leifmadsen | so there is no such thing as an "open source" version of g729 |
| 19:18.17 | zkn | okay |
| 19:19.13 | eject_ck | dlynes 64-bit yes |
| 19:19.33 | Juggie | leifmadsen, all your snow gone? |
| 19:19.55 | eject_ck | p3nguin: I'm not using it illegally |
| 19:20.05 | p3nguin | You didn't pay for it. |
| 19:20.15 | eject_ck | do I need ? |
| 19:20.25 | eject_ck | it's not digium codec |
| 19:20.37 | eject_ck | it's opensource |
| 19:20.47 | p3nguin | (1316.42) <@leifmadsen> so there is no such thing as an "open source" version of g729 |
| 19:20.48 | Juggie | eject_ck, technically yes but it depends on your country of residence. |
| 19:20.59 | eject_ck | UA |
| 19:21.01 | eject_ck | :) |
| 19:21.36 | Juggie | none the less its not the perogitive of #asterisk to play police, it can be used if desired just as mp3's can be downloaded and movies can be bittorrented, but it cannot be supported here. |
| 19:22.20 | WIMPy | But then alternative hardware shouldn't be supported here, either. |
| 19:22.36 | Juggie | hardware has a license |
| 19:22.41 | eject_ck | p3nguin: do you want me to pay sipro lab ? |
| 19:22.42 | p3nguin | What is alternative hardware? |
| 19:22.51 | Juggie | its not that its 'not digiums' its that its unlicensed peroid. |
| 19:23.04 | WIMPy | There is a lot of compied hardware out there. |
| 19:23.41 | p3nguin | I don't know what that is, either. |
| 19:23.44 | Juggie | we are talking specifically about g729, i dont know if there are g729 cards copied as its a pretty niche item but yes there are plenty of t1/e1 copies. |
| 19:23.49 | WIMPy | And as we all know, there are places, where you don't need one. |
| 19:23.54 | Juggie | but there are also legitimate other resllers as well. |
| 19:23.57 | Juggie | c'est la via |
| 19:24.00 | Juggie | -a |
| 19:24.03 | Juggie | +e |
| 19:24.08 | WIMPy | Indeed |
| 19:24.22 | eject_ck | dlynes: openbsd 4.8 |
| 19:24.26 | eject_ck | dlynes: openbsd 4.8 amd64 |
| 19:25.58 | WIMPy | I mean there is a patent for conditional code execution as well. Do you pay a licence every time, you write an if()? |
| 19:27.50 | eject_ck | btw, lol http://www.sipro.com/g729_faq.php |
| 19:27.55 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 19:28.22 | eject_ck | A licensee sells 5,000 channels at $ 1.15 for a total of $5,750.00 during a year. |
| 19:28.45 | *** join/#asterisk fabrianchi (~fabrianch@unaffiliated/fabrianchi) |
| 19:29.07 | eject_ck | how much should I pay for 1 chanell asterisk license ? |
| 19:29.23 | eject_ck | how much should I pay for 1 channel asterisk license ? |
| 19:30.31 | zkn | off the top of my head I remember it being 10USD for the Digium version.. correct me if I'm wrong |
| 19:30.56 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 19:31.02 | *** join/#asterisk puzzled (~patrick@535335AA.cm-6-4a.dynamic.ziggo.nl) |
| 19:32.28 | eject_ck | does digium provide binaries for openbsd ? |
| 19:34.17 | eject_ck | btw, who pays roayalties for g729 codec which is in my nokia e52 codec ? |
| 19:34.28 | eject_ck | Nokia ? |
| 19:35.17 | p3nguin | yes |
| 19:35.34 | eject_ck | zkn: some d-link phones with g729a codec support costs ~ 40$ (retail price). |
| 19:36.14 | eject_ck | I don't think that royalty fees are > 1$ per device |
| 19:36.45 | eject_ck | why digium codec is so expensive (compare to) |
| 19:37.35 | WIMPy | Administrative costs and code supplied? |
| 19:37.47 | eject_ck | 10 times ? |
| 19:38.30 | WIMPy | The licence only gives you the right to write a compatibler codec. You don;t get anything usefull for it. |
| 19:38.44 | eject_ck | ah, okay ;) |
| 19:39.07 | eject_ck | it gives right to use described algo |
| 19:40.06 | puzzled | eject_ck: iirc the cost depends on the license type you get from mpeg-la. if you are unsure how much you are going to sell you go for the entry-level license type which unfortunately makes it more expensive |
| 19:40.11 | eject_ck | so, if we have opensource g729 codec code we can use it but should pay royalties, which seems like difficult task if I wanna buy just one channel |
| 19:40.28 | puzzled | the upfront cost is $50,000 |
| 19:40.56 | WIMPy | "administrative costs" |
| 19:41.33 | puzzled | WIMPy: that golf clinic on Hawai for the mpeg-la board needs to be paid by someone :) |
| 19:41.52 | eject_ck | :) |
| 19:42.04 | eject_ck | thanks for explanations :) |
| 19:42.47 | eject_ck | can I ask you once more, is it possible to get opensource g729 codec for asterisk on openbsd 4.8 amd64 ? |
| 19:43.06 | eject_ck | as I understuud I need get IPP from intel to compile it |
| 19:43.20 | puzzled | I think so but you can look on asterisk.org downloads |
| 19:44.54 | zkn | does anyone know of a way how to use regexten generated extension in the dialplan to dial the respective user that this regexten extension belongs to? or am I expecting too much ... |
| 19:51.19 | *** join/#asterisk Olivier_54 (~Olivier_5@92.90.17.64) |
| 19:54.35 | p3nguin | (1316.42) <@leifmadsen> so there is no such thing as an "open source" version of g729 |
| 19:54.44 | p3nguin | eject_ck: ^^^ still |
| 19:54.47 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 19:55.16 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-jszmcpbvyooyqmip) |
| 20:11.16 | eject_ck | ok |
| 20:11.31 | eject_ck | how long should I wait till patent will expire ? |
| 20:12.10 | *** join/#asterisk dr__ (~duckz@95.76.24.119) |
| 20:17.19 | Juggie | about 50 years :P |
| 20:21.29 | p3nguin | My license expires in 20 years. Does that mean I have to buy another if I want to keep using the codec for the remaining 30 years? |
| 20:21.52 | Corydon76-home | eject_ck: 2014 |
| 20:22.28 | p3nguin | Oh, so in three more years, I can legally use the codec without a license. Interesting. |
| 20:23.07 | Corydon76-home | Well, wait until 2015. The patent expires sometime in 2014, don't remember when |
| 20:23.19 | Corydon76-home | Or rather, the last remaining patent |
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| 20:39.31 | fabrianchi | hi, wanted to know if I POSE help set up a Asterisk version 1.6.2.9-2 asterisk in debian, to make calls from a softphone on a mobile phone in Argentina, I have is my laptop that I fart pass my card details sound if you like, someone help me? |
| 20:41.42 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-yjrbsrwmfjfpnneb) |
| 20:44.18 | p3nguin | Uh, what? |
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| 20:46.04 | fabrianchi | p3nguin: for me ? |
| 20:50.27 | p3nguin | fabrianchi: Yes. Your English is not so good. |
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| 21:21.37 | fabrianchi | p3nguin: sorry , pero me podes ayudar ? |
| 21:21.50 | fabrianchi | sorry, but you can help me? |
| 21:29.33 | carrar | You fart? |
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| 21:31.28 | p3nguin | That's what he said... he farted a laptop. |
| 21:31.38 | carrar | Thats pretty awesome |
| 21:31.43 | carrar | and painfull at the same time |
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| 22:09.03 | CRCinAU_ | yawns |
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| 22:36.24 | CRCinAU_ | p3nguin: my chan-sccp-b bug has... disappeared :\ |
| 22:37.43 | CRCinAU_ | ah - its been moved. :) |
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| 22:40.46 | Aut0Exec | hi guys.... i'm a nub of course... I was wondering... whats a good router to setup asterisk on? and would I need dd-wrt or openwrt? |
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| 22:45.02 | Aut0Exec | any thoughts..? I know this is not the norm per say but I would like to give a stab at it |
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| 22:48.36 | CRCinAU_ | a good router is one that works... |
| 22:48.49 | CRCinAU_ | not sure if dd-wrt has the packages for asterisk? |
| 22:49.02 | Aut0Exec | ok |
| 22:49.06 | Aut0Exec | openwrt right? |
| 22:49.29 | CRCinAU_ | you can certainly give it a go |
| 22:49.35 | SunTsu | you normally can use openwrt's opkg files for dd-wrt |
| 22:50.38 | Aut0Exec | k |
| 22:50.48 | SunTsu | I'd try both dd-wrt and openwrt and use the one that's most stable on my router |
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| 22:51.07 | Aut0Exec | only thing is that dd-wrt has the /etc stored in a non writable area |
| 22:51.15 | Aut0Exec | but openwrt is more flexable with it |
| 22:51.19 | Aut0Exec | stored in ram basically |
| 22:51.21 | Aut0Exec | sux |
| 22:51.39 | Aut0Exec | so prolly gonna go with openwrt even tho its more cli to setup |
| 22:51.47 | SunTsu | Aut0Exec: you're talking about uci? |
| 22:51.56 | Aut0Exec | i just dont want a pc on 24/7 just for asterisk |
| 22:52.00 | Aut0Exec | too much energy |
| 22:52.03 | SunTsu | Aut0Exec: just use Luci |
| 22:52.08 | Aut0Exec | luci? |
| 22:52.11 | Aut0Exec | whassat? |
| 22:52.28 | SunTsu | openwrt's web frontend to uci, that thing totally rocks |
| 22:52.40 | Aut0Exec | oh really? |
| 22:52.42 | Aut0Exec | ok |
| 22:52.44 | Aut0Exec | i'll check it out |
| 22:52.50 | zkn | "Found duplicate exten. Had mariliis found mari-liis", how come Asterisk sees these two extensions as duplicate? |
| 22:52.52 | Aut0Exec | u have asterisk on ur router>? |
| 22:53.08 | SunTsu | I had it some time ago |
| 22:53.39 | Aut0Exec | ok |
| 22:53.42 | Aut0Exec | worked good? |
| 22:54.46 | SunTsu | yeah, but I basically used it as sip proxy, had five internal sip channels and five external ones. Just mapped them against each other |
| 22:54.57 | Aut0Exec | oh ok |
| 22:58.29 | zkn | anyone knows how come Asterisk sees extensions "mariliis" and "mari-liis" as duplicates? |
| 22:59.28 | zkn | upgraded to 1.8.3 |
| 22:59.44 | zkn | did not have this issue before |
| 22:59.50 | wdoekes2 | it's a feature |
| 23:00.25 | zkn | feature? so i can disable it ? |
| 23:00.33 | wdoekes2 | e.g. to "fix" phonenumbers containing dashes, dots, spaces |
| 23:00.46 | wdoekes2 | I think you can |
| 23:01.56 | zkn | i'm familiar with the feature of cleaning up dashes, etc, in caller id's, not extension names, tho.. |
| 23:02.32 | zkn | but i'll have a closer look, maybe i did miss a parameter somewhere |
| 23:03.28 | jkroon | Juggie, @ patent issues, afaik the Digium patent doesn't hold ground in ZA either, however, the quality of the Digium implementation vs the "open source" one is not comparable. I'm willing to pay my $10/channel (and believe me I've forked out my fair share of those) for not having issues. |
| 23:04.09 | jkroon | also, the risk of a judge deciding that the patent does in fact carry weight here is not worth the risk for me. not that it has been tested in court yet afaik. |
| 23:06.13 | wdoekes2 | zkn: is it a _pattern? |
| 23:06.21 | jkroon | zkn, i've found it usually works to put the chars znx inside [] - might work for - too. |
| 23:07.19 | Corydon76-home | zkn: no, you cannot disable that Asterisk ignores dashes in dialplan patterns |
| 23:07.42 | zkn | both "mari-liis" and "mariliis" are named extensions, have not included undercore for any of them for patternmatching |
| 23:08.12 | zkn | just one has the dash in the middle which for some reason is ignored ? |
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| 23:08.34 | zkn | i will try with [] |
| 23:08.50 | Corydon76-home | Yes, character classes are the only way to workaround it |
| 23:08.54 | Aut0Exec | if i'm connecting a sip phone to an asterisk server behind a router... which ports do I need to have open? |
| 23:09.03 | Aut0Exec | or forwarded |
| 23:09.40 | Corydon76-home | Aut0Exec: UDP 5060 and a UDP range that is defined in rtp.conf |
| 23:10.05 | Aut0Exec | ok thanks |
| 23:10.28 | Corydon76-home | There's some additional setup in Asterisk to get the SIP headers correct, but that's it for the firewall itself |
| 23:11.03 | Corydon76-home | It works a lot better in the reverse... i.e. Asterisk has a public IP and the client is behind a NAT |
| 23:16.14 | Corydon76-home | Aut0Exec: btw, as to the number of RTP ports needed, you generally need 2 ports for each concurrent audio call, plus RTP ports cannot be reused for at least 64 * the T1 timer for each peer (T1 timer is generally 500ms, making for 32 seconds) |
| 23:18.13 | zkn | putting square brackets around the dash took care of the duplicate exten warning..hopefully the extension actually works too without the need to specify these brackets when dialing it |
| 23:22.03 | zkn | I suppose no-one has any thoughts on how one could dial the actual user behind the extension created by regexten parameter ? |
| 23:24.23 | jkroon | well, zkn - the docs says it adds a prio1 NoOP() so just have something at prio 2 for each exten, eg exten => foo,2,Dial(SIP/foo) |
| 23:24.31 | jkroon | overly simplistic probably :) |
| 23:27.54 | jkroon | i have alternative mechanisms I use in a distributed way to basically use DUNDI to figure out to which server a trunk is registered and to then request that server to dial a trunk by name, using * chars as matching, so I'd have an extension with _*trunk*.1,Dial(SIP/${CUT(${EXTEN,*,2})}) type of thing. |
| 23:28.09 | jkroon | works quite well for the most part. |
| 23:28.29 | zkn | I was actually looking for more of a dynamic solution... somthing like, for each user in sip.conf I would create a numeric extension (eg. cellphone number with patternmatching) using "regexten" and then have the prio 2 line which will dial the user behind the regexten generated extensions |
| 23:32.14 | zkn | what I'm trying to achieve is that when a particular user is logged in to his/her SIP account, then regexten will add his/her phone number to the "sipregistrations" list, and when his/her number is dialed, Asterisk will first match the call and route to the SIP user before trying to dial out to trunk |
| 23:34.35 | zkn | obviously I could just write a long dialplan context full of cellphone number extensions like that, but this would be too much work in my opinion and not easy to maintain |
| 23:35.30 | zkn | the prio 1 seems to have a way to know exactly what username to display with Noop |
| 23:35.58 | zkn | I wonder how I could I *extract* this inforamtion |
| 23:39.12 | zkn | so that I could write smth like exten => _X.,2,Dial(${SIPUSEROFREGXTEN}) |
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| 23:55.14 | raden | <PROTECTED> |
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