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| 00:20.09 | haps | good evening; i have outgoing calls working fine but incoming isn't working. when i call i get 'not allowed', no rings. there is no noise in the debug info in the cli when i make a call which leads me to believe that it's a nat problem. my router is forwarding port 5060 UDP and 10K-20K UDP to the asterisk box. the connection is via SIP. I assume I'm missing something obvious, any ideas? |
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| 00:49.24 | fatnasty1 | I have a phone registered to my asterisk server, when Im on a call, if i start talking its like its half duplex of something its like it mutes the audio from the other end while Im talking. The phone has VAD turned off. any ideas? |
| 00:54.41 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 00:57.32 | fatnasty1 | so I had VAD enabled in codecs.conf |
| 00:57.36 | fatnasty1 | fixed it |
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| 01:09.54 | Talkradio | anyone have any idea of how many remote users to a pbx on dsl can support? 2 or 5 or ? |
| 01:10.19 | Talkradio | pbx has pri and network has dsl with 768k up |
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| 01:18.31 | mmlj4 | are the sangoma wanpipe drivers needed, or can dahdi handle the card by itself? |
| 01:19.31 | mmlj4 | Talkradio: you can look at the VoIP protocols and decide... there are charts with bandwidth available |
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| 01:22.11 | joako | Can I run asterisk on more than one port on the same IP address? |
| 01:23.29 | mmlj4 | you mean a particular protocol? |
| 01:23.49 | mmlj4 | asterisk runs on zero ports |
| 01:24.08 | joako | Yes, I want to run SIP on two ports |
| 01:24.14 | joako | I tried two IP but it does not work right |
| 01:24.38 | mmlj4 | you can, but A). why? and B). the clients need to be told to look for an alternate port |
| 01:25.02 | *** join/#asterisk eXcAliBuR (n=bob@207.134.8.33) |
| 01:25.04 | joako | Yes I know that. I am having problems with a particular device behind nat staying registered. |
| 01:25.49 | haps | udp session timeout should be >= smallest sipregistration tomeout |
| 01:25.50 | haps | timeout |
| 01:26.16 | joako | udp session timeout where? |
| 01:26.29 | mmlj4 | I'm getting little segfaults on debian lenny... anyone else have trouble? |
| 01:26.31 | haps | in your nat router |
| 01:26.55 | eXcAliBuR | what would be needed to have say 25 sites connected to a asterisk system, the sites are over a vast area. say 200 miles . what i want is to be able to pick up the phone and if calling a number that is local to any site, it be sent there and go out on it's phone line. |
| 01:27.00 | haps | so, what's your sip reg. timeout? |
| 01:27.04 | haps | default is 300 |
| 01:27.13 | haps | you could try lowering it, that might help |
| 01:27.18 | joako | haps: that's not the issue because I have other devices behind the same NAT with no issues. Just this one particular device has a problem (It's a Nokia E71) |
| 01:27.34 | haps | ah |
| 01:27.41 | haps | then i don't know |
| 01:28.24 | joako | haps: maxexpiry=3600, minexpiry=55, defaultexpiry=360 |
| 01:28.39 | joako | Should I try to lower defaultexpiry? |
| 01:29.36 | joako | The interesting thing is since this is behind NAT and there are multiple devices, in sip show peers the port shown is not 5060. And then e.g. if I do a sip debug on this device after a while the sip debug will start showing stuff from a Polycom phone |
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| 01:31.22 | joako | Oh and I did take the polycom offline and that did not help |
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| 01:31.44 | haps | i don't really understand your setup |
| 01:32.02 | haps | but you're port forwarding 5060 on your nat router right? |
| 01:32.10 | haps | it should show 5060 |
| 01:32.28 | haps | what router/nat are you using? |
| 01:32.31 | joako | haps: Asterisk on public internet (colocated server) and then at home I have a wired router (linksys with DD-WRT) and connected to that router I have a handfull of SIP devices |
| 01:33.08 | haps | oh |
| 01:33.09 | joako | I don't forward any ports and it always worked fine. I got the nokia E71 that is a mobile phone with WiFi and SIP support and that device is having problems staying registered, so I will not get inbound calls most of the time |
| 01:33.38 | joako | I'm trying to avoid setting up a local asterisk because I don't have a machine that runs 24/7 |
| 01:34.51 | haps | hmmm. |
| 01:34.56 | haps | i don't know the answer dude |
| 01:35.04 | haps | but yeah different ports sounds like it might work |
| 01:35.30 | haps | in sip.conf you didn't just use [nokia];port=5060 ? |
| 01:35.43 | haps | or port=5061 |
| 01:35.45 | haps | ? |
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| 01:44.53 | Steve_J-obs | someboy please help me with a basic dial question |
| 01:45.33 | Steve_J-obs | so basic, I dont know what's wrong |
| 01:47.08 | mmlj4 | shoot |
| 01:47.20 | haps | *bam* |
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| 01:48.00 | Steve_J-obs | ok... I am trying to dial through a proxy for which I have pre-registered |
| 01:48.58 | Steve_J-obs | with SIP...now... I can dial without registration... but where I get confused is because this seems to be instructions for an ATA and not for asterisk |
| 01:49.30 | Steve_J-obs | they gave the the instructions to register, but not the instructions to dial |
| 01:50.32 | Steve_J-obs | but I dont get it...the on the sip header invite I have: userid@domain |
| 01:51.01 | Steve_J-obs | userid@mydomain |
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| 02:46.37 | thehar | oh do you jbot |
| 02:52.08 | carrar | In Soviet Russia, jbot would have been shot |
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| 03:40.09 | kerx | Hi All, what would the recommended method be to keep call recordings integrated with a database based on what agent answered the phone? |
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| 03:42.57 | joako | If in sip.conf I set maxexpiry=3600, how can sip show peer xxxx show Expire : 3502 |
| 03:43.55 | joako | It can show that because I only read the first and last digit :) |
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| 04:02.36 | pdmmm | ZED |
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| 04:08.14 | scottz | PDM! =) |
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| 05:14.10 | NewCastleScott | hi all, I have been trying to get * up and running with an iaxy and voipstunt. I have pasted my config files as well as some debugging code I got from the cli----> http://gentoo.pastebin.ca/1355022 I havent a clue as to what Im doing wrong. I thank anyone in advance for any help |
| 05:16.37 | haps | hey NewCastleScott i don't know much about iax and i'm a n00b to * as well, so this may be a case of the blind leading the blind |
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| 05:16.49 | *** mode/#asterisk [+o Mog] by ChanServ |
| 05:17.20 | NewCastleScott | heh no worrys |
| 05:17.30 | haps | but i noticed that your voipstunt address is 5060, isn't that sip standard? should you put bindport=5060 in that section to override your general? |
| 05:18.51 | haps | what does iax2 show peers tell you? |
| 05:18.58 | haps | it should list the connected peers |
| 05:19.25 | haps | i only use sip, but when i do 'sip show peers' it lists my phone and provider |
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| 05:20.51 | NewCastleScott | 007/007 192.168.1.69 (S) 255.255.255.255 4569 Unmonitored |
| 05:20.51 | NewCastleScott | voipstunt 194.221.62.198 (S) 255.255.255.255 4569 Unmonitored |
| 05:21.03 | haps | ah cool |
| 05:21.05 | NewCastleScott | thats what I get from iax2 show peers |
| 05:21.13 | haps | that's good |
| 05:21.31 | haps | i was just on the voipstunt website and i didn't see where they support iax? |
| 05:21.34 | NewCastleScott | yeah but its ignoring the port like you noticed, I did change them in both files |
| 05:22.38 | haps | yeah but 5060 is the sip protocol port |
| 05:22.51 | haps | are you sure you should be using iax for voipstunt? |
| 05:23.13 | NewCastleScott | http://wiki.linuxmce.org/index.php/VoIP_with_voipstunt.com |
| 05:23.25 | NewCastleScott | I figured it would work but maybe I dont have the correct device |
| 05:24.06 | haps | asterisk is the right device :) |
| 05:25.33 | haps | NewCastleScott: yeah just remove your voipstunt shit from iax2.conf and put it in sip.conf |
| 05:25.43 | haps | but do it like they say in that wiki |
| 05:25.48 | haps | like, don't just cut and paste it |
| 05:26.10 | haps | also add 'nat=yes' |
| 05:26.52 | NewCastleScott | I have been reading soe of the book all day |
| 05:27.03 | NewCastleScott | s/soe/some |
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| 05:32.06 | haps | NewCastleScott: yeah i'm not a huge fan of the book |
| 05:32.24 | haps | NewCastleScott: don't play around with too many things for now, just get the basics going. |
| 05:32.44 | haps | which means remove voipstunt from your configs (it shouldn't be in iax.conf anyway) |
| 05:34.17 | haps | NewCastleScott: in the book there is a section in chapter 3 or 4 where he talks about a basic sip.conf |
| 05:34.45 | NewCastleScott | but I dont think this is a sip device :( |
| 05:34.52 | haps | NewCastleScott: doesn't matter |
| 05:35.03 | haps | asterisk bridges any different protocol |
| 05:35.09 | haps | the phone hardware talks to asterisk |
| 05:35.14 | NewCastleScott | http://www.google.com/products?q=IAXY&oe=utf-8&rls=org.gentoo:en:official&client=firefox-a&um=1&ie=UTF-8&ei=BgaySZ3ULYrIM9qTzeIE&sa=X&oi=product_result_group&resnum=1&ct=title |
| 05:35.19 | haps | asterisk talks to the voip provider |
| 05:35.29 | haps | so they can both talk different protocols |
| 05:35.54 | NewCastleScott | thats what I thought but Im having a heck of a tie with this one |
| 05:36.08 | haps | NewCastleScott: well, start from the basics |
| 05:36.22 | haps | remove all the voipstunt shit from your configs |
| 05:36.26 | haps | because it's wrong and broken |
| 05:36.40 | NewCastleScott | right on |
| 05:37.51 | haps | then 'iax2 reload' and 'iax2 show peers' should show only your local peer |
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| 05:41.42 | NewCastleScott | haps, that I see |
| 05:42.12 | haps | your phone has a dial tone? and when you dial 500 does that work? |
| 05:43.12 | NewCastleScott | its still on the wrong port though, and no no dial tone |
| 05:43.30 | haps | why is it on the wrong port ? |
| 05:43.48 | haps | the iaxy should be on port 4569 right? |
| 05:44.12 | NewCastleScott | so I guess Im getting a head of myself |
| 05:44.24 | haps | i think you don't understand the point of asterisk |
| 05:44.30 | haps | it sits in the middle |
| 05:44.41 | haps | between your phone and the voip provider |
| 05:45.04 | haps | so the iaxy hardware talks in a different protocol on a different port |
| 05:45.09 | haps | it's independent of the voip provider |
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| 06:07.57 | RichardLynch | Anybody familiar with the Festival Scheme code around?... Or even just scheme programming in general? |
| 06:08.42 | carrar | 10 print "Hello World" |
| 06:08.45 | carrar | 20 goto 10 |
| 06:10.10 | RichardLynch | (let ((log (fopen "/var/log/festival" "a"))) (format log "asterisk voice: %s" voice)) |
| 06:10.56 | RichardLynch | Nada in /var/log/festival But then, I don't have any connect/disconnect messages there either, for days on end now, but have old ones. chmod 777 /var/log/festival has been tried, to no avail. |
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| 06:14.47 | Jared555 | this may sound like a stupid question..... but is there any way to use asterisk as an easy to use remote time clock? where someone could use speed dial # 1 to clock in and speed dial #2 to clock out? |
| 06:14.57 | Jared555 | or something along those lines |
| 06:15.11 | carrar | easy |
| 06:15.22 | Jared555 | basically for a company that has multiple job sites, some with very limited time availability |
| 06:15.30 | carrar | can write your ins and outs to a db |
| 06:15.33 | Jared555 | limited electric* |
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| 06:17.31 | Jared555 | trick would be making it so it could be used quickly and in a way that you know for sure if you are clocking in or out |
| 06:17.53 | carrar | make a voice prompt |
| 06:17.57 | carrar | and voice confirmation |
| 06:18.02 | Jared555 | yeah, what I was thinking |
| 06:18.51 | Jared555 | right now it is only like 3-5 employees..... what sucks for the boss is there is frequently more job sites than employees :) |
| 06:19.58 | RichardLynch | Wouldn't they want to know WHERE they were clocking in/out from? Which site/job? |
| 06:20.20 | Jared555 | RichardLynch, well in that case a voice recording would be possible |
| 06:20.34 | RichardLynch | Or perhaps you could get really slick and tie in GPS :-) |
| 06:21.11 | RichardLynch | So they can't lie and clock in for a job site from the golf course too. :-) |
| 06:21.12 | RichardLynch | I don't think that would qualify as simply anymore though... |
| 06:21.23 | Jared555 | unfortunately it would be using employee phones LOL |
| 06:21.49 | RichardLynch | s/simply/simple/ |
| 06:22.48 | Jared555 | jbot, I know |
| 06:22.56 | jbot | You know? |
| 06:23.09 | Jared555 | btw. not talking about simple for me.... simple for the people having to use it |
| 06:23.10 | RichardLynch | Can I turn of jbot for s/// |
| 06:23.57 | Jared555 | crap.... lol.... anyway..... |
| 06:24.11 | RichardLynch | I think it would be both, actually, if it's just in/out. You could even just use some kind of Log file and grep those with a cron job and call it done. |
| 06:24.41 | Jared555 | yeah, and it is kind of obvious if work has been done or not (construction) |
| 06:25.22 | RichardLynch | Your dialplan would just be exten => clock_in,1,Log(${CID})\nexten=>clock_in,n,HangUp() |
| 06:25.38 | RichardLynch | :-) |
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| 06:32.22 | Jared555 | is there actually a password option or would I just have to put the code into dialplan? |
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| 07:39.38 | dan__t | Ok, so, I'm still trying to wrap my head around this. Using FastAGI, can I literally fire off at a lighty or apache server that has a PHP script which responds in an AGI-ish manner, and go from there? |
| 07:40.11 | dan__t | Like AGI(agi://1.2.3.4:80/myapp) ? |
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| 08:01.52 | halstead | I'd like to setup a system that will accept calls to my current cell phone number, take a voicemail, compress it, and e-mail it to me. I'm tired of my cell and this sounds like fun. Does anyone know a SIP/AIX that offers LNP for a good price? |
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| 08:04.49 | denon | ~itsplist |
| 08:05.00 | denon | ~itsplist-us |
| 08:05.01 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
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| 08:42.27 | EmleyMoor | I have managed to register my N95 with my Asterisk box from another house but, over the same network, cannot register ekiga with it. IPtran and STUN both just time out and connecting without either connects but ekiga won't ring. Why would that happen? |
| 08:43.49 | EmleyMoor | nat=yes in sip.conf for both |
| 08:49.13 | halstead | Out of passing curiosity, does anyone know why at&t, t-mobile, comcast, verizon, etc can port a number in 12-72 hours while all the PSTN DID providers I see need 10-30 days? |
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| 08:49.24 | specialist1 | hi |
| 08:49.38 | halstead | hi |
| 08:50.03 | specialist1 | halstead :wassup pal |
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| 08:51.09 | Frogzoo | is *'s voice quality comparable to cisco's CME? |
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| 08:54.44 | brunner | are there any voip providers that activate instantly? |
| 09:05.01 | siera08 | hi, I have MoH problem in asterisk 1.4.18 now. When i call external phone using cmd dial with 'r' option, i can listen 'connecting...' and call is successed. |
| 09:05.20 | siera08 | When dial with 'm' option, MoH music plays. but i can't hear 'connecting...'. and when called party is busy, i can't hear busy message from telephone provider. |
| 09:05.32 | siera08 | anyone had problem yet? |
| 09:05.42 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 09:05.47 | siera08 | and i applied "internal_timing = yes" in asterisk.conf. but same problem. |
| 09:06.08 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 09:06.49 | siera08 | when i call internal phone with 'm' option, it's good. |
| 09:06.53 | siera08 | ?? |
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| 09:31.29 | Great_Anta_Baka | morning |
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| 09:36.31 | Great_Anta_Baka | I have a golf estate that I am in the process of locking down. Since the site was so big it was decided that the best way to provide broadband access would be to use dsl.. so the place is full of copper infrastructure.. this heads into a DSLAM and then into these things called audiocodes (http://www.truedataonline.com/xq/asp.index/file.home/catID.1899/qx/AudioCodes+Gateways.htm) only thing is the damn things kee |
| 09:37.08 | Great_Anta_Baka | so what can i use in place of those wonderfull hybrex's as analog voip gateways? |
| 10:06.14 | tzafrir_laptop | Great_Anta_Baka, what's wrong with those AudioCode devices? |
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| 11:38.41 | funnymash | i want to have a dedicated * to be a dialer and the other * will be the registrar and storage of voicemail etc etc. I have already create a trunk between the two * but i can't make the outside call |
| 11:40.35 | funnymash | what did i miss? |
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| 12:21.19 | medjr | ... |
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| 16:01.09 | *** join/#asterisk McDarren (n=darren@shrubbery.chris.net.au) [NETSPLIT VICTIM] |
| 16:01.09 | *** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM] |
| 16:01.09 | *** join/#asterisk krdian (i=krdian@195.225.76.25) [NETSPLIT VICTIM] |
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| 16:01.09 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM] |
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| 16:01.09 | *** join/#asterisk bmg505 (n=leon@196-209-76-141-tbnb-esr-2.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 16:01.09 | *** join/#asterisk sevard (n=sev@multimedia.dvc.edu) [NETSPLIT VICTIM] |
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| 16:01.09 | *** join/#asterisk stintel (i=stijn@madwifi/support/stintel) [NETSPLIT VICTIM] |
| 16:01.10 | *** join/#asterisk ingenius (n=alektro@host151.200-117-158.telecom.net.ar) |
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| 16:06.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
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| 16:28.07 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 16:28.16 | brunner | does anyone know of a voip provider who activates instantly? |
| 16:40.19 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 16:43.13 | eppigy | hello |
| 16:43.15 | eppigy | i am dave |
| 16:44.29 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:50.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 17:17.01 | stabler | brunner: flowroute will activate instantly and give you a $.25 credit |
| 17:17.15 | stabler | brunner: to test there service |
| 17:17.35 | brunner | stabler: thanks |
| 17:17.39 | brunner | stabler: are there any others? |
| 17:18.14 | stabler | brunner: there probably are |
| 17:18.23 | *** join/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 17:18.28 | stabler | brunner: but i dont have knowledge of any others |
| 17:18.33 | brunner | okay, thanks |
| 17:18.47 | stabler | I was fully satisfied with flowroute when i tried there service |
| 17:19.08 | stabler | $.0098/min in the US |
| 17:21.55 | *** join/#asterisk leif[mobile] (n=leifmads@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:21.55 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 17:22.00 | leif[mobile] | yo |
| 17:26.32 | brunner | I can't figure out why I can't get calls through IPKall sometimes |
| 17:33.28 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
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| 17:42.19 | brunner | it seems to work sometimes after I restart |
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| 18:01.18 | *** join/#asterisk tehfox (n=tehfox@adsl-dyn-165.95-102-35.t-com.sk) |
| 18:01.34 | tehfox | hello there. |
| 18:01.36 | brunner | I can't tell if it's a problem on my end or if it's IPkall |
| 18:02.02 | tehfox | anybody here tried to interconnect asterisk with IMS (specifically OpenIMSCore)? |
| 18:02.54 | brunner | would someone call brunner@ekiga.net? |
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| 18:11.50 | *** join/#asterisk SparFux (n=raoul@e182024033.adsl.alicedsl.de) |
| 18:12.02 | SparFux | Hello all. My whole dialplan is way too complicated. |
| 18:12.02 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 18:12.19 | SparFux | Too many weirdly named variables and contexts. |
| 18:12.23 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 18:12.51 | NewCastleScott | hey hey all, What would be the *best* way to debug the 500 ext. echo test only giving a busy signal? |
| 18:13.31 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 18:16.53 | haps | NewCastleScott: iax2 set debug |
| 18:17.04 | haps | then watch what scrolls by on the cli when you dial that extension |
| 18:17.27 | haps | does your phone give you an error message, like 404 when you hangup? |
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| 18:40.25 | SparFux | Is there a command line tool to convert letters to predictive text T9 ? |
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| 18:55.03 | *** join/#asterisk harry__ (n=h@imperialglamour.com) |
| 18:58.05 | harry__ | can I play a file *while* receiving input form the user? that is, the audio file continues to play while I receive and handle entered digits? |
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| 19:22.43 | SparFux | Fender: So, you say it is ok to just not use a prefix. In order to be able to simply call an incoming number back. And there should not even be a prefix for local service, just a simple number which does it. |
| 19:25.58 | *** part/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
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| 19:27.59 | *** part/#asterisk tech9iner (n=hacim@unaffiliated/tech9iner) |
| 19:27.59 | unixdawg | ok we have a issue in 1.6.1 |
| 19:28.13 | unixdawg | it does not look to be compiling and installing chan_sip |
| 19:28.26 | unixdawg | I just did a build |
| 19:28.37 | unixdawg | and a install |
| 19:28.44 | unixdawg | and there is no chn_sip |
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| 19:29.05 | unixdawg | chan_sip in /usr/lib/asterisk/modules |
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| 19:39.32 | pdmmm | isnt 1.6.1 beta ? |
| 19:39.47 | unixdawg | its rc |
| 19:41.29 | unixdawg | rc1 |
| 19:41.39 | unixdawg | but I think I found a issue |
| 19:41.41 | unixdawg | lol |
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| 19:53.58 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 19:55.26 | axisys | i am planning to install asterisk on my ubuntu server.. i just got a fxs/fxo card.. which version do I install? 1.4.x or 1.6.x ? |
| 19:56.45 | unixdawg | 1.6.0 |
| 19:56.49 | unixdawg | and the gui |
| 19:57.23 | axisys | unixdawg: what gui are u referring to? sorry i am newbie |
| 19:58.01 | unixdawg | the digium asterisk-gui |
| 19:58.05 | unixdawg | its inthe svn |
| 19:58.23 | Dovid | unixdawg: better if people learn with out the GUI IMHO. That way they "really" learn Asterisk |
| 19:58.28 | unixdawg | we have debian install scipt |
| 19:59.59 | axisys | unixdawg: how about ubuntu ? |
| 20:00.20 | axisys | Dovid: i think so.. which one would u recommend ? |
| 20:00.24 | axisys | Dovid: version wise |
| 20:01.57 | drmessano | 1.6.0 |
| 20:02.04 | drmessano | 1.4.0 is a dinosaur now |
| 20:02.24 | drmessano | Well, 1.4 rather |
| 20:03.17 | axisys | drmessano: ok |
| 20:03.24 | Dovid | i am still using 1.4.x |
| 20:03.43 | Dovid | after 1.4.x came out there was a lot of issues. i am nervous about going to 1.6.X |
| 20:04.28 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 20:04.38 | drmessano | There's no real issues with 1.6 |
| 20:04.49 | drmessano | and 1.4 is soon to be 2 versions back |
| 20:04.57 | drmessano | With 1.6.1 being released |
| 20:05.08 | drmessano | So no point not migrating now |
| 20:06.24 | unixdawg | 1.6.1 seems to have a issue |
| 20:06.35 | unixdawg | it is building chan_sip but not installing it |
| 20:07.08 | unixdawg | I have 3 boxes I have built todat and all of then are missing chan_sip in /usr/lib/asterisk/modules |
| 20:07.16 | unixdawg | todat/today |
| 20:07.31 | *** join/#asterisk sivadnz (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 20:07.42 | unixdawg | so I would stick to 1.6.0.6 |
| 20:08.55 | axisys | unixdawg: 1.6.0.6 have a asterisk gui ? |
| 20:09.18 | axisys | unixdawg: i see only 1.4.x for asterisk gui so far |
| 20:09.36 | unixdawg | yes 1.6.0.6 has the gui also |
| 20:09.42 | unixdawg | and 1.6.1 |
| 20:11.46 | axisys | unixdawg: hmm i have been googling and can find a link |
| 20:11.52 | axisys | can't |
| 20:16.11 | unixdawg | keep looking brb on a call |
| 20:16.31 | axisys | hmm.. zaptel is only 1.4.12 .. will it work with asterisk 1.6.x ? |
| 20:16.34 | axisys | unixdawg: k |
| 20:18.23 | NewCastleScott | hi all, I cant for the life of me get my iaxy box to register...... has anyone been able to resolve tis issue? |
| 20:23.08 | drmessano | axisys: No, Dahdi replaced Zaptel, you use Dahdi with 1.6 |
| 20:23.20 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
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| 20:36.31 | axisys | drmessano: glad i ask that question |
| 20:36.36 | axisys | asked |
| 20:37.02 | axisys | drmessano: i dont see dahdi here http://downloads.digium.com/pub/ |
| 20:38.26 | [TK]D-Fender | NewCastleScott: Issue? You make it sound like a design flaw... |
| 20:39.05 | axisys | drmessano: found it under telephony |
| 20:39.23 | NewCastleScott | [TK]D-Fender, Ill blame user error |
| 20:40.14 | axisys | unixdawg: is this the one http://downloads.digium.com/pub/telephony/asterisk-gui/releases/ that installs to 1.6.x |
| 20:41.41 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
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| 20:48.09 | NewCastleScott | [TK]D-Fender, after I provision it and power cycle the device keep getting an "error" stating "[Mar 7 14:35:11] NOTICE[19608]: chan_iax2.c:6307 register_verify: No registration for peer 'Scott' (from 192.168.5.176)" have you got any insight on it? |
| 20:48.38 | [TK]D-Fender | NewCastleScott: pastebin your iax.conf masking only password |
| 20:49.20 | SparFux | I am working on my dialplan and I am using variables for every sensitive information, so I could simply post the whole dialplan then without the variables, which I put in a file I #include. |
| 20:49.29 | unixdawg | it works on 1.4 and 1.6.0 and 1.6.1 |
| 20:49.34 | unixdawg | axisys, |
| 20:50.57 | [TK]D-Fender | SparFux: You should not have sensitive information in your dialplan normally. |
| 20:51.16 | SparFux | Fender: Well, there are some passowords in it. |
| 20:51.26 | [TK]D-Fender | SparFux: DEFINITELY should not have those. |
| 20:51.29 | SparFux | And I consider my MSN numbers sensitive info. |
| 20:51.36 | [TK]D-Fender | sapMake a peer entry like everyone else |
| 20:52.35 | *** join/#asterisk BuSyAnToS (n=31749@81-208-83-253.fastres.net) |
| 20:52.37 | axisys | unixdawg: thnx |
| 20:52.50 | axisys | unixdawg: i am looking for a installation procedure.. |
| 20:52.51 | SparFux | Fender: well, I have it, like this: Dial(IAX2/${FWD_NUMBER}:${FWD_PASSWORD}@iax2.fwdnet.net/${EXTEN:6},60,r${DDOPT}) |
| 20:53.07 | [TK]D-Fender | SparFux: Like I said, make a PEER ENTRY for them instead |
| 20:53.12 | axisys | do I install dahdi and pri and then asterisk like 1.4.x ? |
| 20:53.15 | SparFux | Ah, ok. |
| 20:53.28 | SparFux | peer entry goes to iax2.conf, right? |
| 20:53.38 | [TK]D-Fender | SparFux: in that case, yes |
| 20:53.46 | SparFux | thx |
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| 20:55.03 | SparFux | Fender: Ok, I have a type=user entry. I think that's the point, right? |
| 20:55.14 | NewCastleScott | [TK]D-Fender, AWESOME!! I looked over it again and I got it to register so again I blame user error. |
| 20:55.21 | [TK]D-Fender | SparFux: "user" is for inbound-only. |
| 20:55.39 | [TK]D-Fender | NewCastleScott: Let me gues.. you tried specifying the host, didn't you? |
| 20:56.09 | harry__ | can I play a file *while* receiving input form the user? that is, the audio file continues to play while I receive and handle entered digits? |
| 20:56.22 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 20:56.23 | NewCastleScott | no, I didnt have all the "names" the same in the file.... |
| 20:56.38 | [TK]D-Fender | NewCastleScott: that would do it... |
| 20:56.56 | [TK]D-Fender | harry__: nothing I can think of off-hand |
| 20:56.59 | axisys | man there is addons,g729 codecs, patch,dahdi .. all kind a stuff.. would be nice to have a step by step install process |
| 20:57.18 | unixdawg | no |
| 20:57.26 | harry__ | damn. I'm replacing an existing VoiceGuide system. that one is able to do it :/ |
| 20:57.27 | unixdawg | dahdi is for zap devices |
| 20:57.46 | unixdawg | and most zapdevices only do ulaw/alaw |
| 20:58.12 | unixdawg | wait a dahdi patch for g729 |
| 20:58.21 | [TK]D-Fender | .... |
| 20:58.27 | unixdawg | sorry miss read |
| 20:58.35 | [TK]D-Fender | sure is a lot of crack going around here these days |
| 20:58.37 | unixdawg | been up for 27 hours |
| 20:58.55 | axisys | unixdawg: heh |
| 20:59.01 | unixdawg | and still have to make it threw today |
| 20:59.07 | axisys | unixdawg: ouch! |
| 20:59.12 | carrar | shares his PRI crack covered g729 codec with [TK]D-Fender |
| 21:00.36 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
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| 21:05.22 | SparFux | Fender: Do I need a user AND a peer entry in iax.conf, or simply a peer entry? |
| 21:06.11 | [TK]D-Fender | SparFux: just a peer |
| 21:06.33 | [TK]D-Fender | usually. Sometimes as a "friend" |
| 21:06.37 | SparFux | So I need a register line in general section and the peer entry. |
| 21:06.51 | [TK]D-Fender | SparFux: You should |
| 21:09.09 | SparFux | And then I have to use context= in the peer section to tell it where to receive the calls in. |
| 21:11.16 | carrar | I swear I saw all that in the book |
| 21:13.01 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 21:13.48 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 21:14.06 | [TK]D-Fender | SparFux: They're going to go SOMEWHERE you know.... |
| 21:14.46 | SparFux | And NOWHERE is somewhere, too! :-P |
| 21:15.01 | brunner | If someone sets CALLERID(ani) to be blank or "Unknown", and the VoIP provider allows this, does that result in the Called Party Number field in the ISUP IAM being blank or invalid as well? |
| 21:16.09 | [TK]D-Fender | SparFux: [nowhere] <- it is now. |
| 21:16.19 | SparFux | LOL :-D |
| 21:16.41 | brunner | sorry, I meant the Charge Number field, of course |
| 21:17.16 | carrar | try it |
| 21:17.47 | brunner | I have no way to get the SS7 field values |
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| 21:20.35 | *** mode/#asterisk [+o bkruse] by ChanServ |
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| 21:29.36 | axisys | failing to compile dahdi on 2.6.27-11 |
| 21:29.37 | axisys | http://pastebin.com/d5d9036f9 |
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| 21:56.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 22:07.25 | *** join/#asterisk eXcAliBuR (n=bob@207.134.8.33) |
| 22:07.55 | eXcAliBuR | can i talk 1-on1 with someone that knows enough to have a asterisk certification but doesn't... |
| 22:08.21 | dan__t | Hello. |
| 22:08.31 | eXcAliBuR | hey dan |
| 22:08.43 | eXcAliBuR | are you able to help me? |
| 22:08.52 | dan__t | I think the combined knowledge you would find here far exceeds a single individual who might know enough to have an asterisk certification but doesn't. |
| 22:09.02 | eXcAliBuR | ok |
| 22:09.06 | pdmmm | yeah |
| 22:09.10 | pdmmm | im smrt |
| 22:09.16 | eXcAliBuR | cisco call manager compared to asterisk |
| 22:09.29 | pdmmm | ask your wallet! |
| 22:09.32 | eXcAliBuR | money not being a issue |
| 22:09.37 | eXcAliBuR | which would you pick? |
| 22:09.45 | pdmmm | Asterisk |
| 22:10.06 | GameGamer43 | I'd pick Asterisk over almost anything out there atm |
| 22:10.43 | pdmmm | its way more versitile |
| 22:10.43 | eXcAliBuR | ok 18 sites, within different calling areas, can asterisk send an outgoing call out from any 1 site, based on where the call is going? |
| 22:10.50 | pdmmm | yes |
| 22:11.08 | eXcAliBuR | so that most calls will be local calls instead of long distance |
| 22:11.12 | pdmmm | yes |
| 22:11.17 | pdmmm | LCR |
| 22:11.23 | pdmmm | least call routing |
| 22:11.25 | eXcAliBuR | how would all them be inter-connected |
| 22:11.33 | eXcAliBuR | all the asterisk boxes |
| 22:11.34 | dan__t | iax2! |
| 22:11.42 | eXcAliBuR | is this easy to do |
| 22:11.44 | pdmmm | SIP |
| 22:11.55 | pdmmm | just like any other would be connected |
| 22:11.58 | dan__t | Doesn't matter, its just another technology. |
| 22:12.08 | eXcAliBuR | cos i have like a 500 page manual ... if i read it.. will i know how to do this? |
| 22:12.21 | dan__t | If the manual tells you how to, sure. |
| 22:12.29 | dan__t | Which manual is it? |
| 22:12.35 | eXcAliBuR | umm one from site |
| 22:12.39 | pdmmm | ha |
| 22:12.40 | eXcAliBuR | asterisk in a nutshell |
| 22:12.43 | eXcAliBuR | or something ike that |
| 22:12.47 | dan__t | I see. |
| 22:12.49 | pdmmm | i think you need to read the manual |
| 22:12.51 | dan__t | I don't know. Read it and let us know. |
| 22:12.58 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 22:13.09 | pdmmm | you're asking questions like "if i install asterisk and go to school, will I know how to dial?" |
| 22:13.14 | pdmmm | a lot of the answers depend on you |
| 22:13.16 | dan__t | I've found the O'Reilly Asterisk book to be fantastic. |
| 22:13.19 | dan__t | The second edition, not the first. |
| 22:13.22 | eXcAliBuR | ok hang on i have it right here |
| 22:13.23 | dan__t | The first was crap compared to the second. |
| 22:13.25 | eXcAliBuR | lets read it together |
| 22:13.39 | eXcAliBuR | page 1 |
| 22:13.41 | dan__t | Also, Asterisk AGI Programming, is fantastic. |
| 22:13.48 | eXcAliBuR | lol |
| 22:13.55 | dan__t | Seriously. |
| 22:14.02 | dan__t | You can even download the O'Reilly book for free. |
| 22:14.03 | dan__t | (legally) |
| 22:14.04 | eXcAliBuR | what is agi ? |
| 22:14.13 | dan__t | Asterisk Gateway Interface |
| 22:14.44 | dan__t | Which I still have some questions on, actually... Any FastCGI fans around? Trying to see how I can use that with Apache or lighty. Can I literally fire requests over the wire to a lighty server running my fancy PHP AGI script |
| 22:14.45 | dan__t | ? |
| 22:17.30 | *** join/#asterisk tehfox (n=tehfox@adsl-dyn-165.95-102-35.t-com.sk) |
| 22:17.44 | eXcAliBuR | don't you love when ur printer says it's got a paper jam and it doesnt |
| 22:17.51 | *** join/#asterisk gulden (n=gulden@av3-84-90-162-232.netvisao.pt) |
| 22:17.55 | eXcAliBuR | will give it something to jam about |
| 22:17.57 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 22:18.25 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 22:18.32 | eXcAliBuR | great, now i need a new printer |
| 22:18.58 | eXcAliBuR | aren't you s'pose to be able to drop these things? |
| 22:19.21 | eXcAliBuR | :P |
| 22:19.28 | eXcAliBuR | ok... now everyone things i'm nuts |
| 22:19.32 | eXcAliBuR | thinks** |
| 22:20.03 | dan__t | Yep. |
| 22:28.41 | eXcAliBuR | how many asterisk boxes can i connect together? |
| 22:28.48 | eXcAliBuR | thinks the answer is as many as i need |
| 22:28.51 | eXcAliBuR | :) |
| 22:29.12 | dan__t | What makes you think you need more than one? |
| 22:29.21 | dan__t | Got like a billion phones or what |
| 22:29.49 | eXcAliBuR | well i have a lot of sites |
| 22:29.56 | eXcAliBuR | over a lot of miles |
| 22:30.01 | dan__t | can't make them all just speak sip to one server? |
| 22:30.04 | eXcAliBuR | so many calling areas |
| 22:30.34 | eXcAliBuR | if i want to call area 1 from area 4 .. |
| 22:30.40 | dan__t | So you're going to use like many FXO cards in each area? |
| 22:30.55 | eXcAliBuR | ummm nops |
| 22:31.11 | eXcAliBuR | 1 live phone line per area |
| 22:31.29 | eXcAliBuR | that makes sence.. right? |
| 22:31.32 | dan__t | Why not just use a SIP or IAX2 provider and tie Asterisk in to that? |
| 22:31.39 | dan__t | I guarantee it would be cheaper. |
| 22:31.40 | haps | yeah |
| 22:31.46 | haps | and better |
| 22:31.54 | haps | does that |
| 22:31.57 | eXcAliBuR | you mean one of them $40 a month things for unlim calling? |
| 22:31.59 | dan__t | To do that you'll need one FXO card per calling area per machine... these other providers already have that established. |
| 22:32.01 | haps | multiple dids |
| 22:32.10 | dan__t | One DID in each diling area. |
| 22:32.12 | dan__t | dialing, too. |
| 22:32.26 | dan__t | It's a lot more than "one of them $40 a month things" |
| 22:32.31 | eXcAliBuR | overall budget is $100k |
| 22:32.32 | haps | dan__t: why fxo cards? |
| 22:32.38 | haps | just use sip phones |
| 22:32.38 | dan__t | haps, how much does a basic 4channel fxo card go for? |
| 22:32.43 | haps | got me |
| 22:32.48 | dan__t | er, fxo == phone line -> asterisk |
| 22:32.48 | dan__t | ? |
| 22:32.50 | haps | i stay the fuck away from fxo/fxs |
| 22:32.53 | dan__t | word |
| 22:33.07 | eXcAliBuR | what do you use then? |
| 22:33.11 | dan__t | I'm just saying. Sounds like he plans on tying a physical phone line in to each asterisk server which then sits in each calling area. |
| 22:33.14 | dan__t | I use Vitelity. |
| 22:33.26 | eXcAliBuR | yups.. thats the plan dan |
| 22:33.33 | dan__t | I've used Teliax before, as well. |
| 22:33.40 | dan__t | Why bother? |
| 22:33.46 | dan__t | That's way too much trouble. |
| 22:34.07 | dan__t | Use a provider, and only $140 of your $100k budget gets spent. |
| 22:34.14 | dan__t | Do you have any idea how much beer that is? |
| 22:34.24 | *** join/#asterisk doolph (n=doolph@200.46.11.40) |
| 22:34.31 | doolph | hell |
| 22:34.33 | haps | heh |
| 22:34.33 | doolph | o |
| 22:34.50 | eXcAliBuR | does voip work with fax now? |
| 22:34.55 | eXcAliBuR | has that been fixed? |
| 22:35.15 | haps | eXcAliBuR: you want to have a bunch of people from different areas able to call the other offices for free, right? |
| 22:35.41 | haps | or do you have one office that you want to be able to call multiple areas locally? |
| 22:35.53 | haps | eXcAliBuR: avoid fax, use email |
| 22:35.53 | dan__t | Where are my FastAGI hackers. |
| 22:35.54 | eXcAliBuR | i want offices to be able to call other areas for free... as in outside the office in such area |
| 22:36.33 | doolph | what you mean with other areas |
| 22:36.39 | haps | well then use a voip provider for calling, and it'll never be 'free', since a local line in each calling area costs, plus you need a colo and a network connection there |
| 22:36.49 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:36.56 | dan__t | And specialized hardware to tie it together. |
| 22:36.58 | eXcAliBuR | we have a fiber link between sites |
| 22:37.02 | haps | eXcAliBuR: but say you want to call anywhere in north america for dirt cheap, then use a voip provider |
| 22:37.04 | dan__t | And, let's be honest. Digium doesn't sell bargain basement stuff. |
| 22:37.04 | dan__t | heh |
| 22:37.06 | haps | they do the hard work for you |
| 22:37.53 | haps | run a * server at each office that deals with the dialplan, connecting to your voip provider for outgoing and connecting to the other offices for inter-branch extensions |
| 22:37.58 | eXcAliBuR | we have locations all over quebec ;( |
| 22:38.05 | eXcAliBuR | big area to cover |
| 22:38.18 | haps | quebec? get out of here you dirty french fuck. |
| 22:38.22 | haps | montreal |
| 22:38.26 | eXcAliBuR | hey am i talking french? |
| 22:38.29 | haps | heh |
| 22:38.44 | haps | i'm just poking because I can. |
| 22:38.56 | eXcAliBuR | é |
| 22:39.07 | eXcAliBuR | my keyboard has french tho |
| 22:39.08 | eXcAliBuR | :) |
| 22:39.22 | haps | dude your idea of hooking up * servers and using local lines to get local calls seems good on the outside but it's very costly when you get down to it |
| 22:39.34 | eXcAliBuR | si jài besion du parle... |
| 22:39.56 | eXcAliBuR | ok, good idea, wrong way of doing it |
| 22:40.11 | haps | each office can have a * server, that's fine |
| 22:41.05 | haps | then that * machine can deal with multiple handsets at the office, and it can connect to a central office * server or all other offices, I don't know the best idea in terms of network topology |
| 22:41.07 | eXcAliBuR | it's required to have a live phone line in each site.. for emergency reasons... so i wanted to make use of it |
| 22:41.09 | haps | best to get a consultant on that |
| 22:41.30 | haps | well, don't make use of it |
| 22:41.37 | haps | leave it alone with a single dedicated handset |
| 22:41.46 | haps | otherwise, it could be in use when you need it |
| 22:42.12 | haps | or, it's plugged into the * box but you had a power outage and the internet is down, etc |
| 22:42.26 | haps | although hopefully the equipment is on backup |
| 22:42.36 | haps | that's my 2cents anyway |
| 22:52.30 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 22:54.45 | *** join/#asterisk HoverHell (n=hell@91.146.50.221) |
| 22:57.29 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 22:58.26 | brunner | what's the cheap way to get a single FXS port? |
| 22:58.32 | brunner | s/the/a |
| 22:59.49 | eXcAliBuR | i would say like a linksys or dlink gateway |
| 22:59.55 | eXcAliBuR | not sure which on makes it |
| 23:00.08 | drmessano | PAP2 |
| 23:00.15 | eXcAliBuR | it ties into your network, then u add it to your asterisk box |
| 23:00.24 | eXcAliBuR | :) |
| 23:00.36 | drmessano | Single port costs more |
| 23:00.42 | drmessano | So get a PAP2 |
| 23:01.09 | brunner | well, I'm looking to build an ATA out of a Mini-ITX board that runs asterisk so I can use ZRTP for encrypted conferencing |
| 23:01.36 | brunner | so I really wanted a PCI card |
| 23:01.54 | eXcAliBuR | search ebay for a cheap knock off |
| 23:02.13 | brunner | eXcAliBuR: knock off of what? |
| 23:02.20 | drmessano | Try being more specific |
| 23:02.31 | drmessano | There is no cheap single FXS PCI card |
| 23:05.06 | eXcAliBuR | http://cgi.ebay.ca/Authentic-X100P-SE-FXO-PCI-for-Digium-Asterisk-VoIP-PBX_W0QQitemZ120385483914QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item120385483914&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1308 |
| 23:05.12 | eXcAliBuR | that is cheap knock off |
| 23:05.44 | eXcAliBuR | it's basically a 56k modem that somehow works with asterisk |
| 23:05.56 | brunner | that's perfect, thanks |
| 23:07.47 | russellb | if you're looking for a way to hook up a phone, that's not it |
| 23:07.58 | brunner | russellb: no? |
| 23:08.05 | russellb | that car is an FXO |
| 23:08.08 | russellb | ~fxofxs |
| 23:08.08 | jbot | fxofxs is, like, An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 23:09.12 | drmessano | Thats NOT an FXS |
| 23:09.22 | brunner | russellb: sorry, I hadn't read carefully enough. I heard it was a voice modem and thought that perhaps the 'phone' port would work as an FXS |
| 23:09.24 | drmessano | Even says it in the LINK and the title |
| 23:09.31 | brunner | drmessano: yes, I understand that |
| 23:10.12 | russellb | if it has a "phone" port, it's just passthrough |
| 23:11.06 | brunner | I understand that. |
| 23:11.21 | brunner | I hadn't thought it through |
| 23:14.02 | haps | while we're on the topic of fxs, let me change the topic a little |
| 23:14.19 | haps | anyone have any good experience with a wifi voip handset? |
| 23:14.54 | haps | or is the trend these days to run software on smartphones? |
| 23:15.14 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:20.01 | stabler | haps: they work well but avoid imlementing too many of them |
| 23:20.29 | drmessano | Wifi voip phones have a horrid battery life |
| 23:20.35 | drmessano | and wifi isnt best for VoIP anyway |
| 23:20.48 | drmessano | ATA + DECT is a good solution |
| 23:21.34 | haps | i just need one :) |
| 23:21.39 | haps | ata + dect ... |
| 23:21.41 | haps | googles |
| 23:21.50 | stabler | one should be fine |
| 23:22.12 | drmessano | Get an ATA |
| 23:22.13 | stabler | unless your wireless network is already overloaded |
| 23:22.16 | drmessano | and a DECT phone |
| 23:22.49 | *** join/#asterisk Tusker (n=tusker@c-24-98-144-94.hsd1.ga.comcast.net) |
| 23:22.50 | drmessano | You dont need to google ATA + DECT unless you know what neither means |
| 23:22.55 | drmessano | Theres no magic to the combo |
| 23:23.41 | Tusker | heya guys, trying to get voxalot inbound trunk working, and voxalot keeps on using a different peer to connect to the asterisk server (eu, then us, then premium)... how do I configure the peer to recognize all IPs ? |
| 23:24.02 | haps | ah dect :) |
| 23:24.08 | drmessano | insecure=port,invite |
| 23:25.32 | Tusker | drmessano: i already have insecure=very |
| 23:25.40 | Tusker | but I still get the "Found no matching peer or user for '64.34.173.199:5060'" error |
| 23:25.44 | stabler | Tusker: what version of * |
| 23:25.55 | Tusker | 1.4 |
| 23:25.58 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:26.42 | stabler | drmessano: could he use host=dynamic? |
| 23:27.07 | drmessano | http://forum.voxalot.com/voxalot-support/328-trixbox-settings-voxalot.html |
| 23:27.09 | drmessano | Follow that |
| 23:27.10 | stabler | or am i thinking cmpletely wrong |
| 23:27.50 | Tusker | stabler: isn't host=dynamic for the asterisk host, not the remote host... |
| 23:28.14 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 23:28.14 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 23:28.19 | stabler | Tusker: yea |
| 23:28.43 | stabler | Tusker: im newer to * so i was just throwing it out there |
| 23:28.46 | stabler | dont listen to me |
| 23:29.00 | drmessano | Ignore stabler, he just got out of prison last monday |
| 23:29.05 | drmessano | "computer fraud" |
| 23:29.13 | Tusker | fun :) |
| 23:29.37 | stabler | lol |
| 23:29.38 | jaytee | he needs to change insecure=very to insecure=port,invite as drmessano said since he's running 1.4. very was deprecated after 1.2 |
| 23:30.18 | *** join/#asterisk giovani2 (n=giovani@unaffiliated/giovani) |
| 23:30.35 | drmessano | Damnit jaytee! |
| 23:31.18 | drmessano | Stop helping him.. hes on 1.4, config just works on 1.4 by guessing |
| 23:31.34 | drmessano | Unless he had nodialplanguess=yes set |
| 23:32.21 | jaytee | hehehe |
| 23:33.02 | Tusker | if i were using 1.2 ? how would I fix it ? (for my benefit) |
| 23:33.32 | jaytee | asterisk core: "hmmm, I don't have extension 4566 in my dialplan so......... I know! I'll dial 2673! Yeah! That's the ticket!" |
| 23:33.50 | jaytee | Tusker, upgrade. |
| 23:33.53 | Tusker | :) |
| 23:33.54 | Tusker | k |
| 23:34.07 | stabler | Tusker: now... "passwd root letstablerin" |
| 23:34.20 | Tusker | ah, i know that one stabler |
| 23:34.22 | Tusker | cheers |
| 23:34.27 | stabler | Tusker: then "/etc/init.d/sshd start" |
| 23:34.28 | *** part/#asterisk harry__ (n=h@imperialglamour.com) |
| 23:34.36 | Tusker | sshd already started... |
| 23:34.44 | stabler | Tusker: JK! |
| 23:34.48 | Tusker | :P |
| 23:34.56 | Tusker | just wanted to see your connection attempt :) |
| 23:35.15 | stabler | heh |
| 23:35.48 | stabler | only reason drmessano knows i was in jail for computer fraud is because he was my cell mate |
| 23:36.15 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:36.34 | stabler | :D |
| 23:36.54 | jaytee | some of the most romantic nights of your life, I'd imagine. |
| 23:37.08 | stabler | lol |
| 23:38.26 | drmessano | Jaytee says, as he's sitting next to me, playing with my hair |
| 23:38.35 | drmessano | giggles and swoons |
| 23:38.42 | *** join/#asterisk stoked (n=df@S01060016b62857a6.vc.shawcable.net) |
| 23:38.56 | jaytee | hahahaa |
| 23:39.30 | stoked | does anyone have links for client behind NAT to asterisk behind NAT ? |
| 23:39.46 | drmessano | ~sipnat |
| 23:39.47 | jbot | [~sipnat] Quick guide on configuring * + 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 |
| 23:39.52 | stabler | lol |
| 23:39.58 | drmessano | Slow old man |
| 23:40.00 | stabler | drmessano: beat me to it |
| 23:40.26 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:40.27 | stoked | thanks guys |
| 23:40.29 | drmessano | :( |
| 23:40.40 | jaytee | ~botsnack |
| 23:40.40 | jbot | thanks, jaytee |
| 23:41.05 | stabler | i couldve added the "^^^^^^" though |
| 23:41.20 | drmessano | Ok, going to petco to continue my quest to modify an off the shelf pet scoop to work with feline pine |
| 23:41.34 | drmessano | Which I plan to wiki |
| 23:41.35 | jaytee | really? I was just thinking of going there too |
| 23:41.46 | drmessano | pinches jaytees ass |
| 23:41.52 | drmessano | See ya there, cutie |
| 23:42.04 | stabler | awww... a date |
| 23:42.13 | stabler | thats sooo cute |
| 23:42.24 | eppigy | oh man |
| 23:42.27 | jaytee | hahaha, in your dreams maybe |
| 23:42.31 | *** join/#asterisk SparFux (n=raoul@e182029148.adsl.alicedsl.de) |
| 23:42.37 | *** part/#asterisk SparFux (n=raoul@e182029148.adsl.alicedsl.de) |
| 23:42.38 | *** join/#asterisk SparFux (n=raoul@e182029148.adsl.alicedsl.de) |
| 23:42.38 | drmessano | Im glad I stayed for that |
| 23:42.46 | drmessano | Sleep on the sofa then, dolphinboy |
| 23:42.51 | drmessano | stomps off |
| 23:42.53 | jaytee | hahahahaa |
| 23:43.02 | SparFux | It seems to me the order in "dialplan show <context>" is NOT the criteria for what is matched first! |
| 23:43.36 | jaytee | SparFux, maybe it's because you don't understand how the dialplan actually matches? |
| 23:44.23 | SparFux | jaytee: I think _00NN. matches 0011 |
| 23:44.45 | *** join/#asterisk path_ (n=path_@pc-15-190-86-200.cm.vtr.net) |
| 23:45.02 | SparFux | jaytee: And I include some contexts, with the includes their order should be key. And their order is reflected in "dialplan show <context>". |
| 23:45.55 | SparFux | http://pastebin.com/d1d0914a0 is my contexts. |
| 23:46.27 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 23:48.07 | SparFux | Ah, I am sure it is because I use _[0-9#+*] in pstn-dialout. It seems to match before _00NN. But on the other hand, I use the include trick, so it should no matter! |
| 23:48.22 | jaytee | SparFux, N does not match against 1 |
| 23:48.44 | jaytee | N matches against 2-9 |
| 23:48.55 | SparFux | Oh. |
| 23:49.05 | jaytee | page 138 of the book, pal |
| 23:50.53 | SparFux | The book? |
| 23:51.03 | jaytee | ~book |
| 23:51.04 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 23:51.16 | jaytee | grab the free PDF and start reading |
| 23:51.40 | SparFux | Ha! |
| 23:51.59 | SparFux | Thx! |
| 23:52.04 | jaytee | in your example you'd want to use either Z which matches 1-9 or X which matches 0-9. N only matches 2-9 |
| 23:52.23 | jaytee | your welcom |
| 23:52.24 | jaytee | e |
| 23:52.37 | jaytee | gotta go play the lottery, back in a few |
| 23:52.50 | SparFux | Have luck! |
| 23:52.54 | doolph | lol |
| 23:54.29 | SparFux | oh, then I made some serious mistakes and in some cases, where I never used 0 and 1 it worked even with N. |
| 23:54.56 | SparFux | Seems like I am in trouble for not having read enough, even though I am peeking on voip-info.org all the time :-\ |
| 23:56.53 | *** join/#asterisk friendly12345 (n=friendly@ppp118-208-145-44.lns10.mel4.internode.on.net) |