| 00:03.27 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
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| 00:55.38 | sky1975 | Hi |
| 00:56.05 | sky1975 | Can anybody help me to set security on skype for asterisk? |
| 00:56.38 | [TK]D-Fender | skyWhat "security"? |
| 00:56.52 | sky1975 | I mean authentication |
| 00:57.25 | sky1975 | I tried auth_policy=accept:martini |
| 00:57.36 | sky1975 | Doesn't work |
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| 01:01.43 | *** join/#asterisk acxty (~acxty@201.220.136.118) |
| 01:01.45 | sky1975 | Shall I put my chan_skype.conf entry here? |
| 01:02.08 | acxty | Hi guys, I define some sip accounts on sip.conf. How can I see which ones are register to the server? |
| 01:02.11 | sky1975 | <[TK]D-Fender> ? |
| 01:02.51 | [TK]D-Fender | acxty: "sip show registry" / "sip show peer [peer]" |
| 01:10.59 | sky1975 | <PROTECTED> |
| 01:15.05 | dinesh___ | well if the order is the same as for the codecs then yes it is wrong |
| 01:15.18 | *** join/#asterisk x303 (~x303@20.226.118.70.cfl.res.rr.com) |
| 01:15.46 | sky1975 | Ahh. is that? I will change and check |
| 01:16.43 | dinesh___ | btw does someone happen to know what codecs the Unified Communication server from Microsoft is using ? no voice is going through the sip calls i'm making from my asterisk |
| 01:16.56 | drmessano | G711 |
| 01:17.04 | drmessano | You need SIP TCP |
| 01:17.37 | dinesh___ | thanks |
| 01:17.52 | dinesh___ | i also think that it only supports TLS (called SIPS) |
| 01:18.29 | dinesh___ | which is probably why i couldn't login from x-lite |
| 01:18.36 | dinesh___ | (which does have SIP TCP) |
| 01:18.49 | drmessano | Nope |
| 01:18.50 | drmessano | http://blogs.technet.com/b/gclark/archive/2010/05/24/3134398.aspx |
| 01:21.30 | *** join/#asterisk SAIDias (~SAID@97-125-143-112.desm.qwest.net) |
| 01:21.34 | SAIDias | Howdy |
| 01:21.51 | SAIDias | whats the best way of streaming audio to shoutcast |
| 01:21.57 | SAIDias | from an extension |
| 01:23.49 | sky1975 | <dinesh___> Will this do the job? [mx-400] secret=xxxxxx context=from-pstn auth_policy=deny auth_policy=accept:martini |
| 01:26.01 | xheliox | ~itsp-usa |
| 01:26.19 | xheliox | ~itsp |
| 01:26.19 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:26.26 | xheliox | ~itsplist-us |
| 01:26.27 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 01:30.32 | [TK]D-Fender | SAIDias: Run a softphone on auto-answer. Wire the speaker-out into Line-In. Have your shotcast app use that |
| 01:30.57 | [TK]D-Fender | SAIDias: Or actually... you ccould use chan_oss and do it more directly |
| 01:31.19 | SAIDias | [TK]D-Fender: I only want to use a single machine todo it |
| 01:32.29 | [TK]D-Fender | SAIDias: I just gave you 2 ways you can. |
| 01:37.15 | SAIDias | thx |
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| 04:33.42 | acxty | Hi guys I am getting this error pbx_spool.c:276 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test.call: Operation not permitted. What this mean? |
| 04:36.33 | acxty | I am getting this when I place a .call file on outgoing |
| 04:36.42 | acxty | chown asterisk:asterisk |
| 04:36.45 | acxty | and chmod 777 |
| 04:39.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:41.53 | acxty | may someone help me with that? |
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| 04:46.22 | pabelanger | acxty: Are you running Asterisk with -U asterisk and -G asterisk? |
| 04:51.15 | acxty | pabelanger, No |
| 04:53.20 | pabelanger | acxty: then why are you chown asterisk:asterisk your call file? |
| 04:56.18 | acxty | Is I use root:root it says permission denied |
| 04:56.28 | acxty | I only did an aptitude install asterisk |
| 04:56.36 | acxty | and only change sip.conf and extension.conf |
| 05:04.18 | jblack | perhaps an overly tight umask |
| 05:18.02 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
| 05:19.25 | p3nguin | Asterisk runs as root on that system? Yuck. |
| 05:20.11 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 05:26.39 | jblack | I think he said he is running asterisk as asterisk, and creating callfiles as root. |
| 05:30.32 | p3nguin | (2346.23) <pabelanger> acxty: Are you running Asterisk with -U asterisk and -G asterisk? |
| 05:30.36 | p3nguin | (2351.15) <acxty> pabelanger, No |
| 05:30.46 | p3nguin | I interpret this as not running it as asterisk. |
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| 05:37.22 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 05:39.40 | Dovid | good morning. anyone know what would casue this and if it is an Asterisk thing or a Linux thing ? |
| 05:45.13 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
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| 06:34.53 | *** join/#asterisk P-Chan (~jason@ip70-162-221-117.ph.ph.cox.net) |
| 06:34.59 | P-Chan | 'ello! |
| 06:36.56 | P-Chan | I've got a system I just set up that I keep getting an immediate "goodbye" and disconnect when calling in. I've got a Digium card w/ 2 lines and I think it's an issue with the incoming context being "default" instead of "from-pstn", but I can't seem to change it |
| 06:37.23 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 06:38.43 | P-Chan | Starting simple switch on 'DAHDI/2-1' |
| 06:38.44 | P-Chan | Executing [s@default:1] Playback("DAHDI/2-1", "vm-goodbye") in new stack |
| 06:39.16 | P-Chan | I'm guessing "s@default:1" means it's coming in on the "default" context, correct? |
| 06:39.41 | P-Chan | 's' => 1. Playback(vm-goodbye) [pbx_config] |
| 06:39.50 | P-Chan | <PROTECTED> |
| 06:39.57 | P-Chan | <PROTECTED> |
| 06:40.03 | P-Chan | that's my "default" dialplan |
| 06:40.32 | P-Chan | but it ignores the "context=from-pstn" line in dahdi_channels.conf |
| 06:41.03 | P-Chan | any ideas on how I can troubleshoot this? Thanks in advance. |
| 06:42.59 | P-Chan | [ Context 'default' created by 'pbx_config' ] |
| 06:42.59 | P-Chan | <PROTECTED> |
| 06:42.59 | P-Chan | <PROTECTED> |
| 06:42.59 | P-Chan | <PROTECTED> |
| 06:43.19 | P-Chan | <PROTECTED> |
| 06:43.20 | P-Chan | <PROTECTED> |
| 06:43.23 | P-Chan | and this is despite: |
| 06:44.24 | P-Chan | ; Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER) |
| 06:44.24 | P-Chan | ;;; line="1 WCTDM/0/0 FXSKS (SWEC: MG2)" |
| 06:44.24 | P-Chan | signalling=fxs_ks |
| 06:44.24 | P-Chan | callerid=asreceived |
| 06:44.24 | P-Chan | group=0 |
| 06:44.24 | P-Chan | context=from-pstn |
| 06:44.24 | P-Chan | channel => 1 |
| 06:44.25 | P-Chan | ;;; line="2 WCTDM/0/1 FXSKS (SWEC: MG2)" |
| 06:44.25 | P-Chan | signalling=fxs_ks |
| 06:44.26 | P-Chan | callerid=asreceived |
| 06:44.26 | P-Chan | group=0 |
| 06:44.27 | P-Chan | context=from-pstn |
| 06:44.27 | P-Chan | channel => 2 |
| 06:44.30 | P-Chan | (that's my dahdi_channels.conf |
| 06:44.35 | P-Chan | sorry for the spamming |
| 06:45.01 | P-Chan | oops... posted to the same chan twice, the second time it was meant for #freepbx... >>; |
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| 06:53.05 | Dovid | P-Chan: please use pb |
| 06:53.07 | Dovid | ~pb |
| 06:53.16 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 06:55.05 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 06:57.40 | ChannelZ | are you sure your chan_dahdi.conf is including that file and not some other version? |
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| 07:27.41 | ChannelZ | gets his groove on |
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| 07:42.18 | joobie | guys is there a way i can limit the RTP ports my sip peer uses? i use a paid sip peer to route some calls and they are usinga ll sorts of ports |
| 07:42.27 | joobie | can teh client somehow specify to the peer that it was the ports limited? |
| 07:43.03 | ChannelZ | Depends on the device |
| 07:43.59 | ChannelZ | As a 'client' (a phone) you can request what RTP port the remote end should send to you at. The reverse is also true (including Asterisk, via rtp.conf) |
| 07:44.22 | joobie | ChannelZ, how can this be done? I'll give it acrack |
| 07:44.31 | joobie | i use pennytel, want to get pennytel to use only a certain range |
| 07:44.41 | joobie | right now they are going all over the shop which makes my firewall rules a little less specific |
| 07:45.25 | *** join/#asterisk kerx_ (~kerxupt@76-240-161-68.lightspeed.irvnca.sbcglobal.net) |
| 07:46.01 | kerx_ | l/join #freeswitch |
| 07:46.04 | kerx_ | err |
| 07:47.56 | ChannelZ | Again it depends on the device.. you'd have to dig around in it's configs |
| 07:48.01 | ChannelZ | what are you using? |
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| 07:49.26 | joobie | i have asterisk setup to connect to pennytel, not a device |
| 07:50.28 | ChannelZ | well then as I said, you can edit /etc/asterisk/rtp.conf and set the port range Asterisk uses |
| 07:51.05 | joobie | isnt that for what asterisk uses? ie it wont affect pennytel as a sip peer to asterisk? |
| 07:51.06 | ChannelZ | You can't control what the other end wants however, but it's usually not a big problem allowing outbound trafficggg |
| 07:51.25 | joobie | ahh |
| 07:51.36 | joobie | that's what i wanted to do :/ doh |
| 07:51.44 | ChannelZ | There's two RTP streams... one from you to someone else, and one from someone else back to you. |
| 07:52.00 | joobie | ya |
| 07:52.06 | ChannelZ | You can only ask the remote site where to send their traffic to you at; You can't control where they want their traffic sent to |
| 07:52.09 | ChannelZ | s/site/side/ |
| 07:52.14 | joobie | yer |
| 07:52.16 | joobie | doh |
| 07:52.24 | joobie | cos thats what i wanted to do |
| 07:52.44 | joobie | i have outbound firewall rules that id have tp loosen if not |
| 07:54.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 07:54.38 | ChannelZ | Do you contact them at random IP addresses? If not it's pretty trivial to just allow all outgoing traffic only to them if you're paranoid about it (or a large chunk of ports) |
| 07:54.42 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 07:55.17 | ChannelZ | 16384-16482 apparently |
| 07:56.02 | joobie | na jsut the one ip |
| 07:56.14 | joobie | but they use very random ports |
| 07:56.21 | joobie | to the point where im thinking to just let all udp out to them |
| 07:56.29 | joobie | .. greater than 1023 |
| 07:56.35 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 07:56.52 | ChannelZ | yeah... according to them, 16384-16482 |
| 07:56.57 | ChannelZ | hardly the world |
| 07:58.34 | joobie | that's bs |
| 07:58.44 | joobie | im getting much more random than that |
| 07:58.56 | ChannelZ | Well then maybe you should yell at them, because that's what their FAQ says |
| 07:58.59 | ChannelZ | "You are behind a firewall. In this case you will need to have UDP ports 5060-5061 and 16384-16482 (default RTP ports for Linksys devices) open and forwarded to be able to register and use the service." |
| 07:59.05 | joobie | maybe it's the AU servers |
| 07:59.11 | joobie | yer |
| 07:59.16 | joobie | they suck for customer service tho |
| 07:59.25 | joobie | gotta be transfered through to people overseas who barely speak englihs |
| 07:59.34 | joobie | always a challenge to explain something like this to them |
| 07:59.35 | joobie | all good |
| 07:59.56 | joobie | ill wokr around it ChannelZ .. was just hoping the protocol had some prevision to specify a prefered port range or sumthen |
| 08:00.08 | ChannelZ | well you can complain here where nobody can do anything about it or complain to them (or switch providers) |
| 08:00.21 | ChannelZ | But if you only contact them on a single IP, it's hardly a big deal |
| 08:00.23 | *** join/#asterisk glam (~glam@58.252.251.247) |
| 08:00.58 | joobie | it's not.. i just dont like to put gay rules like that in my fw |
| 08:01.51 | ChannelZ | suit yoursef |
| 08:01.57 | ChannelZ | have fun not making calls |
| 08:02.48 | wdoekes2 | ~ |
| 08:03.58 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:08.14 | ChannelZ | ^ |
| 08:09.29 | joobie | ChannelZ, making calls fin |
| 08:09.30 | joobie | e |
| 08:09.40 | joobie | just opened up the firewall |
| 08:09.47 | joobie | to the insecure way that you're so keen to do |
| 08:14.03 | ChannelZ | OH NOES I HOAP PENNYTEL DO NOT HACKS YOU! |
| 08:16.06 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 08:17.54 | joobie | obviously your firewall management skills are yet to extend beyond your home network |
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| 08:18.59 | ChannelZ | chuckles |
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| 09:07.19 | k-man | joobie: evening |
| 09:14.34 | joobie | hey k-man |
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| 09:34.46 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
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| 10:02.35 | *** join/#asterisk guza (~guza@91.191.44.31) |
| 10:02.41 | guza | hello |
| 10:02.50 | guza | i was download asterisk now 32 version |
| 10:03.15 | k-man | sorry guza i don't understand what you mean |
| 10:03.17 | guza | after install it i was got error that say kernel missing |
| 10:03.54 | guza | asterisknow was not install kernel in /boot |
| 10:04.06 | guza | so grub can not found it and boot it |
| 10:04.09 | k-man | what was it you were donwloading? |
| 10:04.27 | k-man | that's a linux distro issue not an asterisk issue |
| 10:04.27 | guza | i wass downloading from asterisk.org |
| 10:04.35 | guza | yes it is |
| 10:04.48 | guza | but this is yours distro that u release it |
| 10:05.15 | k-man | wrong on both counts |
| 10:05.18 | guza | http://www.asterisk.org/downloads/asterisknow/i386/asterisknow32.iso this i was downloaded |
| 10:05.42 | k-man | this is a user support channel, not a channel run by digium as far as i know |
| 10:06.22 | guza | ok i know that |
| 10:06.49 | guza | but just to know distro on yours offical site dont work |
| 10:07.16 | guza | now i downloading 64 version |
| 10:07.49 | k-man | guza: on "the" official site not "your" official site |
| 10:09.06 | guza | :) |
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| 10:13.21 | jblack | guza: My suggestion is to find the asterisknow bug tracker and file a bug. |
| 10:13.51 | jblack | That seems like an awfully severe error that nobody's run into it before. Maybe you accidentally told apt to remove all the kenrels. |
| 10:14.58 | jblack | That would be like buying a new car only to find out it doesn't have an engine.. It just doesn't happen. You couldn't get even get it home, so to speak. =) |
| 10:15.19 | ChannelZ | points to #asterisknow |
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| 13:03.48 | *** join/#asterisk Hoi (~hoiung@client-80-3-100-202.bsh-bng-012.adsl.virginmedia.net) |
| 13:10.07 | Hoi | hi, does anyone know whats wrong with the asterisk.org forum? can't register and there doesn't seem to be much activity since May 18 2010! |
| 13:10.10 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 13:14.51 | *** join/#asterisk smooth_penguin (~smoove@59.95.21.3) |
| 13:16.30 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:17.17 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 13:29.43 | *** join/#asterisk Ole_ (ole@54b.pl) |
| 13:37.14 | tzafrir_laptop | Hoi, well at least you have IRC |
| 13:49.27 | *** join/#asterisk Wildy (~simba@83.149.9.23) |
| 13:57.54 | *** join/#asterisk residentgrey (~residentg@pool-71-191-49-141.washdc.fios.verizon.net) |
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| 14:08.36 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:11.48 | *** join/#asterisk jetlag (jetlag@pool-173-61-219-88.cmdnnj.east.verizon.net) |
| 14:16.29 | *** join/#asterisk jetlag (jetlag@pool-173-61-219-88.cmdnnj.east.verizon.net) |
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| 14:56.02 | *** join/#asterisk cusco (~trilili@2001:0:53aa:64c:3877:192a:2ac0:762d) |
| 14:56.05 | cusco | hi |
| 14:56.16 | cusco | what is wrong with: if (${Produto}=460 | ${Produto}=461) |
| 14:56.20 | cusco | in a AEL? |
| 14:56.31 | cusco | the space before and after | |
| 14:56.33 | cusco | ? |
| 14:56.43 | cusco | that is a Logical OR, right? |
| 14:57.01 | Trixboxer | try with || |
| 14:57.07 | Trixboxer | not sure though |
| 14:57.13 | cusco | thats not it |
| 14:57.20 | cusco | [Jun 6 13:55:54] WARNING[11268] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 14:57.21 | Trixboxer | ok.. I never done that |
| 14:57.38 | cusco | ah |
| 14:57.45 | cusco | ${Produto} is empty?! |
| 14:57.51 | cusco | err |
| 14:57.56 | domi | how can i make asterisk answer incoming calls directly to its IP without any SIP-Provider? |
| 14:58.29 | cusco | domi: who's making the call? |
| 14:58.30 | cusco | sip client? |
| 14:58.34 | domi | yes |
| 14:58.44 | cusco | that client is registered with asterisk? |
| 14:58.46 | domi | but the client should not register to my box |
| 14:58.52 | domi | foreign clients |
| 14:59.38 | cusco | not sure |
| 14:59.38 | domi | so that any IP-Phone can call my box with its IP |
| 14:59.46 | cusco | isn't that inseure? |
| 15:00.07 | cusco | it must be possible, I don't know how, tho |
| 15:00.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:00.21 | domi | why? this context can nat dial out, only for calling my internal phones |
| 15:00.26 | domi | s/nat/not/ |
| 15:00.36 | domi | lol nice bot ;) |
| 15:00.58 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:05.18 | domi | ah... will try this: http://lists.digium.com/pipermail/asterisk-users/2007-April/184131.html |
| 15:05.45 | *** join/#asterisk Benwa (~Benwa@dyn.83-228-160-169.dsl.vtx.ch) |
| 15:16.48 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:30.18 | *** part/#asterisk ruben23 (~unit41@119.93.217.243) |
| 15:37.29 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:44.24 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 15:50.20 | *** part/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 15:53.07 | *** join/#asterisk kallisti5 (~kallisti5@kallisti5-2-pt.tunnel.tserv9.chi1.ipv6.he.net) |
| 15:53.25 | kallisti5 | is there a way to detect an offline sip extension before dialing in extensions.conf? |
| 15:54.28 | kallisti5 | i was thinking GotoIf... but i have no clue on how to use it |
| 15:55.01 | p3nguin | If you mean an offline device, you can use ChanIsAvail(). |
| 15:56.44 | *** join/#asterisk onats (~onats@unaffiliated/onats) |
| 15:57.37 | *** join/#asterisk onats (~onats@unaffiliated/onats) |
| 15:57.39 | kallisti5 | p3nguin: thanks! that was just what i was looking for ! |
| 16:05.32 | domi | hmm i cannot connect to gtalk. get "aji_act_hook: JABBER: encryption failure. possible bad password." but the credentials are correct |
| 16:05.58 | domi | asterisk 1.6.2.6-1 on debian squeeze |
| 16:17.48 | *** join/#asterisk BANSAL (~bansal@117.199.127.95) |
| 16:32.48 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 16:43.18 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 16:45.18 | p3nguin | Does this romb thing ever do anything other than join, ping timeout, and change nicks? |
| 16:53.06 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 17:05.04 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 17:05.13 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
| 17:08.22 | *** join/#asterisk Alagar (~Administr@122.164.42.143) |
| 17:09.42 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 17:11.38 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 17:11.41 | [sr] | hi guys |
| 17:11.49 | [sr] | spent some time reading about BRI and PRI |
| 17:12.02 | [sr] | so i need to configure a BRI 2B+D |
| 17:12.11 | [sr] | where can I find some good docs about it? |
| 17:12.52 | WIMPy | [sr] What hardware? |
| 17:13.21 | [sr] | openvox, HFC4 |
| 17:13.35 | [sr] | i just realized that cant access the machine :( |
| 17:13.59 | [sr] | have to see why its not up tomorow, anyway that the card, or at least part of the description |
| 17:14.36 | [sr] | it has two ports |
| 17:15.02 | WIMPy | Do they supply drivers? |
| 17:15.13 | [sr] | nothing! |
| 17:15.14 | WIMPy | Only two ports? |
| 17:15.22 | [sr] | but its supported by dahdi, i remember that |
| 17:15.52 | [sr] | yes, two ports only |
| 17:16.07 | WIMPy | If you only need TE mode, that's fine. NT mode and dahdi doesn't seem to work with all cards. |
| 17:16.46 | [sr] | well, i'll have to use it with the NT.. |
| 17:16.50 | [sr] | just hope ir works :| |
| 17:16.55 | WIMPy | I'm using LCR now. All other options seem to be dead ends. |
| 17:17.19 | [sr] | what LCR stands for? |
| 17:17.33 | WIMPy | Linux Call Router |
| 17:18.44 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 17:18.45 | [sr] | going to investigate on the net... sec |
| 17:20.57 | [sr] | i see on docs here that the HFC-S (that the correct model i rememebr now) works in NT mode |
| 17:22.03 | WIMPy | They all do, but dahdi doesn't seem to do it with all cards. |
| 17:22.24 | [sr] | HFC-4S i mean |
| 17:22.30 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 17:22.45 | WIMPy | So that would be four ports. |
| 17:23.33 | [sr] | well yes, for connect it only has 2 fisical ports, |
| 17:23.39 | [sr] | thats why i called it 2 prots |
| 17:27.02 | [sr] | i think i may try this LCR |
| 17:27.12 | [sr] | does it has freepbx builtin also? |
| 17:27.33 | TJNII | Oh, physical ports. I read that as fiscal ports and was quite confused. |
| 17:27.48 | tzafrir_laptop | [sr], which specific HFC-4S card? |
| 17:27.55 | tzafrir_laptop | oh, the openvox one? |
| 17:28.10 | [sr] | tzafrir_laptop: yes |
| 17:28.31 | tzafrir_laptop | . How do can you tell that "NT mode is not supported"? How did you set NT mode? |
| 17:28.48 | [sr] | tzafrir_laptop: i'm just investigating for now |
| 17:29.00 | [sr] | doing some homework |
| 17:29.21 | tzafrir_laptop | AFAIK it should work. Though NT/PtMP will require asterisk trunk (NT PTP should work well) |
| 17:29.51 | tzafrir_laptop | What do you connect to it? |
| 17:30.22 | WIMPy | [sr]: LCR connects to Asterisk via chan_lcr. |
| 17:30.22 | [sr] | does asterisk 1.6.2.8 doesn't have that feature yet? |
| 17:30.34 | tzafrir_laptop | sadly, no |
| 17:31.46 | [sr] | ok asterisk trunk wont be a problem |
| 17:32.15 | [sr] | WIMPy: i see, i was checking and the 11MB size of the file couldn't be a distro!! |
| 17:33.07 | WIMPy | [sr]: No, it's an applications. Chose whatever distro, you like. |
| 17:35.25 | [sr] | im going to setup a box to compile asterisk from source |
| 17:35.35 | [sr] | right now i have a trixbox with a 4FXO port card |
| 17:35.41 | [sr] | that confighuration is easy |
| 17:35.45 | [sr] | and simple |
| 17:36.21 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:37.43 | [sr] | the 1st part i already cleared that, i have a BRI interface, 2B+D |
| 17:37.52 | [sr] | so i have to focus on that conf type |
| 17:37.59 | [sr] | with 10 MSN's |
| 17:40.12 | [sr] | ah one thing i didn't found |
| 17:40.20 | [sr] | what's the type/protocol |
| 17:40.31 | [sr] | for the normal central PBX |
| 17:40.38 | [sr] | for a isdn fone? |
| 17:41.01 | [sr] | i mean, if I want to connect an isdn fone to this card, what's the name of the configuration? |
| 17:42.00 | WIMPy | That's the NT mode. The mentioned NT-ptmp to be exact. |
| 17:42.04 | *** join/#asterisk Trixboxer (~Trixboxer@datacenter3.supportdepartment.net) |
| 17:42.45 | [sr] | hum wait, i'm a bit confused now |
| 17:42.47 | Trixboxer | Hi, Is there a way to append the chosen IVR number to recording file ? |
| 17:42.59 | Trixboxer | I mean User call in and presented with an IVR to press 1,2,3,etc |
| 17:43.22 | [sr] | 1st scenario: connect from NT to the card, NT=>CARD using BRI |
| 17:43.40 | [sr] | 1nd scenario, connect a isdn fone to the card, ISDN phone=>card |
| 17:43.49 | [sr] | that's the same configuration? |
| 17:44.00 | WIMPy | Nope |
| 17:44.48 | WIMPy | 1) requires TE mode, 2) NT mode |
| 17:45.34 | WIMPy | If you want to connect phones as well, I'd recommend LCR. |
| 17:46.46 | [sr] | nice |
| 17:46.49 | [sr] | going to read the docs! |
| 17:46.54 | [sr] | thank you for now!! |
| 17:48.16 | [sr] | ah wait |
| 17:48.23 | [sr] | for the card list i'm seeing, |
| 17:48.42 | [sr] | an ISDN modem (normaly used for internet connection in the past) can also work here, am i correct? |
| 17:50.16 | [sr] | misdn.org is down, normal? |
| 17:50.38 | WIMPy | LCR uses misdn2 from the kernel which at least supports anything HFC based. Not sure about other stuff. |
| 17:51.06 | cusco | hi... |
| 17:51.08 | [sr] | hum if it's already on the main kernel nice |
| 17:51.31 | cusco | im fiddling with peer to peer comunication with the canreinvite options |
| 17:51.41 | cusco | it works OK but... |
| 17:52.30 | cusco | there are 2 asterisk boxes. If one of them has a queue and then dials, the comunication will always stay between the asterisk queueing and the peer |
| 17:53.34 | cusco | asterisk1 -> dial -> asterisk2 -> queue, dial -> peer |
| 17:53.54 | cusco | comunication between peer and asterisk1 is not direct, instead goes from peer to asterisk2 |
| 17:54.17 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-126-104.mia.bellsouth.net) |
| 17:58.19 | [sr] | hey WIMPy, in which part of the kernel are the isdn required modules? |
| 18:01.13 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 18:01.52 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 18:02.49 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 18:08.57 | [sr] | WIMPy: found it |
| 18:13.14 | *** join/#asterisk telecos (~sergio@171.Red-81-43-200.staticIP.rima-tde.net) |
| 18:14.14 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 18:29.56 | *** join/#asterisk xEBIx (~ebi@188-194-83-64-dynip.superkabel.de) |
| 18:30.13 | xEBIx | hello |
| 18:30.57 | xEBIx | I'hve got a telefonie problem with my asterisk |
| 18:31.48 | xEBIx | ive some telephones on my local asterisk server, most of them are in the same net. One is connected from another net via NAT and the internet |
| 18:33.27 | xEBIx | only with that one ive some problems, a connections is opening but we can't here each other, but he can hear waitmusic when activate it, after i switch that off i can here him very short |
| 18:33.37 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:33.52 | xEBIx | what kind of problem might that be? |
| 18:33.57 | cusco | xEBIx: in the peer info set nat=yes |
| 18:34.03 | TJNII | ~sipnat |
| 18:34.03 | infobot | [~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 |
| 18:34.11 | xEBIx | cusco, yes thats is set |
| 18:34.53 | TJNII | xEBIx: Is the server behind NAT, or does it have a public, static IP? |
| 18:35.13 | xEBIx | the server is not NAted |
| 18:35.24 | xEBIx | only the peer telephone |
| 18:35.33 | TJNII | hmmm... And you have your IP ranges set properly in sip.conf? |
| 18:35.45 | TJNII | externip and localnet, iirc..... |
| 18:35.46 | xEBIx | the server has a static ip |
| 18:36.37 | xEBIx | aa that could be a problem, i didn't think of having telephone out of my net when i set it up |
| 18:36.50 | cusco | so the IP is listening on port 560 and the udp ports range... |
| 18:36.52 | xEBIx | thanks for that firstofall |
| 18:38.15 | xEBIx | yes it is listening on 0.0.0.0 and the contactpermit rule is commented out |
| 18:38.47 | xEBIx | i think you think about port 5060? |
| 18:38.56 | cusco | yes sorry |
| 18:40.30 | xEBIx | where do i have to set my IP of the server? |
| 18:41.34 | *** join/#asterisk Jaxyeh (~jax@c-69-250-52-161.hsd1.md.comcast.net) |
| 18:41.38 | cusco | sip.conf |
| 18:41.42 | cusco | externalip |
| 18:43.47 | xEBIx | cusco, is that the exact directive? |
| 18:44.18 | cusco | hold |
| 18:44.54 | cusco | externip=213.63.137.210 |
| 18:45.03 | cusco | localnet=192.168.2.0/255.255.255.0 |
| 18:45.14 | cusco | thats what I have |
| 18:47.06 | xEBIx | is a dns name for externip working also? |
| 18:47.47 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 18:47.57 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 18:49.49 | cusco | yes |
| 18:50.53 | xEBIx | i set that and relaoded with the command, nothing changed... |
| 18:52.58 | cusco | sip reload |
| 18:55.51 | WIMPy | [sr |
| 18:55.56 | WIMPy | ups |
| 18:56.13 | WIMPy | [sr]: drivers>isdn |
| 18:56.23 | xEBIx | hmm no change |
| 18:59.27 | xEBIx | do you have a working open stun server for me |
| 19:07.58 | xEBIx | now the echo test is working but, its no change with my telephones |
| 19:09.05 | xEBIx | if i switch music on hold on and off i can here him for a very short time... |
| 19:09.22 | xEBIx | any further idea? |
| 19:15.18 | TJNII | Is only one external phone not working? Can you try another phone from outside? |
| 19:16.03 | xEBIx | no i cant, at this time |
| 19:16.34 | xEBIx | can we try it? |
| 19:17.39 | TJNII | I'm sure you can. You don't have access to a network behind another public IP? |
| 19:18.02 | xEBIx | no i do nat have |
| 19:18.09 | xEBIx | s/a/o |
| 19:18.46 | TJNII | So your asterisk server is also your NAT gateway for your private network? |
| 19:18.55 | xEBIx | yes thats right |
| 19:20.25 | xEBIx | TJNII yes right |
| 19:21.38 | TJNII | So you obviously have a firewall on that box. Are you sure it is configured properly? Not blocking RTP traffic on the public port? |
| 19:22.00 | *** join/#asterisk Tim_Toady (~moi@193.92.244.148.dsl.dyn.forthnet.gr) |
| 19:22.10 | xEBIx | TJNII, no i do not have a firewall active on the public port |
| 19:22.46 | TJNII | Yes, you do. You have to have a firewall running to have NAT working. |
| 19:22.50 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 19:22.55 | TJNII | iptables on Linux. |
| 19:23.44 | xEBIx | TJNII, yes thats true, but no further rules and policies on ACCEPT |
| 19:24.32 | TJNII | Well, then I would try a softphone from another network to try and isolate the problem. |
| 19:24.46 | TJNII | Not the network the suspect phone is behind. |
| 19:25.09 | xEBIx | yes i would do that if i could |
| 19:27.15 | xEBIx | do you have a softphone? would you connect to me, for testing? |
| 19:28.15 | TJNII | I don't want to install a softphone and I don't have a microphone. |
| 19:28.52 | TJNII | Plus, my attitude is that you should do your own testing. No offense. |
| 19:29.01 | xEBIx | aah ok |
| 19:29.17 | xEBIx | so any other idea? |
| 19:29.48 | TJNII | You should be able to find another network to test from easily. Work, a friends house, a coffee shop, a WiFi hotcpot, etc. |
| 19:30.05 | *** join/#asterisk pabelanger (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 19:30.31 | xEBIx | ok thanks or your time |
| 19:32.25 | *** join/#asterisk x303 (~x303@187.159.121.70.cfl.res.rr.com) |
| 19:33.22 | TJNII | Haha, I just found a job listing on Craigslist that pays in chickens. |
| 19:37.07 | *** join/#asterisk m_tadeu (~quassel@173.191.19.95.dynamic.jazztel.es) |
| 19:40.50 | m_tadeu | hi all...I'm trying to use a agi php script but I'm getting a "broken pipe" error |
| 19:45.49 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 19:46.16 | ChannelZ | TJNII: Is it in Mexico? |
| 19:46.39 | ChannelZ | m_tadeu: the app is either not getting called right or is terminating in a not-so-nice way |
| 19:46.58 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 19:47.33 | p3nguin | I guess that PAP2T with the authentication problem is a pile of crap, because a softphone on that same LAN works just fine. |
| 19:47.47 | ChannelZ | I think the firmware on it was ancient |
| 19:47.54 | p3nguin | Even after a firmware upgrade, the problem still exists. |
| 19:48.02 | ChannelZ | really |
| 19:48.18 | p3nguin | It was at 3.1.something, now it is using 5.1.6. |
| 19:48.31 | p3nguin | User-Agent: Linksys/PAP2T-5.1.6(LS) |
| 19:48.37 | m_tadeu | ChannelZ: I'm sure it's getting called...gonna check the other situation |
| 19:48.41 | TJNII | ChannelZ: No here in Colorado. I'll let you make the obligatory immigrant jokes. |
| 19:48.42 | ChannelZ | I suppose it could be something in the config but I don't know what -- it just seemed to be totally ignoring the digest auth request |
| 19:49.12 | ChannelZ | wonders if chickens are taxable income |
| 19:49.16 | TJNII | "natural fram" or something similar. So hippies. |
| 19:49.28 | p3nguin | Twinkle works fine using the same peer information. Ekiga was working when he went direct to VoIP.ms, too. |
| 19:50.27 | bluOxigen | is now known as TheBird |
| 19:50.31 | *** join/#asterisk linkd (~switch@unaffiliated/linkd) |
| 19:51.01 | ChannelZ | Well plenty of other people seem to have it working so it has to be some config or another. |
| 19:51.10 | *** join/#asterisk wayne (~wayne@ool-ad03ce08.dyn.optonline.net) |
| 19:53.14 | b14ck | Yo. |
| 19:53.21 | p3nguin | If there was a known-good PAP2T that he could plug in and make work, that would prove that his ATA is borked. |
| 19:53.24 | b14ck | How often do the developers touch code in pbx/? |
| 19:53.36 | b14ck | I've been doing some code review, there are several things that could be updated there. |
| 19:53.44 | b14ck | I was wondering if that is on any sort of development schedule or not. |
| 19:54.32 | p3nguin | If you have a patch, you can submit it for inclusion. |
| 19:54.33 | ChannelZ | submit patches |
| 19:54.36 | b14ck | For example, in pbx_spool.c there are numerous hard-coded constants, and routines which do thinks like trim whitespace. They just seem to be copy+pasted around. But the include/asterisk/strings.h contains functions that should be used for that stuff :x |
| 19:55.11 | ChannelZ | probably better to take it up in #asterisk-dev |
| 19:57.12 | m_tadeu | ChannelZ: ok, no broken pipe anymore...but I'm outputing some garbage...shouldn't it show up in the console? |
| 20:01.03 | ChannelZ | outputting how? stdout is for sending commands to Asterisk, it doesn't wind up on the console |
| 20:01.21 | ChannelZ | you can turn on AGI debug and see what commands it's doing IIRC |
| 20:02.47 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 20:04.28 | *** part/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 20:07.00 | m_tadeu | ChannelZ: ah ok...I thought I would see some error or something |
| 20:08.41 | m_tadeu | it's working :) |
| 20:08.43 | ChannelZ | with debug off all you'll see is the script being called, and its exit code when it's done |
| 20:10.08 | m_tadeu | ChannelZ: Thanx a lot :) |
| 20:19.51 | *** join/#asterisk seanjohn (~shanebrig@ns2.sheltoncomputers.com) |
| 20:20.02 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:20.31 | seanjohn | I finally decided to use Cepstral and have compiled it manually to get swift() application. Anyone know how to slow this B@tch down? |
| 20:21.16 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 20:28.01 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 20:42.24 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:42.55 | [sr] | people, WIMPy, when compiling dahdi-linux from trunk, or last 2.3.0.1 i get this: /usr/local/src/asterisk/dahdi-linux/drivers/dahdi/voicebus/GpakCust.h:114: error: field 'sem' has incomplete type |
| 20:43.03 | [sr] | GCC 4.4 |
| 20:43.53 | Chainsaw | [sr]: Ah yes, you need to talk to tzafrir_laptop. |
| 20:43.58 | Chainsaw | [sr]: He said that bug doesn't exist. |
| 20:44.10 | [sr] | it does exist :P |
| 20:44.19 | [sr] | tzafrir_laptop: can you help? ;) |
| 20:44.30 | Chainsaw | [sr]: Or wait, I may have a bug open about that. There are about 3. |
| 20:44.32 | Chainsaw | [sr]: Sec. |
| 20:44.41 | WIMPy | [sr]: I have eliminated dahdi |
| 20:45.26 | Chainsaw | [sr]: Here you go: https://issues.asterisk.org/view.php?id=17382 |
| 20:45.43 | tzafrir_laptop | https://issues.asterisk.org/view.php?id=17382 ? (initial patch there by Chainsaw ) |
| 20:45.49 | Chainsaw | [sr]: I need to redo that patch as it breaks kernels below 2.6.26 |
| 20:45.51 | [sr] | WIMPy: only LCR? |
| 20:45.54 | Chainsaw | [sr]: But it should work for you. |
| 20:46.19 | WIMPy | [sr]: LCR with Asterisk |
| 20:47.48 | [sr] | ChanServ: tzafrir_laptop, hum a missing include |
| 20:47.53 | [sr] | works OK now |
| 20:48.27 | [sr] | WIMPy: I'll do that also, going to do my 1st test with dahdi |
| 20:49.37 | Chainsaw | tzafrir_laptop: https://issues.asterisk.org/view.php?id=17383 should be safe to apply now, we're at RC2. |
| 20:49.43 | m_tadeu | how can I make the agi_calleridname known? |
| 20:50.07 | Chainsaw | tzafrir_laptop: I need to wrap that include in an ifdef, which I know you dislike. But I can't think of a portable way that is compatible with <2.6.26 otherwise. |
| 20:50.29 | tzafrir_laptop | Chainsaw, I'll try to look at it tommorow |
| 20:50.39 | Chainsaw | tzafrir_laptop: Thanks, I'll do my best to have them both ready for you. |
| 20:51.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:54.00 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-056-202.pools.arcor-ip.net) |
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| 20:57.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:59.41 | [sr] | WIMPy: who maints LCR? doesn't compile :S |
| 21:02.34 | WIMPy | Jolly |
| 21:03.30 | ChannelZ | m_tadeu: what do you mean? |
| 21:03.40 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:04.56 | [sr] | ok emailed him |
| 21:05.11 | m_tadeu | ChannelZ: the vars that are passed inside an agi script...one of them is agi_calleridname, and it's value is "unknown". I was expecting the caller's username, or something |
| 21:05.45 | ChannelZ | well it only knows what Asterisk knows.. are you sure you're getting callerID over the channel in question? |
| 21:06.30 | ChannelZ | IE put a NoOp(${CALLERID(all)}) in your dialplan and see what you're getting |
| 21:08.29 | *** join/#asterisk boodu (~antoine@175.158.129.128) |
| 21:09.12 | boodu | hello |
| 21:10.00 | ChannelZ | ohell |
| 21:10.34 | m_tadeu | ChannelZ: I get the name of the peer I'm calling...not the caller |
| 21:12.58 | ChannelZ | pastebin the console output of a complete call |
| 21:13.04 | ChannelZ | core set verbose 4 |
| 21:13.25 | *** join/#asterisk MiserySoft (~LND@89.193.239.100) |
| 21:14.29 | m_tadeu | ChannelZ: http://pastebin.com/Y57gaevz |
| 21:16.02 | *** join/#asterisk gnude (~andre@muedsl-82-207-249-193.citykom.de) |
| 21:16.17 | ChannelZ | is s,2 the NoOp(${CALLERID(all)}) ? (right before the AGI) |
| 21:16.51 | *** join/#asterisk cesar_CR (~cesar@201.199.168.170) |
| 21:17.05 | ChannelZ | and where is this call coming from, just another local device? |
| 21:18.21 | m_tadeu | yes...right before the agi call, and also yes, local device, registered on sip.conf,I mean |
| 21:19.13 | ChannelZ | and does that device have a "callerid=Foo <123>" in sip.conf? |
| 21:20.51 | *** join/#asterisk Squeeb (~squirt@eggwee.co.uk) |
| 21:21.08 | m_tadeu | nop...gonna take care of that |
| 21:21.18 | Squeeb | Hi, Quick question about AEL: How can I assign the output of an application (System() in this case) to a variable. |
| 21:21.24 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 21:21.57 | ChannelZ | System() as an application might not give you app output |
| 21:22.05 | Squeeb | ah. |
| 21:22.54 | *** join/#asterisk Benwa (~Benwa@dyn.83-228-160-169.dsl.vtx.ch) |
| 21:22.55 | Squeeb | Hmm, it's executing a small bash script that spits out a number. |
| 21:23.02 | Squeeb | I need to take that value.. and pass it to another Application |
| 21:23.09 | ChannelZ | you'll porbably have to write an AGI in some other language to do what you need.. problem is a program could output all sorts of crazy things |
| 21:23.31 | Squeeb | Well I started with the bash script being called by AGI |
| 21:23.43 | Squeeb | but then I had the same problem, how do I pass the return of the AGI to another App? |
| 21:24.29 | ChannelZ | If it's an app in the dialplan, you'd just have to set a channel variable from the AGI |
| 21:25.01 | Squeeb | Aha.. of course. |
| 21:25.02 | Squeeb | Thanks :) |
| 21:25.35 | ChannelZ | sho thang |
| 21:25.58 | [sr] | brb |
| 21:26.15 | Squeeb | meh, I should stop trying to write AGI scripts in bash tbh :/ |
| 21:26.58 | m_tadeu | ChannelZ: same thing...didn't change...I should have something like "agent1" calling "agent_pbx" |
| 21:27.38 | *** part/#asterisk gnude (~andre@muedsl-82-207-249-193.citykom.de) |
| 21:28.15 | ChannelZ | does 'sip show peer xxx' show the Callerid you set for the peer? (xxx being whatever its name was.. agent1 or something based on what you just said maybe) |
| 21:29.35 | m_tadeu | yes |
| 21:29.54 | m_tadeu | the same string I set in sip.conf |
| 21:29.59 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 21:31.18 | ChannelZ | and that is the device you were calling _from_ and it's still showing up totally blank? |
| 21:33.26 | m_tadeu | yup...still showing "unknown" |
| 21:34.25 | boodu | I need help with sccp, my device can't register on the server. In the phone, I can found in DeviceConfiguration>UnifiedCMConfiguration "Unified CM 1 : unavail." but it's the right ip address. |
| 21:34.39 | m_tadeu | but I think the caller name should be in agi_calleridname, which I'm saying is unknown |
| 21:35.02 | boodu | if you have an idea |
| 21:35.04 | ChannelZ | hmmm no makey sense |
| 21:44.41 | *** part/#asterisk Ole_ (ole@54b.pl) |
| 21:44.44 | m_tadeu | I'm out of ideas...gotta read more |
| 21:45.41 | ChannelZ | well I dunno why it's being erased in the dialplan |
| 21:45.47 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 21:46.27 | *** part/#asterisk SAIDias (~SAID@97-125-143-112.desm.qwest.net) |
| 21:47.02 | *** join/#asterisk MiserySoft (~LND@89.193.48.145) |
| 21:47.12 | *** join/#asterisk Mango (~iMango@S0106001bfc77a834.vc.shawcable.net) |
| 21:47.18 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE000c41445085-CM001225409602.cpe.net.cable.rogers.com) |
| 21:47.24 | m_tadeu | maybe tomorrow something will popup :) thanx for everything |
| 21:47.30 | Mango | Word of advice. |
| 21:47.32 | Squeeb | *sigh* .. I wish Cisco had better documentation for their XML configs :( |
| 21:47.38 | Mango | Don't do exten => h,n,NoOp(${CHANNEL(peername)}) - it crashes Asterisk. |
| 21:47.40 | Mango | er |
| 21:47.47 | Mango | what I meant to paste was: exten => h,n,NoOp(${CHANNEL()}) |
| 21:48.11 | cdahmedeh | hello, i want to setup a simple voip getaway.. just want to access my home phone from the internet anywhere.. currently looking for the proper hardware... looking for the cheapest usb to rj11 adapter possible.. can anyone help ? |
| 21:48.45 | Mango | Sangoma has one though I'm not sure of the cost. http://www.sangoma.com/products/hardware_products/analog_telephony/usb_fxo.html |
| 21:48.54 | Mango | I don't believe it has hardware echo cancellation. |
| 21:49.08 | Chainsaw | That's the politically correct version of "this is going to suck". |
| 21:49.17 | cdahmedeh | oh ok |
| 21:49.28 | Mango | giggles at Chainsaw |
| 21:49.45 | cdahmedeh | like i'm planning to buy used or something |
| 21:50.48 | Chainsaw | cdahmedeh: It can be done cheap, but I'd recommend that you use a VoIP gateway. You can get those for quite reasonable prices. It'll do echo cancellation and... you won't have to leave a PC on. |
| 21:51.53 | cdahmedeh | like i just want to setup my own server.. and this is only for a couple of months |
| 21:52.14 | Mango | Chainsaw, what would you use for that? I've heard bad things about the SPA3102 |
| 21:52.40 | Chainsaw | Mango: I wouldn't use cheap kit anywhere, but my reliability concerns and budget are completely different. |
| 21:52.42 | Squeeb | cdahmedeh: If it's just to make calls over the internet for a month or so, why not buy a USB handset and use Skype? |
| 21:52.59 | cdahmedeh | i want to use the home phone line |
| 21:53.00 | Squeeb | £10 from InsertCheapVendorHere |
| 21:53.34 | Chainsaw | Mango: The prospect of having to log into something remotely, and using cheap USB hardware... I'm not going to lie to you. It fills me with dread. And nightmares of having to send people in to "fix" stuff. |
| 21:54.11 | Mango | Aye. |
| 21:54.29 | Squeeb | So let me get this straight, you want to call through your standard land line, from your computer? |
| 21:54.35 | Squeeb | I believe the term is "Modem" |
| 21:54.37 | Mango | I haven't heard anyone who claims they have a reliable, cheap way of doing that. |
| 21:55.10 | *** join/#asterisk Lantizia (~Lantizia@93-97-23-110.zone5.bethere.co.uk) |
| 21:55.36 | Lantizia | Cheap but good single or dual SIP to FXS adapters that arn't a PAP2T :) Hit me :P |
| 21:56.02 | Lantizia | I've seen Pattons range and they're stuuuupidly priced, any better ideas would be most appreciated and thanked :) |
| 21:56.04 | Squeeb | Lantizia: http://www.byfarthecheapest.com/products/Cisco-Small-Business-Linksys-VoIP-Adaptor-%252d-1-FXS,-1-FXO-Ports,-2-Ethernet-with-NAT.html ? |
| 21:56.31 | Chainsaw | Lantizia: Yes, Patton is telco-grade. It'll cost ya. |
| 21:56.32 | Lantizia | umm thats basically an SPA3102 |
| 21:56.34 | cdahmedeh | maybe something like that |
| 21:56.41 | Squeeb | Lantizia: pretty much |
| 21:56.49 | cdahmedeh | just very basic.. all i need is just plug in my phone line into the computer.. |
| 21:57.01 | cdahmedeh | so i can use the phone line via internet |
| 21:57.04 | cdahmedeh | via an sip client |
| 21:57.08 | cdahmedeh | just looking for the cheapest hardware |
| 21:57.34 | Lantizia | Chainsaw, well I'm considering Patton for BRI and PRI gateways... but thats only because they're well known and the cheapest. If they're telco-grade, who isn't telco-grade and doing BRI/PRI gateways :P |
| 21:57.34 | Squeeb | cdahmedeh: that link I posted is probably also what you're looking for |
| 21:58.02 | cdahmedeh | ok.. so it's called a pstn gateway ? |
| 21:58.18 | Squeeb | Plug your phone line into the FXO port, then connct to it over the internet (obviously after configuring your network) |
| 21:58.38 | cdahmedeh | is it possible to directly plug it into the computer ? |
| 21:58.41 | cdahmedeh | server i mean |
| 21:59.07 | ChannelZ | USB, ethernet, what difference does it make? |
| 21:59.10 | Squeeb | You don't need the server, but you can attach it to a PBX |
| 21:59.36 | Lantizia | I always wondered if old internal dial-up modems can be used for FXS/FXO SIP use |
| 21:59.45 | cdahmedeh | in fact.. can they ? |
| 22:00.03 | Squeeb | http://voip.weblogsinc.com/2005/07/14/use-a-v92-modem-as-an-fxo-card-on-asterisk/ |
| 22:00.06 | Squeeb | some can |
| 22:00.08 | Mango | cdahmedeh: You could consider a FXO card that you install in the computer. It'd be MUCH better quality than the USB ones, but not cheap. |
| 22:00.09 | Lantizia | They have a port lol, and you can generate noise down it |
| 22:01.12 | cdahmedeh | so for the ethernet one.. i connected to the network.. and have the server detect it ? |
| 22:01.20 | Squeeb | what? no |
| 22:01.30 | Squeeb | What's with this "Server" what's the "Server" doing? |
| 22:01.30 | cdahmedeh | so what do i do ? |
| 22:01.32 | cdahmedeh | ok |
| 22:01.37 | cdahmedeh | the server will be running asterisk |
| 22:01.45 | cdahmedeh | and the devices will connect to the server via sip |
| 22:01.57 | cdahmedeh | and i want the server to connect to the phone line |
| 22:02.18 | Squeeb | right, well that's one option, if you're server has PCI slots then I strongly suggest purchasing a dedicated FXO card. |
| 22:02.26 | Squeeb | which will provide physical PSTN sockets |
| 22:02.31 | Squeeb | that you can plug your land line in |
| 22:02.39 | Squeeb | and, to be quite honest, possibly the easiest to configure |
| 22:02.50 | cdahmedeh | hmm.. ok |
| 22:02.51 | ChannelZ | $2xx |
| 22:02.59 | cdahmedeh | can i get those cheap ? |
| 22:03.02 | Squeeb | meh, there's some Digium clones on eBay |
| 22:03.38 | Squeeb | http://cgi.ebay.co.uk/Authentic-X100P-SE-FXO-PCI-Digium-Asterisk-VoIP-PBX-/130319896399?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item1e57abcf4f#ht_3808wt_1139 |
| 22:03.42 | Squeeb | These are terrible |
| 22:03.44 | Squeeb | but cheap |
| 22:03.57 | cdahmedeh | terrible in what sense ? |
| 22:03.59 | drmessano | OpenVOX single port is better than that crap |
| 22:04.05 | Squeeb | drmessano: true |
| 22:04.09 | Squeeb | cdahmedeh: really bad echo cancellation |
| 22:04.10 | Squeeb | static |
| 22:04.10 | cdahmedeh | all i need is one port |
| 22:04.16 | Squeeb | People sounding the terminator |
| 22:04.17 | Squeeb | etc.. |
| 22:04.19 | drmessano | They're poor quality |
| 22:04.19 | cdahmedeh | ok |
| 22:04.23 | Lantizia | Why not just a standard 56k/V90 modem? |
| 22:04.28 | Squeeb | basically, ring Dell customer support.. it'll sound like that |
| 22:04.34 | Squeeb | Lantizia: depends on what modem |
| 22:04.36 | Mango | LOL |
| 22:04.37 | cdahmedeh | ok makes sense |
| 22:04.42 | Lantizia | Squeeb, why does it have to? |
| 22:04.47 | Lantizia | I mean I don't get it why someone hasn't made some software to use them |
| 22:05.02 | drmessano | Lantizia: Because they're crap hardware. |
| 22:05.05 | Squeeb | I don't know to be honest.. but it's only some that support it |
| 22:05.08 | Lantizia | Something in userland not alternative drivers |
| 22:05.28 | Squeeb | something along the lines of vgetty may have to, I don't know about latency though |
| 22:05.30 | drmessano | Lantizia: Someone made software for the MODEM chipset that is the X100P, and you see the end result. Crap |
| 22:05.40 | *** join/#asterisk Gos (~jhoekman@ip154-92-210-87.adsl2.static.versatel.nl) |
| 22:05.42 | Gos | #welkommentilnederlandersCSS |
| 22:05.48 | Squeeb | Sod off. |
| 22:05.52 | drmessano | Modems do not make decent voice cards.. Different design goal |
| 22:06.04 | Squeeb | That and they're designed for a completely different use :P |
| 22:06.15 | drmessano | ^^^ |
| 22:06.21 | Lantizia | drmessano, if they're accurate enough for modem use - surely voice quality should be fine |
| 22:06.22 | Squeeb | Use the right tool for the right job |
| 22:06.24 | Squeeb | is the golden motto |
| 22:06.29 | Gos | spam worlds best canned ham |
| 22:06.49 | drmessano | Lantizia: Negative.. That makes no sense. They're made for passing data.. Not transcoding audio |
| 22:06.55 | Squeeb | well.. |
| 22:07.01 | Gos | spam worlds best canned ham |
| 22:07.11 | Squeeb | *ACTUALLY*.. that's exactly what they're designed for .. just not in a vocal sense |
| 22:07.13 | Lantizia | drmessano, they don't need to transcode audio - software can do that it just needs to listen/send audio to the modem |
| 22:07.25 | Lantizia | Just like any "phone dialer" software can place calls with a dial-up modem |
| 22:07.27 | Squeeb | more in a 'working out ones and zeros' kind of way |
| 22:07.31 | Lantizia | and that call quality is fine |
| 22:07.49 | cdahmedeh | i'm planning to use some old laptop.. and has a dial-up modem.. if it helps that's fine |
| 22:07.53 | cdahmedeh | i'm not looking for perfect quality |
| 22:08.02 | drmessano | Lantizia: The call quality is not fine, which is what you run into here |
| 22:08.09 | Lantizia | I'm not sold on the whole, "the quality is bad" argument |
| 22:08.11 | Chainsaw | cdahmedeh: You're just looking to save some money with a soldering iron and some old modems? |
| 22:08.15 | Gos | spam worlds best canned ham |
| 22:08.23 | drmessano | Lantizia, go buy one and come back in a week |
| 22:08.36 | Lantizia | I've got plenty of old modems lying about |
| 22:08.52 | cdahmedeh | like if can find usb to rj11 to connect into the pc for cheap.. i'm all good |
| 22:08.59 | drmessano | Lantizia: So go for it.. make it happen. Prove us wrong.. We've only been doing this 24 hours too |
| 22:08.59 | cdahmedeh | can't echo cancellation be done with asterisk ? |
| 22:09.23 | drmessano | ~x100p |
| 22:09.31 | infobot | well, x100p is an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 22:09.31 | drmessano | ~x100 |
| 22:09.32 | drmessano | :( |
| 22:09.32 | Squeeb | cdahmedeh: not quickly |
| 22:09.57 | Lantizia | drmessano, I don't *need* it I'm just putting the case forward that it shouldn't be that hard/problematic. I mean why did Windows come with a "Phone Dialer" for all those years that could be used for placing calls using your internal modem! |
| 22:09.59 | Squeeb | certain codecs support it |
| 22:10.15 | Gos | spam worlds best canned ham |
| 22:10.15 | Squeeb | Lantizia: that was for precisely that, Dialing |
| 22:10.24 | Squeeb | you'd hook your handset up to the "Phone" port on the modem |
| 22:10.25 | Squeeb | and boom |
| 22:10.30 | Lantizia | but they supported headsets! |
| 22:10.33 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 22:10.35 | Squeeb | yes, the modem did |
| 22:10.37 | Squeeb | not the computer |
| 22:10.41 | Lantizia | ah as a passthrough? |
| 22:10.44 | Squeeb | exactly |
| 22:10.52 | Lantizia | hmm ok, see I never tried it :) |
| 22:11.03 | Squeeb | Handy when you had an address book |
| 22:11.07 | Lantizia | righty |
| 22:11.09 | Squeeb | click, dial, connect |
| 22:11.15 | Corydon76-dig | Additionally, many of the internal modems were half-duplex |
| 22:11.40 | Squeeb | But now we're not in the 90's :D So thankfully we don't have to deal with that stuff any more |
| 22:11.43 | Squeeb | yay |
| 22:11.47 | Chainsaw | Corydon76-dig: Would you mind taking care of Gos btw? |
| 22:11.51 | Squeeb | yea that too |
| 22:12.17 | Corydon76-dig | Chainsaw: no idea what you're talking about |
| 22:12.29 | Squeeb | Corydon76-dig: there's a spam bot in here called Gos |
| 22:12.54 | Corydon76-dig | Oh |
| 22:13.09 | cdahmedeh | ok.. so let's go back to my original call. .some old laptop converted to an old server.. i want to connect to the server a phone line.. and use that phone line via the internet |
| 22:13.22 | Squeeb | wait a second.. |
| 22:13.24 | Squeeb | "Laptop" ? |
| 22:13.28 | cdahmedeh | yes |
| 22:13.30 | cdahmedeh | no pci |
| 22:13.31 | *** mode/#asterisk [+b *!*@ip154-92-210-87.adsl2.static.versatel.nl] by Corydon76-dig |
| 22:13.31 | Gos | is is not bot i just like ht ham |
| 22:13.44 | Chainsaw | Corydon76-dig: Much appreciated. |
| 22:14.18 | cdahmedeh | so i'm gonna have to connect a ethernet to fxs (i think it's fxs) converter to the next |
| 22:14.20 | cdahmedeh | network* |
| 22:14.21 | cdahmedeh | right ? |
| 22:14.26 | Squeeb | FXO and yes. |
| 22:14.44 | cdahmedeh | so i'm looking for a basic fxo converter that connects to the network |
| 22:14.47 | cdahmedeh | just one port for the phone line |
| 22:14.47 | Squeeb | Laptop <--> VoIP / FXO convertor <----> PSTN |
| 22:14.54 | *** part/#asterisk Gos (~jhoekman@ip154-92-210-87.adsl2.static.versatel.nl) |
| 22:14.57 | cdahmedeh | yes ! |
| 22:15.02 | cdahmedeh | exactly |
| 22:15.14 | cdahmedeh | so all i need is one pstn/rj11 port, and one ethernet port for that thing |
| 22:15.14 | *** mode/#asterisk [-b *!*@ip154-92-210-87.adsl2.static.versatel.nl] by Corydon76-dig |
| 22:15.22 | cdahmedeh | and looking for a good cheap option for that hardware |
| 22:15.24 | Squeeb | http://www.voipuser.org/review_8.html |
| 22:15.26 | Squeeb | ^^ CHEAP |
| 22:15.29 | Squeeb | and very very shit quality |
| 22:15.55 | Squeeb | phone line in, ethernet in .. bish bash bosh |
| 22:16.11 | Squeeb | Set the thing up as a trunk in users.conf or sip.conf |
| 22:16.13 | Squeeb | and you're away |
| 22:16.17 | cdahmedeh | sounds good |
| 22:16.32 | cdahmedeh | the sipura 3000 is discontinued ? |
| 22:16.35 | Squeeb | ea |
| 22:16.37 | Squeeb | yea |
| 22:16.38 | Squeeb | hella old |
| 22:16.46 | Lantizia | Does anything even rival the Sipura/Linksys/CSB range of ATA's in terms of price and popularity? |
| 22:17.01 | cdahmedeh | that's why i can only find one on ebay |
| 22:17.01 | Lantizia | Something less... Linksys lol |
| 22:17.13 | Squeeb | cdahmedeh: here's an almost relevant HOWTO |
| 22:17.13 | Squeeb | http://www.freepbx.org/support/documentation/howtos/howto-linksys-spa-3102-sipura-spa-3000-freepbx |
| 22:17.55 | cdahmedeh | ok there we go |
| 22:17.55 | cdahmedeh | found the 3000 |
| 22:17.58 | cdahmedeh | cheap stuff |
| 22:18.10 | cdahmedeh | http://shop.ebay.ca/i.html?_nkw=SPA-3000&_sacat=0&_sop=2&_odkw=SPA-3102&_osacat=0&bkBtn=&_trksid=m270 |
| 22:18.12 | cdahmedeh | this is it ? |
| 22:19.27 | Lantizia | cdahmedeh, any old Sipura/Linksys/CSB (been rebranded alot) SPAxxxx/PAPxx type device with FXS will do fine :) |
| 22:19.29 | Squeeb | that's the newer one |
| 22:19.35 | cdahmedeh | perfect |
| 22:19.42 | Squeeb | wait no.. that's THE one |
| 22:19.45 | Squeeb | not the newone one |
| 22:19.48 | Squeeb | misread it |
| 22:19.54 | Squeeb | but yes, that will work |
| 22:19.59 | Lantizia | an old Sipura SPA3000 |
| 22:20.10 | Squeeb | tbh, if it's just for one connection at a time.. you probably don't even have to use asterisk :P |
| 22:20.27 | cdahmedeh | oh i can connect directly to it ? |
| 22:20.28 | Squeeb | but still, you can if you want to, which I think is the main point |
| 22:20.33 | Squeeb | yea, just point your sip client at it |
| 22:20.39 | cdahmedeh | oh wow |
| 22:20.40 | cdahmedeh | that's very nice |
| 22:21.10 | Squeeb | Not sure what they mean by "UNLOCKED" in the description though |
| 22:21.14 | Squeeb | *shrug* |
| 22:22.13 | TJNII | Squeeb: You're talking about an "UNLOCKED" ATA? |
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| 22:25.12 | cdahmedeh | ok.. i found some local store in canada that sells this type of stuff |
| 22:25.20 | cdahmedeh | would the spa-2102 work ? |
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| 22:27.14 | Mango | Does anyone here use a carrier in Canada called ISP Telecom? If so, have you had any termination problems over the past week? |
| 22:36.32 | cdahmedeh | ok.. so in the end.. i have decided for a Cisco SPA2102 |
| 22:36.37 | cdahmedeh | is that what i am looking for ? |
| 22:37.59 | Mango | cdahmedeh, that doesn't have an FXO port. You want a SPA3102. |
| 22:38.37 | cdahmedeh | what's the difference between fxo and fxs ? |
| 22:38.44 | Squeeb | google is your friend |
| 22:39.07 | jaytee | fxo is for POTS lines from the telco, FXS is for lines to phones |
| 22:39.10 | Chainsaw | cdahmedeh: FXS ports are for plugging phones into. FXO ports are for plugging into a phone network (like BT, KPN, etc). |
| 22:39.32 | cdahmedeh | oh ok |
| 22:40.23 | Squeeb | what you're looking for is an FXO gateway |
| 22:40.28 | cdahmedeh | http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=012331 |
| 22:40.30 | cdahmedeh | ok this like this |
| 22:40.30 | Squeeb | hense the SPA3xxx |
| 22:40.42 | Squeeb | yes |
| 22:40.55 | cdahmedeh | wow .. not too expensive |
| 22:41.09 | Squeeb | We have sort of being saying this for about an hour |
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