| 00:04.06 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
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| 00:31.42 | jsjc | I know i will need to integrate myself but spandsp does it come with res_fax or res_fax_digium? if not... where can I find that? |
| 00:32.39 | *** join/#asterisk manxpower (~ewieling@216.186.151.147) |
| 00:32.40 | jsjc | so that will be included in fax for asterisk? |
| 00:33.40 | *** join/#asterisk adam_g (~adam@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 00:34.33 | adam_g | hi, does asterisk do any kind of preservation of connection state or something similar by default? trying to re-route packets out a secondary ISP to SIP provider, but asterisk seems to not want to play nice with second network |
| 00:34.42 | jsjc | found spandsp! ;) |
| 00:35.16 | p3nguin | adam_g: You'll have to formulate your dialplan and SIP peers appropriately. |
| 00:35.52 | TJNII | Reroute packets? So you want to be able to switch the active interface mid-call? |
| 00:36.43 | TJNII | thinks this is a OS level routing problem, not a Asterisk problem. |
| 00:36.52 | *** join/#asterisk KavanS (~KavanS@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 00:36.54 | adam_g | not necessarily mid-call, even restarting asterisk would be fine |
| 00:36.56 | *** join/#asterisk Brian_H (~Brian_H@173-12-184-89-oregon.hfc.comcastbusiness.net) |
| 00:37.14 | TJNII | Why don't you expand on _exactly_ what you are trying to accomplish. |
| 00:38.11 | adam_g | asterisk server has a default gw of a local router, which then has two connections to the internet |
| 00:38.22 | carrar | adam_g, run BGP |
| 00:38.26 | vader-- | what would be the best way to stress test my asterisk box for phone calls |
| 00:38.41 | vader-- | i have a 23 channel PRI line and i want to try and get as many calls going |
| 00:38.44 | vader-- | to test for echo |
| 00:38.45 | vader-- | call drop |
| 00:38.47 | vader-- | etc |
| 00:39.14 | adam_g | we'd like to be able to simply re-route packets from the asterisk host thru secondary provider if needed |
| 00:39.31 | carrar | thats just you configuring your router |
| 00:39.36 | carrar | nothign to do with asteirsk |
| 00:39.46 | TJNII | That isn't a asterisk problem, save for the external IP changing. |
| 00:39.57 | adam_g | right, thing is.. when we switch routes at the router, all traffic except SIP works just fine |
| 00:40.10 | WIMPy | last time I tried two uplinks I ran into problems with Asterisk as well, but that's some time ago. |
| 00:40.19 | TJNII | Do you have externip set in your sip.conf? |
| 00:41.00 | adam_g | TJNII, nope |
| 00:41.45 | TJNII | Are you configured for outside sip clients to connect in? Does outbound calling work? |
| 00:42.20 | adam_g | the issue as far as i can see, is that when we do re-rooute out the second interface, tcp shows SIP traffic leaving thru the appropriate inteface but with a source ip of the first interface |
| 00:43.03 | carrar | I thought your router has the second interface? |
| 00:43.06 | jsjc | so spandsp does not have nothing to do with Digium's Fax for Asterisk? |
| 00:43.08 | carrar | not asterisk |
| 00:43.10 | WIMPy | adam_g: But Asterisk is not running on the router, is it? |
| 00:43.17 | adam_g | sorry, wording it porrly |
| 00:43.19 | adam_g | *poorly |
| 00:43.35 | adam_g | tcpdump'ing on the router |
| 00:43.57 | carrar | router is linux? |
| 00:43.58 | carrar | heh |
| 00:44.05 | adam_g | (on router) re-route all traffic thru second external interface.. asterisk's gateway stays the same (via router) |
| 00:44.11 | adam_g | yes, linux |
| 00:44.19 | carrar | thats probably you're problem |
| 00:44.22 | carrar | missconfig |
| 00:44.48 | TJNII | snaps his fingers |
| 00:44.59 | TJNII | How are you implementing the NAT gateway? |
| 00:45.00 | WIMPy | adam_g: looks like a more general problem. Are you usind SNAT or Masquerading? |
| 00:45.33 | adam_g | like i said, when things are re-routed, all traffic from asterisk host reroutes accordingly, works fine.. except SIP which is still leaving with the old external IP |
| 00:45.40 | adam_g | WIMPy, SNAT |
| 00:46.02 | WIMPy | adam_g: Then try flushing the routing cache. |
| 00:46.17 | WIMPy | But it might be something more fundamental. |
| 00:46.30 | carrar | or buy a real router |
| 00:46.43 | adam_g | WIMPy, tried, haven't had much luck |
| 00:47.50 | adam_g | WIMPy, `ip route get proxy.siprovider.com` reports correct, ping from asterisk host to sip provider look fine, except the sip traffic |
| 00:47.58 | *** part/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 00:48.00 | WIMPy | In that case, I'd suspect your setup. Multihomed can be a little tricky. Especially when policy routing is involved. |
| 00:48.38 | WIMPy | I'd suspect routing table selection, but this is not the time of day for me to think about this kind of stuff. |
| 00:50.16 | adam_g | :) |
| 00:52.32 | adam_g | ive tried lots of things with ip route2, fwmark, etc |
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| 00:59.31 | WIMPy | What do you use fwmarks for? I'd suggest to look at 'ip rule'. |
| 01:00.03 | vader-- | is there a way to see the callerID that is inside a sip connection coming in on the console? |
| 01:00.12 | vader-- | the caller id shows up fine on our ip phones |
| 01:00.23 | vader-- | but in the asterisk console all i see is the SIP connections |
| 01:00.25 | KavanS | vader--, yes, increase asterisk verbosity....should show in SIP headers |
| 01:00.54 | carrar | adam, I would use a cisco router running NAT with a policy based route that uses IP SLA to ping the next hop to set a valid default route |
| 01:01.09 | carrar | or bgp |
| 01:01.13 | vader-- | hmmm |
| 01:01.14 | vader-- | nope |
| 01:02.12 | vader-- | http://pastebin.com/T22N4ScC |
| 01:02.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:03.11 | KavanS | vader--, errr my bad...google around a bit for "sip debug" levels, that should do it |
| 01:03.20 | TJNII | vader--: You can always throw a noop in your dialplan, too. |
| 01:03.28 | vader-- | i have that |
| 01:03.33 | vader-- | but i dunno what variable to pull |
| 01:04.31 | vader-- | i got it |
| 01:04.37 | vader-- | i was just calling CALLERID |
| 01:04.47 | vader-- | im upgrading from 1.2 to 1.6 they removed that |
| 01:04.51 | vader-- | you have to specify it now |
| 01:05.13 | p3nguin | What are you trying to find out? |
| 01:05.25 | vader-- | i just wanted to see it in my log |
| 01:05.26 | p3nguin | Oh, nevermind. I see... |
| 01:05.30 | vader-- | in the console |
| 01:05.46 | p3nguin | Verbose(${CALLERID(all)}) |
| 01:06.14 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 01:06.16 | p3nguin | or num or name |
| 01:06.22 | vader-- | ya got it now |
| 01:06.48 | vader-- | what would be a decent way to test an asterisk setup? |
| 01:07.02 | vader-- | im going to try and get a bunch of people to call in and max out our PRI |
| 01:07.11 | p3nguin | You can use a verbose level to determine when it is displayed, too. Verbose(3,${CALLERID(all)}) would provide the detail at core set verbose 3 or more. |
| 01:07.39 | vader-- | any thoughts? |
| 01:08.07 | KavanS | vader--, I've seen slides on people scripting such a scenario...but i've not done it myself |
| 01:08.35 | carrar | vader, SIPp? |
| 01:08.52 | TJNII | List your DID as a free sex line number on 4chan. |
| 01:08.57 | vader-- | haha |
| 01:08.58 | p3nguin | heh |
| 01:09.02 | KavanS | lol TJNII |
| 01:09.02 | TJNII | That might have long term reprocussions, though. |
| 01:09.06 | vader-- | i was going to create a DID |
| 01:09.09 | vader-- | for people to call in |
| 01:09.17 | KavanS | correction: List your DID as a *underage* sex line number on 4chan. |
| 01:09.23 | vader-- | but im not sure what to put on the asterisk system that will be a good reresentation |
| 01:09.24 | KavanS | then 4chan would be going crazy... |
| 01:09.25 | vader-- | rep |
| 01:09.26 | TJNII | ZING! |
| 01:09.30 | KavanS | oh snap! |
| 01:09.47 | vader-- | because in the real world it would be PRI to Asterisk to IP Phone |
| 01:10.00 | vader-- | actually |
| 01:10.11 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
| 01:10.17 | vader-- | PRI <--SIP--> Asterisk <--SIP--> IP Phone |
| 01:10.34 | vader-- | if i just terminate at the asterisk box it won't be generating as many sip connections |
| 01:10.43 | p3nguin | If you are using PRI, do you really think there is SIP between it and Asterisk? |
| 01:10.50 | vader-- | just PRI <--SIP--> Asterisk |
| 01:10.51 | vader-- | yes |
| 01:11.04 | vader-- | ;-) |
| 01:11.05 | p3nguin | More like PRI <-> Asterisk |
| 01:11.07 | vader-- | nope |
| 01:11.09 | vader-- | not in my setup |
| 01:11.16 | TJNII | You could set up another * box and relay the call a couple dozen times. |
| 01:11.18 | p3nguin | Why would there be SIP between PRI and Asterisk? |
| 01:11.30 | carrar | mediagateway |
| 01:11.31 | vader-- | my PRI line terminates in an Adtran 924e |
| 01:11.35 | vader-- | carrar bingo |
| 01:11.36 | p3nguin | d'oh |
| 01:11.43 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 01:11.50 | vader-- | the unit then presents it as a sip connection to asterisk |
| 01:12.05 | p3nguin | Sure. You left out that piece of hte puzzle, though. |
| 01:12.10 | TJNII | I had a dialplan error allow one bad call to open a couple hundred SIP connections between two * boxes once. Was fun. |
| 01:13.33 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 01:20.14 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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| 01:31.17 | vader-- | so any thoughts on what i could do to test asterisk? |
| 01:32.01 | Maliuta | order 3000 pizza's from dominos in Iraq? |
| 01:32.16 | vader-- | i mean what i could program into the dialplan |
| 01:32.17 | Maliuta | ask it hard questions about quantum physics? ;P |
| 01:32.33 | vader-- | so once people connect it could give me a good representation of whats going on |
| 01:32.44 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 01:32.45 | Maliuta | <PROTECTED> |
| 01:41.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:42.49 | jsjc | is it opossible to dial a few extensions at same time? |
| 01:42.56 | vader-- | yes |
| 01:42.57 | jsjc | for example passsing to the first available agent... |
| 01:43.09 | jsjc | how can I perform taht in my dialplan? |
| 01:43.45 | vader-- | exten => 1072,2,Macro(stdexten,157,SIP/001759E5591E-02&SIP/001759E55612-02) |
| 01:44.01 | vader-- | just put a & between the dialing things |
| 01:44.48 | jsjc | so could I do exten => 1000,2,Dial(DAHDI/1&DAHDI/2&DAHDI3) |
| 01:44.57 | vader-- | yes should work |
| 01:45.01 | jsjc | and then the phones connected in dahdi 1 2 and 3 will be ringing |
| 01:45.28 | vader-- | i grabbed my example from the wrong config |
| 01:45.30 | vader-- | but that will work |
| 01:45.51 | jsjc | ok |
| 01:46.08 | jsjc | because now i am starting to get confused when it comes to simple stuff |
| 01:46.24 | jsjc | where can I find dialplan examples so I can learn a bit more? |
| 01:46.33 | vader-- | voip-info.org? |
| 01:46.46 | vader-- | thats usually where i find all my stuff |
| 01:47.44 | jsjc | ok going there! thanks |
| 01:49.48 | jsjc | what is the difference between exten => _1,1 and exten => 1,1 |
| 01:51.42 | TJNII | ~book |
| 01:51.43 | infobot | [~book] 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://astbook.asteriskdocs.org or see ~buybook |
| 01:53.33 | TJNII | jsjc: The _ means pattern match. So exten => _1,1,app() shoule be the same as exten => 1,1,app() |
| 01:53.42 | *** join/#asterisk OrNix (~ornix@178.49.0.149) |
| 01:54.02 | vader-- | can you loop the playback of a sound file? |
| 01:56.05 | TJNII | jsjc: Do you understand why? |
| 01:56.08 | jsjc | i thought that because in book just appears in the matching patterns |
| 01:56.17 | jsjc | but if I put _1X then is a different story |
| 01:56.20 | TJNII | Exactly |
| 01:56.26 | *** join/#asterisk snot (~snot@unaffiliated/snot) |
| 01:56.31 | jsjc | it is anything that matches 1 and any other digit there |
| 01:56.40 | jsjc | two digit starting by one |
| 01:56.44 | TJNII | exten => 1X,1,app() is not the same as exten => _1X,1,app() |
| 01:56.47 | jsjc | but extension 1X will be extension 1X |
| 01:57.04 | TJNII | Correct |
| 01:57.22 | jsjc | good at least i am getting somethings clearer... |
| 01:58.00 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 01:58.19 | TJNII | Extensions don't have to be numbers. I have some alpha extensions in my dialplan that are used as goto targets from other locations. |
| 01:58.21 | jsjc | i dont knwo how it is this called in telephony but i know there is... in hotels when teh cleaners finish clenaing rooms they dial in the phones to change the status of the roomo |
| 01:58.39 | jsjc | i was thinking to use something like this to check in staff at work |
| 01:58.47 | TJNII | You can do that with AGI |
| 01:58.55 | TJNII | I have an AGI that controls my stereo. |
| 01:59.39 | jsjc | hehe |
| 02:00.32 | TJNII | One of my friends had an AGI that was linked to an online calendar. The "Hootenany Hotline." You'd call in and it would tell you when and where the next party was. |
| 02:00.40 | TJNII | I was disappointed when he let that lapse. |
| 02:02.31 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:03.22 | vader-- | hehe |
| 02:03.30 | vader-- | i need something to test this :-/ |
| 02:14.21 | jsjc | asterisk can do some fancy shit... |
| 02:14.30 | jsjc | but before fancy need to get it just working... |
| 02:14.49 | jsjc | if I use Background() can I dial an extension while background is playing? |
| 02:15.13 | manxpower | yes. also look at Read() |
| 02:19.21 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:19.27 | jsjc | haha what is the use of what-are-you-wearing!? |
| 02:19.29 | jsjc | that is funny... |
| 02:28.05 | ChannelZ | There are lots of random gems |
| 02:36.53 | jsjc | hehehe |
| 02:40.41 | ChannelZ | screaming monkeys |
| 02:53.10 | *** part/#asterisk djMax (~chatzilla@66.92.91.132) |
| 02:57.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 02:57.12 | vader-- | hmm |
| 02:57.20 | vader-- | why isn't my realtime voicemail working :-/ |
| 02:57.24 | vader-- | im gettting |
| 02:57.25 | vader-- | [Apr 8 22:48:08] WARNING[6666]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk (check res_mysql.conf) |
| 02:57.44 | vader-- | but if i run realtime mysql status i get general connected to asterisk@localhost, port 3306 with username root for 0 seconds |
| 02:57.45 | Corydon76-dig | vader--: do you have [asterisk] in res_mysql.conf? |
| 02:58.00 | Corydon76-dig | Or do you have [general]? |
| 02:58.20 | jsjc | I have configured defaultlanguage to es but sounds still sounding in en |
| 02:58.25 | vader-- | general |
| 02:58.29 | jsjc | why coudl be this issue? |
| 02:58.39 | Corydon76-dig | vader--: then that's your DB name you should be using |
| 03:00.07 | vader-- | here is what my extconfig.conf line looks like |
| 03:00.07 | vader-- | voicemail => mysql,asterisk,voicemail_users |
| 03:00.13 | vader-- | my database is called asterisk |
| 03:00.19 | vader-- | my table is called voicemail_users |
| 03:01.14 | vader-- | the context in my voicemail file is general |
| 03:01.19 | Corydon76-dig | vader--: According to your configs, your dbname is general |
| 03:01.24 | vader-- | and the context in res_mysql.conf is general |
| 03:01.41 | vader-- | in res_mysql.conf it's dbname = asterisk |
| 03:01.44 | Corydon76-dig | but as extconfig is looking for [asterisk], it's not finding it |
| 03:02.06 | Corydon76-dig | vader--: nope, extconfig is looking for a context name in res_mysql.conf |
| 03:02.19 | Corydon76-dig | It ain't finding it |
| 03:03.14 | Corydon76-dig | either change extconfig to point to general or change res_mysql.conf to say [asterisk] |
| 03:09.12 | vader-- | weird |
| 03:09.15 | vader-- | that worked |
| 03:09.16 | vader-- | thanks |
| 03:09.29 | vader-- | hmm now i wonder why it isn't emailing the voicemails |
| 03:09.32 | vader-- | :-/ |
| 03:12.51 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 03:15.31 | vader-- | could be no mail server running too |
| 03:15.32 | vader-- | hehe |
| 03:15.52 | vader-- | hmm sendmail is running |
| 03:15.54 | vader-- | :-/ |
| 03:36.40 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 03:50.08 | *** join/#asterisk tvaughn (tyler@admin.of.tylivia.com) |
| 03:52.49 | jsjc | I have 3 dahdi channels for three users but I want just to have one common voicemail it is viable? |
| 03:53.12 | jsjc | I do not want every person to have a voicemail it is pointless in our setup. |
| 03:53.42 | jsjc | so I am trying to have all extensions ringing at same time and send to voicemail if none answered |
| 03:53.50 | jsjc | but cannot get it to send to voicemail. |
| 03:54.07 | ChannelZ | why not |
| 03:54.35 | ChannelZ | Dial(DAHDI/1&DAHDI/2&DAHDI/3,20) then VoiceMail(1) or whatever |
| 03:55.17 | jsjc | exten => s,n,Dial(DAHDI/1&DAHDI/2&DAHDI/3,20) |
| 03:55.18 | jsjc | exten => s,n,VoiceMail(100,u) |
| 03:55.20 | jsjc | like that? |
| 03:55.40 | jsjc | mhnm lets test again something might not be working... |
| 03:58.06 | ChannelZ | one ringy-dingy |
| 03:58.12 | ChannelZ | two ringy-dingies |
| 03:59.22 | jsjc | hehe |
| 03:59.34 | jsjc | ohh ohhh yeap.. was something stupid... |
| 04:06.01 | patrb-afk | So im still a bit green as an asterisk admin...ive never needed to connect/disconnect our PRI's. Assuming the server is already configured for the PRI, will I need to run any scripts/restart any services for the connection to initiate? |
| 04:06.51 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
| 04:09.19 | *** join/#asterisk tvaughn (tyler@66.55.71.171) |
| 04:10.03 | jsjc | why when I call to a sip account that asterisk is registered to (asterik is client of that sip) then asterisk sends it to the context I have set on sip.conf but as well starts ringing in one of my asterisk sip clients |
| 04:10.21 | patrb | I should mention that is a sangoma card running w/ the wanpipe/wanrouter software |
| 04:10.32 | jsjc | I think i have not said anything anywhere so the asterisk client gets the ring.... |
| 04:11.25 | jsjc | oh no no |
| 04:11.31 | jsjc | is this sip client has both sign up hehe |
| 04:11.33 | jsjc | what a nightmare! |
| 04:11.40 | jsjc | i was going nuts... |
| 04:16.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:22.14 | jsjc | I have a confused asterisk... hehehe I have two folders with sounds /var/lib/asterisk and /usr/share/asterisk ... |
| 04:22.42 | jsjc | at the moment it is not reading es from /var/lib/asterisk but en from /usr even I have es mentioned everywhere. |
| 04:23.17 | jsjc | in asterisk.conf I have mentioned astdatadir => /var/lib/asterisk |
| 04:23.22 | jsjc | isnt good enough? |
| 04:24.33 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
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| 04:34.21 | jsjc | root@localhost:/etc/asterisk# asterisk -U asterisk -G asterisk -C asterisk.conf |
| 04:34.22 | jsjc | Unable to open specified master config file 'asterisk.conf', using built-in defaults |
| 04:34.24 | jsjc | mhnm..... |
| 04:34.28 | jsjc | might have something to do... |
| 04:34.35 | jsjc | always trying to load the built in... |
| 04:37.45 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 04:41.08 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 04:41.46 | joobie | hey guys.. got a few polycom 320's.. just wondering if there's something decent to use (preferably hardware based) that can allow a supervisor / trainee to listen in on a converstaion on the phone itself? |
| 04:42.05 | joobie | polycom's have a 2.5mm jack.. but doesnt seem to run concurrently with the headset.. it's either or |
| 04:42.27 | joobie | was thinking about getting a 2.5mm splitter and plugging in an extra set of headphones - but just not sure if there's a more elegant solution |
| 04:44.56 | patrb | ok, ive got the PRI's coming into my 1.6 box...testing my dial plan im getting the following error:http://pastebin.com/bkwg2ZEh I think its probably just syntax. Any suggestions? |
| 04:45.07 | patrb | 1.6.2.6 |
| 04:46.43 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-blvoqltskhiyyccc) |
| 04:51.27 | ChannelZ | joobie: well if you're talking about * and the media stream is running through it, you can ChanSpy from another phone |
| 04:53.58 | ChannelZ | patrb: yeah show us the actual dialplan |
| 04:55.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 04:58.20 | joobie | ChannelZ, i'm talking about * but i'm trying to go with a hardware based solution for other reasons |
| 05:03.10 | jsjc | I am trying to change the bloody default language but no chance.. this asterisk it is reading shit from somewhere.... |
| 05:13.19 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
| 05:34.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 05:45.43 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 05:50.44 | sawgood | Hey! |
| 05:57.53 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 06:03.00 | *** join/#asterisk Nombrandue (~Satan@ip174-71-68-157.om.om.cox.net) |
| 06:05.43 | ChannelZ | HEY! If you were a hotdog, and were starving, would you eat yourself? |
| 06:06.10 | Nombrandue | Does anyone have any idea's how to get Asterisk to see a Sipura SPA-3201 ATA as a trunk device? |
| 06:08.31 | ChannelZ | Do you mean 3102? |
| 06:08.49 | Nombrandue | yes sorry |
| 06:09.03 | ChannelZ | And what do you mean bu 'trunk device' |
| 06:10.01 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:10.15 | Nombrandue | to show up as a trunk in, say the Asterisk-GUI, or some other flavor of GUI for Asterisk. Namely to replicate what is seen with having a provider such as Broadvoice, or having an analoge card in place |
| 06:10.34 | ChannelZ | http://forums.digium.com/viewtopic.php?t=19261 |
| 06:11.56 | Nombrandue | that is a lot like how I have it set up now (Background on this, I am migrating from Callweaver to Asterisk, due to feature sets) |
| 06:12.50 | Nombrandue | my issue is I am trying to re-write my dial plan in a way I can use the other tools, like freepbx, or asterisk-gui, and change the configs there, instead of by hand every time I am working on a new line. |
| 06:12.53 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 06:13.20 | *** join/#asterisk MaliutaLap (~biteme@kiev.lusan.id.au) |
| 06:13.30 | ChannelZ | One you configure the 3102 it registers like any other device |
| 06:14.18 | kaldemar | Nombrandue: if you want to use a GUI, then use a GUI. don't modify the configs by hand. if you're asking what kind of configs a GUI makes, you'll be better off asking in a GUI channel. |
| 06:15.23 | Nombrandue | kaldemar: That is one thing I will be working on over the weekend, is pestering people in the GUI channels about that. If I go that route, I will need to convert my existing dial plan to something I can import, and I have, so far, found much that would do that |
| 06:16.13 | ChannelZ | These GUIs cause more headaches than they solve |
| 06:16.39 | ChannelZ | But I'm really confused because your question seems to have absolutely nothing to do with the SPA3102 |
| 06:17.05 | Nombrandue | ChannelZ : just a seperate part of my total migration problem |
| 06:17.51 | kaldemar | asterisk itself really does not separate "trunk" devices and other devices. they're all pretty much the same. |
| 06:18.17 | Nombrandue | the SPA3102 works, perfectly, with my old setup using Callweaver. I stop Callweaver, and start Asterisk, with a like setup (configs copied over and modified for the new syntaxes) and everything works fine, for about an hour. Then no inbound calls ever register, only outbound work |
| 06:19.33 | Nombrandue | If I restart Asterisk, it works again, for a while, then no more calls come in from the SPA. But I do see the RTP traffic start for a moment, but nothing happens across the dial plan, and the line just keeps ringing with no pickup |
| 06:19.36 | Polysics | i have realtime queues defined. how can i force a reload without restarting * when i change something? |
| 06:19.42 | Polysics | so far only restart fixed it |
| 06:20.04 | Nombrandue | I am 80% sure it is something to do with SIP, SIP registration, and timeouts |
| 06:20.21 | ChannelZ | module reload app_queue ? |
| 06:21.03 | ChannelZ | Nombrandue: probably. After an hour does 'sip show peers' show the SPA with no IP and/or unreachable? |
| 06:22.05 | Nombrandue | I am not sure, I will check that though. If that is the case, I would think I need to adjust that on Asterisk. I am pretty damned sure so much has changed on the SIP side with Asterisk from what the CW was, that is part of my problem |
| 06:22.55 | ChannelZ | It depends. |
| 06:25.21 | ChannelZ | There's half a dozen different things it could be |
| 06:26.23 | Nombrandue | in a nutshell, I am moving from Asterisk 1.2 to 1.6.1, configuration wise |
| 06:26.27 | ChannelZ | Is the SPA local to the * box? |
| 06:26.41 | ChannelZ | Does it really have a dynamic IP address? |
| 06:26.43 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.84) |
| 06:27.19 | Nombrandue | no, the SPA is on the network, static IP's on both the SPA and the * box, and they are connected at 100MB/s on a gig switch |
| 06:28.04 | kaldemar | Nombrandue: there are quite a few changes between 1.2 and 1.6.1. read UPGRADE*.txt in an 1.6.1 source package to see the configuration syntax changes. |
| 06:28.06 | ChannelZ | Then why not just config the IP of the SPA in sip.conf and forget registering? |
| 06:28.45 | Nombrandue | I have the SPA set up as a user in sip.conf, for inbound and outbound, right now |
| 06:29.58 | ChannelZ | What I'm saying is that if they're both on the same network, there's no NAT between them, and they both have static IPs, there's no real benefit in setting it up as a dynamic user in sip.conf and making the SPA register. |
| 06:31.38 | ChannelZ | Although you said outbound works, inbound doesn't.. which would imply * knows how to read the SPA but the SPA is not knowing how to reach * |
| 06:32.45 | ChannelZ | which doesn't make much sense but either way I'm not going to sit here guessing every possible scenario with no real information to go on |
| 06:33.06 | Nombrandue | inbound works, it just times out after a while |
| 06:33.56 | Nombrandue | when I am running callweaver, inbound and outbound work, all the time, with the SPA in the sip.conf file, as the inbound extension and the outbound device (as outlined in the documentation you posted earlier) |
| 06:34.03 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 06:35.15 | ChannelZ | great. This isn't #callweaver and we're not clairvoyant enough to see your configs and console output |
| 06:35.41 | Nombrandue | is there a pastebin I can use, if that would help? |
| 06:35.52 | ChannelZ | ~pb |
| 06:35.53 | 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. |
| 06:39.04 | Nombrandue | the SPA config that works is: http://asterisk.pastey.net/135060 |
| 06:39.24 | Nombrandue | I am working on getting the other ones up as well. |
| 06:42.46 | kaldemar | Nombrandue: monitor is not a valid parameter, and insecure does not take "very" as a value. |
| 06:42.58 | kaldemar | in asterisk, that is. |
| 06:43.45 | ChannelZ | I think that's from 1.2 |
| 06:44.09 | kaldemar | "very" was valid until 1.4. |
| 06:44.13 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 06:44.48 | kaldemar | but there's not monitor parameter in 1.2 either. |
| 06:44.49 | ChannelZ | I meant that sip.conf - he said it was the "config that works" which I am assuming is this old setup |
| 06:45.27 | Nombrandue | Callweaver was a branch off 1.2, which is what that is from. Yes, what I am getting right now is from the old setup |
| 06:45.49 | ChannelZ | I'd never heard of it but never ran 1.2. But I"m waiting for the config that _doesn't_ work since it's more useful fix a problem than fix a not-problem |
| 06:46.28 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 06:46.32 | ChannelZ | but I"m thinking I'd really rather go play video games. |
| 06:47.31 | ChannelZ | I do see 'qualify' is turned off, perhaps the SPA is getting bored and going to sleep. |
| 06:48.59 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-172-249.clienti.tiscali.it) |
| 06:49.30 | Nombrandue | that is possible |
| 06:50.20 | Nombrandue | I will work on the config and remove the old stuff out, and finish transcribing it, instead of trying to limp along until I got more time to do it completely. Thanks for the help so far |
| 06:53.50 | sawgood | I have an Asterisk box 1.6.2.6 which I added dhcpd to (so my phones could grab and IP address from the IP PBX) ... This part is working, but I do not think the phones can 'reach' the Internet (there is two NICS in this box) (one NIC has a static live public IP ... and the 2nd NIC is a 172.16.x.x NIC issuing DHCP) |
| 06:54.01 | ChannelZ | you say it happens after an hour, that's the default registration expire in the 3102 |
| 06:54.04 | sawgood | Is there anyway to 'know' if the phones can reach the Internet? |
| 06:54.23 | ChannelZ | sawgood: like, testing... |
| 06:54.39 | sawgood | I am trying to have the phones do f/w updates from Snom, but they fail ... I think there is no route out (I do not have a SIP trunk setup on the box yet) |
| 06:55.01 | sawgood | I have the correct statements in sip_nat.conf ... |
| 06:55.11 | sawgood | But I think the IP PBX is not a 'router' ... |
| 06:55.24 | ChannelZ | well that's up to how you have your server setup |
| 06:55.35 | sawgood | do you mean dhcpd.conf by chance? |
| 06:56.13 | ChannelZ | no, the thing as a whole. Are the phones on the same LAN segment as *? Is there a router on that segment connected to the net? Is it setup to NAT local traffic out, etc etc etc |
| 06:56.50 | sawgood | The * box has two NICs (one NIC has a static IP out to the Internet ) with no router/firewall in front of it. |
| 06:57.07 | sawgood | The * box can reach the Internet just fine, and I can SSH into the * box from another Internet host |
| 06:57.30 | ChannelZ | What is the other NIC, LAN? Private address space? |
| 06:57.37 | sawgood | I 'think' when the phones (getting 172.16.50.x IP addresses from DHCP) try to go out to the Internet ... there is no route |
| 06:57.59 | sawgood | 2nd NIC is 172.16.50.100 (DHCP from .10 to .25) |
| 06:58.50 | sawgood | I have a default gw statement in route for the public NIC (eth0) ... and it is working ... but (eth1) is static 172.16.50.x |
| 06:58.52 | ChannelZ | So the server is the router. And it sounds like you have no NAT setup |
| 06:59.17 | kaldemar | sawgood: your dhcpd needs to send an "option routers" to the clients. |
| 06:59.21 | ChannelZ | so the phones send traffic to the server which has no idea what to do with it and drops it on the floor |
| 06:59.25 | sawgood | ChannelZ: I think you are right ... I probably need NAT setup (I thought doing SIP setting NAT=yes) would work for this |
| 06:59.37 | ChannelZ | no, it has nothing to do with the phones |
| 06:59.56 | sawgood | oh ... the option routers should NOT be 172.16.50.100, but rather the LIVE static eth0 NIC IP address |
| 07:00.00 | sawgood | let me try that |
| 07:00.03 | ChannelZ | well.. it COULD have something to do with the phones but you have other problems |
| 07:00.45 | sawgood | my option routers statement = 172.16.50.100 .... not the WAN NIC for eth0 (maybe that is the concern) |
| 07:00.49 | ChannelZ | That won't get you much further. If you're not running NAT the traffic has nowhere to come back to |
| 07:01.49 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.244.174) |
| 07:03.28 | ChannelZ | iptables --table nat -A POSTROUTING --out-interface eth0 --jump SNAT --to the.real.ip |
| 07:03.31 | kaldemar | sawgood: it needs to be the router of your network, whatever that is. |
| 07:04.26 | sawgood | Do you mean the 'default gateway' ... there is no router .... just the phone, the IP PBX, and straight out to to the Internet via a Comcast cable gateway |
| 07:04.37 | ChannelZ | Your linux box IS the router |
| 07:05.06 | ChannelZ | You just haven't configured it to NAT machines on the local side of your network out to the rest of the world |
| 07:05.21 | sawgood | yes, I agree the Linux box is acting as a router ... it has two NIC cards (one public and one private IP address) |
| 07:05.33 | sawgood | How do I setup NAT on the Linux box (the router)? |
| 07:05.41 | ChannelZ | And unless you have multiple IPs coming from comcast and are giving each phone a real IP, and have it routed as such on the Linux box, this won't work without using NAT. |
| 07:05.53 | ChannelZ | Read back 4 lines |
| 07:06.01 | kaldemar | i guess there's a channel for linux networking too. :P |
| 07:06.11 | sawgood | iptables --table nat -A POSTROUTING --out-interface eth0 --jump SNAT --to the.real.ip |
| 07:06.38 | ChannelZ | yes assuming eth0 is your WAN interface, and the.real.ip should be it's real-world ip address |
| 07:06.40 | sawgood | ChannelZ: is this what you meant? |
| 07:06.51 | sawgood | right ... but I think iptables is 'off' ... |
| 07:06.55 | sawgood | firewall is stopped |
| 07:07.03 | ChannelZ | It's never off, it's just not doing anything. |
| 07:07.25 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 07:07.48 | ChannelZ | I must say you are bold for putting a box on the net with no firewall up to this point! I hope you know what all is running on it |
| 07:08.50 | sawgood | when I typed in your iptables command (the only thing the statement did not 'like' was the IP address at the very end) |
| 07:09.21 | sawgood | I forgot the word --to (and then the IP) |
| 07:09.22 | sawgood | sorry |
| 07:09.58 | joobie | patrb, http://www.asteriskguru.com/tutorials/no_application_for_extension.html |
| 07:10.58 | sawgood | how to I 'see' if iptables is running the command I just typed in |
| 07:11.23 | ChannelZ | iptables -t nat -L |
| 07:11.24 | ChannelZ | sort of |
| 07:11.51 | sawgood | looks good ... so far |
| 07:12.05 | ChannelZ | does 'route' show a default route with a comcast gateway? |
| 07:12.51 | sawgood | yes |
| 07:13.04 | ChannelZ | ok so your box should masquerade any LAN traffic it can't otherwise figure out how else to route out your cable |
| 07:13.32 | sawgood | dw = 173.13.158.30 eth0 = 173.13.158.20 (Which IP should I use for NAT .30 or .20)? |
| 07:13.54 | ChannelZ | Do you have a block of multiple IPs? |
| 07:14.03 | sawgood | yes 255.255.255.240 |
| 07:14.52 | ChannelZ | well you can use any one you want so long as they are routed to and you have an interface listening. But for the sake of ease use whatever IP you already are using |
| 07:15.09 | ChannelZ | which is .20 I guess |
| 07:15.30 | sawgood | yeah ... the phone fails to 'go out to the net' and get a f/w update ... |
| 07:15.43 | sawgood | I do not think traffic can leave the phones and get to the Internet |
| 07:16.05 | ChannelZ | pastebin the output of "iptables -L -v -n" and "route" |
| 07:18.31 | sawgood | http://pastebin.com/6ExqvDXR |
| 07:19.19 | sawgood | http://pastebin.com/nnVFD3GK |
| 07:21.14 | ChannelZ | hmm |
| 07:21.57 | ChannelZ | eth0 is 173.13.158.20 ? |
| 07:22.08 | sawgood | yes |
| 07:22.24 | ChannelZ | and eth1 is what? |
| 07:22.43 | sawgood | 172.16.50.100 |
| 07:23.03 | sawgood | I have another customer with this same setup ... I am SSHing into their box to see thier iptables statement |
| 07:23.32 | ChannelZ | and you have 172.16.50.100 set as the IP for 'option routers' in your DHCP that is serving the phones? |
| 07:24.15 | sawgood | no I changed it to 173.13.158.20 |
| 07:24.21 | sawgood | I can change it back if you think it will work |
| 07:24.38 | ChannelZ | yeah |
| 07:24.56 | ChannelZ | and you'll have to reboot the phones too after restarting dhcpd |
| 07:25.28 | ChannelZ | also double-check iptables -t nat -L POSTROUTING |
| 07:26.28 | sawgood | Chain POSTROUTING (policy ACCEPT) |
| 07:26.28 | sawgood | target prot opt source destination |
| 07:26.28 | sawgood | SNAT all -- anywhere anywhere to:173.13.158.20 |
| 07:26.43 | sawgood | I did dhcpd restart |
| 07:26.49 | sawgood | should I reboot the phone? |
| 07:26.55 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:27.05 | ChannelZ | oh.. and "cat /proc/sys/net/ipv4/ip_forward" |
| 07:28.03 | sawgood | 0 |
| 07:28.06 | sawgood | was the output |
| 07:28.07 | ChannelZ | ah. |
| 07:28.14 | ChannelZ | echo "1" > /proc/sys/net/ipv4/ip_forward |
| 07:28.33 | ChannelZ | then reboot your phones so they get the right info and try again |
| 07:29.43 | ChannelZ | And if it still doesn't work it's hard to say without being able to dig around in your machine, but do you have any other computers/laptops you could put on the LAN and see if they work |
| 07:30.06 | ChannelZ | (and why, just as an aside, do you have 14 IPs but apparently nothing but 1 server on your cable?) |
| 07:31.14 | sawgood | I have other routers and devices using up all 14 IP address (only 1 spare) |
| 07:31.32 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 07:31.37 | sawgood | This setup I have now ... was 'working' until I had to rebuild the box the other day |
| 07:31.54 | sawgood | I did not document how to 'make the Linux box into a router' ... I am trying to do that now |
| 07:32.14 | sawgood | <PROTECTED> |
| 07:32.18 | sawgood | still 0 |
| 07:32.32 | ChannelZ | uhm.. so let me get this straight then... physically, from the perspective of this box, it has one NIC going directly to the cable modem.. and another which goes into a hub or switch, and the ONLY other things on that hub/switch are the phones? |
| 07:32.45 | ChannelZ | you probably have to be root to set it to 1 |
| 07:33.20 | sawgood | 100% correct ... in the network layout you described |
| 07:33.50 | sawgood | 3 phones plugged into a PoE switch and a 4th cable running to the 2nd NIC in the Linux box |
| 07:34.13 | sawgood | super clean simple Ethernet switch and cable run to a Comcast gateway |
| 07:34.36 | sawgood | I am definitely root .... |
| 07:35.16 | ChannelZ | hmm |
| 07:35.38 | ChannelZ | you echo'd 1 to /proc/sys/net/ipv4_ip_forward and then cat'd it and it still said 0? |
| 07:36.41 | *** join/#asterisk greysd (~oae2@mail.inter-test.ru) |
| 07:37.11 | sawgood | I did it two more times ... it goes through with no error, but when I check it ... it still says zero |
| 07:37.19 | ChannelZ | try "sysctl net.ipv4.ip_forward=1" |
| 07:37.48 | greysd | Hi! Please, how can i trace a call transfers? |
| 07:38.11 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 07:38.21 | ChannelZ | greysd: sip debug and watch the messages? |
| 07:38.38 | ChannelZ | or maybe not since you don't say transfers between what |
| 07:39.11 | *** join/#asterisk samy^ (~samy@cpe-76-173-222-231.socal.res.rr.com) |
| 07:39.11 | sawgood | now it has a 1 in the statement |
| 07:39.15 | *** join/#asterisk PhoenixMage (~Phoenix@49.71.96.58.static.exetel.com.au) |
| 07:39.26 | ChannelZ | ok |
| 07:41.27 | sawgood | Its working now, Bob! |
| 07:41.38 | sawgood | I am getting firmware updates from Snom on the phone |
| 07:41.49 | ChannelZ | I think I just shit my pants |
| 07:42.07 | sawgood | So, did I just make this box into a DHCP server and a router? |
| 07:42.26 | ChannelZ | yes.. a masquerading router |
| 07:42.39 | sawgood | NAT, right? |
| 07:42.43 | ChannelZ | yes |
| 07:43.35 | greysd | A call B, B answer, B call C, C answer, B transfer A to C, A talk with C. i want to record in DB that A call B and B transfer it to C. |
| 07:44.17 | sawgood | So, If I scroll back in this channel window and make notes ... I should be able to document what we did to make it work? |
| 07:46.32 | ChannelZ | Yes.. you can turn on ipv4 forwarding in /etc/sysctl.conf to make it 'permenant', or write a simple firewall script that does the sysctl bit and sets up iptables for doing NAT |
| 07:47.59 | sawgood | net.ipv4.ip_forward = 0 |
| 07:48.09 | ChannelZ | well you want 1 to turn it on |
| 07:48.11 | sawgood | This is the statement in my /etc/sysctl.conf file now |
| 07:48.27 | sawgood | strange? |
| 07:48.35 | sawgood | we set this to a 1, but the file has 0 |
| 07:49.14 | ChannelZ | the file is a config file, not a current state |
| 07:50.03 | ChannelZ | sysctl sets kernel parameters. sysctl.conf is a config file to set a whole bunch of parameters the same way every time the system boots |
| 07:50.04 | sawgood | I edited the file to change it to a 1 |
| 07:50.08 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 07:51.48 | Tim_Toady | i have a real strange prob with an asetrisk box.I running 1.6.0.26 with latest dahdi, my sip phones are spa921 and i have a B410P card connectin to bri lines |
| 07:52.12 | Tim_Toady | half of my calls are failing |
| 07:53.06 | Tim_Toady | when the other side answers the call i get a SIP/2.0 503 Service Unavailable |
| 07:53.27 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 07:53.34 | Tim_Toady | it seems to happen randomly |
| 07:53.42 | Tim_Toady | and i cant think of something |
| 07:54.51 | ChannelZ | codecs? |
| 07:55.01 | Tim_Toady | g729,alaw,ulaw |
| 07:55.03 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
| 07:55.37 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:55.44 | Tim_Toady | internal calls work perfectly |
| 07:55.59 | Tim_Toady | i get this only in outgoint call to the bri lines |
| 07:56.23 | sawgood | ChannelZ: can you see if these step by step notes look right: http://pastebin.com/Kx0YNbEh |
| 07:57.05 | ChannelZ | And does it happen from the save device to the same number all the time or that's random? |
| 07:57.12 | Tim_Toady | random |
| 07:57.43 | Tim_Toady | prob appeared after i upgraded to the latest linksys fw, 5.1.8 |
| 07:58.28 | Tim_Toady | but i have the same devices deployed in many sites with the same configuration same asterisk without having the same prob |
| 07:58.29 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 07:58.31 | ChannelZ | have you looked harder at SIP debugs to see what messages are going back and forth to try and see if it's a negotiation problem between * and the phone? |
| 07:58.47 | ChannelZ | sawgood: more or less yes |
| 07:58.47 | Tim_Toady | thats what im looking now |
| 07:58.59 | sawgood | ChannelZ: thank you!!! |
| 07:59.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:00.41 | ChannelZ | yup |
| 08:05.21 | sawgood | when one is working strickly with extensions.conf (and you want to make a company wide 'number' for all extensions to dial to reach voicemail ... would it look something like this: |
| 08:05.42 | sawgood | exten => 404,1(VoiceMailMenu) |
| 08:06.31 | ChannelZ | nope |
| 08:06.45 | ChannelZ | exten => 404,1,VoiceMailMain |
| 08:07.00 | sawgood | oh ... the () mean something else, huh? |
| 08:07.26 | ChannelZ | well () is for arguments. You also missed the , after the priority (1) and there is no application called VoiceMailMenu |
| 08:07.30 | ChannelZ | but other than that.. |
| 08:07.38 | ChannelZ | :P |
| 08:07.55 | sawgood | I was 'close' ... not bad for less then a few days from breaking away from FreePBX! |
| 08:08.28 | sawgood | ChannelZ: even without using FreePBX ... does Asterisk src require MySql to work? |
| 08:08.36 | ChannelZ | no |
| 08:08.39 | tuxx- | NEIN |
| 08:08.41 | sawgood | Or is MySql only something for GUI |
| 08:08.45 | sawgood | excellent! |
| 08:08.49 | ChannelZ | it's for 'realtime config' |
| 08:08.53 | tuxx- | and for cdr's |
| 08:09.01 | tuxx- | :X |
| 08:09.07 | ChannelZ | being able to put some configs in a database instead of static config files |
| 08:09.12 | ChannelZ | and logging as tuxx says |
| 08:09.14 | sawgood | So, the 'Asterisk Manager' is still part of the build though, right? |
| 08:09.15 | tuxx- | time to get some asphalt in my longues. |
| 08:09.40 | ChannelZ | Manager is "standard", yes |
| 08:10.11 | sawgood | Seem to me that Asterisk 1.6.2.6 was about about 25Mb tops ... not very large of an application |
| 08:11.04 | sawgood | These 'extra' things like FOP, ARI ... are not part of Asterisk ... they come from the 'distro's' adding them, right? |
| 08:11.36 | ChannelZ | yeah.. FOP is a totally separate thing, 3rd party.. ARI I'm not sure what that is |
| 08:11.47 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 08:11.50 | sawgood | Asterisk recording Interface |
| 08:11.50 | ChannelZ | but FOP gets put in as part of FreePBX I think |
| 08:12.58 | sawgood | In FreePBX (when one wants to ADD a system recording for an IVR menu) ... the *77 process from a phone is used ... how does one create the .wav files from scratch in Asterisk for IVR menus? |
| 08:13.38 | ChannelZ | make an extension that does Record(whatever.ulaw) |
| 08:13.57 | sawgood | oh ... neat ... one extension for making recordings only so to speak |
| 08:14.37 | sawgood | ChannelZ: you've never been tempted to try out a GUI? |
| 08:14.55 | ChannelZ | yeah just make a bunch if you have a few to record, make them all different extensions for whatever filenames... or put a bunch in a row on the same extension, and everytime you hit # to end the recording it'll go on to the next one |
| 08:15.30 | ChannelZ | I have FreePBX running in a VM to see what it looked like but I haven't played with it a lot because it seemed a little pointless to |
| 08:16.06 | sawgood | what about call recording on demand ... it is *1 with FreePBX ... what do you do to launch this in Asterisk only? |
| 08:16.30 | ChannelZ | features.conf |
| 08:16.57 | ChannelZ | that's 'automon', and you have to have Dial()ed with proper flags to allow recording |
| 08:17.08 | sawgood | Do you think the 'distros' have hurt the Asterisk business or made it more open to others to get involved? |
| 08:17.27 | ChannelZ | I have no idea |
| 08:18.08 | sawgood | I got ExtenSpy working on a few IP PBX boxes with whisper mode and barge mode ... thanks to your help! |
| 08:18.15 | sawgood | The customer's love this feature! |
| 08:18.32 | ChannelZ | There are some companies who have created "telephone appliances" with them, like a phone-system-in-a-box. I don't know how well they sell vs hiring a consultant or a company who builds a system themselves |
| 08:19.11 | ChannelZ | You're not the one that owns a pizza place are you? |
| 08:19.19 | sawgood | Plus ... there is several OEM makers of IP PBX boxes like Allworx, TalkSwitch, and Epygi which are not based on Asterisk |
| 08:19.28 | sawgood | no pizza place for me |
| 08:19.54 | ChannelZ | I think that was someone else with an 's' nick |
| 08:19.59 | sawgood | The PIKA appliance |
| 08:20.06 | ChannelZ | Sluggs or something maybe. so much for my brain |
| 08:20.14 | sawgood | and trixbox "Rhino" solutions |
| 08:20.42 | sawgood | I guess its off to sleep for me ... take care! |
| 08:21.16 | ChannelZ | yeah there's a lot out there. I got into * because I needed a small phone system for my business.. my partner was looking at an old Merlin system on eBay for some crazy amount of money, and I was like NOOOO don't you dare |
| 08:24.47 | voxter | Anyone have any idea why after converting one side of an asterisk trunk to 1.4, an IAX peer is now expiring its registration after 20 seconds of registering, every time? |
| 08:29.06 | Polysics | what is the proper way in 1.6 to have users be able to take only 1 call at once? |
| 08:29.16 | Polysics | is there a setting, or do i need logic? |
| 08:29.44 | Polysics | callcounter only enables me to actually KNOW the SIP device's status |
| 08:29.47 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.244.174) |
| 08:29.53 | Polysics | talking about SIP devices here, to be correct |
| 08:30.00 | Polysics | they have no extension associated |
| 08:30.36 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-vlbemxpuwnfozkpk) |
| 08:33.20 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
| 08:33.53 | Polysics | anyone? |
| 08:34.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:34.30 | Tim_Toady | Polysics disable callwaiting on ur sip phones |
| 08:35.18 | ChannelZ | indeed |
| 08:35.47 | Polysics | i don't see the option in the db, sorry |
| 08:36.00 | Polysics | is it a sip.conf option? |
| 08:36.09 | ChannelZ | no, on the phone its self |
| 08:36.22 | Polysics | i can't do that :-( |
| 08:36.42 | Polysics | complicated to explain why, so i won't bore you |
| 08:36.48 | ChannelZ | Most SIP phones, even so-called "single line" phones have a call-waiting feature that lets them accept another call. You should be able to turn that off in the phone's actual config. |
| 08:36.50 | Polysics | but doesn't call-limit do exactly that? |
| 08:37.47 | ChannelZ | call-limit is being deprecated |
| 08:38.28 | Polysics | and replaced by? |
| 08:40.11 | *** join/#asterisk kruemeltee (~Maddin@port-92-198-62-82.static.qsc.de) |
| 08:40.12 | Polysics | might it be busylevel? |
| 08:40.18 | kruemeltee | hello all together :-) |
| 08:40.34 | Polysics | although the wiki says you need call-limit for busylevel to work, which is confusing :-) |
| 08:40.34 | ChannelZ | apparently hosing it up in the dialplan yourself with the GROUP_COUNT function or something |
| 08:40.58 | Polysics | ChannelZ, i find that not very smart, but hey, i don't code * |
| 08:40.58 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:42.30 | ChannelZ | Didn't we go over this the other day? |
| 08:43.42 | jsjc | when I play exten => s,n,Background(welcome) plays in english should I set exten => s,n,Background(es/welcome) so plays spanish? |
| 08:43.58 | ChannelZ | At least for me setting 'callcounter=1' in sip.conf works and manually checking the status with the DEVICE_STATE function |
| 08:44.05 | kruemeltee | do I have to see the CIDName within "database show" of every phone thats registered? for instance callerid is set to "CCA 1 <601>" ... do I have to see "CCA 1" within "database show"? |
| 08:44.18 | Gido-E | jsjc, it is also possible to set language to es |
| 08:44.29 | ChannelZ | jsjc: no you'd set the language for the channel to es |
| 08:44.38 | jsjc | Gido-E: I have set language to es all around and does not play es.... |
| 08:44.45 | Polysics | ChannelZ, yes, we did, and that works, but there are a number of settings that might do that instead - i was basically trying to find out why non of them work :-) |
| 08:45.18 | ChannelZ | you should just fix your phones but nevermind |
| 08:45.37 | jsjc | I am receiving the call through SIP and in general has language=es and in the actual [user] has language=es as well |
| 08:45.39 | Polysics | Zoiper web doesn not have a provisionable setting for that :-( |
| 08:45.43 | Polysics | that is the problem |
| 08:45.53 | jsjc | I actually would like to setup whole asterisk to es unless otherwise mentioned |
| 08:46.54 | ChannelZ | jsjc: where have you changed the language? Channels have a default (sip.conf, chan_dahdi.conf, etc depending) |
| 08:47.07 | jsjc | sip.conf |
| 08:48.45 | *** join/#asterisk chendy_ (~chatzilla@204.152.211.137) |
| 08:49.50 | ChannelZ | jsjc: did you set it globally? |
| 08:50.21 | *** join/#asterisk mahiti-irc (~mahiti@122.166.105.215) |
| 08:50.36 | mahiti-irc | hi |
| 08:51.23 | mahiti-irc | i installed asterisk 1.4.27 and ooh323c in a centos machine |
| 08:51.37 | jsjc | ChannelZ: I set it up in [general] |
| 08:51.55 | jsjc | any way to set it up global? |
| 08:51.57 | mahiti-irc | i need asterisk to talk with a hypermedia GSM server |
| 08:52.01 | ChannelZ | well that's what I meant |
| 08:52.15 | mahiti-irc | when i use a softphone like sjphone |
| 08:52.17 | ChannelZ | I just set mine to es, put a file in the es directory and it worked |
| 08:52.22 | mahiti-irc | which is working |
| 08:52.36 | ChannelZ | Is it saying "Playing 'xxxx' (language 'es')" on the console? |
| 08:52.37 | mahiti-irc | but asterisk is not able to make a dial ouut |
| 08:52.48 | jsjc | can I set it up in asterisk.conf as a default language? |
| 08:52.50 | mahiti-irc | can anyone help me on this |
| 08:54.46 | ChannelZ | jsjc: well you could either 'hotwire' it by turning off the language prefix, and setting the sounds directory to the 'es' subdirectory, or move them all out of the language directory |
| 08:55.08 | ChannelZ | jsjc: Are all your calls only coming in/out via SIP? |
| 08:55.16 | jsjc | coming in via SIP |
| 08:55.18 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 08:55.42 | jsjc | but if I open the voicemailmain it is in es |
| 08:56.05 | ChannelZ | that's what I tested |
| 08:56.22 | jsjc | I have an extensions that execs VoicemailMain() and appears to be spanish |
| 08:56.30 | jsjc | but the Background(welcome) it is english... |
| 08:57.41 | ChannelZ | Are the permissions on your sounds such that * is able to access them? And you didn't answer my other question |
| 08:58.01 | ChannelZ | does the console show (language 'es') when it's playing back |
| 08:58.10 | jsjc | I am confused now I have two asterisk sound folders /var/lib/asterisk/sounds and in /usr/share/asterisk/sounds one of them is missing the spanish... so I am confused... |
| 08:58.46 | ChannelZ | default is /var/lib/asterisk/sounds/xx/ where xx is the language |
| 08:59.12 | ChannelZ | do 'core show settings' |
| 08:59.34 | jsjc | says defualt language english |
| 08:59.36 | ChannelZ | actually that doesn't show you the sound dir |
| 08:59.42 | jsjc | nop it does not |
| 08:59.44 | *** join/#asterisk Z_God (~julius@wlan228007.mobiel.utwente.nl) |
| 08:59.47 | jsjc | i been trying that for a while |
| 09:00.01 | *** join/#asterisk contrabanda (~contr@188.123.128.2) |
| 09:00.31 | contrabanda | Hellooo |
| 09:03.04 | mahiti-irc | can anyone help me |
| 09:03.42 | contrabanda | Please i need hepl with dahdi. http://pastebin.com/QecQKJ4a |
| 09:04.25 | jsjc | ChannelZ: will do some investigation if not just will leave using my custom made prompts |
| 09:04.49 | ChannelZ | well it's pretty easy to figure out |
| 09:04.57 | ChannelZ | where are your custom prompts? |
| 09:07.28 | *** join/#asterisk aceio (~90fe6658@gateway/web/freenode/x-kfscnydknzorrztv) |
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| 09:11.17 | jsjc | the structure of sounds folder is language en/allthesounds and then folders within en for follome phonetics..... |
| 09:11.37 | jsjc | or sounds/es and sounds/phonetic/es |
| 09:13.13 | contrabanda | Please i need hepl with dahdi. http://pastebin.com/QecQKJ4a |
| 09:13.22 | *** join/#asterisk TimeRider (steve@5ac31894.bb.sky.com) |
| 09:13.23 | ChannelZ | it should all be /var/lib/sounds/xx/* |
| 09:13.37 | ChannelZ | like /var/lib/sounds/en/phonetic/* and /var/lib/sounds/es/phonetic/* |
| 09:14.05 | ChannelZ | If you tell it to play a sound that doesn't exist in the current language, it falls back to trying the default language |
| 09:14.43 | ChannelZ | You're probably getting english because your spanish files are not structured right or asterisk otherwise can't read them |
| 09:14.45 | mahiti-irc | waiting |
| 09:15.06 | jsjc | ChannelZ: I think that is the issue I will sort it out and test |
| 09:15.15 | ChannelZ | Are you trying to use the default asterisk 'es' sound pack? |
| 09:15.35 | jsjc | I cannot find default es soundpack |
| 09:15.39 | ChannelZ | What version of asterisk? |
| 09:15.54 | jsjc | when I compiled the asterisk 1.6.2.6 looks like did not put |
| 09:16.15 | jsjc | maybe default is the colombian accent one... and I really didnt like it so I am getting the spain version |
| 09:16.26 | jsjc | nothing wrong with colombianS!? My housemate is one of them hehe |
| 09:16.30 | ChannelZ | go into your build directory and do "make menuconfig" |
| 09:16.43 | jsjc | ChannelZ: all spanish are selected |
| 09:16.53 | ChannelZ | then go into "Core Sound Packages", turn on CORE-SOUNDS-ES-ULAW, save, and make install |
| 09:17.11 | jsjc | yeap that is like htat |
| 09:18.35 | contrabanda | Please i need help with dahdi. http://pastebin.com/QecQKJ4a |
| 09:18.38 | mahiti-irc | can anyone help me debug openh323 with asterisk 1.4 on centos |
| 09:18.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:18.48 | sulex | please, what's the right way to completely stop using extensions.conf in favor of ael? (apart from having a 0 size .conf) |
| 09:21.19 | jsjc | Playing 'welcome.ulaw' (language 'en') why!? |
| 09:21.47 | ChannelZ | jsjc: you have something not configured right |
| 09:21.56 | ChannelZ | jsjc: paste your sip.conf |
| 09:22.01 | ChannelZ | pastebin |
| 09:22.02 | jsjc | ChannelZ: I know but looking around and no luck |
| 09:22.16 | ChannelZ | cuz I just downloaded and installed the es core sounds and this bitch is yelling at me in spanish |
| 09:22.27 | Tim_Toady | sulex just load pbx_ael.so module and put ur extensions.ael in place |
| 09:23.22 | jsjc | http://pastebin.com/EkKQ7FD7 |
| 09:24.20 | ChannelZ | and you're not setting it to something else somewhere in your dialplan |
| 09:24.29 | jsjc | mnm let me check |
| 09:25.00 | jsjc | in [general] says language=es |
| 09:25.04 | *** part/#asterisk jsjc (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 09:25.13 | *** join/#asterisk jsjc (~chatzilla@219-90-149-184.ip.adam.com.au) |
| 09:25.22 | jsjc | that is the only place I have mentioned language |
| 09:26.01 | sulex | Tim_Toady: yep and it's working, but since now extensions.conf is useless and i see its loading as a waste of resources(yes I'm in paranoid land), i was wondering if there's a way to get rid of its loading. but maybe it's the * core in charge of its loading, and so is not possible to avoid its parsing? |
| 09:26.13 | ChannelZ | I mean you're not using the LANGUAGE function anywhere... this isn't freepbx or something doing something crazy in another config file.../ |
| 09:26.26 | jsjc | ChannelZ: if I do cat * |grep language in /etc/asterisk just replies me with language=es |
| 09:26.46 | mahiti-irc | can anyone help me debug openh323 with asterisk 1.4 on centos?? |
| 09:26.48 | Tim_Toady | sulex delete it and remove pbx_confic.so module |
| 09:26.49 | ChannelZ | grep -i |
| 09:27.04 | kaldemar | jsjc: NoOp(${CHANNEL(language)}) in a call |
| 09:27.28 | jsjc | I have not mention bloody language... |
| 09:27.29 | Tim_Toady | sulex sorry i mean pbx_config.so module |
| 09:27.38 | jsjc | I am just thinking this thing of having to sound directories |
| 09:27.47 | jsjc | and it is reading from one of them not the other? |
| 09:28.00 | jsjc | but the same in asterisk.conf i have set datadir to be /var/lib |
| 09:28.04 | ChannelZ | no even if it's playing a file out of 'en' if it thinks the channel is 'es' it should say that |
| 09:28.17 | ChannelZ | at least in 1.6.1 |
| 09:28.54 | sulex | Tim_Toady: I try that, thanks. i thought pbx_config was in charge of the parsing not onlx of extensions.conf but from the source i see iwas thinking wrong. thanks |
| 09:29.35 | ChannelZ | you have something crazy going on with your configs or something, barring some sort of bug in your version which I tend to doubt |
| 09:30.22 | ChannelZ | I could probably ssh in and figure it out if you were so inclined but else I need to go to bed |
| 09:30.27 | jsjc | I tend to rule out bugs and more something i am doing... but definetly even if I specify Playback(es/welcome) should be spanish right? |
| 09:30.52 | mahiti-irc | ok |
| 09:31.13 | jsjc | ChannelZ: just go to bed will mock around a bit more if not will ask this again tomorrow |
| 09:31.15 | mahiti-irc | which is latest supported openh323 driver for installation on asterisk |
| 09:31.31 | ChannelZ | no not unless the file was .../sounds/en/es/welcome.ulaw |
| 09:31.59 | *** join/#asterisk alhafoudh (~alhafoudh@77.93.192.244) |
| 09:32.00 | alhafoudh | hi guys |
| 09:32.53 | mahiti-irc | which is latest supported openh323 driver for installation on asterisk 1.4 |
| 09:33.13 | alhafoudh | anyone can help me with this? Avaya has LAR feature, look ahred routing, to use this, i need to send reason code 34 back to avaya when circuit has congestion but i also need to preserve call progress setup fields which are controlled by progress_setup=8 setting in h323.conf but it does not work, can anyone help? |
| 09:34.22 | jsjc | ok so that is why that does not work neither... |
| 09:34.38 | jsjc | any possibility of setting whole default language of asterisk to es |
| 09:34.46 | jsjc | because when checkin core show settings |
| 09:34.56 | jsjc | appears default language = en |
| 09:35.09 | jsjc | where shoul I address to put in es? |
| 09:38.30 | alhafoudh | anyone please can help? |
| 09:38.42 | alhafoudh | mahiti-irc: 1.19.1 |
| 09:39.08 | mahiti-irc | alhafoudh, which one openh323 or h323plus ? |
| 09:39.13 | mahiti-irc | which is better? |
| 09:39.29 | alhafoudh | openh323 |
| 09:43.13 | alhafoudh | anyone please? |
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| 09:48.04 | mahiti-irc | alhafoudh, these days replies came very late |
| 09:48.12 | mahiti-irc | u have to wait :) |
| 09:57.37 | sulex | the underscore in front of a variable used in Read(), does it make the variable inheritable by the child channels as in SetVar()? |
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| 10:23.19 | *** join/#asterisk _markh_ (~chatzilla@fentech.gotadsl.co.uk) |
| 10:23.58 | _markh_ | 'm trying to build a system where a user (with no headphone/speakers) can play a video to their PC but listen to the audio via a phone. I envisage a system whereby the user visits a web page, is presented with a phone number/pin. They dial the number, enter the pin and then press play on their onscreen player. |
| 10:23.59 | _markh_ | I'm thinking that we can adapt a video streamer to connect to an asterisk conference, playing the audio into the conference which contains the (only) user. Or is there a better way? |
| 10:25.02 | Zeeek | what is the advantage of the phone here? |
| 10:25.36 | _markh_ | The use has no sound card on their PC, but they do have a phone on their desk |
| 10:25.39 | Zeeek | oh, no audio system at all |
| 10:25.54 | Zeeek | but it would never be in sync |
| 10:26.24 | Zeeek | I wonder where you see the need for such a thing? |
| 10:26.31 | _markh_ | It doesn;t really need to be in sync (i.e. there's no lip sync involved), just a commentary |
| 10:26.37 | Zeeek | ok |
| 10:26.49 | _markh_ | It's a software demo |
| 10:26.55 | Zeeek | what population segment needs this? |
| 10:27.17 | Zeeek | most computers have a least speakers |
| 10:27.36 | _markh_ | This is in a health service environment |
| 10:27.43 | _markh_ | Reception PC |
| 10:27.50 | Zeeek | patient side or professional side |
| 10:27.53 | Zeeek | ok |
| 10:27.59 | _markh_ | Professional |
| 10:28.19 | _markh_ | Even if they have a sound card (on the mobo) they won;t have headphones/speakers |
| 10:28.28 | Zeeek | and the video in question is from a commection you contriol or any potentiazl video anywhere? |
| 10:28.52 | Zeeek | IOW what videos, where are they coming from |
| 10:28.53 | _markh_ | Our video |
| 10:28.58 | Zeeek | all in one place? |
| 10:29.05 | _markh_ | yes |
| 10:29.07 | Zeeek | how many? |
| 10:29.12 | Zeeek | less than 100 ? |
| 10:29.29 | _markh_ | probably only one |
| 10:29.43 | Zeeek | I'd recommend you extract the audio and just initiate a call to the requester |
| 10:30.06 | Zeeek | There are many free tools to extract the audio to WAV or whetevr you need |
| 10:30.21 | _markh_ | the only trouble there is that the we can;t always DID to the user |
| 10:30.34 | Zeeek | then just call the number or whatever, and play theaudio file or use an IVR to allow them to start the play |
| 10:30.39 | _markh_ | and how would we launch the sound at the same time as the video |
| 10:31.27 | Zeeek | You'd need some progralmming to start |
| 10:31.33 | Zeeek | oops |
| 10:31.51 | Zeeek | you would call a normal phone? |
| 10:32.00 | _markh_ | Yes |
| 10:32.25 | Zeeek | I'd say they're looking at the page, the click on a button that calls them |
| 10:32.46 | Zeeek | when they answer the IVR asks them to hit the # key when the are reay to start the video |
| 10:33.24 | Zeeek | if there werer several videos it could be like "Press 1 for geriatic assination |
| 10:33.42 | Zeeek | press 2 for diahretic analisys" |
| 10:33.45 | Zeeek | etc |
| 10:33.57 | Zeeek | "to restart the audi at any time, hit *" |
| 10:34.08 | Zeeek | I need to go to lunch |
| 10:34.14 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 10:34.16 | Zeeek | sounds like an interesting project, tho |
| 10:34.31 | _markh_ | Thx for te ideas |
| 10:34.37 | _markh_ | the |
| 10:34.54 | _markh_ | I'm just not sure how to start the cide from * |
| 10:35.09 | _markh_ | cide ;) = video |
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| 10:47.11 | Zeeek | _markh_: I think the only way is on the site itself, the sperson starts the video when they get the calls |
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| 10:53.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:08.34 | nagios1111 | hi, i get this error in asterisk pbx_spool.c: Call failed to go through, reason (3) Remote end Ringing |
| 11:08.45 | nagios1111 | any body have an idea?? |
| 11:08.58 | nagios1111 | how to resolve it |
| 11:09.02 | nagios1111 | please |
| 11:13.03 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 11:20.00 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:21.40 | _markh_ | Zeekek: I'm wondering about using MOH to play the audio. If we take an opensource streaming app, break out the audio (probably easier said than done), route it through ffmpeg and make it available to asterisk as a MP3 streaming source, I can connect the user that calls in to an extension that plays that MP3 source as audio. That way, when the user hits play on their browser they will hear... |
| 11:21.41 | _markh_ | ...the audio as MOH. |
| 11:22.06 | _markh_ | Zeeek: :) |
| 11:22.46 | _markh_ | One day I'm going to learn to type... |
| 11:26.23 | alhafoudh | anyone can help please? |
| 11:26.30 | alhafoudh | anyone can help me with this? Avaya has LAR feature, look ahred routing, to use this, i need to send reason code 34 back to avaya when circuit has congestion but i also need to preserve call progress setup fields which are controlled by progress_setup=8 setting in h323.conf but it does not work, can anyone help? |
| 11:27.15 | Zeeek | _markh_: it might work, if it does maybe yiu can sell the idea :) |
| 11:28.01 | Zeeek | is leaving to shop for food and beer for the VUC (http://vuc.me #vuc on Freenode.net - starts in 4.5 hours) |
| 11:31.03 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 11:32.50 | *** join/#asterisk TSM (~the_softw@fw-lon1.wenn.com) |
| 11:33.18 | TSM | any good serial port bods here mind to answer a few questions for me, pvt if needed instead of the room |
| 11:34.09 | nagios1111 | can you help me to resolve this error pbx_spool.c:347 attempt_thread: Call failed to go through, reason (3) Remote end Ringing |
| 11:37.28 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
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| 12:11.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
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| 12:21.08 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 12:21.11 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:21.18 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 12:22.35 | ariel_ | Morning |
| 12:22.47 | carrar | Evening |
| 12:23.57 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:24.05 | *** part/#asterisk CVirus (~Satan@196.205.193.191) |
| 12:29.50 | Skeeter- | Yesterday's gun talk made me go crazy so i buy a new carrying case for my G17 |
| 12:31.43 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:34.14 | ariel_ | I do own a gun, but that is due to it was my dad's. But overall it's locked in a safe box. I really don't like them. |
| 12:34.47 | Skeeter- | why |
| 12:35.18 | Skeeter- | Probably the only thing that makes me Zen |
| 12:36.03 | [TK]D-Fender | Skeeter-: 'cause nothing says eastern philosophy & peace... like guns |
| 12:36.53 | Skeeter- | i dont get anything u just said, but your always right so... +1 |
| 12:37.19 | ariel_ | I was in the Military and I know how to shoot, but still think there dangerous and most don't have the respected needed to own one. |
| 12:37.31 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-wbjfhufdnwwjxbkg) |
| 12:40.06 | [TK]D-Fender | prefers the feel of his katanas. Anything worth doing is worth making personal. |
| 12:40.08 | Skeeter- | im in Canada, so obviously, if you own a pistol/revolver legaly, you only shoot at the range like me |
| 12:40.57 | Skeeter- | wonders if katanas can dodge gun's bullets... |
| 12:41.06 | pentanol | hi there. why can be happening this error... ast_func_read: Function CHECK_DST not registered |
| 12:41.11 | jblack | Oh of course. Canadians are so sensible in that way |
| 12:41.18 | Skeeter- | i cant remember that movie where they could |
| 12:41.27 | Skeeter- | jblack, bullsh1t |
| 12:41.38 | jblack | Guns only at the range, trash only in receptacles. Sex only in bed. |
| 12:42.04 | jblack | and everyone that can sing or dance sent across the border |
| 12:42.13 | Skeeter- | Harder for citizen to defend themself, but easier for criminal |
| 12:42.15 | pentanol | this is defined into func_odbc.conf http://codepad.org/F9MdB9Ec |
| 12:45.17 | *** join/#asterisk coppice (~chatzilla@202.64.176.93) |
| 12:48.53 | *** join/#asterisk dewinda (~sahan@123.231.98.226) |
| 12:49.09 | Skeeter- | how can i see if i have installed the asterisk-addons |
| 12:51.49 | [TK]D-Fender | Skeeter-: see any of the modules loaded? |
| 12:51.55 | *** join/#asterisk rbd_ (~rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 12:52.13 | Skeeter- | i tried to load cdr_addons_mysql.so, and its not there |
| 12:52.53 | Skeeter- | What would installing asterisk-addons affect my productivity |
| 12:54.51 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 12:54.57 | [TK]D-Fender | Skeeter-: Grammar fail. Try again. |
| 12:56.37 | Skeeter- | If i stop asterisk, then install asterisk-addons, start asterisk. Will it crash |
| 12:57.12 | *** join/#asterisk btsteve_ (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 13:03.02 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:06.03 | Skeeter- | [TK]D-Fender, still grammar fail? |
| 13:06.12 | Skeeter- | i ordored the poster BTW |
| 13:07.16 | [TK]D-Fender | Skeeter-: Why would you think * would crash if you add a module to it? |
| 13:07.38 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:07.45 | Skeeter- | [TK]D-Fender, its asterisk, which cause 0.1% of the problem |
| 13:07.46 | [TK]D-Fender | Skeeter-: If I turn off my TV and then connect a DVD player to it and turn it on, do you think it will die? |
| 13:07.58 | Skeeter- | im behind it which is 99.9% of the problem |
| 13:08.05 | [TK]D-Fender | Skeeter-: You are being neurotic for nothing again |
| 13:10.03 | Skeeter- | gret |
| 13:10.21 | manxpower | Skeeter-: show us a pastebin of asterisk starting as "asterisk -cvvv" |
| 13:10.27 | manxpower | (when it fails) |
| 13:10.34 | manxpower | I doubt that's the problem, but you never know. |
| 13:10.51 | Skeeter- | i didnt installed them, im pretty sure of it |
| 13:11.18 | [TK]D-Fender | Skeeter-: are the modules THERE? |
| 13:11.34 | manxpower | Skeeter-: nevermind. I thought you had a valid question. |
| 13:11.55 | Skeeter- | i dont know any module beside cdr_addon_mysql.so, and its not there, LIKE i mentioned BEFORE |
| 13:12.12 | manxpower | Skeeter-: What is your ACTUAL problem? |
| 13:12.17 | Skeeter- | ok maybe i didnt |
| 13:12.24 | manxpower | missing cdr_addon_mysql.so? |
| 13:12.39 | Skeeter- | asterisk-stats looks into the mysql asteriskcdrdb to get infos |
| 13:13.04 | Skeeter- | mysql isnt loading/parsing data from /var/log/asterisk/cdr-csv/master.csv |
| 13:13.04 | manxpower | Ah, Now the truth comes out. You have a question about asterisk-stats and asteriskcdb. |
| 13:13.12 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 13:13.13 | Skeeter- | sigh |
| 13:13.27 | manxpower | does the module show when you do "core show modules" |
| 13:13.34 | Skeeter- | manxpower, thats the goal, not the problem |
| 13:14.03 | manxpower | so the problem is "mysql isnt loading/parsing data from /var/log/asterisk/cdr-csv/master.csv"? |
| 13:14.31 | Skeeter- | nope |
| 13:14.34 | contrabanda | Please i need help with dahdi. http://pastebin.com/QecQKJ4a |
| 13:14.35 | manxpower | A problem is something like "When I do X, then Y happens, this is a problem". |
| 13:14.41 | Skeeter- | manxpower, but the module does it? |
| 13:14.46 | manxpower | Skeeter-: when you have an actual problem feel free to come back. |
| 13:15.01 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.84) |
| 13:15.05 | manxpower | Skeeter-: I have no idea what that silly CDR thing does. I'm trying to help with Asterisk, not billing your clients. |
| 13:15.15 | [TK]D-Fender | [09:13]<Skeeter->mysql isnt loading/parsing data from /var/log/asterisk/cdr-csv/master.csv <--- .... why would MySQL give a shit about some random TEXT FILE? |
| 13:15.15 | Skeeter- | i mean the module if called cdr_addon_mysql.so for some reason? |
| 13:15.37 | manxpower | Skeeter-: I wish you the BEST of luck. |
| 13:15.55 | Skeeter- | manxpower, thats not about billing, i just wanna have some stats |
| 13:16.06 | manxpower | Skeeter-: I wish you the BEST of luck. |
| 13:16.09 | [TK]D-Fender | Skeeter-: If you set your CDR up properly it will log to MySQL. There is no "in between" |
| 13:16.32 | Skeeter- | [TK]D-Fender, tell me whats cdr_addon_mysql.so used for then? |
| 13:20.39 | contrabanda | hello |
| 13:20.49 | Skeeter- | salut |
| 13:21.29 | *** join/#asterisk VEc (~Vector@84.12.253.146) |
| 13:21.47 | VEc | Does anyone have any tips for running Asterisk in VMware ? |
| 13:22.16 | contrabanda | Please i need help with dahdi. http://pastebin.com/QecQKJ4a |
| 13:22.55 | VEc | contrabanda : what help do u need ? |
| 13:23.15 | Skeeter- | VEc, what tips do you want |
| 13:23.16 | Zeeek | It's official! [TK]D-Fender is NOT our special guest today on #vuc in 2.5 hours |
| 13:24.16 | Skeeter- | Zeeek, whats is that conference about? |
| 13:24.25 | VEc | Skeeter- : does it work :O ? is it true that accurate high resolution timing is not possible in VMware ? is there away to resolve it ? |
| 13:24.29 | *** join/#asterisk TJ^ (~tjq@host86-142-46-156.range86-142.btcentralplus.com) |
| 13:24.36 | contrabanda | Vec: from log which i have posted you will find outmy chan_dahdi.conf and asterisk output when i try to reload it |
| 13:24.38 | Zeeek | Skeeter-: VoIP |
| 13:24.43 | TJ^ | so fed up of dealing with dell |
| 13:24.47 | TJ^ | they piss me off |
| 13:24.53 | Skeeter- | Zeeek, why is it on lunch time? |
| 13:25.02 | TJ^ | why move everything to india where no one can help you |
| 13:25.13 | Skeeter- | VEc, i dont know what ur talking about, only thing i know is PCI card wont work |
| 13:25.56 | Skeeter- | TJ we use Dell for customer that we hate, those that will nvr come back |
| 13:26.03 | Zeeek | It isn't in my time zone |
| 13:26.44 | Zeeek | Skeeter-: it is 12 Noon EDT, 9AM PAcific, 5PM UK, 6PM Europe evening in India, very early Saturday in Hong Kong |
| 13:26.51 | [TK]D-Fender | [09:16]<Skeeter->[TK]D-Fender, tell me whats cdr_addon_mysql.so used for then? <- Giving * the ability to store CDR's in MySQL |
| 13:27.03 | Skeeter- | lol |
| 13:27.17 | Zeeek | For the VUC in your time zone, see http://vuc.me/next |
| 13:27.38 | [TK]D-Fender | Zeeek: I love your exclusion of UK from Europe :) |
| 13:27.42 | Skeeter- | [TK]D-Fender, just wondering, does it take it directly from the file or directly from asterisk? |
| 13:27.58 | Zeeek | Because the UK is not in the same time zone! |
| 13:28.06 | [TK]D-Fender | Skeeter-: Its an asterisk module.... * uses it directly |
| 13:28.18 | WIMPy | It's not only the timezone. |
| 13:28.35 | Skeeter- | [TK]D-Fender, once installed, it will not store older calls details...? |
| 13:28.51 | Zeeek | btw, next week, starting Wednesday if you are in Europe (even the UK) AstriEurope is on for three days, it's free and several people are there from Digium and from the VUC |
| 13:29.15 | Zeeek | Several of us are meeting up there |
| 13:29.53 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.87.110) |
| 13:29.54 | Zeeek | The amazing tie-dye shirts will be there |
| 13:30.22 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:31.11 | [TK]D-Fender | Skeeter-: * doesn't do time travel. |
| 13:31.38 | [TK]D-Fender | hides his pre-release copy of res_fluxcapacitor.so |
| 13:31.55 | VEc | Skeeter- : have u ever run it on Vmware ? |
| 13:34.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:37.06 | *** join/#asterisk The_Canuck (~The_Canuc@adsl-67-38-81-158.dsl.sfldmi.ameritech.net) |
| 13:37.35 | The_Canuck | hello |
| 13:37.37 | slashtom | i'm linking up asterisk-1.4 (ooh323 module) and gnugk, so that sip clients can call h323 clients. calls from sip clients to h323 always appear, however most of the time the call is immediately terminated (each side says the other one disconnected). where should i start looking? |
| 13:37.42 | Skeeter- | VEc, i got 1 right next to me |
| 13:38.58 | *** join/#asterisk Slugs_ (~yeah@173-8-52-45-Jacksonville.hfc.comcastbusiness.net) |
| 13:40.37 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:41.52 | VEc | Skeeter- : what verison of Asterisk and Dahdi u running on it ? |
| 13:42.13 | Skeeter- | 1.6.2.6 with the latest dahdi |
| 13:42.15 | VEc | Skeeter- : how many calls go though it per hour or day ? |
| 13:42.21 | Skeeter- | about 5 |
| 13:42.29 | VEc | per hour ? or per day ? |
| 13:42.41 | VEc | what version of VMWare ? |
| 13:42.48 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:42.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.31 | Skeeter- | per day |
| 13:43.36 | *** join/#asterisk gr0mit (~tim@router0.txrx.org.uk) |
| 13:43.42 | Skeeter- | vmwamre sphere 4 |
| 13:44.24 | Skeeter- | i dont recommend vmware for asterisk simply cuz its preferable to have a dedicated box |
| 13:44.24 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:44.25 | contrabanda | Please i need hepl with dahdi. I have configured chan_dahdiwith pre and ss7 signalling. when i restart asterisl it gives me dahdi errors. http://pastebin.com/QecQKJ4a Please help to fix this problem. |
| 13:44.56 | *** join/#asterisk rgsteele (~rgsteele@207.106.239.81) |
| 13:45.29 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
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| 13:47.00 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 13:47.45 | TJ^ | System uptime: 10 years, 14 weeks, 2 days, 9 hours, 34 minutes, 3 seconds ??? |
| 13:47.46 | TJ^ | lol? |
| 13:47.50 | VEc | contrabanda : I never knew Dahdi support SS7, so not going to be able to help u there |
| 13:48.04 | TJ^ | thats not right |
| 13:48.16 | Naikrovek | 10 years. sacre bleu |
| 13:48.27 | TJ^ | wonders how to fix that... |
| 13:48.33 | VEc | <PROTECTED> |
| 13:48.38 | TJ^ | lol |
| 13:48.49 | TJ^ | u got it too then... |
| 13:48.51 | TJ^ | wonder why that is |
| 13:49.00 | VEc | TJ^ : I was just kidding |
| 13:49.05 | TJ^ | oh :( |
| 13:49.18 | Skeeter- | Naikrovek, another french folks here, oh bin caliss |
| 13:49.32 | Naikrovek | i'm not french |
| 13:49.38 | Skeeter- | quebecer |
| 13:49.40 | Naikrovek | i just wish i spoke french for some reason |
| 13:49.46 | Naikrovek | always wanted to speak french |
| 13:49.47 | Skeeter- | oh |
| 13:49.50 | Naikrovek | i'm a fat-ass from illinois |
| 13:49.56 | Skeeter- | hahaha |
| 13:50.17 | Skeeter- | hands Rosetta Stone to Naikrovek |
| 13:51.06 | Skeeter- | Naikrovek, u gimme asterisk courses, i give u french courses |
| 13:52.48 | Naikrovek | i should probably learn asterisk then |
| 13:52.56 | Naikrovek | i know asterisk "am pu" |
| 13:53.04 | Naikrovek | spelling unknown for those words, btw |
| 13:53.21 | Skeeter- | un peu |
| 13:53.27 | Naikrovek | (french for "a little") whatev.. okay |
| 13:54.00 | Skeeter- | but if u read am pu in englsh, it almost make sense |
| 13:54.16 | Skeeter- | like u said, speeling unknown but still make sense |
| 13:54.41 | Naikrovek | only makes sense when he misspells in french |
| 14:00.47 | *** join/#asterisk Coin_Ope_Boy (~coin@bl6-161-78.dsl.telepac.pt) |
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| 14:07.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:09.35 | *** join/#asterisk wdbl (~not@ool-44c0668f.dyn.optonline.net) |
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| 14:18.43 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 14:19.36 | jhirley | o/ |
| 14:20.00 | Faustov | does anyone know of something like teamspeak but based on sip/asterisk? |
| 14:20.09 | Naikrovek | heh |
| 14:20.20 | Naikrovek | those are called softphones |
| 14:20.37 | Naikrovek | or just "phones" |
| 14:20.39 | Faustov | softphones into permanent conference call (meetme)? |
| 14:20.43 | Naikrovek | and conference rooms |
| 14:20.57 | Naikrovek | yeah the phone still has to dial into the room somehow |
| 14:21.02 | Naikrovek | but the room can persist |
| 14:21.06 | Naikrovek | with or without callers |
| 14:21.09 | Faustov | sure |
| 14:21.26 | Faustov | now what I'm still missing is a) push-to-talk b) indication who is speaking atm |
| 14:21.36 | Faustov | any chance some solution could give such functionality? |
| 14:22.00 | Naikrovek | the notification of who is talking, that's not technically possible with telephones as far as i know |
| 14:22.12 | Naikrovek | push to talk is a feature on some softphones i think |
| 14:22.36 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 14:22.59 | Faustov | I was wondering if some softphone would allow such functonality, hence asking around |
| 14:23.06 | Faustov | and googling in the meantime |
| 14:23.07 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 14:25.19 | Skeeter- | Anyone ever played with auto-installer |
| 14:29.17 | *** join/#asterisk [SySteM] (~antoine@aqu33-6-88-168-80-163.fbx.proxad.net) |
| 14:29.43 | [SySteM] | Hi, anyone got a swissvoice IP10S (sip) on asterisk ? i need some helps please |
| 14:30.24 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:30.28 | ijpalmer | hi if I'm creating an iax2 channel to another server what is the failure code and what is the name of the variable the code is set against |
| 14:31.25 | [SySteM] | i can call with my swissvoice IP10S and it can receive call .. but 10 minutes after do nothing.. phone return a : SIP response 486 "Busy Here" |
| 14:34.17 | Naikrovek | never even heard of swissvoice |
| 14:35.14 | [SySteM] | :( |
| 14:35.35 | [TK]D-Fender | [10:21]<Faustov>now what I'm still missing is a) push-to-talk b) indication who is speaking atm <- b = EASY TO WRITE A SCRIPT FOR |
| 14:36.15 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:36.32 | [TK]D-Fender | ijpalmer: ${DIALSTATUS} |
| 14:36.32 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:36.48 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:37.21 | Faustov | [TK]D-Fender: do you have a specific client in mind? |
| 14:37.43 | *** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk) |
| 14:37.50 | ijpalmer | Thanks [TK]D-Fender |
| 14:38.28 | [TK]D-Fender | ~toywy |
| 14:38.29 | infobot | i guess toywy is The one you write yourself. |
| 14:38.31 | [TK]D-Fender | Faustov: ^^^ |
| 14:38.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:39.52 | Faustov | [TK]D-Fender: I'm under the impression I'd be doubling someone's efforts, I'm in fact quite sure someone already tried writing a softphone which has ptt and a gui display of conf call members |
| 14:40.22 | [TK]D-Fender | Faustov: Sure.. everyone is trying to turn one product into a direct equivalent of another. |
| 14:40.37 | [TK]D-Fender | Faustov: best of luck in your search for it of course... |
| 14:40.45 | Faustov | thanks |
| 14:41.07 | [TK]D-Fender | Faustov: A softphone that also can specifically tap into * to check confrence speakers... |
| 14:41.32 | [TK]D-Fender | Faustov: This sould be something commercial at best. They pieces are eay, the package someone will want real money for |
| 14:42.01 | Faustov | yup, there's money in it ;) |
| 14:42.32 | [TK]D-Fender | Faustov: Beacuse the only people who'll want it are pretty much for commercial use |
| 14:43.14 | *** join/#asterisk micols (~mio@cl-85.cph-01.dk.sixxs.net) |
| 14:43.34 | Faustov | [TK]D-Fender: Maybe, I'd say whoever does not want a closed, slowly evolving solution like teamspeak or ventrilo, wich such client would be able to craft whatever he needs |
| 14:43.54 | Faustov | so whoever wants to play around with asterisk possibilities will be interested |
| 14:45.49 | Skeeter- | [TK]D-Fender, how would you provide buddy watch support ith polycoms over an IAX trunk, u already figured me sth out but i cant remember what it was |
| 14:46.18 | [TK]D-Fender | Skeeter-: I don't |
| 14:46.46 | Skeeter- | it was sth about making another iax trunk and send device_stats via dialplans to the new trunk |
| 14:46.56 | Skeeter- | device_state |
| 14:47.14 | [TK]D-Fender | Skeeter-: http://www.russellbryant.net/blog/2008/06/10/asterisk-16-now-with-distributed-presence/ |
| 14:47.38 | Skeeter- | there is some prerequist that i cant find for that |
| 14:48.03 | [TK]D-Fender | Skeeter-: And your problems are nameless as always... |
| 14:48.44 | *** join/#asterisk dewinda (~sahan@123.231.97.117) |
| 14:48.50 | Skeeter- | when settings make menuconfig, under ressoucres_module res_ais is marked [XXX] cuz i need some prerequist |
| 14:49.25 | [TK]D-Fender | Skeeter-: Novel idea : FULFILL THE DAMN PRE-REQUISITE |
| 14:57.05 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 14:59.14 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.00 | Skeeter- | [TK]D-Fender, i did apt-get install openais |
| 15:00.34 | Skeeter- | i wonder why no one else wants that feature so badly |
| 15:01.23 | [TK]D-Fender | Skeeter-: You mean badly enough to install the pre-reqs? |
| 15:01.49 | [TK]D-Fender | Skeeter-: Apparently we HAVE this feature. |
| 15:02.08 | Skeeter- | http://svnview.digium.com/svn/asterisk/branches/1.6.1/doc/distributed_devstate.txt?view=markup i followed this |
| 15:02.15 | [TK]D-Fender | Skeeter-: Do I ahve to "badly want" the basic SIP phone support I already have? Should I be raving why no-one else wants it |
| 15:03.36 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 15:03.38 | [TK]D-Fender | Skeeter-: or is the fact that no-one is talking about it a testament tot he fact it work, therefor why discuss it? |
| 15:04.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:04.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:04.08 | Skeeter- | i dont understand what u just said |
| 15:04.26 | Gido-E | Skeeter- /me to :-) |
| 15:04.30 | [TK]D-Fender | [11:00]<Skeeter->i wonder why no one else wants that feature so badly <- Who says they don't want it? We already HAVE it. |
| 15:04.36 | mort_gib | Skeeter-: Don't worry you'll get used to TK before long! :-) |
| 15:04.46 | Gido-E | i think [TK]D-Fender is in a happy mood |
| 15:04.47 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 15:04.57 | Skeeter- | Gido-E, better then ever |
| 15:05.00 | [TK]D-Fender | hasn't killed anyone in over a WEEK |
| 15:05.08 | Gido-E | :-) |
| 15:05.12 | Skeeter- | see |
| 15:05.21 | freezey | hey when dialing a external number... in order to not use the 1 before the number would the dialplan look like NXXNXXXXXX |
| 15:05.23 | freezey | ? |
| 15:05.37 | Skeeter- | [TK]D-Fender, we have it , it doesnt work for me |
| 15:05.50 | Skeeter- | no one SEEMS to be using it |
| 15:11.11 | Kobaz | anyone familiar with adtran ta904 |
| 15:12.52 | *** join/#asterisk blaines (~blaines@c-98-213-119-125.hsd1.il.comcast.net) |
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| 15:14.28 | leifmadsen | Skeeter-: what feature? |
| 15:14.45 | Skeeter- | leifmadsen, i dont wanna get another person into that |
| 15:14.52 | Skeeter- | device_state over IAX |
| 15:15.04 | [TK]D-Fender | Skeeter-: Who says hundreds of people aren't using it? |
| 15:15.07 | Skeeter- | goal is to have buddy watch working on polycoms over IAX |
| 15:15.20 | Skeeter- | [TK]D-Fender, me |
| 15:15.30 | [TK]D-Fender | Skeeter-: Based on what? |
| 15:15.33 | leifmadsen | define your topology, because obviously Polycom doesn't support IAX |
| 15:15.34 | Zeeek | Official time is BEER O CLOCK |
| 15:15.36 | Skeeter- | 's opinion doesnt count |
| 15:15.49 | [TK]D-Fender | [11:05]<Skeeter->[TK]D-Fender, we have it , it doesnt work for me <- you haven't shown us the problem. |
| 15:15.51 | Zeeek | topical anesthetic |
| 15:16.01 | Katty | peeks in |
| 15:16.04 | Katty | hugs on Zeeek |
| 15:16.06 | [TK]D-Fender | pokes out |
| 15:16.18 | Skeeter- | [TK]D-Fender, i cant even install/compile it |
| 15:16.31 | Zeeek | {{{{{Katty}}}}} |
| 15:16.38 | Skeeter- | leifmadsen, well, device_state can be sent via iax with res_ais |
| 15:16.46 | [TK]D-Fender | Skeeter-: You've shown us nothing. Results are likely to scale accordingly. |
| 15:16.46 | leifmadsen | it's not sent via IAX |
| 15:16.50 | leifmadsen | it's sent via AIS |
| 15:16.51 | Naikrovek | [TK]D-Fender: lol |
| 15:16.59 | Naikrovek | i'm using that |
| 15:17.04 | Zeeek | UPS works best for us |
| 15:17.15 | Zeeek | no ports needed |
| 15:17.21 | hardwire | Just sign here. |
| 15:17.22 | [TK]D-Fender | Naikrovek: CreativeCommons Licensed |
| 15:18.07 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 15:18.51 | leifmadsen | M16573 |
| 15:18.54 | MuffinMan | [assigned] [Asterisk] Core/General 0016573: [patch] [regression] iaxclient (tested with Zoiper) registered to asterisk shows devicestate Unavailable instead Not-InUse reported by nenadr https://issues.asterisk.org/view.php?id=16573 |
| 15:19.04 | leifmadsen | note that device_states with IAX2 are not nearly as robust at SIP device_states |
| 15:19.43 | leifmadsen | M16573#0119180 |
| 15:19.46 | MuffinMan | [16573#0119180] Commented by russell on 2010-03-09 at 12:52: While the text says "unknown", app_queue should continue to work properly. If that is not the case, please clarify the call scenarios that are not working properly. As indicated in previous updates, this is a side effect of the device |
| 15:19.46 | MuffinMan | ..state handling for IAX2, but will not cause problems as far as I know. |
| 15:19.46 | MuffinMan | .. |
| 15:19.46 | MuffinMan | ..The patch to the devicestate core is not something we can do as it will change behavior in many cases. |
| 15:20.57 | Skeeter- | so its in the working |
| 15:21.02 | leifmadsen | no |
| 15:21.08 | leifmadsen | read the entire issue |
| 15:21.09 | Skeeter- | this is too technical for me |
| 15:21.11 | leifmadsen | that issue should really just be closed |
| 15:21.25 | leifmadsen | iax2 does not support device states like SIP does |
| 15:21.38 | Scotty | We're givin' it all we can Captain! |
| 15:21.39 | Qwell | Scotty: O.o |
| 15:21.48 | Skeeter- | Morgan! |
| 15:21.50 | Scotty | She's gonna blow soon |
| 15:22.02 | Skeeter- | anyway |
| 15:22.13 | leifmadsen | if it works, it works, if it doesn't, then it doesn't. |
| 15:22.30 | Bones | damn it Jim, I'm just a country doctor |
| 15:23.03 | James_Tiberius_K | STand down! |
| 15:23.57 | bmoraca_work | yay for sed! |
| 15:24.15 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:25.01 | Skeeter- | http://img717.imageshack.us/img717/79/40614165.png : installing openais, with the right configs AND starting openais gives the same result |
| 15:27.34 | leifmadsen | you don't have the OpenAIS development libraries installed then |
| 15:27.51 | leifmadsen | Asterisk has to be able to compile against the development libraries |
| 15:28.29 | leifmadsen | starting OpenAIS doesn't actually mean anything to Asterisk. There is a guide in the doc/ directory I followed and it worked pretty much the first time. Have you been following that documentation? |
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| 15:32.27 | Skeeter- | i used |
| 15:33.16 | Skeeter- | http://svnview.digium.com/svn/asterisk/branches/1.6.1/doc/distributed_devstate.txt?view=markup |
| 15:36.22 | *** part/#asterisk _penfold1972_ (~penfold19@66-194-25-11.static.twtelecom.net) |
| 15:36.38 | snot | http://pastie.org/911615 - asterisk wont start. thats what I have in my log. any ideas? please let me know if additional information is needed |
| 15:36.54 | snot | standard 1.4 from the debian lenny repo |
| 15:37.08 | *** part/#asterisk aceio (~90fe6658@gateway/web/freenode/x-kfscnydknzorrztv) |
| 15:39.29 | Qwell | snot: Did you do something silly like install multiple voicemail packages? |
| 15:39.34 | Qwell | (hint: you did) |
| 15:39.37 | leifmadsen | snot: looks like it's trying to load multiple versions of app_voicemail -- look in /usr/lib/asterisk/modules/ for app_voicemail* and see if you have multiple values. Somethign like app_voicemail_odbc.so or whatever |
| 15:39.42 | leifmadsen | Qwell: ;) |
| 15:40.03 | leifmadsen | snot: use modules.conf to noload => app_voicemail_odbc.so (or whatever the modules names the package installed) |
| 15:40.09 | bmoraca_work | i thought that menuselect prevented you from doing that... |
| 15:40.22 | leifmadsen | bmoraca_work: there is no menuselect when installing from packages |
| 15:40.26 | bmoraca_work | oh |
| 15:40.30 | bmoraca_work | reading ftw |
| 15:40.32 | leifmadsen | he installed from the debian lenny repo |
| 15:40.42 | leifmadsen | yet another reason it is better to install from source :) |
| 15:40.46 | bmoraca_work | indeed |
| 15:40.59 | Zeeek | Last call for beer: join us anytime on #vuc - http://vuc.me or call in and talk via SIP. YOu're VoIP users after all |
| 15:41.04 | *** part/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
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| 15:44.07 | snot | asterisk:~# ls /usr/lib/asterisk/modules/app_voicemail* |
| 15:44.07 | snot | /usr/lib/asterisk/modules/app_voicemail_imap.so /usr/lib/asterisk/modules/app_voicemail_odbc.so /usr/lib/asterisk/modules/app_voicemail.so |
| 15:44.11 | snot | appeasrs so :) |
| 15:45.05 | *** join/#asterisk mhaddog (~mhaddog@83.33.30.137) |
| 15:45.57 | snot | I'm a bit confused... what do I need to purge? |
| 15:45.58 | snot | http://pastie.org/911635 |
| 15:47.15 | snot | asterisk-sounds-extra was the key to it |
| 15:48.05 | snot | damn! it wasnt! |
| 15:48.09 | snot | still not working |
| 15:48.50 | snot | asterisk:~# /etc/init.d/asterisk start && ps aux | grep aster |
| 15:49.02 | snot | that returned asterisk 5560 0.0 2.4 241324 6220 ? Rsl 17:47 0:00 /usr/sbin/asterisk -p -U asterisk |
| 15:49.10 | snot | but it wasnt running for long |
| 15:49.14 | idespinner | anyone here know of a zoip port to 1.6? |
| 15:50.59 | *** join/#asterisk elzid (~IceChat7@host81-143-42-174.in-addr.btopenworld.com) |
| 15:52.24 | elzid | hey - can someone shed some light on the extent to which dns srv lookups are implemented in asterisk 1.6? |
| 15:53.12 | *** join/#asterisk Ad-Hoc (~nimbus@194.219.186.16.dsl.dyn.forthnet.gr) |
| 15:56.52 | elzid | voip-info doc says dns srv lookups are partially implemented - how partial is partial? |
| 15:57.17 | elzid | priorities/weights? Yet the functions/logic exists in 1.4/1.6 code... |
| 15:57.19 | Qwell | voip-info is rarely accurate/up-to-date |
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| 16:00.02 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
| 16:00.19 | elzid | yes it seems - but the last update date is farely recent... |
| 16:00.38 | elzid | has anyone tried and tested dns srv lookups - successfully? |
| 16:03.13 | leifmadsen | define "successfully" |
| 16:03.25 | leifmadsen | I'm using 1.6.2 and DNS SRV lookups seem to work for me |
| 16:09.43 | elzid | @leifmadsen: sorry - are you able to dial to an FQDN via a dns srv lookup, selecting the correct DNS target applying the priorities and weightings set? i.e. can you send traffic to a peer who is load balancing via SRV records in DNS? |
| 16:10.14 | leifmadsen | elzid: weightings don't work in Asterisk because it'll just select the first one returned |
| 16:10.26 | snot | leifmadsen: I added noload => app_voicemail_odbc.so to the end of modules.conf but it's still the same warnings in the log |
| 16:10.45 | leifmadsen | snot: perhaps you have other modules also listed |
| 16:11.07 | snot | leifmadsen: I'll nopaste, 2 sec |
| 16:11.33 | snot | http://pastie.org/911706 |
| 16:11.35 | snot | leifmadsen: there |
| 16:11.43 | snot | leifmadsen: danish btw.? |
| 16:12.00 | leifmadsen | danish heritage, but only speak english :) |
| 16:12.25 | snot | leifmadsen: oh I see. very common name(s) in denmark |
| 16:12.44 | leifmadsen | yep :) I'm John Smith in DK :) |
| 16:13.10 | elzid | @leifmadsen: so only priorities are implemented? so noway to load balance? |
| 16:13.18 | leifmadsen | right |
| 16:13.35 | leifmadsen | as is my understanding anyways |
| 16:13.42 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 16:13.43 | snot | leifmadsen: indeed, my last name is jensen which is just as common |
| 16:13.44 | elzid | "srv_callback() function puts the |
| 16:13.44 | elzid | records returned by the DNS lookup into priority order (lowest numbersfirst), and then the process_weights() function sorts by weight within each priority." |
| 16:13.51 | leifmadsen | full DNS SRV weighting and priority stuff is not implmented -- just the first record returned is used |
| 16:14.03 | snot | leifmadsen: did you have an idea about what I nbeed to adjust in my modules.conf? |
| 16:14.14 | leifmadsen | snot: I don't -- doing too many things at once |
| 16:14.37 | elzid | @leifmadsen: thanks - can I take this as the "current" situation? |
| 16:14.39 | Naikrovek | wernerjagermanjenson |
| 16:15.01 | leifmadsen | elzid: do you also have dnsmgr enabled? |
| 16:15.02 | snot | leifmadsen: np mate, if you find the time please take a look at http://pastie.org/911706 - otherwise I'll try/ask later |
| 16:15.40 | leifmadsen | elzid: the lookup won't be done each time you place a call -- the DNS SRV records will be updated whenever the dnsmgr is set to update the records in memory |
| 16:15.52 | leifmadsen | elzid: you could potentially do it if you have a low refresh time |
| 16:16.38 | elzid | @leifmadsen: but how would multiple lookups effect anything if srv lookup returns the first line always...? |
| 16:16.51 | leifmadsen | not sure |
| 16:17.02 | leifmadsen | I've never tried doing anything failover with SRV records |
| 16:17.11 | leifmadsen | I don't believe Asterisk is going to do what you want though |
| 16:17.22 | leifmadsen | I know SRV record implementation is minimal |
| 16:19.17 | elzid | @leifmadsen: mmm - I'm more interested in load balancing rather than failover... its frustrating that the situation's not changed in at least the last 5/6 years... I see posts from 2004 mentioning incomplete srv lookup features! Anyway, perhaps openser (or its newer branches) is the way to go for load balancing? frustrating because it needs more budget as its a new network component... |
| 16:19.58 | leifmadsen | elzid: well apparently no developer has been interested enough to move SRV records further, so.... |
| 16:20.12 | Qwell | what he's saying is: patches are welcome |
| 16:20.20 | leifmadsen | elzid: using opensips or kamailio or something may be the way to do that... just send the call to that application, and have it do the load balancing stuff |
| 16:21.24 | elzid | @leifmadsen: yeh - I 'spose that's the only way it's going to happen - many thanks for your time mate |
| 16:21.51 | snot | leifmadsen: in case you or anyone else care... a few more noloads was the key to it |
| 16:21.57 | snot | (more errors to come though) |
| 16:27.11 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 16:32.34 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:43.00 | *** join/#asterisk shinao1 (~shinao1@41.219.152.42) |
| 16:43.05 | shinao1 | hi community, i am suddenly getting 'all-circuits-are-busy' errors when i try to make calls over my dahdi trunks.. and in the logs i see this error message: 'Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)' which leaves me utterly confused. How do i deal with this please? |
| 16:44.10 | *** join/#asterisk RobH (~robh@2620:0:860:2:21e:c2ff:fe03:2465) |
| 16:44.30 | paulc | shinao1: Are you using POTS lines or T1/E1? |
| 16:45.02 | shinao1 | POTS--fxo lines |
| 16:46.04 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
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| 16:49.56 | Qwell | shinao1: Asking the same question in multiple channels is very rude. |
| 16:50.38 | shinao1 | my apologies.. i shall curb my enthusiasm from now on. |
| 16:51.38 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 16:52.13 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:52.15 | cusco | hi |
| 16:52.36 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 16:54.14 | cusco | hi... |
| 16:54.35 | cusco | Telco is stating that we mark our outboud calls as international calls even tho they are national |
| 16:55.35 | *** join/#asterisk wimt (wimt@freenode/staff/wikipedia.wimt) |
| 16:55.50 | jblack | Are they staying they're too stupid to handle the routing correct? |
| 16:56.15 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 16:57.02 | *** join/#asterisk btsteve (~tstevens@24-196-234-39.dhcp.gwnt.ga.charter.com) |
| 16:58.44 | hluesea | hello, i want to run a cli command via php on website but when i try to run it turns me (Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) ) error... When i try to run it from command panel it is working when i tried to run from my php it is successfull from same server but i can't run it from web. Can anyone know the solution ? |
| 17:00.03 | cusco | jblack: maybe... thy say it messes their accounting... |
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| 17:00.12 | cusco | anyway this is what they sen't me http://i42.tinypic.com/2hxssz.jpg |
| 17:00.21 | cusco | they complain about TYPE_B number |
| 17:00.52 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 17:01.05 | Qwell | they're double-charging you? |
| 17:01.21 | cusco | Qwell: they are not, only asking me to change that... |
| 17:02.12 | cusco | Im not sure what is the dahdi parameter to set it right... |
| 17:02.35 | cusco | I had switchtype=dms100 |
| 17:02.40 | cusco | just changed it to national |
| 17:03.08 | cusco | not sure if that's it tho... |
| 17:04.08 | Corydon76-dig | Most of the switchtypes are ni-2, just variants thereof |
| 17:05.05 | Corydon76-dig | That's probably not what you want. What it looks like is that they want you to change the pridialplan |
| 17:06.05 | Corydon76-dig | In most cases, you can set it to 'unknown' and it will just work. If they're adamant that it be correct, set it to 'dynamic' |
| 17:06.48 | Corydon76-dig | and the prefix sent to dahdi will determine the Type Of Number sent |
| 17:07.11 | Corydon76-dig | Prefix of 1 is national, 011 is international, anything else is local |
| 17:08.13 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:09.31 | Corydon76-dig | Just under the pridialplan section in chan_dahdi.conf are the exact settings for which prefix sets which TON |
| 17:09.43 | Corydon76-dig | Just know that the prefix is stripped |
| 17:11.22 | anonymouz666 | Corydon76-dig: the ISDN book you bought is really good! :P |
| 17:12.33 | *** join/#asterisk Fubard (~brawr@173.10.235.205) |
| 17:12.42 | Corydon76-dig | Eh? |
| 17:13.58 | Corydon76-dig | I'm relying on memory of how it's implemented in chan_dahdi, having spent a good amount of time in that particular code |
| 17:14.18 | Corydon76-dig | and also having read the spec |
| 17:14.58 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 17:22.11 | Katty | i joined curves. |
| 17:23.32 | hardwire | Katty: congrats |
| 17:23.41 | hardwire | I joined the get my ass outside and walk the dogs club. |
| 17:24.51 | *** join/#asterisk aidinb (~Aidin@66-214-28-176.dhcp.lnbh.ca.charter.com) |
| 17:27.09 | Katty | hardwire: that's great! |
| 17:27.35 | hardwire | I do it often enough as is.. but now that we have sun again they are a lot longer :) |
| 17:30.35 | jblack | I stopped eating carbs. ;) |
| 17:30.43 | jblack | I have lost 17 pounds so far |
| 17:31.28 | anonymouz666 | no carbs, no energy |
| 17:32.08 | jblack | That's about 1/3 true. |
| 17:32.39 | ariel_ | I need my bread. |
| 17:32.47 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 17:33.33 | jblack | ariel_: Yeah. A life without bread, potatos, fruits, veggies, pasta, pancakes and most veggies is not easy. |
| 17:33.35 | ariel_ | hardwire: you up to some t/s |
| 17:34.08 | ariel_ | ohhhh god.... pancake (Simpson tone) |
| 17:34.42 | *** join/#asterisk Cherebrum (jgarland@209.9.237.93) |
| 17:35.57 | Cherebrum | I would like to donate some $$$ to the asterisk project in exchange for having something contributed to the code. Who can I talk to? |
| 17:36.30 | florz | there is no such thing as "the asterisk project" |
| 17:36.43 | Cherebrum | that sucks |
| 17:37.07 | florz | there is digium, who publish the canonical asterisk version, and who certainly can do development work for money |
| 17:38.29 | florz | and there are third parties who do provide similar services, just without any guarantees that things will end up in the canonical version, obviously |
| 17:38.55 | Cherebrum | This would go into the dialplan |
| 17:39.11 | Cherebrum | the example dialplan |
| 17:39.13 | Cherebrum | not the code itself |
| 17:39.37 | TJ^ | u promoting something in the dialplan? |
| 17:39.59 | Cherebrum | not really... |
| 17:40.05 | Cherebrum | I run tollfreetollfree.com |
| 17:40.08 | Cherebrum | it's a free service |
| 17:40.15 | TJ^ | florz canonical, are the same guys who own ubuntu? |
| 17:40.29 | Cherebrum | and I am willing to donate to have my service used by default in the example for calling tollfree numbers |
| 17:40.42 | Cherebrum | it's mutually benefitial. :) |
| 17:40.45 | florz | TJ^: no, canonical is just an english word |
| 17:40.51 | p3nguin | How could it be used by default when there is no working default dialplan? |
| 17:41.02 | Cherebrum | there is a demo |
| 17:41.05 | p3nguin | No default. |
| 17:41.18 | TJ^ | Cherebrum take the source and make ur own? |
| 17:41.25 | p3nguin | The sample extensions.conf is not usable. |
| 17:41.57 | Cherebrum | I have not interest in using asterisk for anything myself |
| 17:42.01 | Cherebrum | er no |
| 17:42.58 | Cherebrum | I just want to make it easy for people to discover and make use of my free service |
| 17:42.59 | TJ^ | Cherebrum looks like a good idea |
| 17:43.05 | TJ^ | what is it exactly tho |
| 17:43.06 | shinao1 | um also, how do you clear these kinds of errors? I noticed this as well 'WARNING[3671] chan_dahdi.c: Detected alarm on channel 24: Red Alarm' |
| 17:43.16 | Qwell | Cherebrum: you aren't really familiar with how open source projects work, are you? |
| 17:43.17 | shinao1 | my channels are all congested |
| 17:43.24 | p3nguin | tj^: It's free toll free termination. |
| 17:43.25 | TJ^ | a sip trunk to your servers that gives free calls to toll free's? |
| 17:43.36 | Cherebrum | exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@tollfreetollfree.com) |
| 17:43.36 | idespinner | shinao1, you fix that by calling the telco or plugging the t1 cable back in... |
| 17:43.49 | Cherebrum | something like that in the example dialplan |
| 17:44.26 | shinao1 | its fxo ports |
| 17:44.31 | shinao1 | but ok ill try that |
| 17:44.44 | idespinner | same deal... |
| 17:45.08 | idespinner | fxo ports detecting no carrier signal. have you hooked up a buttset to the lines? |
| 17:45.43 | TJ^ | shinao1 did u just install the fxo card? |
| 17:47.26 | Cherebrum | Qwell: I am familiar. Asterisk also isn't exactly opensource either |
| 17:47.27 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:47.38 | p3nguin | It isn't? |
| 17:47.39 | Qwell | ... |
| 17:47.45 | p3nguin | How do I have the source of it? |
| 17:47.51 | p3nguin | Theft? |
| 17:47.51 | Qwell | I'd love to hear the answer to this |
| 17:47.52 | Cherebrum | You have to sign away rights to your code in order to contribute |
| 17:48.04 | Qwell | Cherebrum: how about you actually *READ* the license agreement? |
| 17:48.06 | TJ^ | im new to asterisk (less than a week working on it) and I know its opensource |
| 17:48.28 | TJ^ | Qwell u know as well as i do no one reads those :P |
| 17:48.32 | Cherebrum | Asterisk has two license agreements available, and they aren't compatible with each other |
| 17:48.33 | idespinner | ABE isnt open source |
| 17:48.37 | idespinner | he has a technicality |
| 17:48.42 | Deeewayne | signs away Qwell |
| 17:48.44 | idespinner | its dual licensed |
| 17:48.47 | p3nguin | Asterisk is still open, though. |
| 17:48.57 | p3nguin | ABE isn't Asterisk. |
| 17:48.59 | Qwell | Asterisk Business Edition != Asterisk |
| 17:49.02 | Qwell | p3nguin: exactly |
| 17:49.11 | TJ^ | what is asterisk business edition? |
| 17:49.20 | Corydon76-dig | Well, it is, but it's not Asterisk Open Source |
| 17:49.21 | shinao1 | no TJ^ |
| 17:49.21 | Cherebrum | but ABE uses most of the asterisk code base |
| 17:49.23 | idespinner | err well, thats not quite how didigum marketed it |
| 17:49.38 | *** part/#asterisk The_Canuck (~The_Canuc@adsl-67-38-81-158.dsl.sfldmi.ameritech.net) |
| 17:49.38 | p3nguin | tj^: A non-free, commercial implementation of Asterisk, I guess. |
| 17:49.49 | idespinner | its all semantics really |
| 17:49.54 | TJ^ | shinao1 in the asterisk CLI have you tried "dahdi show status" ? |
| 17:50.02 | p3nguin | Semantics? Really? |
| 17:50.08 | idespinner | yea |
| 17:50.11 | idespinner | semantics |
| 17:50.26 | Corydon76-dig | ABE is marketed under a different license and also contains some code that is not in open source |
| 17:50.30 | Cherebrum | I guess maybe I need to contact the Digium business office |
| 17:50.44 | idespinner | abe is EOL... |
| 17:50.45 | shinao1 | yes i have TJ^ |
| 17:50.52 | TJ^ | alarm? |
| 17:50.56 | Qwell | idespinner: i don't think so |
| 17:51.02 | idespinner | digium has moved to 'open source asterisk' with support contracts |
| 17:51.14 | idespinner | Qwell, no to ABE going eol? |
| 17:51.16 | shinao1 | alarms section shows OK on all lines |
| 17:51.29 | TJ^ | 0o |
| 17:51.40 | Corydon76-dig | idespinner: it's simply not being marketed to the small business community |
| 17:51.47 | TJ^ | shinao1 woudldn't know what to do past that im too much of a noob |
| 17:51.54 | shinao1 | ok |
| 17:51.58 | TJ^ | but you should make a call and pastebin the log from asterisk -r |
| 17:52.00 | shinao1 | thanks anyway |
| 17:52.18 | idespinner | the understanding I was given from our reps was that ABE was going EOL in a year or so... |
| 17:52.37 | Corydon76-dig | idespinner: As a product, it is |
| 17:53.38 | Corydon76-dig | idespinner: as a branch which is source-licensed to others to modify and build their own products, it's not |
| 17:53.39 | idespinner | Corydon76-dig, maybe theres something i'm missing. Is abe being rebranded, sold to only specific markets, something else? |
| 17:54.11 | idespinner | you mean by how its now referenced as a 'telephony toolkit' |
| 17:54.37 | TJ^ | guys for a small business less than 10 extensions which "flavour" of asterisk would u reccommend? ie, asterisk now, plain old vanilla, vanilla + freepbx, trixbox... etc... |
| 17:54.47 | Corydon76-dig | ABE as a binary-only product is EOL. ABE as a source-product is very much alive |
| 17:54.56 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-172-249.clienti.tiscali.it) |
| 17:55.06 | [TK]D-Fender | TJ^: depends what you want to do |
| 17:55.58 | TJ^ | [TK]D-Fender got the FXO working |
| 17:56.01 | TJ^ | it was the RJ11 |
| 17:56.13 | Corydon76-dig | idespinner: note that there's also stuff in open source that is not currently in ABE, either, but may be merged at some future time |
| 17:56.16 | [TK]D-Fender | TJ^: UK wiring FTL? |
| 17:56.45 | idespinner | Corydon76-dig, i'm curious as i've installed ABE here on test machines. What does ABE have that open source doesnt? |
| 17:56.51 | TJ^ | [TK]D-Fender yea went to the store bought an RJ11 that was dedicated for modems and it worked straight away |
| 17:57.03 | [TK]D-Fender | TJ^: \o/ |
| 17:57.06 | idespinner | to me it's only a 'more regression tested source' |
| 17:57.08 | Qwell | idespinner: support |
| 17:57.16 | idespinner | yes, and support |
| 17:57.45 | Corydon76-dig | idespinner: support and methods to ensure that ABE support is given only to those with proper ABE licenses |
| 17:57.58 | idespinner | lol, well yea |
| 17:57.58 | TJ^ | [TK]D-Fender might sound like an obvious question, but what makes more expensive FXO cards better than a xp100se? |
| 17:58.15 | idespinner | but I meant, anything feature wise... |
| 17:58.31 | idespinner | yea we get an extra ./registerbe |
| 17:58.34 | Corydon76-dig | idespinner: a few extra connectors to commercial-only backends |
| 17:58.39 | [TK]D-Fender | TJ^: duplex issues, PCI stability (X100 = shit), CID is flakey on those, etc |
| 17:58.54 | TJ^ | 0o... |
| 17:59.12 | TJ^ | duplex? PCI as in the interface? |
| 17:59.18 | [TK]D-Fender | TJ^: hardware echo cancellation options, etc |
| 17:59.34 | Corydon76-dig | idespinner: I don't think anything that you couldn't get by purchasing those commercial modules and linking them to Asterisk Open Source, though |
| 18:00.02 | [TK]D-Fender | TJ^: X100p are nasty when sharing interrupts, los of clocking can cause timeing issues and all sorts of other bad stuff |
| 18:00.12 | TJ^ | so which FXO card would u recommend for a small business that had less than 10 users? |
| 18:00.43 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 18:00.46 | idespinner | Corydon76-dig, i'm pulling the latest abe documentation now... |
| 18:00.58 | Corydon76-dig | and to be clear, those commercial modules are items where our hands are tied... licensing of those technologies are what make them commercial-only |
| 18:01.32 | idespinner | the only few i can think of are cepestral and lumenvox |
| 18:01.40 | TJ^ | can anyone recommend a good fxo carD? |
| 18:02.13 | Corydon76-dig | Usually because there are some patents with vicious maneating lawyers behind them |
| 18:02.14 | ariel_ | hello Devon_ |
| 18:02.29 | idespinner | TJ^, TDM410p or AEX410 |
| 18:02.31 | Devon_ | hello Ariel |
| 18:02.49 | [TK]D-Fender | TJ^: don't count users.. count lines |
| 18:03.10 | TJ^ | 1 line |
| 18:04.30 | *** join/#asterisk blaines (~blaines@209.94.61.126) |
| 18:05.08 | idespinner | looks like the ABE manual has turned into mostly and asterisk-gui manual... :/ |
| 18:05.32 | TJ^ | idespinner AEX410 <---- $615!!! |
| 18:05.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:05.37 | TJ^ | thats a lot |
| 18:05.54 | TJ^ | for such a small company |
| 18:06.00 | idespinner | for 1 port of fxo? |
| 18:06.07 | idespinner | you could get by with a spa3102 |
| 18:06.09 | Qwell | TJ^: feel free to price "other" PBX vendors |
| 18:06.19 | Qwell | if you think that's a lot, you'll cry when you see theirs |
| 18:06.33 | idespinner | Qwell, thats certainly true |
| 18:06.35 | TJ^ | ive seen some cards in the 1000's |
| 18:06.58 | TJ^ | but this is for a small company with one line and a low budget |
| 18:07.01 | Deeewayne | I have some cards that used to be in the 15,000's |
| 18:07.12 | Qwell | TJ^: then you don't need a quad module.. |
| 18:07.19 | idespinner | a spa3102 is pretty cheap both in cost and in function so dont use it for mission critical stuff... |
| 18:07.22 | TJ^ | and everyone flammed me when i asked bout the x100's lol |
| 18:07.31 | Qwell | and you probably don't need HWEC either |
| 18:07.35 | [TK]D-Fender | TJ^: You seem to need 1 little line. Stick witht eh card you have until it isn't good enough |
| 18:07.59 | Qwell | Deeewayne: you should ebay them for full former retail |
| 18:08.33 | Deeewayne | yes, that would be awesome, but they were given to me for 'research purposes only' |
| 18:08.40 | Qwell | bah! :) |
| 18:08.48 | Deeewayne | and these days, they are worth much less |
| 18:09.09 | TJ^ | there isnt an entry level FXO for under £100? |
| 18:09.45 | Deeewayne | I need a certain canadian to help me out :-) |
| 18:09.55 | [TK]D-Fender | TJ^: Closest is the Linksys SPA-3102 |
| 18:13.23 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 18:16.09 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:17.10 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:21.28 | TJ^ | openvox any good? |
| 18:21.40 | Qwell | TJ^: are any chinese clones good? |
| 18:21.48 | Qwell | (hint: no0 |
| 18:21.52 | TJ^ | lol |
| 18:22.06 | TJ^ | Qwell im asking cos i dont know |
| 18:22.09 | skeptikal | Linksys SPA-3102 |
| 18:22.10 | TJ^ | openvox are clones? |
| 18:24.40 | anonymouz666 | I think so |
| 18:25.03 | Deeewayne | puts another dime in Qwell |
| 18:25.15 | Deeewayne | Qwell, I'm enjoying your comments this afternoon |
| 18:26.06 | Corydon76-dig | TJ^: most of their cards are |
| 18:26.07 | Qwell | Deeewayne: don't mind me - I'm just procrastinating breaking a bunch of systems |
| 18:26.24 | Qwell | wonders how many people really use 1.6.0 with AsteriskNOW |
| 18:26.42 | *** join/#asterisk rubbs (~rubbs@cpe-71-72-56-140.neo.res.rr.com) |
| 18:26.43 | Deeewayne | I'm procrastinating working on a tcl script because I have to test on windows |
| 18:27.16 | anonymouz666 | TJ^: the openvox cards uses the same drivers as digium ones |
| 18:27.24 | TJ^ | Corydon76-dig so buying an official digium card is the way to go? |
| 18:27.35 | TJ^ | anonymouz666 so what makes em so bad? |
| 18:28.07 | Corydon76-dig | TJ^: not to mention that most of their cards are clones of the previous generation of Digium cards, cards which hit EOL due to various issues with them |
| 18:28.22 | TJ^ | ok thats good to know |
| 18:28.47 | TJ^ | so from what i gather getting a proper digium card seems to be the way to go |
| 18:29.40 | rubbs | Ok, so I was given a server (asteriskNow) and a linksys phone (boo I know), and I have an incoming SIP account. When I call the incoming SIP number from my cell I can see the Active Calls go up one on FreePBX, but I get the "cannot connect to that number" message. I'm guessing its from the asterisk box. I can access the Voicemail of the ext from the phone. What am I missing? |
| 18:30.50 | TJ^ | am i going to get grilled for asking what the best phones to buy are? |
| 18:30.59 | Qwell | TJ^: polycom |
| 18:31.08 | Naikrovek | agreed |
| 18:31.09 | Qwell | rubbs: #freepbx |
| 18:31.10 | rubbs | TJ^: I asked last week, I got Polycom as an answer all the time |
| 18:31.14 | Naikrovek | POL-Y-COM |
| 18:31.23 | rubbs | Qwell: ah, thanks I'll check there. |
| 18:31.32 | TJ^ | they seem nicely priced |
| 18:31.39 | hardwire | anybody think of a way to add dialplan from the dialplan? |
| 18:31.41 | Qwell | they definitely are |
| 18:31.49 | hardwire | yes.. I'm evil |
| 18:31.54 | hardwire | maybe I shold be using a realtime switch |
| 18:31.58 | hardwire | shold/should |
| 18:31.58 | TJ^ | Qwell why they so good? |
| 18:32.05 | Qwell | hardwire: asterisk -rx "extension add ..." |
| 18:32.07 | Qwell | rather |
| 18:32.15 | Qwell | hardwire: exten => _foo,1,System(asterisk -rx "extension add ...") |
| 18:32.22 | Qwell | meta. |
| 18:32.32 | Qwell | TJ^: because they're polycom |
| 18:32.42 | Qwell | that's like asking why cheetos are so amazing |
| 18:32.43 | hardwire | Qwell: I thought about that. |
| 18:32.47 | Naikrovek | TJ^: they're just really, really well designed. easy to configure. easy to diagnose. look good. work good. sound good. they behave. once, my phone bought my dinner. |
| 18:32.53 | hardwire | Qwell: fine.. I'll do it that way |
| 18:33.02 | Qwell | hardwire: that's such a terrible way to do it |
| 18:33.16 | TJ^ | Naikrovek thanks |
| 18:33.30 | Naikrovek | TJ^: these are not uninformed opinions, mind you. |
| 18:33.49 | hardwire | Qwell: bite me in the toe. |
| 18:33.50 | Naikrovek | TJ^: I love polycom because I use them a lot and they've *never* let me down |
| 18:33.56 | TJ^ | cool |
| 18:34.05 | TJ^ | i was going to go for snom or cisco |
| 18:34.05 | hardwire | Qwell: sigh.. you're right.. dialplan reload would be fail. |
| 18:34.09 | hardwire | I need more coffee |
| 18:34.20 | TJ^ | good to know that polycom are good |
| 18:34.47 | Naikrovek | TJ^: others will recommend aastra. those are fine too, i'm sure. just avoid cisco and grandstream |
| 18:34.59 | Naikrovek | one is too expensive, the other too cheap |
| 18:35.11 | TJ^ | we currently have grandstream budgetones as part of the trial phase |
| 18:35.16 | TJ^ | their not too bad |
| 18:35.21 | Naikrovek | not yet they're not |
| 18:35.27 | TJ^ | got 2 of them |
| 18:35.28 | Naikrovek | mine caused all kinds of problems |
| 18:35.54 | hardwire | now to see if I can use astdb from queues.conf as an "extension state" hint. |
| 18:37.56 | TJ^ | oh polycom look good 2 lines |
| 18:38.44 | *** join/#asterisk sun28 (~light@sun28.ipfw.su) |
| 18:38.48 | *** join/#asterisk SaiSoma (~Anth8708@client105.jdcc.edu) |
| 18:38.50 | *** join/#asterisk kuku (~ingo@c-67-175-3-155.hsd1.il.comcast.net) |
| 18:39.06 | kuku | Is asterisk a good tool to send/receive faxes over ip ? |
| 18:40.32 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 18:40.44 | chuckf | if by good you mean it works when it wants to on occasion with some prodding, then yes |
| 18:41.44 | *** join/#asterisk DennisG (~DennisG@84.30.136.208) |
| 18:41.52 | *** join/#asterisk scottsmith7 (~ssmith@64.201.141.80) |
| 18:43.27 | idespinner | was browsing throught the Asterisk 1.4 source and noticed this: "#define NUM_DCHANS 4 /*!< No more than 4 d-channels */" Does this mean we can only have 4 dchannels(4 pri's) max per asterisk instance? |
| 18:43.34 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 18:46.55 | freezey | ok question... so the dundi does a lookup everything works perfect between machines but 1 machine cant transfer it is getting rejected for some reason |
| 18:48.00 | freezey | what i do notice is that on one machine their is a T next to the port when i run iax2 show peers on the other machine thre is not |
| 18:49.23 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 18:50.48 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:51.29 | Fubard | jdoe: i'm noticing the same issue you had yesterday with MWI now retaining message notification; but this is on *1.6, on my *1.4 server it manages MWI correctly. which ver are you using? |
| 18:52.39 | *** join/#asterisk Tech_Travis (~Travis@mail.techglia.com) |
| 18:56.57 | AndyGraybeal | is there a general telecom channel? |
| 18:57.22 | jdoe | Fubard: 1.6.2.6 |
| 19:00.38 | Fubard | jdoe: well atleast we know you're not the only one experiencing it. dunno how to fix it but i know 1.4 doesnt have this issue. likely linked to a feature new to 1.6.2 |
| 19:01.51 | idespinner | Fubard, are these sip channels? |
| 19:02.09 | idespinner | do you have mailbox=1234 in your peer definition in sip.conf? |
| 19:02.19 | idespinner | just a thought, as thats something new in 1.6 |
| 19:03.55 | jdoe | idespinner: yes. |
| 19:06.11 | *** join/#asterisk xpot-mobile (~james@66.60.101.91) |
| 19:08.29 | jdoe | Fubard: well if this is a regression and not just my shitty phones, maybe I'll take a look at how mwi works since I currently have no idea. |
| 19:11.44 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:13.10 | Knightfal | Hey guys if I am changing a single app can I just recompile that particular app |
| 19:13.50 | leifmadsen | yes |
| 19:13.50 | leifmadsen | just run 'make' |
| 19:13.51 | Knightfal | thx |
| 19:13.56 | Knightfal | I thought so just wasnt sure |
| 19:15.19 | *** join/#asterisk DelphiWorld (~Miranda@41.104.108.54) |
| 19:15.25 | *** part/#asterisk DelphiWorld (~Miranda@41.104.108.54) |
| 19:18.09 | jdoe | Fubard: I'll let you know if I find anything. |
| 19:18.11 | *** join/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com) |
| 19:18.17 | *** part/#asterisk Tobarja (~chatzilla@user-0c8h5rb.cable.mindspring.com) |
| 19:18.31 | *** part/#asterisk scottsmith7 (~ssmith@64.201.141.80) |
| 19:19.37 | Knightfal | one more ... how can i get the inbound leg of the calls channel from app_queue peer->name returns Agent/888 but I want the inbound leg such as SIP/Yadda-00000xx |
| 19:30.36 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 19:33.15 | *** join/#asterisk azertyui (~chatzilla@bgl93-7-88-189-218-150.fbx.proxad.net) |
| 19:33.22 | azertyui | hi there |
| 19:33.52 | azertyui | is there anyone here ? |
| 19:34.38 | azertyui | well i need kind of billing management module for my callshop which one do you advice me ppl ? |
| 19:36.39 | Fubard | jdoe: same here, knowing there are 'ghost VMs' floating around somewhere angers me in a way i cant articulate |
| 19:37.34 | bmoraca_work | anyone here use callcentric? |
| 19:37.54 | azertyui | what is that ? bmoraca_work |
| 19:38.06 | bmoraca_work | it's an obscure-ass VOIP provider |
| 19:38.21 | [TK]D-Fender | bmoraca_work: They're pretty big. A few of my clients have used them |
| 19:38.24 | *** join/#asterisk githogori (~githogori@catmint.mail-abuse.org) |
| 19:38.35 | *** join/#asterisk endemic (~endemic@lynx.ipv6.onvox.net) |
| 19:38.44 | bmoraca_work | [TK]D-Fender: i meant that their service delivery is obscure, not that the company was small |
| 19:38.50 | azertyui | does they offer a good service ? |
| 19:39.02 | [TK]D-Fender | bmoraca_work: meanint? |
| 19:39.21 | bmoraca_work | [TK]D-Fender: i don't like the way the talk SIP |
| 19:39.42 | [TK]D-Fender | bmoraca_work: Yeah, thier heading mangling needed to parse the DID is a PITA |
| 19:39.48 | [TK]D-Fender | bmoraca_work: Fugly. |
| 19:39.55 | [TK]D-Fender | bmoraca_work: But functional |
| 19:40.00 | bmoraca_work | if only just |
| 19:40.07 | azertyui | PITA ? |
| 19:40.14 | bmoraca_work | pain in the ass |
| 19:40.26 | azertyui | lol |
| 19:40.47 | TJ^ | anyone know a good phone supplier in the uk? |
| 19:41.02 | azertyui | i was thinking it as in new service like DID |
| 19:41.18 | *** join/#asterisk githogori (~githogori@catmint.mail-abuse.org) |
| 19:41.19 | *** join/#asterisk iheffner (~iheffner@sea02-v600-nat.marchex.com) |
| 19:41.43 | azertyui | good phone supplier ? TJ |
| 19:41.55 | azertyui | good phone supplier ? TJ^ |
| 19:42.04 | TJ^ | yes... |
| 19:42.14 | azertyui | what phone do you need ? |
| 19:42.17 | TJ^ | polycoms specifically |
| 19:42.40 | azertyui | for what purpose ? |
| 19:42.57 | Fubard | presumably to make calls with |
| 19:43.00 | TJ^ | lol |
| 19:43.57 | azertyui | do you need any specific model on polycom ? |
| 19:44.00 | TJ^ | 321 |
| 19:44.02 | TJ^ | 331 |
| 19:44.04 | jdoe | Fubard: a reload triggers the mwi for me as well. For some reason the send_mwi (and adding MWI subs) isn't being triggered on SUBSCRIBE |
| 19:44.39 | Fubard | jdoe: i dont want to jinx it but i think i found it... one sec... |
| 19:44.45 | azertyui | how many do you need ? |
| 19:44.54 | TJ^ | 4 |
| 19:45.34 | kuku | chuckf: Anything that will handle Faxes over ip really well ? |
| 19:47.36 | Fubard | jdoe: nvm, it still does it. when the phone comes back up, i get a notice but when the phone is turned off and receives a message it forces mwi into an amnesic state. completely forgets i had a message waiting. |
| 19:47.44 | *** join/#asterisk Wildy (~simba@91.205.147.94) |
| 19:47.59 | Fubard | but if it knows its gotta message and it reboots, it comes back up knowing there's a message waiting |
| 19:48.35 | jdoe | mine doesn't. comes up as a blank slate (presumably because it's wiping its fs every time) |
| 19:49.13 | Fubard | thats odd. what makes you think its wiping each time? |
| 19:49.15 | [TK]D-Fender | TJ^: how many phones do you need? What kind of call volume? |
| 19:49.48 | TJ^ | four phones |
| 19:49.57 | Fubard | usually you gotta tell it specifically to wipe fs |
| 19:50.03 | jdoe | Fubard: the display on the phone asking me to please wait while it formats ;) |
| 19:50.13 | jdoe | Fubard: yeah, it's a known issue with this firmware. |
| 19:50.21 | Fubard | jdoe: thats a fair assumption to make then lol |
| 19:50.23 | [TK]D-Fender | TJ^: and the call volume & functionality requirements? |
| 19:50.39 | jdoe | Fubard: it's annoying, and more annoying because it's likely the last they'll release for this phone. |
| 19:50.43 | [TK]D-Fender | jdoe: Really... which? |
| 19:52.02 | jdoe | [TK]D-Fender: polycom's 2.1.3 firmware. 2.1.2 doesn't do it. |
| 19:52.14 | [TK]D-Fender | jdoe: What model of phone? |
| 19:52.18 | jdoe | ip500 |
| 19:52.34 | Naikrovek | i think 500s go newer than that. maybe i'm thinking of 501 |
| 19:52.48 | jdoe | 501 goes newer |
| 19:52.50 | [TK]D-Fender | Naikrovek: 2.2 is where they droped the 500/300 |
| 19:52.57 | Naikrovek | oh wow |
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| 19:53.02 | Naikrovek | how old is 2.2 |
| 19:53.03 | [TK]D-Fender | Naikrovek: My home IP 501 is on 3.1.3 currently |
| 19:53.14 | bmoraca_work | i've got a 501 on 3.2 i think |
| 19:53.18 | bmoraca_work | maybe 3.1 |
| 19:53.22 | bmoraca_work | don't remember |
| 19:53.26 | jdoe | yeah. Unfortunately we have a ton of 500s, so 500s are what we're stuck with. |
| 19:53.33 | Naikrovek | 3.1.3 (3.1.6?) is the highest they go on 501 |
| 19:53.38 | leifmadsen | elzid: ping |
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| 19:53.59 | jdoe | so my options are either "phone formats and re-downloads its settings every boot" or "occasional one-way audio" |
| 19:54.03 | jdoe | talk about a crappy choice ;) |
| 19:54.03 | [TK]D-Fender | Naikrovek: SIP 2.2.0 = Aug 17, 2007 |
| 19:54.08 | Naikrovek | wow |
| 19:54.21 | [TK]D-Fender | jdoe: Or pick ANOTHER firmware release |
| 19:55.30 | leifmadsen | elzid: you want func_srv I think |
| 19:55.45 | jdoe | [TK]D-Fender: if I go too early I lose some of the config changes that make management more pleasant. |
| 19:55.56 | leifmadsen | elzid: Set(RESULT=${SRVQUERY(_sip._udp.example.com)}) |
| 19:56.07 | TJ^ | [TK]D-Fender im going after the polycoms 321 |
| 19:56.18 | [TK]D-Fender | jdoe: Keep in mind how old those phoes are. |
| 19:56.34 | [TK]D-Fender | TJ^: Where are they to be placed? |
| 19:56.36 | leifmadsen | elzid: Set(NUM_OF_RECORDS=${SRVRESULT(${RESULT},getnum)}) |
| 19:56.57 | [TK]D-Fender | TJ^: and what kind of call volume? |
| 19:57.05 | leifmadsen | elzid: Set(FIRST_RECORD=${SRVRESULT(${RESULT},1)}) |
| 19:57.14 | leifmadsen | elzid: if you wanted, you could cycle through with a loop I suppose |
| 19:57.22 | hardwire | woa |
| 19:57.25 | hardwire | func_srv |
| 19:57.28 | hardwire | where've you been? |
| 19:57.46 | manxpower | hardwire: in trunk-land |
| 19:57.57 | leifmadsen | hardwire: I think it's new within the last couple of weeks |
| 19:58.11 | jdoe | [TK]D-Fender: believe me, well aware. Unfortunately like I said, we have a bunch so I'm kinda stuck with them. |
| 19:58.11 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:58.11 | Corydon76-dig | leifmadsen: btw, SRVQUERY is not necessary at all |
| 19:58.15 | leifmadsen | elzid: note func_srv is only in trunk |
| 19:58.18 | hardwire | I've been thinking about how to deal with SRV all week |
| 19:58.21 | Corydon76-dig | leifmadsen: the result ID is the same as the query |
| 19:58.23 | jdoe | boss sees "they're cheap", I'm the one that has to fight with them. |
| 19:58.29 | [TK]D-Fender | jdoe: Be prepared to tell management they might not be "happy" |
| 19:58.33 | jdoe | haha. |
| 19:58.57 | [TK]D-Fender | jdoe: How much was "cheap"? |
| 19:58.58 | leifmadsen | Corydon76-dig: assuming you don't want to hit SRVRESULT() a few times |
| 19:59.00 | jdoe | that's probably a non-starter, trying to move away from an overpriced proprietary system that works with the same phones :/ |
| 19:59.14 | Corydon76-dig | leifmadsen: The result is cached |
| 19:59.25 | [TK]D-Fender | jdoe: Old Shoretel? |
| 19:59.26 | jdoe | [TK]D-Fender: I forget off the top of my head. They all came from ebay. |
| 19:59.31 | jdoe | [TK]D-Fender: you know it. |
| 19:59.41 | Corydon76-dig | leifmadsen: always cached, whether you start with the QUERY or not |
| 19:59.51 | [TK]D-Fender | jdoe: Passingly aware of.... never had to personally touch. |
| 19:59.55 | hardwire | leifmadsen: yeh.. I was trying to figure out a way to use the result of SRV/DNS lookups as the peer ID |
| 20:00.00 | hardwire | that will help dramatically. |
| 20:00.13 | jdoe | [TK]D-Fender: it's unpleasant. I mean the phones work well enough, the system is a monstrosity and written entirely in js as near as I can tell. |
| 20:00.18 | [TK]D-Fender | jdoe: Still good phones... but the firmware is at the end of their road, so its like buying anything else used |
| 20:00.34 | hardwire | wishes he could see the dnsmgr cache however |
| 20:00.35 | jdoe | that's my take on it too. |
| 20:00.42 | [TK]D-Fender | jdoe: Yes, trash the core and the phones should be OK I guess |
| 20:00.43 | TJ^ | [TK]D-Fender in an office and call volume is your average small business so i dunno 50-60 calls/day |
| 20:00.52 | TJ^ | maybe even 40 calls/day |
| 20:00.56 | jdoe | [TK]D-Fender: that's what I'm hoping, I imagine my ass is somewhat on the line if not ;) |
| 20:01.07 | [TK]D-Fender | TJ^: I would go with Linksys SPA in your case |
| 20:01.09 | leifmadsen | Corydon76-dig: so I can do: Set(FIRST_RESULT=${SRVRESULT(${SRVQUERY(_sip._udp.example.com)},1)}) |
| 20:01.29 | leifmadsen | Corydon76-dig: so I can do: Set(SECOND_RESULT=${SRVRESULT(${SRVQUERY(_sip._udp.example.com)},2)}) |
| 20:01.51 | [TK]D-Fender | TJ^: FAR cheaper in the UK, and the Polycom IP 321 requires you to have a PoE switch unless you buy it with a power brick (extra). |
| 20:02.16 | Corydon76-dig | leifmadsen: Set(FIRST_RESULT=${SRVRESULT(_sip._udp.example.com,1)}) |
| 20:02.24 | TJ^ | ooh... shit PoE switch 4got bout that |
| 20:02.37 | leifmadsen | then why even have SRVQUERY() |
| 20:02.49 | [TK]D-Fender | TJ^: do you have an extra ethernet jack each place you want a phone by? |
| 20:02.51 | TJ^ | ty [TK]D-Fender |
| 20:02.55 | Corydon76-dig | leifmadsen: to mirror DUNDIQUERY and ENUMQUERY |
| 20:03.00 | leifmadsen | right... |
| 20:03.06 | TJ^ | extra jack, yeap |
| 20:03.28 | Corydon76-dig | leifmadsen: I think we'll eventually change those two functions to make the QUERY irrelevant (but backwards compatible), too |
| 20:03.51 | [TK]D-Fender | TJLinksys SPA-941 should do |
| 20:04.13 | Corydon76-dig | leifmadsen: but my criterion for modeling that way was to reduce the need to train anybody on using it |
| 20:05.19 | iheffner | I'm having some difficulty finding documentation on setting up sip friends through ODBC. I'm using 1.6.2.6 and extconfig.conf essentially says that just about any .conf can be loaded from realtime storage. |
| 20:05.26 | iheffner | it points to a file doc/extconfig.txt for specific table formats, but I don't find this under doc/ in my source tree (or under SVN trunk or 1.6.2 branch). |
| 20:05.33 | iheffner | The best I can find is this file [ https://svn.sunlabs.com/svn/solaris-asterisk/asterisk-1.4/1.4.4/doc/extconfig.txt ] but it looks like it is for 1.4. Should I expect the same to work in 1.6? |
| 20:05.33 | TJ^ | [TK]D-Fender the 941 is the same price as the polycom |
| 20:06.34 | [TK]D-Fender | TJ^: I advise you to shop around. A LOT. go collect a bunch of local resellers and then link them in for us after so we can give you a more balanced opinion. |
| 20:07.05 | TJ^ | there are only 3-4 good sellers in the uk |
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| 20:20.22 | michaely | I seem to recall a dial plan application that would let you build custom AMI events, NOT UserEvent. It was something like BuildEvent. Does anyone remember what it was? |
| 20:21.02 | azertyui | well about my problem |
| 20:21.18 | azertyui | well i need kind of billing management module for my callshop which one do you advice me ppl ? |
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| 20:30.46 | *** join/#asterisk endemic (~endemic@lynx.ipv6.onvox.net) |
| 20:30.56 | azertyui | hello |
| 20:30.59 | azertyui | anyon ehre ? |
| 20:33.07 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:33.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:33.59 | ChannelZ | We were playing the Quiet Game. |
| 20:34.01 | ChannelZ | You just lost. |
| 20:34.59 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 20:35.24 | hardwire | Is quite game over? |
| 20:35.32 | hardwire | quiet |
| 20:35.33 | hardwire | haha |
| 20:35.48 | ChannelZ | :) |
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| 20:39.05 | azertyui | ;) |
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| 20:54.10 | jblack | ChannelZ++ |
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| 20:56.59 | ChannelZ | ahoyhoy |
| 21:09.06 | manxpower | iheffner: have you looked in the doc/ directory of the Asterisk source? |
| 21:11.12 | ChannelZ | ok wtf is wrong with Macbooks' wireless |
| 21:11.43 | Chainsaw | ChannelZ: It's just an Atheros 5000 or 9000 series adapter in most cases. |
| 21:11.52 | Chainsaw | ChannelZ: Nothing particularly special about it. |
| 21:12.07 | ChannelZ | I have clients here working all day, and then all of a sudden after hours of no issues, they get knocked off the network.. the signal blinks in and out |
| 21:12.29 | Chainsaw | Use 802.11A if the macs are suitable for it. |
| 21:12.30 | iheffner | manxpower: I looked in doc/ under where I unpacked the tarball. I also looked online under svn trunk/ and 1.6.2/ |
| 21:12.31 | ChannelZ | Yet I'm in the same room and nothing of interest is happening on my computer |
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| 21:27.32 | jdoe | Fubard: ... damnit, I could have *sworn* I'd already checked this, but if you don't have a msg.mwi.1.subscribe="1234" it won't actually send the subscribe request. That's annoying. |
| 21:33.54 | jdoe | Fubard: anyway, it works for me now. |
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| 21:44.24 | *** part/#asterisk bsaxon (~bsaxon@12.68.234.174) |
| 21:45.38 | bjhaid | I am having serious problems connecting my client software to asterisk, i tried x-lite would not connect, and i tried with twinkle too, it wouldnt, i cannot get to call myself, i am not on a network, just trying all this out locally, can i not get to connect without been on a network? |
| 21:51.15 | *** join/#asterisk aidinb (~Aidin@66-214-28-176.dhcp.lnbh.ca.charter.com) |
| 21:51.25 | ChannelZ | like you're running the softphones on the same computer * is running on? |
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| 22:08.10 | hardwire | soooooooooooooooooooooooooooooooooooooooooooooooo |
| 22:08.23 | hardwire | apt-get install memlockd |
| 22:08.25 | hardwire | find /usr/share/asterisk/sounds/ -type f >> /etc/memlockd.cfg |
| 22:08.35 | hardwire | <PROTECTED> |
| 22:08.37 | hardwire | aaaaaand yay |
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| 22:13.28 | ChannelZ | YU AERE USING SOME RAMS NOW, YES! |
| 22:19.05 | Qwell | hardwire: neat |
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| 22:21.37 | hardwire | Qwell: my IVR is much more responsive |
| 22:21.43 | hardwire | under heavy disk load :) |
| 22:22.36 | riddlebox | hey all, is there a setting to tell asterisk to only allow sip devices to connect from your internal network? or to only allow like SIP/535 to come from a specific IP Address? |
| 22:23.50 | Chainsaw | riddlebox: Yes, there is support for access control lists. |
| 22:26.51 | bmoraca_work | riddlebox: check the allow and deny settings |
| 22:27.24 | riddlebox | bmoraca_work, Chainsaw, ok thank you |
| 22:27.57 | bmoraca_work | allow and deny are distinctly different from disallow and allow, however |
| 22:32.00 | Qwell | I think you mean permit/deny |
| 22:33.35 | riddlebox | dissallow and allow are for codecs right? |
| 22:33.47 | bmoraca_work | blah yes |
| 22:33.57 | bmoraca_work | it's friday |
| 22:34.10 | bmoraca_work | and they're still too similar |
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| 22:37.18 | Fubard | i got an hour to kill and i dont know what to do |
| 22:37.32 | riddlebox | so in your general settings you could say deny=0.0.0.0/0.0.0.0 then permit 192.168.0.0/255.255.255.0? |
| 22:37.49 | riddlebox | and it would only allow connections for the 192.168.0.0 network correct? |
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| 22:57.20 | jdoe | Fubard: you saw the message earlier? I don't suppose you explicitly subscribe? |
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