| 00:00.31 | Ikarus | Yes, you could literally make mpg123 a link to ogg123 if you want to do it hard style |
| 00:01.58 | Mavvie | hmmm... uplink provider says that the eventduration of my RFC2833 events is 0 ms |
| 00:02.18 | *** join/#asterisk radhios (n=bob@200.122.118.41) |
| 00:02.28 | SparFux | Ikarus: ok, that style seems to be too hard for me because it would mean my asterisk cannot play mp3 anymore :-( |
| 00:03.05 | SparFux | I think I will have to manipulate this MP3Player thingy. |
| 00:03.25 | SparFux | Ok, Im off in bed now. Thanks for help :-) |
| 00:03.38 | Ikarus | Yup, or replace mpg123 with a script that launches mpg123.real or ogg123 based on extension |
| 00:03.58 | SparFux | Ikarus: yeah, that's a good idea :-) |
| 00:04.09 | hertell | can asterisk btw play midi-files? |
| 00:04.19 | Ikarus | hertell: not by default |
| 00:04.32 | SparFux | hertell: with a script which will call timidity based on extension ;-) |
| 00:04.35 | Ikarus | see my answers above for possible solutions but s/ogg123/somemidiplayer/ |
| 00:04.36 | hertell | ikarus: is there any guide in how to do that? |
| 00:04.52 | hertell | ok |
| 00:04.54 | hertell | thnx! |
| 00:05.02 | SparFux | k, Im off. bye! |
| 00:05.16 | *** join/#asterisk voip470 (n=A_mail@pool-71-123-48-33.phlapa.fios.verizon.net) |
| 00:07.40 | *** join/#asterisk propagandhi (n=opera@210.8.161.140) |
| 00:09.29 | *** join/#asterisk Habakuk (n=andy@65.89.31.126) |
| 00:09.46 | propagandhi | hey people, can anyone give me a clue as to where to start looking for the ability to do a 3 way conference call not a conference room |
| 00:09.56 | [av]bani | hmm.. xml config doesnt work |
| 00:10.09 | mogorman | what type of device propagandhi |
| 00:10.27 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:10.42 | propagandhi | its a sip deskphone, but it doesnt have a button or option to do it |
| 00:11.31 | Ikarus | DAMN FSCKING voip-info.org, the wiki doesn't accept un-wikified data in the comments (~np~ is ignored) |
| 00:11.59 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 00:12.13 | Tagor | How do I use this command in Dial()?: |
| 00:12.14 | Tagor | 'm' -- provide hold music to the calling party until answered. |
| 00:12.14 | [av]bani | bah i want sip |
| 00:12.31 | _Sam-- | [av]bani : aside from making you look busy at work, why bother with sccp? <im just curious again, the advantage you think> |
| 00:12.33 | Tagor | Should I use it like this? Dial(....,m(mysoundfile)) |
| 00:12.50 | [av]bani | _Sam--: because this phone doesnt _do_ sip |
| 00:12.58 | _Sam-- | it does if you get the software? |
| 00:13.00 | [av]bani | no |
| 00:13.01 | _Sam-- | or no? |
| 00:13.04 | [av]bani | 7970 = no sip |
| 00:13.17 | [av]bani | 7960 = sip |
| 00:13.33 | [av]bani | which i bet, cisco only grudgingly did, with a lot of whine and complaints |
| 00:13.46 | _Sam-- | what does sccp.conf look like? |
| 00:13.58 | [av]bani | its not sccp.conf, its the phone isnt provisioning at all |
| 00:14.01 | [av]bani | its not taking my config |
| 00:14.12 | _Sam-- | no im curious what it looks like for myself, for learning :) |
| 00:14.14 | [av]bani | which i took 1:1 from the examples |
| 00:14.17 | _Sam-- | sccp.conf that is |
| 00:14.26 | [av]bani | its a real mess |
| 00:14.57 | propagandhi | I want to be able to conference a call that has come in to a third party but one particular phone doesnt have an option, so I'd like to have a function that can be dialed etc to conference it |
| 00:15.13 | propagandhi | and I cant find where to start |
| 00:15.29 | propagandhi | just a little shove in the right direction would be helpful |
| 00:15.56 | _Sam-- | sounds like a gxp: 3. If you are configuring a 7960 to work with Asterisk and SCCP you will also need to create SEP<MAC>.cnf.xml and place on the TFTP server, the 7960 (SCCP firmware 5.0.5) will continuously reboot until it finds this file. |
| 00:18.02 | ManxPower | ~docs |
| 00:18.04 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:19.56 | hertell | ikarus: which mp3-script where you refering to for playing music with *? |
| 00:20.13 | lunaphyte | how can i remove some of the folders used by default in voicemail (i.e. work, etc.) |
| 00:20.40 | Ikarus | hertell: something you hack up yourseld |
| 00:23.15 | hertell | sometthign like #/bin/sh /usr/bin/timidity /root/mozart_nachtmusik.mid ? |
| 00:23.44 | hertell | or does it have to hook up to a pipe/socket etc? |
| 00:24.00 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 00:25.00 | hertell | ikarus: does timidity need the freepats audio synthesis? |
| 00:25.10 | hertell | or can it be used without it? |
| 00:25.31 | Ikarus | hertell: I haven't used it in about 5 years |
| 00:26.56 | hertell | ikarus: whatabout that script? does the script have to hook up against a socket or similar? |
| 00:27.37 | Ikarus | hertell: uhmmmm, afaik, asterisk expects it to write raw audio to stdout at a rate of 8khz @ 16 bit |
| 00:31.01 | hertell | ikarus: tnx.. i'll try to see if i manage to do that :-) |
| 00:31.41 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 00:31.47 | [av]bani | _Sam--: i was able to php up the gxp config, transparent and automatic. doesnt look like the cisco can do it |
| 00:31.56 | [av]bani | and yay, got it working i think |
| 00:32.40 | [av]bani | nope :< |
| 00:33.57 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-149-59.nwrk.east.verizon.net) |
| 00:34.06 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com) |
| 00:37.29 | *** join/#asterisk GyrosGeier (n=geier@port-212-202-193-172.dynamic.qsc.de) |
| 00:37.36 | GyrosGeier | hi |
| 00:41.22 | kuku5 | How can I connect 2 asterisk servers together... any url to read? |
| 00:41.34 | Mavvie | kuku5: search voip-info.org for iax |
| 00:43.00 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 00:43.07 | Goral | hey all |
| 00:43.54 | Goral | i'm not getting any incomming audio from a call can anyone help me out? |
| 00:44.10 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 00:44.20 | Tagor | I am trying to get musiconhold working. It says everything is fine but I just don't hear anything if I call to the asterisk server |
| 00:44.28 | Tagor | Any idea what's wrong? |
| 00:44.57 | syzygybsd | what do you mean by it says everything is fine |
| 00:46.01 | syzygybsd | my guess of the top of my head is that you don't have mpg123 installed |
| 00:46.18 | *** part/#asterisk propagandhi (n=opera@210.8.161.140) |
| 00:46.54 | ManxPower | What version does "mpg123 -v" say? |
| 00:48.17 | syzygybsd | I am trying to debug some time issues on a polycom phone in a remote office, is there a way to get the time, status, etc remotly? |
| 00:50.16 | Tagor | syzygybsd >> I mean I don't get any errors |
| 00:50.36 | Tagor | ManxPower >> Version 0.59r |
| 00:50.48 | ManxPower | I'm trying to test out the Tellabs echo canceler we just put in, but I don't know anyone that still lives in New Orleans (which is a local call so it's good to test) that has an analog phone |
| 00:50.57 | ManxPower | Tagor, what version of Asterisk |
| 00:51.08 | Tagor | 1.2.4 |
| 00:51.10 | ManxPower | LD calls are echo canceled by the carrier as far as I can tell |
| 00:51.28 | ManxPower | Tagor, are you using the 1.2 format for the musiconhold.conf file or the 1.0 format? |
| 00:51.51 | Tagor | I'm not sure what you mean, ManxPower |
| 00:52.15 | ManxPower | Tagor, 1.2 and 1.0 use different formats for the config file. |
| 00:52.38 | ManxPower | did you get the sample from the mailing list or wiki, or did you use the sample that came WITH 1.2? |
| 00:52.46 | syzygybsd | tangor they are described http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 00:53.08 | Tagor | I used the samples of voip-info.org (wiki) |
| 00:53.14 | Tagor | But I also tried other things |
| 00:53.23 | Tagor | Because it has another format |
| 00:53.27 | ManxPower | the BEST place to see is musiconhold.conf,sample in the Asterisk source dir |
| 00:53.31 | Goral | i'm not getting any incomming audio from a call can anyone please tell me what might be wrong. |
| 00:53.36 | ManxPower | the WIKI frequently has outdated info |
| 00:53.43 | ManxPower | Goral, NAT |
| 00:54.07 | Tagor | I already thought something like that, ManxPower |
| 00:54.09 | Goral | how do i get passed that? |
| 00:54.22 | Tagor | Do you know where I can find the correct command, ManxPower? |
| 00:54.22 | ManxPower | Goral, NAT is one of the hardest things to deal with in voIP. |
| 00:54.32 | ManxPower | Tagor, do you have the source code for Asterisk? |
| 00:54.46 | ManxPower | Goral, the Wiki has a fair number of examples for NAT. |
| 00:54.53 | Tagor | No, I don't have it, ManxPower |
| 00:54.59 | ManxPower | Tagor, THEN GET IT |
| 00:55.01 | kizmet | ManxPower, I just use DMZ heh. |
| 00:55.24 | *** join/#asterisk Umaro (n=umaro@68.142.142.105) |
| 00:55.26 | Goral | well i am using a linksys BEFW11S4 |
| 00:55.32 | Goral | what wiki is that? |
| 00:55.36 | Umaro | hey guys.. anyone know a digium/sangoma reseller in las vegas? |
| 00:55.37 | ManxPower | ~docs |
| 00:55.39 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 00:55.39 | ManxPower | that one |
| 00:55.41 | Tagor | Euh, and what should I do with it, ManxPower? :P |
| 00:56.05 | ManxPower | Tagor, look in the configs directory for the correct sample config files. |
| 00:56.31 | Goral | ManxPower ty |
| 00:56.48 | ManxPower | Goral, In SIP and many other protocols the IP addresses and port numbers of the two devices is in the DATA part of the packet and so routers can't correctly do NAT for those protocols |
| 00:57.37 | ManxPower | they will try to do the NAT magic based on the IP addresses and port numbers in the HEADER of the packet, but unless they actually can decode the SIP (and RTP) protocols, they can't do NAT correctly for those protocols |
| 00:57.43 | Goral | well i have everything wide open and pointing to the linux box |
| 00:58.12 | ManxPower | Goral, Yeah, and the endpoint will try to send responses to the internal address, not your external address |
| 00:58.12 | Goral | wow that sucks |
| 00:58.14 | syzygybsd | Goral: do you have incomming packets? |
| 00:58.15 | Tagor | Ok, I just tried using the sample config, ManxPower. But I still don't hear anything |
| 00:58.24 | syzygybsd | usign tcpdump to watch |
| 00:58.28 | GyrosGeier | heh, and my flatmate bought a Cisco 2500 and a 2900 to get the NAT stuff right |
| 00:58.30 | ManxPower | Goral, Yeah. If I ever meet the people that designed that I will kill them. |
| 00:58.38 | GyrosGeier | (well, that and BGP) |
| 00:58.43 | ManxPower | Tagor, post it to pastebin.ca |
| 00:58.55 | Tagor | Sure, one second :) |
| 00:59.03 | GyrosGeier | ManxPower: SIp was designed with IPv6 in mind |
| 00:59.04 | Goral | synthetiq may i pm you? |
| 00:59.07 | ManxPower | GyrosGeier, I just disable any special NAT fixup stuff in the router and then do NAT the way Asterisk. |
| 00:59.18 | ManxPower | GyrosGeier, then they are morons |
| 00:59.24 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-168-16.mad.east.verizon.net) |
| 00:59.31 | Goral | syzygybsd may i pm you? |
| 00:59.34 | GyrosGeier | I should forward port my v6 patches one day |
| 00:59.49 | GyrosGeier | and the mighty res_stun |
| 00:59.50 | syzygybsd | Goral: ask in the channel, other people may have the same questino |
| 01:00.17 | Goral | my linux is real rusty |
| 01:00.32 | Goral | that was tcpdump? |
| 01:00.40 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 01:00.41 | syzygybsd | and I am on vactaion on the other side of the world from where i work... |
| 01:00.42 | *** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 01:00.56 | syzygybsd | Goral: you have root, i am guessing yes... |
| 01:00.56 | Tagor | ManxPower >> http://pastebin.ca/43481 |
| 01:01.03 | Goral | yes |
| 01:01.18 | syzygybsd | tcpdumpdst port 5060 |
| 01:01.25 | syzygybsd | tcpdump dst port 5060 |
| 01:01.27 | ManxPower | mode=quitmp3 |
| 01:01.31 | ManxPower | try spelling it correctly |
| 01:01.57 | xmark | can I have a asterisk answer() the PSTN port without useing exten=>s ? |
| 01:02.03 | syzygybsd | should catch all incomming packets, depending on what is sending them, but you could have an outgoing problem instead... between asterisk and you phone, |
| 01:02.04 | ManxPower | Goral, UDP/5060 is only for the SIGNALLING, audio uses RTP and is on a port number decided by the two endpoints |
| 01:02.11 | ManxPower | xmark, yes |
| 01:02.15 | syzygybsd | xmark: yes |
| 01:02.37 | Tagor | ManxPower >> Oh, didn't notice that. But it still doesn't do anything |
| 01:02.39 | xmark | I should be more specific. How? |
| 01:02.49 | ManxPower | but unless asterisk knows the number dialed (it can't on analog ports) the call will go to exten => s because that's where calls go when asterisk doesn't know what number was dialed. |
| 01:03.03 | ManxPower | analog ports == analog FXO ports |
| 01:03.06 | Goral | syzygybsd i'm using sjphone on another computer on the same network the astrisk server is on |
| 01:03.29 | xmark | Manx -- that's the problem I'm running into |
| 01:03.41 | ManxPower | Goral, if the server and client are on the same LAN then you have a firewall issue, not a NAT issue. |
| 01:03.48 | xmark | manx -- yes FXO port |
| 01:03.53 | ManxPower | xmark, then you can't. |
| 01:03.55 | Goral | calling from computer sjphone to voipbu$ter to a lan line |
| 01:04.30 | *** join/#asterisk Corydon-w (i=midnight@pdpc/supporter/sustaining/Corydon76-home) |
| 01:04.37 | GyrosGeier | is it a good idea to remove the "context" setting from the defaults in sip.conf? |
| 01:04.40 | *** part/#asterisk _DAW (n=bob@adsl-6-66-81.msy.bellsouth.net) |
| 01:04.42 | ManxPower | Im SO tempted to go to the French Quarter, but I really don't want to deal with the hassle. |
| 01:05.03 | GyrosGeier | I mean, do I get a warning then when a context does not set it, or will it crash like 1.0.0 did? |
| 01:05.13 | syzygybsd | Tagor: i noticed your default context doesnt' have an application |
| 01:05.19 | GyrosGeier | ManxPower: do it. |
| 01:05.36 | ManxPower | GyrosGeier, I've attended 10 Mardi Gras. |
| 01:05.41 | syzygybsd | goral, what is the end to end setup of your phone call |
| 01:05.44 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 01:05.46 | GyrosGeier | ManxPower: I believe honestly that this one is pretty special. |
| 01:05.58 | ManxPower | GyrosGeier, Yeah -- depressing |
| 01:06.04 | syzygybsd | what is the source of the sound, and the destination |
| 01:06.27 | Tagor | syzygybsd >> Can I use the one that [manual] uses? |
| 01:06.30 | GyrosGeier | ManxPower: well, from what I've heard this is the party of the millenium, and the millenium just started |
| 01:06.52 | ManxPower | If nothing else, I have to work tomorrow |
| 01:06.53 | GyrosGeier | ManxPower: also it beats configuring VoIP |
| 01:06.54 | syzygybsd | Tagor: yes, if you have the configuration in the other places correct |
| 01:06.55 | ManxPower | and parking sucks |
| 01:07.03 | Goral | i'm calling from this computer using sjphone to the asterisk server routing it to voipbuster then to a landline |
| 01:07.06 | ManxPower | I have 2 pints of guinness where I'm staying..... |
| 01:07.29 | Tagor | syzygybsd >> What do you mean in 'other places'? |
| 01:07.32 | GyrosGeier | ManxPower: using a car to get to a party is always a bad idea |
| 01:07.57 | MRH2 | hey anyone know if it is possible to keep asterisk out of the initial media when sip reinvites are on and asterisk knows it won't need to stay in the middle |
| 01:07.59 | ManxPower | GyrosGeier, exactly. |
| 01:08.01 | syzygybsd | Tagor: extensions.conf where you say which moh to use |
| 01:08.12 | Goral | i can hear myself on the landline phone when i talk but i cannot hear what is said on the phone |
| 01:08.15 | ManxPower | MRH2, that is the default |
| 01:08.26 | Goral | to the computer |
| 01:08.41 | Tagor | syzygybsd >> I told extensions.conf to use the default context |
| 01:08.52 | MRH2 | aye it seems to stay in there for a bit until the reinvites go through |
| 01:09.15 | syzygybsd | Tagor: and the default context doesnt' have an application, put one in, yes you can use the same one as manua |
| 01:09.17 | syzygybsd | l |
| 01:09.28 | *** join/#asterisk Weezey (n=weezey@206.210.109.233) |
| 01:09.34 | syzygybsd | goral, your asterisk box is dmz right? |
| 01:09.50 | Weezey | how do I make a call generate two CDR records? |
| 01:09.54 | syzygybsd | can you call into voicemail from the landline and leave yourself a voicemail? |
| 01:10.01 | [av]bani | [TK]D-Fender around? |
| 01:10.03 | MRH2 | let me just double check the dial options.... |
| 01:10.16 | syzygybsd | take out part of the question to see if that part works |
| 01:10.41 | Goral | i cannot accept incomming calls |
| 01:10.47 | *** join/#asterisk essaredee (n=srd@cpc1-lich2-0-0-cust257.brhm.cable.ntl.com) |
| 01:10.47 | Tagor | syzygybsd >> Ok tried that, but I still don't hear anything |
| 01:10.58 | Goral | i don't subscript to a phone number |
| 01:11.41 | syzygybsd | Tagor: if you do a ps aux |
| 01:11.45 | essaredee | I'm using freebsd, asterisk is uing 96% cpu and I already have noload => wilcalu.so in modules.conf |
| 01:11.51 | syzygybsd | is mpg123 in the list |
| 01:11.55 | Tagor | Let me try, syzygybsd |
| 01:12.03 | *** join/#asterisk forao (n=fasdfasd@pool-138-89-168-16.mad.east.verizon.net) |
| 01:12.14 | Goral | syzygybsd yes it is DMZ |
| 01:12.16 | Grizzy | What kind of daemon is running at fwdnat.pulver.com:5082 ? |
| 01:12.47 | Tagor | No, it isn't, syzygybsd |
| 01:12.55 | syzygybsd | Grizzy: looks close to manager |
| 01:13.08 | essaredee | anyone know what would be causing this? |
| 01:13.19 | *** part/#asterisk radhios (n=bob@200.122.118.41) |
| 01:13.24 | Grizzy | "manager"? |
| 01:13.31 | syzygybsd | essaredee: what does the console say it is doing? |
| 01:13.40 | Grizzy | some kind of RTP forwarder? |
| 01:13.41 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 01:13.52 | syzygybsd | manager is a remote way of controling asterisk.. kinda like telnet |
| 01:14.11 | essaredee | «syzygybsd» I see no activity in console |
| 01:15.16 | syzygybsd | if you start asterisk from the console... ie `asterisk -vvvvvvvvvvvvvvvvvvvvvvc` are there error messages? |
| 01:15.44 | syzygybsd | Goral: do you have 2 sip phones that call call eachother? |
| 01:15.44 | MRH2 | would the Dial timeout cause asterisk to stay in the media path for about 500-1000ms with reinvite....s as in Dial(SIP/${AGENT(${EXTEN}:exten):0:4},500) |
| 01:15.53 | Goral | syzygybsd no |
| 01:15.59 | essaredee | «syzygybsd» i've done that already, I'm not seeing any messages at all |
| 01:16.03 | Goral | i have no one to test it all out |
| 01:16.28 | Goral | just buster and landlines |
| 01:17.19 | syzygybsd | have your sip phone call into voicemail and leave a message, make sure you can hear yourself |
| 01:17.21 | *** part/#asterisk buZz (n=bastiaan@213-136-16-131.adsl.bit.nl) |
| 01:17.36 | Goral | i don't think thats setup |
| 01:17.38 | syzygybsd | wait, wrong way... |
| 01:17.59 | syzygybsd | can your sip phone here prompts if you add them to your dialplan? |
| 01:18.55 | Goral | do i have to auth a sip phone to use my dial plan? no |
| 01:19.02 | kuku5 | Why is cisco 871 better than cisco 831 ? |
| 01:19.23 | Tagor | Any suggestion what could be wrong, syzygybsd? |
| 01:19.36 | Grizzy | Where's a cookbook tutorial in how to traverse NAT on the side distant from the unNATted asterisk box? |
| 01:19.54 | syzygybsd | Goral: sorry, I meant if you add something like s,1,Playback(deadbeat) adn dialin can you hear it on yoru sip phone |
| 01:20.20 | MRH2 | get the nated to start the connection? |
| 01:20.51 | syzygybsd | Tagor: moh is never starting, are there no error messages in the asterisk console, and how verbose is yoru console? |
| 01:20.58 | *** join/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 01:20.59 | MRH2 | and then qualify = yes |
| 01:21.12 | syzygybsd | kuku5: bigger number means it is more advanced? |
| 01:21.15 | Goral | syzygybsd i don't know what youmean by that noob |
| 01:21.26 | Grizzy | MRH5 - was that for me? |
| 01:21.33 | MRH2 | yep |
| 01:21.58 | Goral | syzygybsd i'm really new and my linux is rusty |
| 01:22.08 | MRH2 | sorry ii can;t get used to chatzilla |
| 01:22.12 | Grizzy | Is natd (you said nated? ) something I build separate from asterisk? |
| 01:22.40 | essaredee | «syzygybsd» know of any reason why asterisk would be eating the cpu? |
| 01:23.06 | Abydos313 | so if my server is sitting live in the dmz what ports need to be open on iptables? same ones? |
| 01:23.07 | Goral | i don't even know if i would be able to call sip to sip accounts setup on this box |
| 01:23.19 | syzygybsd | Goral: i assume you have your extensions.conf configured somewhat, you are getting a call out, I want you to create a new extension where you playback one of the soundfiles so you can tell if you can hear on your sip phone at all |
| 01:23.32 | Tagor | syzygybsd >> 26, no error messages |
| 01:23.39 | Tagor | syzygybsd >> But it also doesn't tell me what file it plays |
| 01:23.50 | Tagor | syzygybsd >> So I guess it doesn't even take one |
| 01:24.06 | GyrosGeier | Hm |
| 01:24.23 | syzygybsd | Tagor: paste the relavent lines in pastebin.ca |
| 01:24.30 | GyrosGeier | Is there a good SIP echo or test server? |
| 01:24.40 | Goral | syzygybsd what would i enter and where in the extentions.conf? |
| 01:25.06 | syzygybsd | you know which context your sip ophone goes into right? |
| 01:25.22 | Goral | no |
| 01:25.34 | *** join/#asterisk dash (n=washort@adsl-147-84-117.bhm.bellsouth.net) |
| 01:25.48 | Tagor | syzygybsd >> The only relevants I see are: |
| 01:25.49 | Tagor | <PROTECTED> |
| 01:25.49 | Tagor | <PROTECTED> |
| 01:25.58 | dash | ahoy. has asterisk's response to SDP-less INVITEs changed recently? |
| 01:27.11 | [av]bani | yay chan_sccp.so |
| 01:27.13 | syzygybsd | pastebin the lines from yoru extensions .conf of the context that the moh is in |
| 01:27.15 | file | what good is an INVITE without SDP? |
| 01:27.48 | kkun | What about chan_sccp.so |
| 01:27.57 | dash | file: Is that a serious question? =/ |
| 01:28.07 | file | yes |
| 01:28.08 | syzygybsd | Goral: in your sip.conf file there is a context for your sip phone, what is it |
| 01:28.11 | Weezey | ugh! Why is everyone who uses RAR a moron? |
| 01:28.11 | kkun | Is was for banj |
| 01:28.37 | dash | file: the expected result is that the reply will contain an SDP offer |
| 01:28.43 | dash | and the ACK will contain the naswer |
| 01:28.45 | dash | answer |
| 01:28.47 | syzygybsd | Weezey: because they want somethign for nothing, and want it now |
| 01:28.54 | file | can't say I've ever seen that |
| 01:29.12 | Goral | syzygybsd you mean this... context=sip_phones |
| 01:29.15 | dash | file: it's standard for 3pcc scenarios |
| 01:29.21 | syzygybsd | Goral: yep |
| 01:29.25 | Tagor | syzygybsd >> http://rafb.net/paste/results/1BI6hB32.html |
| 01:29.35 | Weezey | syzygybsd: good point. |
| 01:29.50 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 01:30.18 | syzygybsd | ok, now in your extentions.conf, under [sip_phones] you will add 123,1,Playback(lyrics-louie-louie) |
| 01:30.52 | Goral | okay give me a second |
| 01:33.47 | Tagor | Any idea what's wrong, syzygybsd |
| 01:34.11 | *** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-46.west.biz.rr.com) |
| 01:34.12 | syzygybsd | yes, your configuration is wrong, I am looking through to find out what is missing |
| 01:34.28 | Tagor | Ok, thanks :) |
| 01:35.04 | kizmet | Oompa loompa dippedy doo have i got a really crummy story for you.... |
| 01:35.34 | *** join/#asterisk Eggplant (i=No@dsl-216-155-210-079.cascadeaccess.com) |
| 01:35.45 | kkun | kizmet, do tell |
| 01:35.56 | kizmet | kkun, nar im rite. |
| 01:36.27 | essaredee | any chance on getting some help here? |
| 01:36.44 | [av]bani | kizmet: share, so that we may laugh at your misfortune |
| 01:37.20 | kkun | essaredee, what help do you need? |
| 01:37.31 | essaredee | I'm using freebsd, asterisk is uing 96% cpu and I already have noload => wilcalu.so in modules.conf |
| 01:37.39 | Goral | syzygybsd now what do i do |
| 01:37.43 | essaredee | no activity is showing in console |
| 01:38.19 | kkun | What is the specs. Of your system, what version of asterisk are you running? |
| 01:38.19 | *** part/#asterisk dash (n=washort@adsl-147-84-117.bhm.bellsouth.net) |
| 01:38.30 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-6-66-81.msy.bellsouth.net) |
| 01:38.37 | essaredee | even with verbose at a high level and setting debug on all peers and iax2 |
| 01:38.38 | syzygybsd | Tagor: add a ,m to the end of your dial command |
| 01:38.42 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 01:38.52 | syzygybsd | dial(????,20,m) |
| 01:39.03 | syzygybsd | Goral: from yoru sip phone call 123 |
| 01:39.24 | Tagor | Ok, let me try that, syzygybsd |
| 01:39.34 | essaredee | freebsd 5.4, asterisk 1.2.4, 2.8GHz 768MB ram |
| 01:39.40 | Goral | address incomplete |
| 01:39.57 | Goral | call rejected: 484 adress incomplete |
| 01:40.12 | syzygybsd | lol.. stupid sip phone |
| 01:40.16 | kkun | Interesting indeed, are you having call quality problems? (expected) |
| 01:40.36 | syzygybsd | ok, change your extensions.conf to be 1234567 from 123 |
| 01:40.43 | essaredee | call quality is fine |
| 01:40.43 | Tagor | Still nothing, syzygybsd |
| 01:41.00 | syzygybsd | does it ring? or just silence |
| 01:41.06 | kkun | So you are just concerned of the high usage |
| 01:41.23 | kkun | What are you using to monitor cpu usage |
| 01:41.47 | essaredee | I use the machine for other things than asterisk and I'd like to keep cpu usage to minimum for when it's needed |
| 01:41.50 | essaredee | I'm using top |
| 01:42.55 | Goral | going over kill and restarting the server |
| 01:43.06 | essaredee | it's hovering around 94-97% |
| 01:43.09 | kkun | I am running asterisk 1.2.4, centos (Asterisk@home 2.5). I have a dual 450 mhz processor system with only 512mb ram |
| 01:43.21 | kkun | 1% cpu usage |
| 01:43.35 | essaredee | CPU states: 56.6% user, 0.0% nice, 43.0% system, 0.4% interrupt, 0.0% idle |
| 01:45.11 | kkun | What about the asterisk process itself? |
| 01:45.26 | essaredee | 28941 root 122 0 3808K 2528K RUN 105:57 96.19% 96.19% asterisk |
| 01:45.31 | essaredee | PID USERNAME PRI NICE SIZE RES STATE TIME WCPU CPU COMMAND |
| 01:45.33 | syzygybsd | Tagor: is mpg123 in yoru ps list yet? |
| 01:46.34 | syzygybsd | essaredee: did you compile asterisk yourself? |
| 01:46.43 | syzygybsd | or install it from ports |
| 01:47.15 | [av]bani | blah |
| 01:47.19 | [av]bani | 7970 deregistered |
| 01:47.29 | essaredee | i compiled it myself |
| 01:47.36 | Goral | syzygybsd okay now call that extention? |
| 01:47.44 | kkun | What linux kernel are you using? |
| 01:47.44 | syzygybsd | ya, 1234567 |
| 01:47.59 | syzygybsd | kkun: he is freebsd |
| 01:48.07 | kkun | gotcha |
| 01:48.16 | Goral | number does not exist |
| 01:48.26 | Goral | Call rejected: 404 not found |
| 01:48.29 | syzygybsd | is your phone saying that? |
| 01:48.34 | Goral | yes |
| 01:48.47 | syzygybsd | from the command line, do asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvr |
| 01:48.58 | syzygybsd | then try it again |
| 01:49.37 | Goral | now |
| 01:49.58 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 01:50.06 | syzygybsd | now....? |
| 01:50.19 | kkun | essaredee, how long have you been running since restarting asterisk? |
| 01:50.22 | Goral | i did that in the command line |
| 01:50.49 | syzygybsd | oh.. now dial 1234567 |
| 01:50.59 | essaredee | asterisk has been running for 18 minutes now |
| 01:51.15 | Goral | syzygybsd same thing |
| 01:51.24 | Ariel_ | hello folks |
| 01:51.25 | syzygybsd | what does the console say? |
| 01:51.30 | syzygybsd | hello Ariel_ |
| 01:51.41 | Goral | nothing |
| 01:51.46 | Ariel_ | syzygybsd, thanks |
| 01:51.50 | syzygybsd | grr, stupid sip phones |
| 01:52.08 | syzygybsd | change it to 7654321 in extensions.conf |
| 01:52.14 | syzygybsd | can't start with a1 |
| 01:52.14 | Vyeperman | Does anyone know if there is a way to make Polycom 501 or Graindstream BT100 play a message that is being received via speakerphone as it's being received? |
| 01:52.50 | syzygybsd | Vyeperman: by message, do you mean voicemail? |
| 01:53.07 | Vyeperman | syzygybsd, yes, sorry. |
| 01:53.42 | syzygybsd | I am not sure if there is a way to tell the polycom to auto answer a call, and put it on speakerphone, I have never heard of one |
| 01:53.50 | kkun | Sorry essaredee, given that (and BSD) I can't help you much, maybe someone in the channel is watching and can offer sme help |
| 01:53.56 | Goral | what do i do to restart asterisk |
| 01:54.17 | syzygybsd | Goral: run asterisk -vvvvvvvvvvvvvvvr |
| 01:54.19 | syzygybsd | then reload |
| 01:54.22 | Vyeperman | I don't want it to answer it though :-/ Unfortunately I'm still pretty new so I don't have many idea's. |
| 01:54.33 | *** join/#asterisk shakuhashi (n=shaku@200.163.3.125) |
| 01:54.57 | syzygybsd | well, the phone has to answer something, if nto the actuall call the call from voicemail with a copy of the message |
| 01:55.42 | syzygybsd | there may be some i ntercom features |
| 01:55.48 | Goral | same thing |
| 01:56.09 | Grizzy | How is it not finding the file? exten => s,2,Playback(/usr/local/share/asterisk/sounds/demo-thanks.gsm) and the file is readable by world? |
| 01:56.10 | Vyeperman | Well, it doesn't have to be those phones. Do you know of any piece of hardware that can be setup to do that? |
| 01:56.26 | syzygybsd | Grizzy: drop the .gsp |
| 01:56.31 | syzygybsd | .gsm |
| 01:56.31 | Tagor | syzygybsd >> No, it's not yet in the ps list |
| 01:56.43 | Grizzy | thanks, syzgy. |
| 01:57.00 | *** part/#asterisk shakuhashi (n=shaku@200.163.3.125) |
| 01:57.15 | Goral | syzygybsd same thing |
| 01:57.35 | Goral | Call rejected: 404 not found |
| 01:57.44 | syzygybsd | you dialed 7654321 right? |
| 01:57.52 | Goral | yes |
| 01:57.55 | Grizzy | Yow, it actually works! |
| 01:58.39 | Grizzy | They're using the voice of some bimbette. |
| 02:00.13 | *** join/#asterisk joat (n=joat@69.15.59.67) |
| 02:03.39 | Flauto | is there any free iax softphone? |
| 02:04.17 | *** join/#asterisk vmedrano (n=vmedrano@196.32.128.206) |
| 02:04.18 | Ariel_ | yes |
| 02:04.33 | Flauto | would you show me where? |
| 02:04.36 | Ariel_ | Flauto, look at diax, |
| 02:05.02 | Flauto | search? |
| 02:05.09 | *** join/#asterisk websae (i=websae@CPE-24-167-204-30.wi.res.rr.com) |
| 02:05.17 | Ariel_ | also idefisk |
| 02:05.53 | Ariel_ | idefisk is at asteriskguru.com |
| 02:06.00 | Ariel_ | ~diax |
| 02:06.01 | jbot | somebody said diax was http://www.laser.com/dante/diax/diax.html |
| 02:06.01 | Flauto | thanks |
| 02:06.14 | Ariel_ | ~iax softphones |
| 02:06.23 | Flauto | thanks |
| 02:06.34 | Flauto | i have an entry under iax.conf |
| 02:06.38 | Ariel_ | Flauto, any time |
| 02:06.38 | Flauto | guest |
| 02:07.01 | Flauto | but when i tried to register my phone number through e164.org |
| 02:07.03 | Ariel_ | there is also one from downunder I think it's called firefox??? or something like that |
| 02:07.11 | Flauto | for some reason, the calls can not come in |
| 02:07.23 | Grizzy | Is there a "cheat sheet" (quick reference) for all the asterisk directives? |
| 02:07.56 | Ariel_ | Grizzy, no |
| 02:08.13 | *** join/#asterisk dash (n=washort@adsl-147-84-117.bhm.bellsouth.net) |
| 02:08.13 | joat | (heh) i think the short version is published by o'reilly |
| 02:08.15 | Ariel_ | ~docs |
| 02:08.16 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:08.18 | dash | So uh |
| 02:08.22 | Flauto | ariel, would you try to call my iax guest on my server? |
| 02:08.22 | Grizzy | good, thanks. |
| 02:08.27 | dash | what version of asterisk is recommended for running on production systems at the moment |
| 02:08.36 | Ariel_ | Flauto, ok |
| 02:08.55 | Flauto | can i give you my server info privately? |
| 02:09.04 | Ariel_ | ok pm me the info |
| 02:09.20 | Grizzy | is the function "Delay(msec)" or sleep? |
| 02:09.26 | *** join/#asterisk SPoon_TSX (n=Kit@199.243.99.212) |
| 02:09.30 | Ariel_ | wait(5) |
| 02:09.38 | Grizzy | thanks. |
| 02:10.34 | SPoon_TSX | Hi there, May i know if I use fxotune and created a fxotune.conf. Do I still need to config the zapata.conf rxgain and txgain or just leave them 0 should be fine? |
| 02:11.01 | Ukyo | hey, is anyone here really familiar with the cisco 79xx series phones ? |
| 02:11.52 | mogorman | ask Ukyo |
| 02:12.09 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:12.21 | Habakuk | hi is there a command similar to Playback that will take a URL as an argument instead of filename? |
| 02:12.43 | Ariel_ | opps |
| 02:12.45 | Ariel_ | my kid hit my network cable... |
| 02:13.06 | Ukyo | mogorman: I have sip firmware, and asterisk shows the phone registered. It can receive calls, and open lines and place them. Using the phone, I can hear what anyone / asterisk mailbox says, but the other end cannot hear me |
| 02:14.06 | Ariel_ | SPoon_TSX, don't know never used fxotune |
| 02:14.22 | mogorman | you have nat and canreinvite on? |
| 02:14.24 | joat | Habakuk: (i'm a beginner) if there isn't, you could probably rig something via the AGI (wget possibly?) |
| 02:14.27 | Ariel_ | Ukyo, when you only have one way audio it's mostly due to network firewall/nat problems |
| 02:14.33 | SPoon_TSX | Ariel_: You never use it? You never have echo problem? |
| 02:14.50 | Ariel_ | SPoon_TSX, yes but I have not had to use it |
| 02:15.01 | Habakuk | hi joat yeah I was thinking that. or just right an app to do it myself. just didn't want to reinvent the wheel |
| 02:15.07 | joat | Habakuk: i imagine that there'd be a delay though while the file downloads |
| 02:15.11 | Habakuk | s/right/write/ |
| 02:15.17 | Grizzy | Does "canreinvite" force the extension to originate RTP? |
| 02:15.24 | Ukyo | mogorman: the server is behind a nat. The phone is on a public IP. If I set the server as dmz on the nat, it still does it |
| 02:15.45 | xachen | server behind nat? phone on public IP? |
| 02:15.47 | SPoon_TSX | Ariel_: May I know are you from Canada too? Since I am wondering my Aastra phone got no echo but the remote party say they cannot heard me clearly and they say they hear static. What do you tink? |
| 02:15.48 | xachen | bad bad mix |
| 02:15.55 | Ukyo | xachen: it happens. |
| 02:15.57 | Ukyo | either way |
| 02:16.01 | xachen | the phone can be behing phone |
| 02:16.01 | Habakuk | joat: right possible, if there is some network latency |
| 02:16.06 | xachen | er, behind nat |
| 02:16.11 | xachen | but the server needs to be public :) |
| 02:16.11 | Ukyo | both sides are fully forwarded atm |
| 02:16.25 | Ariel_ | SPoon_TSX, no far from there. But if they hear static I would make sure I had the latest firmware first of all |
| 02:16.28 | Ukyo | as far as networking is concerned, being the dmz, it is public |
| 02:16.45 | SPoon_TSX | Ariel_: You mean the firmware from the phone? |
| 02:16.45 | nanogeek | Ukyo, what about on the server side? |
| 02:16.46 | Grizzy | It was hard enough with my server being bare naked on the net. |
| 02:16.51 | Habakuk | Ukyo: ethereal is you friend |
| 02:17.03 | Ukyo | HabakukL > |
| 02:17.03 | Ukyo | ? |
| 02:17.05 | SPoon_TSX | Ariel_: I got 1.3.1 loaded. |
| 02:17.06 | Ariel_ | SPoon_TSX, yes |
| 02:17.19 | Ukyo | Habakuk: the game? |
| 02:17.25 | Ariel_ | servers don't need to be naked |
| 02:17.33 | Ariel_ | iptables works fine |
| 02:17.37 | Ukyo | nanogeek: what about server side ? |
| 02:17.37 | *** part/#asterisk nanogeek (n=nanogeek@207.244.10.65) |
| 02:17.38 | Habakuk | Ukyo: ethereal.com <-- network sniffer. #1 tool you should learn to use |
| 02:17.47 | Ukyo | oh, that |
| 02:17.49 | Ukyo | n/m |
| 02:17.51 | Ariel_ | ngrep |
| 02:17.59 | SPoon_TSX | Ariel_: May i know what result should I get if I run zttest? Should I all most of the time in 100%? |
| 02:18.10 | Ariel_ | sounds good to me |
| 02:18.19 | Ukyo | mogorman: canreinvite is the only thing I haven't touched |
| 02:18.33 | Habakuk | Ariel_: ngrep may not help in this case. won't show rtp streams easily |
| 02:18.59 | Ariel_ | Habakuk, ture but it's a start |
| 02:19.00 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 02:19.00 | SPoon_TSX | Ariel_: Are you also using aastra 480i phone? |
| 02:19.02 | Psykick | hi guys |
| 02:19.22 | *** join/#asterisk leenuxg33k (n=bpeck@71-10-248-241.dhcp.oxfr.ma.charter.com) |
| 02:19.47 | Grizzy | fronts and ends of messages are getting chopped off. |
| 02:19.54 | Psykick | I'm using AAH version 2.5 ... and I'm having issues with the queue disconnecting callers after different periods of time |
| 02:19.55 | Ariel_ | SPoon_TSX, aastra phones that is |
| 02:20.16 | SPoon_TSX | Ariel_: Which firmware you are using on your phone now? |
| 02:20.38 | Ariel_ | SPoon_TSX, I have not tried any updates since about 6 month ago |
| 02:20.42 | Ariel_ | so it's not current |
| 02:20.46 | Ariel_ | I gave up on them |
| 02:20.49 | Psykick | the other issue I'm having is sometimes calls are not coming through even if there is an agent not on a call |
| 02:20.51 | SPoon_TSX | Ariel_: Oh, I see. |
| 02:21.06 | SPoon_TSX | Ariel_: Why you gave up on them? the quality? |
| 02:21.28 | Ariel_ | firmware kept loosing it's self if I did not keep the tftp server up |
| 02:21.42 | Ariel_ | so phones would loose settings |
| 02:21.48 | Grizzy | even with a wait ahead of the first Playback, I lose stuff. Any recommendations? |
| 02:22.09 | Psykick | if anyone was to recommend a commercial * product what would it be? |
| 02:22.36 | Psykick | and of course why |
| 02:22.38 | Ariel_ | answer |
| 02:22.42 | leenuxg33k | has anyone ever seen asterisk -rx 'some command' hangs. It never returns back to the shell prompt |
| 02:22.53 | leenuxg33k | doesn't matter what command |
| 02:22.56 | Qwell | mogorman, [av]bani: ALL UI are useless, yes :p |
| 02:23.04 | Ariel_ | leenuxg33k, see what the log says |
| 02:23.08 | Grizzy | ariel - I do an Answer |
| 02:23.24 | Ariel_ | leenuxg33k, also you can do asterisk -vvvvgc to see it in direct mode |
| 02:23.29 | SPoon_TSX | Ariel_: May I know what will be the best result when I run zttest? |
| 02:24.05 | Ariel_ | SPoon_TSX, hehehe, as long as it's above 98.9 it's all good |
| 02:24.23 | *** join/#asterisk Eggplant (i=No@dsl-216-155-210-079.cascadeaccess.com) |
| 02:24.45 | *** join/#asterisk Druken (n=druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 02:24.59 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 02:25.01 | Grizzy | Any hint on how to trigger the "s" clauses from a softphone? (it doesn't do anything unless I dial some number). |
| 02:25.19 | diclophis | howdy all |
| 02:25.28 | diclophis | so... i have a question regarding AGI and set_callerid |
| 02:25.58 | SPoon_TSX | Ariel_: Can you help me to take a look of this post http://pastebin.com/576007 if you see I may have IRQ problem? |
| 02:26.13 | leenuxg33k | Ariel_: where/how can I spam you some ouput? |
| 02:26.28 | Ariel_ | pastebin.ca |
| 02:26.32 | Ariel_ | spammmmmmmmm |
| 02:27.04 | Ariel_ | SPoon_TSX, wow 201 |
| 02:27.20 | Ariel_ | looks like it's not sharing anything. |
| 02:27.25 | leenuxg33k | Ariel_: http://pastebin.ca/43492 |
| 02:27.32 | leenuxg33k | I've done an strace as well |
| 02:27.50 | leenuxg33k | and it shows it polling on the /var/run/asterisk.ctrl socket |
| 02:27.55 | SPoon_TSX | Ariel_: Mmmm.. so my hardware setting is good right? |
| 02:28.16 | Ariel_ | SPoon_TSX, looks ok |
| 02:28.25 | leenuxg33k | every morning I have hung asterisk processes from logrotate |
| 02:29.04 | SPoon_TSX | Ariel_: Mmmm... then why people say they have trouble to hear me talking? I have no echo at all..... Any idea? |
| 02:29.30 | Ariel_ | trouble hearing is nothing to do with echo |
| 02:29.47 | Ariel_ | now are you too low, garbled? |
| 02:32.04 | lunaphyte | when asterisk has a sip phone (set up as a friend) talking to it, who sends INVITE to whom? |
| 02:32.23 | Mavvie | the person who makes the call. |
| 02:32.26 | Mavvie | well, phone :-) |
| 02:32.53 | SPoon_TSX | Ariel_: I am wondering if I should increase the tx gain. |
| 02:33.31 | Ariel_ | well it too low? |
| 02:33.46 | lunaphyte | what if the call is going to the phone, not coming from the phone? does asterisk send an INVITE in that case? |
| 02:34.42 | SPoon_TSX | Ariel_: Do you use ztmonitor to adjust your gain? SHould the level always the 50% or 100% when you speak normally? |
| 02:35.03 | Ariel_ | SPoon_TSX, depends... But I like mine around 75% |
| 02:35.33 | SPoon_TSX | Ariel_: Problem I got is, if I tune up the tx, I go echo. |
| 02:35.45 | Ariel_ | keep them apart |
| 02:35.58 | Ariel_ | if you use txgain at 4.0 then rxgain 2.0 |
| 02:36.30 | lunaphyte | ah - so when a phone registers with asterisk, it tells asterisk what user@ipaddress to send invites to when a call for that phone comes in...? |
| 02:36.39 | Ariel_ | tx sends sound out. rx is for sound coming into your system |
| 02:36.51 | Ariel_ | lunaphyte, well yes |
| 02:37.21 | kuku5 | <PROTECTED> |
| 02:37.30 | kuku5 | It doenst want to choose ulaw - anyone know why ? |
| 02:37.49 | Ariel_ | someone is sending you a call in another codec you dont have |
| 02:37.57 | lunaphyte | so if i trust that a phone will always register, there's no need for me to tell asterisk user/ip to go find it at, right? |
| 02:38.23 | Ariel_ | lunaphyte, most people only put ip or host for turnks |
| 02:38.27 | Ariel_ | trunks |
| 02:39.35 | kuku5 | Ariel_: Its me, Im sending the call, it only wants to take gsm for some stupid reason |
| 02:39.53 | Olobola | any nufone users? I just moved to a new office and can't tell if it's the new network or my provider. |
| 02:40.21 | Goral | anyone know what this error means? |
| 02:40.23 | Goral | <PROTECTED> |
| 02:40.28 | Ariel_ | kuku5, then you have your settings incorrect |
| 02:40.40 | Chotaire | hi all.. short question.. it seemed the behaviour changed from 1.0.3 to 1.2.4... is there any way that e.g. when I hit # on a call and transfer the call to e.g. "33#" I can have "#" act as immediate dial instead of passing "#" to the dialstring? |
| 02:40.44 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 02:40.51 | Chotaire | I would like to set that globally |
| 02:41.10 | Ariel_ | Chotaire, features.conf |
| 02:41.15 | Chotaire | let me check. |
| 02:41.24 | kuku5 | Ariel_: how so ? |
| 02:41.26 | lunaphyte | what is a reasonable registration interval for well-connected phones? (i.e. lan)? i see asterisk defaults to 60 - is it wise to extend that? |
| 02:41.41 | Ariel_ | kuku5, disallow=all allow=ulaw only use that for testing |
| 02:41.53 | Chotaire | ariel: ? |
| 02:41.54 | kuku5 | Ariel_: what about bandwith |
| 02:42.06 | Ariel_ | kuku5, your testing correct |
| 02:42.10 | kuku5 | yes |
| 02:42.16 | Ariel_ | test with the best first |
| 02:42.19 | Ariel_ | then try others |
| 02:42.59 | Chotaire | ariel: sorry I don't see that feature there.... anything I can do about it? |
| 02:43.15 | kuku5 | Ariel_: that doenst work - same error |
| 02:43.30 | Chotaire | all I want is "#" in any dialstring to be ignored but to be taken as "dial everything I dialed before right now" |
| 02:43.37 | kuku5 | The server I am calling, receives from other peers via ulaw, but from this server, it doesnt want to. |
| 02:43.41 | *** join/#asterisk Q-At-Home (n=Queue@S0106000c41bb87af.ed.shawcable.net) |
| 02:43.47 | *** join/#asterisk xmark (n=brg@c-68-43-129-244.hsd1.mi.comcast.net) |
| 02:44.22 | Chotaire | like I have an extension "33"... I call someone, hit "#" to transfer the call, then hit "33#"... instead of acting like before, it will try to dial "33#", die and disconnect the caller now. |
| 02:44.45 | xmark | My exten=>s is not passing callerid to the device its ringing |
| 02:44.55 | *** part/#asterisk Habakuk (n=andy@65.89.31.126) |
| 02:45.02 | xmark | exten => s,1,Answer() |
| 02:45.04 | xmark | exten => s,2,Dial,Zap/1|20 |
| 02:45.44 | xmark | I wonder if I need to pass an argument to answer? |
| 02:46.03 | SPoon_TSX | Ariel_: You mean if my rxgain is 0, then my tx should be 2 then? |
| 02:47.16 | Ariel_ | not really 0.0 is fine for most setups |
| 02:47.30 | Ariel_ | but if you increase one you need to then change the other in most cases |
| 02:48.26 | *** join/#asterisk tuxinator_linux (n=tuxinato@m190e36d0.tmodns.net) |
| 02:50.22 | kuku5 | Ariel: The server I am calling, receives from other peers via ulaw, but from this server, it doesnt want to. |
| 02:58.44 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 02:58.54 | forao | does anybody know when the final release of mysql 5.0 is gonna be out? |
| 03:00.19 | Olobola | Anyone else using nufone? Are your incoming calls ok? |
| 03:01.49 | *** join/#asterisk Pids (n=pids@dsl081-072-084.sfo1.dsl.speakeasy.net) |
| 03:04.14 | glm2k | Olobola: i use nufone, don't have a did with them though sorry |
| 03:04.23 | *** join/#asterisk kizmet (n=kizmet@rizzo.argon.net.au) |
| 03:05.48 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 03:07.41 | Olobola | oh well. Wish they had status page or something. |
| 03:13.27 | *** join/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:13.37 | *** part/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:14.20 | Flauto | what is slin? |
| 03:17.19 | *** join/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:18.35 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:19.36 | *** part/#asterisk kkun (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:19.46 | Qwell | signed linear audio |
| 03:20.11 | Flauto | is it one of the codec |
| 03:24.52 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4155505.sympatico.ca) |
| 03:25.10 | *** join/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:25.37 | GeneG | Is there a way to write v-mail in MP3 format? |
| 03:26.01 | AAH_USER | yes |
| 03:26.51 | GeneG | Nice. Can you point me to a config file or reference of some sort? |
| 03:27.55 | Qwell | voicemail.conf with asterisk-addons |
| 03:28.33 | *** join/#asterisk jskcr (n=zaphod@90-pool1.ras14.floca.alerondial.net) |
| 03:28.54 | GeneG | Thank you. I will look into that. |
| 03:28.54 | SPoon_TSX | exit |
| 03:35.42 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:38.39 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 03:41.02 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 03:41.18 | riddlebox | should asterisk be using /dev/dsp all the time? |
| 03:41.53 | AAH_USER | I don't even have /dev/dsp on my box |
| 03:42.39 | riddlebox | its wierd, if I start asterisk nothing on the system can use audio, if I kill it I can use audio right away |
| 03:43.00 | fu3 | sounds like your card isnt full duplex? |
| 03:43.49 | riddlebox | I used to do this when I was using knoppmyth, I switched to ubuntu and now it wont work? |
| 03:43.52 | fu3 | anyone else hear clicking on a phone connected to a TDM400P when switching consoles? |
| 03:45.52 | *** join/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:46.00 | *** part/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:46.43 | Flauto | i have a guest entry in my iax.conf |
| 03:46.54 | Flauto | i was able to get incoming calls before |
| 03:47.00 | Flauto | but now, it does not |
| 03:47.50 | *** join/#asterisk bmg505 (n=leon@dsl-165-148-45.telkomadsl.co.za) |
| 03:47.56 | *** join/#asterisk mattwj2005 (n=Matt@dialup-4.159.62.152.Dial1.Chicago1.Level3.net) |
| 03:49.06 | Flauto | i see the call comes in |
| 03:49.17 | Flauto | but it hangs up right away |
| 03:49.28 | *** join/#asterisk AAH_USER (n=none@c-67-186-207-234.hsd1.ut.comcast.net) |
| 03:49.47 | Flauto | <PROTECTED> |
| 03:49.47 | Flauto | <PROTECTED> |
| 03:49.47 | Flauto | <PROTECTED> |
| 03:49.47 | Flauto | <PROTECTED> |
| 03:49.47 | Flauto | <PROTECTED> |
| 03:49.48 | Flauto | <PROTECTED> |
| 03:49.50 | Flauto | <PROTECTED> |
| 03:49.52 | Flauto | <PROTECTED> |
| 03:49.54 | Flauto | <PROTECTED> |
| 03:49.56 | Flauto | <PROTECTED> |
| 03:49.57 | mattwj2005 | well....I actually might be getting voip soon....a company might be interested in hiring me |
| 03:49.58 | Flauto | <PROTECTED> |
| 03:50.00 | Flauto | <PROTECTED> |
| 03:50.02 | Flauto | <PROTECTED> |
| 03:50.04 | Flauto | this is what i see |
| 03:50.17 | kizmet | Flauto, its called past bin PLEASE! http://www.pastebin.ca |
| 03:50.31 | Flauto | sorry |
| 03:50.44 | fu3 | god damn canada, stealing the worlds notes. |
| 03:51.34 | f^bar | How to spell canada C-ay-N-ay-d-ay |
| 03:51.40 | fu3 | ha ha ha |
| 03:51.46 | Ukyo | anyone here ever had rtp type issues with a cisco 7960 ? |
| 03:51.46 | fu3 | we say 'eh' |
| 03:51.54 | fu3 | C-eh-N-eh-D-eh |
| 03:52.23 | f^bar | gotcha |
| 03:52.52 | mattwj2005 | I really need to decide on a provider |
| 03:53.06 | f^bar | I like iax.cc / sixTel |
| 03:53.07 | mattwj2005 | I like broadvoice....but I hear the quality isn't that great |
| 03:53.33 | f^bar | Telasip gets good reviews from customers |
| 03:53.46 | fu3 | yeah.. those customers are their employees. |
| 03:53.51 | f^bar | Broadvoice customers complain |
| 03:54.27 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 03:54.39 | f^bar | See nerdvittles.com, they really like telasip |
| 03:55.00 | *** join/#asterisk Trevor_b (n=FreePBX5@64-142-72-32.dsl.static.sonic.net) |
| 03:56.37 | f^bar | Voxee is good for outbound, I use them for outbound and iax.cc for inbound (DIDs) |
| 03:56.40 | [av]bani | \o> |
| 03:56.41 | justinu | everyone here says sixtel is satan |
| 03:56.53 | [av]bani | hmm.. polycom ip601 has better sound than cisco 7970 |
| 03:57.00 | [av]bani | shame no backlight :| |
| 03:57.07 | mattwj2005 | yeah I am looking for unlimited incoming and outgoing in the us....I thought maybe I would go iax....but I have been reconsidering sip |
| 03:57.20 | trixter | I havent had any problems with sixtel |
| 03:57.24 | f^bar | Why is that (sixtel = satan) , I happen to like them |
| 03:57.27 | mattwj2005 | this would be my personal residental setup |
| 03:57.45 | kizmet | Does anyone other than IPKall provide free USA DID's ? |
| 03:57.52 | f^bar | Not too many iax providers |
| 03:57.55 | f^bar | stanaphone |
| 03:57.58 | glm2k | i don't have any problems with sixtel either |
| 03:58.03 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 03:58.04 | f^bar | Stanaphone - free dids |
| 03:58.14 | diclophis | how can i verify i have callerid setup correctly on my PRI? |
| 03:58.23 | fu3 | can you see the caller id? |
| 03:58.24 | mattwj2005 | didn't stanaphone stop doing that? |
| 03:58.59 | trixter | stanaphone is still free inbound only |
| 03:59.03 | f^bar | Not for me, fax is no longer free, but you ccan get a free did, I have one |
| 03:59.08 | trixter | I am 1ms from them |
| 03:59.24 | *** join/#asterisk axscode (n=asdfasdf@203.213.217.123) |
| 03:59.25 | glm2k | lucky! |
| 03:59.32 | kizmet | trixter, Do you have a Asterisk config i can use ? |
| 03:59.33 | trixter | same data center :P |
| 03:59.34 | Flauto | trixter, i am using your service now |
| 03:59.34 | axscode | ~FXP |
| 03:59.36 | jbot | rumour has it, fxp is a program that uses FTP PASV mode to transfer between two FTP servers, ask about 'quftp' for software |
| 03:59.37 | Flauto | it works great |
| 03:59.39 | axscode | ~FXO |
| 03:59.40 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 03:59.44 | trixter | kizmet: config for what? |
| 04:00.00 | trixter | Flauto: that box you connect to is at the same data center as stanaphone, NYIIX |
| 04:00.06 | axscode | ~FXS |
| 04:00.07 | jbot | fxs is probably foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 04:00.13 | f^bar | trixter: what is your service (url?) |
| 04:01.07 | trixter | http://www.trxtel.com -> tollfree termination |
| 04:01.10 | axscode | hi guyz.. what do i need for my asterisk... TDM04B or TDM40B? i want to connect my TELCO LINE to my ASTERISK.. |
| 04:01.27 | diclophis | so... no love for the callerid ? |
| 04:01.29 | fu3 | I like my TDM400P, but it's got shitty echo cancellation :| |
| 04:01.35 | fu3 | or, more likely, i cannot config it right :| |
| 04:01.44 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 04:01.51 | kizmet | trixter, to get the stanaphone did to my asterisk ? |
| 04:02.52 | *** join/#asterisk arctic_import (n=jason@209-112-170-172-cdsl-rb1.nwc.acsalaska.net) |
| 04:03.03 | Flauto | trixter, i am using the toll free service now. it is all good. |
| 04:03.25 | axscode | TDM04B <-- is this the one use for connection from TELCO to my ASTERISK? |
| 04:04.15 | *** join/#asterisk dofear (n=arodef@146-142-222-203.rev.techex.net.au) |
| 04:04.54 | dofear | can i use an asterisk box to receive IAX calls and then route those using h323? |
| 04:05.49 | mattwj2005 | someone was asking about free DIDs....I found this |
| 04:05.51 | mattwj2005 | http://www.voip-info.org/wiki/view/Cheapest+ATAs+and+Service |
| 04:05.52 | f^bar | trixter, glad to hear of your service, I will start sending toll-free your way |
| 04:08.55 | trixter | thanks |
| 04:09.12 | Ukyo | I just don't know what do on this phone anymore |
| 04:09.17 | Ukyo | I've ruled out everything anyone can think of |
| 04:10.37 | Flauto | how did you get the service up running, it is great to be able to get free toll free number |
| 04:10.56 | Flauto | i use voipstunt, so it is good to use your service |
| 04:11.11 | Flauto | mike j told me about your service |
| 04:11.21 | Flauto | trixter |
| 04:11.21 | diclophis | what would be causing random hangups? |
| 04:11.33 | diclophis | and causing asterisk to crash? |
| 04:13.25 | trixter | internetcalls.com > voipstunt (same company) |
| 04:15.42 | diclophis | wtf does this mean: Channel 2/22, span 1 received AOC-E charging 859189561 units |
| 04:16.17 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 04:17.34 | docelmo | I know this is the wrong channel but any knowegable SER people in here? |
| 04:18.06 | *** join/#asterisk kpettit (n=keith@user-0cet2g6.cable.mindspring.com) |
| 04:18.41 | Flauto | but it is different in the free countries, trixter |
| 04:20.41 | mattwj2005 | what do you guys think of viatalk? |
| 04:22.59 | Ukyo | hey trixter, ever worked with a cisco 7960? :P |
| 04:25.41 | mattwj2005 | quality any good? anyone ever use it? |
| 04:25.56 | kpettit | anybody know how I can use ztdummy and still specify stuff in /etc/zapata.conf and not have it puke? |
| 04:26.08 | *** join/#asterisk kb1_kanobe (n=jsmith@h24-207-96-51.cst.dccnet.com) |
| 04:26.25 | GeneG | I've compiled and installed format_mp3 from asterisk-addons, but when I changed my voicemail.conf to format=mp3, asterisk is reporting a ast_play_and_record: Error creating write stream error. Any suggestions? |
| 04:26.41 | kpettit | trying to use a Sangoma PRI card, and the zttest scores are bad, so it was suggested that I use the ztdummy driver. But then there is a issue with my zaptel.conf stuff |
| 04:26.45 | *** join/#asterisk _mountie (n=mountie@trb229.travel-net.com) |
| 04:38.40 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 04:40.29 | mattwj2005 | what about telasip or thinkbright? |
| 04:40.59 | fu3|gone | why dont you list every possible provider? |
| 04:41.09 | f^bar | trixter, my I PM you about your service? |
| 04:41.22 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 04:41.57 | mattwj2005 | I am just trying to find a good unlimited service |
| 04:42.40 | fu3 | fair enough. |
| 04:42.43 | docelmo | GeneG, ya format is READ ONLY |
| 04:44.17 | docelmo | err format_mp3 |
| 04:45.28 | dofear | Can i use an asterisk box to receive IAX calls and then route those using h323? |
| 04:45.39 | docelmo | yes |
| 04:45.53 | docelmo | however h323 is unstable on more than 10 calls |
| 04:45.59 | trixter | f^bar: yes but I am on the phone so I wont respond imediatly |
| 04:47.59 | *** join/#asterisk WAudette (n=WAudette@c-67-170-156-3.hsd1.or.comcast.net) |
| 04:48.01 | DiscusZ | Hi all |
| 04:48.09 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4156109.sympatico.ca) |
| 04:48.47 | DiscusZ | still trying to get outbound FWD calls working.. In from FWD works as call me connects no problem.. |
| 04:48.50 | GeneG | Is it possible to write voice-mail messages as MP3 format rather than WAV? I've installed asterisk-addons (format_mp3.so) and changed format=mp3 in voicemail.conf but I am getting an error message. |
| 04:49.12 | *** part/#asterisk dash (n=washort@adsl-147-84-117.bhm.bellsouth.net) |
| 04:49.57 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 04:55.26 | *** join/#asterisk Eggplant (i=No@dsl-216-155-210-079.cascadeaccess.com) |
| 04:59.51 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 05:03.49 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:04.53 | mogorman | gnite |
| 05:04.54 | *** part/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 05:05.00 | kizmet | GeneG, format_mp3 allows decoding but not encoding. |
| 05:11.03 | mattwj2005 | yeah I think I will probably go with telasip |
| 05:11.20 | mattwj2005 | it sounds like a lot of good reviews....plus a good price :) |
| 05:14.35 | *** join/#asterisk GeneG (n=GeneG@toronto-HSE-ppp4159887.sympatico.ca) |
| 05:14.56 | GeneG | Kizmet, is there any way to have messages recorded in mp3 format? |
| 05:15.10 | kizmet | GeneG, Not really |
| 05:15.22 | kizmet | GeneG, Record them in wav |
| 05:15.31 | kizmet | GeneG, then convert them to mp3 manually |
| 05:15.39 | kizmet | (They will be about the same size anyway) |
| 05:16.41 | GeneG | Hmmm, the reason I want to convert to MP3 is to save size when e-mailing the messages. Right now, a 1 min v-mail takes about 1 MB of space. I figure an MP3 will be about 1/5 the size. Am I figuring wrong? |
| 05:17.51 | *** join/#asterisk propagandhi (n=opera@210-8-161-140.dsl.officelink.net.au) |
| 05:19.05 | propagandhi | can anybody think of a way to provide three way calling if the phone has no function to do it (IE a keypad code or similar) |
| 05:20.32 | Snake-Eyes | make a extension maybe |
| 05:21.36 | propagandhi | Snake-Eyes: sorry but I dont understand can u elaborate |
| 05:24.33 | Snake-Eyes | propagandhi, try making a extension that connects to a three way call, if you want to get really fancy maybe you could do some menu/IVR setup |
| 05:25.00 | DiscusZ | I am trying to get outbound FWD calls working and when I try and call I get a recording that all circuits are busy (I believe this is from my Asterisk box) what am I missing? |
| 05:27.09 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 05:30.40 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 05:31.06 | glm2k | DiscusZ: have you tried the FWD 612 extension? plays back the current datetime? |
| 05:32.08 | *** join/#asterisk axonxorz (n=axonxorz@198.161.102.243) |
| 05:32.28 | DiscusZ | thats what I have been trying and the call is not leaving my asterisk box I get a recording saying all circuits are busy now try again later |
| 05:32.52 | DiscusZ | I even got my Asterisk Book in the mail today and tried its FWD example and can not make it go.. |
| 05:33.02 | axonxorz | hey, i dont know where to start with this, im trying to configure a SIP-only asterisk box with realtime support |
| 05:33.13 | axonxorz | asterisk compiles fine, but NONE of the addons even compile |
| 05:33.19 | axonxorz | any insight? |
| 05:33.43 | DiscusZ | Anyways I have to run into work for a quit hardware swap /msg me anything I can try. back in 1 hour |
| 05:35.07 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 05:37.06 | *** join/#asterisk clive- (n=pirch@dsl-145-56-246.telkomadsl.co.za) |
| 05:37.52 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 05:46.06 | *** join/#asterisk angom_h (n=angom@red-corp-200.38.17.106.telnor.net) |
| 05:47.14 | *** part/#asterisk propagandhi (n=opera@210-8-161-140.dsl.officelink.net.au) |
| 05:50.46 | clive- | anyone using ztdummy on the 2.6 kernel? I am wondering what result you get from ztcfg .... |
| 05:52.14 | SwK | clive- you dont get any result as you arent really configuring anything |
| 05:52.19 | SwK | its just there for a timing source |
| 05:52.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:53.58 | clive- | Swk thanks, I thought I was doing something wrong, as previously I used a f100x card and ztcfg gave a result |
| 05:58.38 | *** join/#asterisk xygoat (n=xygoat@71.194.90.204) |
| 05:58.42 | xygoat | hi all |
| 05:58.50 | MikeJ[Laptop] | baaaahhh |
| 05:58.57 | x86 | anyone happen to use Asterisk with Stanaphone? |
| 05:59.27 | xygoat | does anyone know a linux software sip phone that doesn't have a gui |
| 05:59.58 | Corydon76-home | Nope |
| 06:00.14 | *** part/#asterisk mattwj2005 (n=Matt@dialup-4.159.62.152.Dial1.Chicago1.Level3.net) |
| 06:00.21 | xygoat | yea. me neither |
| 06:00.30 | Corydon76-home | but you might could use the chan_alsa console driver |
| 06:00.30 | *** part/#asterisk axonxorz (n=axonxorz@198.161.102.243) |
| 06:00.37 | xygoat | hmm.. |
| 06:01.17 | Corydon76-home | Nice nick, btw. Reminds me of my favorite mag |
| 06:01.26 | xygoat | which mag is that |
| 06:01.32 | Corydon76-home | XY Magazine |
| 06:02.14 | xygoat | that is a good magazine |
| 06:02.20 | xygoat | and a really nice domain name |
| 06:02.36 | glm2k | x86: tried it before. never got it to work |
| 06:05.07 | xygoat | http://www.linphone.org/ - linphonec will do it ;) |
| 06:05.33 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 06:05.39 | x86 | glm2k: me either :( |
| 06:06.21 | x86 | glm2k: i have IPKall and IPStar.us both working just fine.... I have 2 DID's from IPKall and one from IPStar |
| 06:08.11 | brockj49464 | Boy, I just fixed a funny problem where some of the DB records for AMPUSER/ext/device were missing... |
| 06:09.17 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-18624ce49782a91e) |
| 06:18.27 | arctic_import | is it possible to ease drop on a channel? Meaning redirect the channel audio to another channel? |
| 06:21.20 | Rhizome | arctic_import, http://www.voip-info.org/wiki/view/MixMonitor , that will dump it to a file. |
| 06:23.33 | *** join/#asterisk SPoon_TSX (n=Kit@70.54.197.99) |
| 06:24.26 | SPoon_TSX | Hi there, I hope someone can help me out on this.... I everything setup, but when I try to dial out, it say no Zap Channel are avariable. May I know what would be the problem? |
| 06:24.47 | Rhizome | :/ |
| 06:24.59 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 06:25.35 | SPoon_TSX | Also, May I know if I really need a Sound Card for Music On-Hold? |
| 06:27.14 | SPoon_TSX | Anyone? |
| 06:28.23 | [av]bani | http://bani.anime.net/cat_look.jpg |
| 06:29.48 | x86 | SPoon_TSX: you shouldnt need a sound card |
| 06:29.49 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 06:29.59 | hellop | what package contains wcfxo? |
| 06:30.03 | x86 | SPoon_TSX: it sets mpg123 up to output to stdout |
| 06:30.21 | hellop | I forget what I need to install for it.. I already installed zaptel and libzap1 |
| 06:31.47 | SPoon_TSX | x86: I am wondering if I enable the sound that cause me the channel unavarilable problem. |
| 06:32.26 | SPoon_TSX | x86: Do you know how do I check if the Zap channel is being used? |
| 06:32.27 | hellop | I thought wcfxo is part of zaptel |
| 06:32.38 | hellop | SPoon_TSX zap show channels |
| 06:32.57 | tronix | hellop: yes, for Gentoo, wcfxo is part of the zaptel package. I don't know about other distributions, though. |
| 06:33.44 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:36.40 | *** part/#asterisk xygoat (n=xygoat@71.194.90.204) |
| 06:42.11 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.adnc.com) |
| 06:43.41 | hellop | tronix, I'm experimenting with installing asterisk from the default apt sources in debian sarge. |
| 06:43.56 | hellop | I mean, why is it in there unless it can be done? |
| 06:44.21 | hellop | tronix, but apt-get install zaptel does not include wcfxo. neither libzap1 |
| 06:44.27 | hellop | tks for the help, tho. |
| 06:46.37 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 06:49.31 | *** join/#asterisk EnErGy[CSDX] (n=energy@193.242.114.14) |
| 06:50.47 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 06:51.35 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 06:52.56 | Telamon | Does anyone know of a good VOIP enabled LD reseller in Canada? Currently looking at a US company, but I'm thinking that a Canadian one would be cheaper for in-Canada calls. |
| 06:55.39 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-118-15.dsl.tulsok.swbell.net) |
| 06:59.44 | *** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net) |
| 07:00.23 | hellop | aparently the zaptel package contains only ztcfg not the zaptel or wcfxo modules.. anyone know if they are apt-gettable? |
| 07:01.22 | Telamon | hellop: There should be two packages, a zaptel tools and a zaptel modules. You need to apt-get the modules one. Sorry, but I don't use Debian, so I don't know what the particular package name would be. |
| 07:07.10 | hellop | Telamon, thanks.. I'll let ya know when I figure it out. |
| 07:07.28 | *** part/#asterisk obsidian-studios (n=obsidian@c-24-129-16-31.hsd1.fl.comcast.net) |
| 07:07.46 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 07:08.09 | Telamon | hellop: This might help http://packages.debian.org/unstable/devel/zaptel-source |
| 07:08.11 | emergion | Hello all I needed some help with making calls from my provider which uses g729 to my gsm server I get the following error |
| 07:08.50 | emergion | chan_iax2.c:7900 iax2_request: Unable to create translator path for unknown to g729 on IAX2/pauly-3 |
| 07:08.59 | Telamon | emergion: You can't do gsm->g729 without installing a g729 module on your GSM server, which means that you need to buy a license. |
| 07:09.21 | Telamon | Or rather, to be legal, you have to buy a license. There are modules out there that work without the license file. |
| 07:09.45 | tzafrir | try pdo.debian.net instead of packages.debian.org for searches and alike |
| 07:09.54 | hellop | Telamon yeah I guess I have to build from source instead of just apt-get zaptel I dunno why tho.. |
| 07:10.15 | emergion | Got the names of any? |
| 07:10.20 | Telamon | emergion: http://kvin.lv/pub/Linux/Asterisk/ |
| 07:10.51 | emergion | Thanks ill have a look now |
| 07:10.54 | tzafrir | Anyway, http://pkg-voip.buildserver.net/ has the latest Debian packages (much untested) built for various platforms |
| 07:10.57 | Telamon | emergion: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ Is the website with more info. |
| 07:11.05 | Telamon | BTW: You a fellow Gentoo user? |
| 07:11.17 | emergion | Yes |
| 07:11.40 | Telamon | Ah, figured from the nick. :) |
| 07:11.55 | emergion | hehe was hoping your were going to give me some portage action |
| 07:12.06 | emergion | but yeah thats where it came from ;-) |
| 07:12.24 | emergion | So i should be able to just download and load one of those modules in yes? |
| 07:13.01 | Telamon | emergion: Yep, copy it into /var/lib/asterisk/modules and then restart asterisk, or enter 'module load chan_g729.so' from the console. |
| 07:13.10 | tzafrir | hellop, I have also uploaded yesterday packages to http://rapid.dotsrc.org/experimental/ (deb http://rapid.dotsrc.org/ experimental/ ) |
| 07:13.17 | emergion | will do now thanks mate |
| 07:13.31 | Telamon | np |
| 07:14.06 | emergion | Oh no I am using asterisk on a soekris board ;-( |
| 07:14.10 | emergion | have to do some wild googling |
| 07:14.39 | DiscusZ | How can I debug why my calls out to FWD do not work? when I dial 393612 i get a recorded message which I am sure is my asterisk box (it says all circuits are busy) |
| 07:14.47 | tzafrir | hellop, the latter has zaptel-modules built (if you don't use a custom kernel) |
| 07:15.24 | tzafrir | DiscusZ, start with 'set verbose 3' in the CLI and looking at the output there |
| 07:15.55 | hellop | tzafrir, tks for the info |
| 07:16.20 | DiscusZ | perfect thanks very new to asterisk |
| 07:22.16 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-229-203.dsl.irvnca.pacbell.net) |
| 07:22.55 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 07:28.08 | DiscusZ | <PROTECTED> |
| 07:37.19 | hellop | tzafrir, thanks again man, all those deb packages look great. gtg ttyl |
| 07:39.36 | *** join/#asterisk Olobola (n=casper_s@216.218.221.166) |
| 07:40.40 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 07:43.43 | *** join/#asterisk lalito (n=erg@201.154.26.12) |
| 07:45.51 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 07:46.20 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 07:46.33 | *** join/#asterisk Huynh (n=chatzill@adsl-71-131-15-10.dsl.sntc01.pacbell.net) |
| 07:46.55 | Huynh | hi |
| 07:47.43 | DiscusZ | hi |
| 07:48.04 | kram | hi |
| 07:48.12 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-51.claranet.co.uk) |
| 07:48.21 | DiscusZ | I am getting frustrated :) |
| 07:48.42 | DiscusZ | I can not figure out why FWD calls do not leave my asterisk box... |
| 07:48.52 | Qwell | kram: y0 |
| 07:49.00 | Qwell | kram: Shouldn't you be in bed? ;) |
| 07:49.04 | kram | i am |
| 07:49.09 | kram | just not off the computer yet |
| 07:49.11 | Qwell | damn |
| 07:49.32 | Qwell | kram: nice job with the iax thread stuff |
| 07:49.35 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 07:49.51 | kram | thanks |
| 07:51.29 | kram | working with jerjer to try to isolate a last few things |
| 07:53.40 | Huynh | what is the diff between a channel and a trunk? |
| 07:54.06 | Telamon | Huynh: A trunk can contain multiple channels, I believe. |
| 07:54.38 | Telamon | IE, a PRI is a trunk, that can contain up to 23 channels. A plain old phone line (ie, an X100) is a trunk that contains only 1 channel. |
| 07:55.24 | Huynh | ok, thx...i've been reading so many docs; some seem to use the terms interchangibly |
| 07:55.48 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 07:57.02 | Qwell | kram: yeah, he's probably one of the better people to test |
| 07:58.14 | Qwell | off to bed.. |
| 07:58.15 | glm2k | DiscusZ: does iax2 show peers tell you FWD is ok? |
| 07:58.42 | DiscusZ | yes |
| 08:00.23 | DiscusZ | I have 2 peers listed under that command |
| 08:01.12 | DiscusZ | when verbose is set to 3 it looks like its using dialtrunk 2 |
| 08:01.35 | NotFreak | yawn ¬_¬ too early to wake up |
| 08:02.41 | glm2k | have you tried the Call Me feature on FWD? |
| 08:02.56 | DiscusZ | yes it worked thje last time i tried it.. let me try again |
| 08:03.35 | DiscusZ | Yup works |
| 08:03.48 | glm2k | so it's just outgoing .... |
| 08:03.57 | DiscusZ | Yup |
| 08:04.18 | DiscusZ | and I just can not find why.. |
| 08:09.36 | *** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it) |
| 08:10.36 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 08:15.00 | *** join/#asterisk tsume (n=tsume@zanshin.tsumelabs.com) |
| 08:15.44 | tsume | heres a good question. I'm setting up an asterisk box in a lodge with 10 cabins. Each cabin has 2 phones. How do I designate an extension to ring on both phones |
| 08:15.52 | tsume | without making a queue. they shouldn't need to log in ;) |
| 08:17.49 | *** join/#asterisk sunil (n=sunil@202.54.37.186) |
| 08:18.30 | Skid | exten => blah,1,Dial(SIP/phone1&SIP/phone2) |
| 08:18.37 | Skid | or whatever else you use to dial |
| 08:19.05 | tsume | wow.. |
| 08:19.11 | tsume | I didn't know you could use & |
| 08:19.33 | tsume | voicemail is kept on the file basis, yes? |
| 08:19.46 | Skid | yep |
| 08:19.49 | sunil | is there any body who can help me in voice cut through in Asterisk |
| 08:19.49 | tsume | can I just delete files, or does it keep a data structure somewhere in asterisk? |
| 08:20.03 | tsume | Skid: basically.. its a lodge.. I want to create software for them |
| 08:20.05 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:20.05 | Skid | /var/spool/asterisk/voicemail or somthing |
| 08:20.20 | tsume | Skid: so just deleting the files makes everything magical? :) |
| 08:20.27 | Skid | depending on the context that the voicemail is in, depends what dir the files are stored in |
| 08:20.33 | Skid | yep |
| 08:20.49 | tsume | magical.. I haven't used asterisk at a lodge, so I never looked in to the internals :) |
| 08:21.18 | tsume | hopefully these grandstream budget tone phones are simple enough =] |
| 08:21.46 | tsume | polycom phones just wont fly.. The questions I was asked for just the 501s at the business locations where just.. lots |
| 08:23.05 | Skid | ya, budget tones are alright |
| 08:23.11 | Skid | cheap enough |
| 08:23.15 | Skid | I know a few ISPs that use them |
| 08:23.32 | tsume | Skid: well, they are more complex than nortel 2840s. but still ;) |
| 08:23.49 | tsume | Skid: I don't think a person who "just wants to use the phone" wants a polycom |
| 08:24.01 | Skid | :) |
| 08:24.13 | Skid | I use 7940's but damn, never buying any again |
| 08:24.15 | Skid | cheap things |
| 08:24.22 | tsume | 7940s.. :( |
| 08:24.24 | Skid | I've had to basically fix, myself |
| 08:24.26 | tsume | I'm so forgetful |
| 08:24.36 | Skid | cisco 7940 |
| 08:24.50 | tsume | I can't remember :) |
| 08:24.52 | Skid | :) |
| 08:24.55 | tsume | maybe it was 2140 |
| 08:25.35 | Skid | their so cheap, yet seem to be a market competitor :o |
| 08:25.51 | Skid | gotta be some down fall :) |
| 08:25.56 | tsume | damn I'm forgetful |
| 08:25.58 | tsume | 9310 |
| 08:26.01 | tsume | and 9320 |
| 08:26.11 | tsume | :) I hate those phones |
| 08:26.23 | tsume | I really hate them. they cost 80 dollars each, and they break frequently |
| 08:26.34 | Skid | thats cheap really i guess |
| 08:26.35 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:26.43 | Skid | budgettones here are like 50 quid, which is i guess 100 usd |
| 08:26.55 | Skid | brb.. |
| 08:26.57 | tsume | Skid: what? |
| 08:27.06 | tsume | I get my budgettones for 40USD |
| 08:27.28 | tsume | Skid: I'm a bulk buyer though, don't listen to my prices :) |
| 08:28.01 | Skid | :) |
| 08:28.09 | tsume | I'm also a talker :) |
| 08:28.15 | tsume | I'll sale any system ; |
| 08:28.30 | x86 | you can get retail budgettones for $40 USD :P |
| 08:29.12 | x86 | you're crazy if you pay that in bulk lol |
| 08:29.16 | tsume | x86: black ones? I doubt it |
| 08:29.26 | x86 | sure |
| 08:29.26 | tsume | maybe the crappy white ones which sunbleach in 40 days |
| 08:29.27 | x86 | ebay.com |
| 08:29.31 | tsume | x86: pfft |
| 08:29.38 | tsume | x86: I buy them from the stock |
| 08:29.41 | x86 | new in box, lot of 1000 if you want |
| 08:29.54 | tsume | hmm |
| 08:30.05 | tsume | I wouldn't mind buying them off ebay if I don't get screwed again |
| 08:30.13 | tsume | the last guy I tracked down :) |
| 08:30.26 | x86 | i've never got screwed on ebay |
| 08:30.35 | x86 | i always check their feedback... |
| 08:30.40 | x86 | good feedback == good seller ;) |
| 08:31.02 | tsume | x86: I did too, but the asshole signed up multiple accounts and used a merry go round feedback |
| 08:31.14 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:31.26 | tsume | didn't look like it until you checked the user's feedback who left feedback |
| 08:39.01 | Skid | right, shower time before test driving two new motors :-) |
| 08:40.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:42.00 | *** join/#asterisk TallAndy (i=AndyD@82.163.6.217) |
| 08:42.23 | *** join/#asterisk marta (n=a@host40.natpool.mwn.de) |
| 08:42.47 | TallAndy | Hi, can I get this message back into an AGI script --> -- Got SIP response 486 "Busy here" back from 83.245.6.81 |
| 08:43.01 | TallAndy | Need to detect if a line is busy |
| 08:43.48 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:46.41 | *** join/#asterisk puzzled (n=yeahrigh@62.45.11.228) |
| 08:47.32 | puzzled | morning |
| 08:47.36 | Winkie | morning |
| 08:47.39 | Winkie | you a brit then? :) |
| 08:48.26 | NotFreak | goedemorgen |
| 08:48.41 | puzzled | Winkie: nope |
| 08:48.47 | Winkie | puzzled: or close? |
| 08:48.49 | Winkie | :) |
| 08:48.56 | puzzled | hu NotFreak. how' the patch and solder holding up? |
| 08:49.03 | puzzled | s/hu/hi |
| 08:49.06 | NotFreak | haha |
| 08:49.11 | NotFreak | the soldering is going ok |
| 08:49.17 | NotFreak | the patching is driving me nuts |
| 08:49.33 | NotFreak | damn my inferior c skills :( |
| 08:49.34 | puzzled | yes, many have gone insane before you trying to patch zaptel |
| 08:49.36 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 08:50.24 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:51.38 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 08:51.46 | NotFreak | would be nice if i could make a nice visual flow of what files/functions are called throughout the process |
| 08:52.39 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 08:53.14 | puzzled | NotFreak: did you try 'make progdocs' in the source for an overview of all the functions? |
| 08:53.55 | Skid | +w 17 |
| 08:54.16 | puzzled | -x/y+3 |
| 08:54.17 | NotFreak | i use the doxygen site |
| 08:54.22 | puzzled | ok |
| 08:54.25 | NotFreak | but still gotta go through it manually ofcos |
| 08:54.54 | puzzled | yup |
| 08:55.43 | Skid | damn numlock on, couldn't change irssi windows ;) |
| 08:56.15 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:56.23 | *** join/#asterisk mkl1525 (n=daniel@82.100.204.245) |
| 09:06.17 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-70-63.dsl.irvnca.pacbell.net) |
| 09:07.16 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 09:13.16 | *** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au) |
| 09:13.27 | emergion | anyone got any sites on setting up ring groups ? |
| 09:15.16 | x86 | you mean ACD? |
| 09:15.31 | x86 | like hunt groups? |
| 09:16.21 | emergion | yeah |
| 09:16.27 | emergion | so all the extensions ring when a DID calls |
| 09:17.01 | x86 | just make an extension that rings multiple locations |
| 09:17.16 | emergion | need abit of help with the syntax |
| 09:17.26 | emergion | cant seem to find much on it |
| 09:17.35 | x86 | exten => 6000,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103) |
| 09:17.55 | x86 | where 6000 is the hunt group |
| 09:18.07 | emergion | Oh ok thanks ;-0 |
| 09:18.11 | emergion | ;-) |
| 09:18.48 | x86 | you can do other things too, like play a "Please hold for the next available operator" sound or something, before it rings any agents :) |
| 09:19.10 | X-Rob_ | 600,1,Playback(you-suck) |
| 09:19.14 | x86 | i'm sure there are better ways to do it, where agents can login/logout, but not sure how to go about doing all that... this is plain and simple ;) |
| 09:19.14 | X-Rob_ | 600,2,Dial(SIP...) |
| 09:19.36 | x86 | hmm |
| 09:19.46 | x86 | my asterisk is telling me it doesnt know what Playback is.... |
| 09:20.10 | x86 | i think i might have screwed it up heh |
| 09:20.30 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:26.32 | *** part/#asterisk Grizzy (i=Generic@adsl-68-127-29-121.dsl.pltn13.pacbell.net) |
| 09:28.24 | *** join/#asterisk Grizzy (i=Generic@adsl-68-127-29-121.dsl.pltn13.pacbell.net) |
| 09:29.48 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 09:31.39 | x86 | hmm |
| 09:32.19 | x86 | i'm thinking about buying a SIP softphone for my PDA.... anyone ever do that? |
| 09:33.55 | Winkie | what PDA? |
| 09:33.59 | Winkie | also PM you bastard :) |
| 09:34.03 | x86 | ah ;) |
| 09:34.14 | Winkie | emergion: either do what x86 says or use a queue with 'ringall' |
| 09:35.45 | *** join/#asterisk Conductor (n=thomas@62.8.240.185) |
| 09:35.48 | Conductor | hi all! |
| 09:36.52 | Grizzy | Grizmorning. |
| 09:37.41 | Conductor | how would i do this: someone calls a number, let's say 222. After this he gets an hangup and is called back (by his ${callerId}) and transferred to a specific extension, like 765. |
| 09:39.46 | *** join/#asterisk pawal (n=pawal@212.247.14.36) |
| 09:40.11 | *** join/#asterisk kll (i=kll@insomnia.juniks.net) |
| 09:42.29 | kll | anyone seen this message: Unable to calculate sample length for format unknown |
| 09:45.20 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 09:45.25 | *** join/#asterisk TallAndy (i=AndyD@83.104.196.69) |
| 09:45.43 | emergion | exten => _9.,1,Dial(SIP/${EXTEN:1}@exetel-out,30,r) how do I dial with that extension just 9 then the number? |
| 09:50.12 | TallAndy | Is there a way of getting the hangupcause back when using the originate manager command to make a call? |
| 09:50.41 | kll | emergion: please explain further what you are trying to accomplish |
| 09:53.06 | *** join/#asterisk usam (n=alx@203.156.48.250) |
| 09:54.24 | usam | when calling from china, dont they send out the CLI? got a call from china, and the CLI was from HK ... it is just like this? |
| 09:57.23 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:57.33 | *** join/#asterisk blkremedy (n=ur3rdeye@083M13.oasis.mediatti.net) |
| 09:57.38 | blkremedy | hello all |
| 09:57.44 | Winkie | hey |
| 09:58.27 | heroine | hi |
| 09:58.50 | *** join/#asterisk Qber (n=Qbera@c-24-6-80-84.hsd1.ca.comcast.net) |
| 09:59.02 | blkremedy | has anyone here tried the wbp54g wireless bridge for phone adapters from linksys? |
| 09:59.03 | Qber | how do I know that g729 is registered on 1.2.4 |
| 09:59.12 | Qber | somehow the command is missing show codec g29 |
| 09:59.15 | Qber | g729 |
| 09:59.34 | Qber | when i try to connect with an endpoint using g729, its failing |
| 09:59.42 | Qber | i have registered a codec from digium |
| 09:59.50 | TallAndy | I can get back originate sucessful or orginiate failed, but I need to differentiate between hungup / no answer etc. |
| 10:00.03 | Qber | its a dual xeon 64bit, with 1.2.4 on FEDORA core 4 running |
| 10:00.08 | Qber | what could I be missing |
| 10:00.19 | Qber | anyone knows??? |
| 10:01.40 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 10:04.41 | Conductor | how would i do this: someone calls a number, let's say 222. After this he gets an hangup and is called back (by his ${callerId}) and transferred to a specific extension, like 765. |
| 10:08.47 | *** join/#asterisk _fenlander (n=fenlande@82.152.81.57) |
| 10:11.34 | mkl1525 | Hi, just read on http://www.voip-info.org/wiki/view/Asterisk+manager+experience that if you make heavy use of the manager interface it will deadlock * at least once a day - is this still true with current versions or is it history? |
| 10:13.00 | *** join/#asterisk X-Rob_ (n=Rob@dsl-220-235-91-96.vic.westnet.com.au) |
| 10:19.48 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
| 10:20.06 | Skumling | hi there |
| 10:20.34 | Skumling | anyone want to join me in killing the developers of Cisco 7970? ;) |
| 10:20.53 | vgster | ill help you bring down cisco as a whole |
| 10:21.11 | Skumling | vgster: mmm, bring down :) |
| 10:21.32 | Skumling | their phones are really really not built to run without their CallManager |
| 10:21.47 | vgster | yes thats the point |
| 10:22.00 | vgster | and dont ask for support unless you have the maintenance contract |
| 10:22.14 | Skumling | seems like one can't get hold of the needed software |
| 10:22.29 | vgster | use grandstreams |
| 10:22.58 | Skumling | even though I know people who has access to CCO, some of the "localization" stuff seems to be present only in CallManager |
| 10:23.59 | Winkie | i quite like the sipura 941 |
| 10:24.13 | vgster | i have 1 of those for testing |
| 10:24.18 | Winkie | yeah we have a couple |
| 10:24.21 | Winkie | they seem quite nice |
| 10:24.25 | Skumling | I would really love to have my Cisco 7970 online on asterisk :) |
| 10:24.40 | Winkie | we also have grandstreams, aastras and an 841 |
| 10:24.46 | Skumling | I'm sure that it's a nice phone, if I could just get it to bootup |
| 10:24.46 | Winkie | and i prefer the gxp2000 and the 841 |
| 10:24.48 | Winkie | uh, 941* |
| 10:24.49 | vgster | i prefer the 9133i |
| 10:24.58 | vgster | but the gxp firmware is getting there |
| 10:25.08 | Winkie | the build quality on the 9133i is laughable though |
| 10:25.13 | Winkie | the volume button is the best part |
| 10:25.17 | Winkie | just a single bar of plastic that's barely held in |
| 10:25.37 | Winkie | at least the gxp feels like it's made from plastic you wouldn't find encasing a drink |
| 10:25.50 | vgster | i thin kthe 9133i is quite a solid phone |
| 10:26.03 | Winkie | well the 480i is laughable too |
| 10:26.11 | vgster | the buttons arent the best but the handset and general feel of the phone is ok |
| 10:26.12 | Winkie | 'register with asterisk? hah you must be joking' |
| 10:26.33 | Winkie | i LOVE the screen on the gxp though, although i haven't tested it with a headset yet |
| 10:27.19 | Winkie | still we'll see how i like it when i upgrade the firmware, gotta get devstate working for logins etc |
| 10:29.17 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:35.30 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:38.07 | *** join/#asterisk fourcheeze (n=rich@82.153.215.21) |
| 10:38.22 | fourcheeze | any know why * doesn't support g726-16 ? |
| 10:38.48 | fourcheeze | only seems to handle g726-32 |
| 10:39.13 | fourcheeze | trying to explain to a customer why he can't have any of the other varieties |
| 10:39.54 | vgster | tell him he doesnt need them |
| 10:39.58 | RoyK | :) |
| 10:40.02 | RoyK | use g.729 instead |
| 10:40.09 | vgster | tell him he only needs them if he is in bulgaria |
| 10:40.39 | RoyK | ? |
| 10:40.56 | vgster | excuses to tell customers |
| 10:41.01 | fourcheeze | hehe |
| 10:41.15 | fourcheeze | RoyK: never tried g729 - isn't there a royalty with that? |
| 10:41.16 | RoyK | no, that only works with cheese |
| 10:41.33 | RoyK | fourcheeze: $10 per call to digium |
| 10:41.45 | fourcheeze | per call??? |
| 10:42.22 | RoyK | concurrent call |
| 10:42.25 | fourcheeze | phew |
| 10:42.27 | RoyK | :) |
| 10:42.27 | fourcheeze | :-) |
| 10:42.39 | RoyK | we use that |
| 10:42.39 | fourcheeze | how do you rate the sound quality compared to ulaw / gsm / g726-32 ? |
| 10:42.41 | RoyK | works well |
| 10:42.55 | RoyK | about the same as g.726 |
| 10:43.01 | fourcheeze | so pretty good then |
| 10:43.07 | RoyK | but we're dropping g.729 now |
| 10:43.13 | RoyK | getting the new sip jb we don't need it |
| 10:43.31 | RoyK | also the new sip jb has excellent PLC, something that's lacking everywhere else |
| 10:43.53 | fourcheeze | sip jb ? |
| 10:44.14 | fourcheeze | did I miss the memo? |
| 10:44.29 | RoyK | http://bugs.digium.com/view.php?id=3854 |
| 10:44.58 | fourcheeze | how does that relate to codecs? |
| 10:45.09 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 10:45.11 | *** join/#asterisk fulgas (n=fulgas@209.8.233.247) |
| 10:46.22 | RoyK | fourcheeze: for dialup users, g.729 is used for those with low bandwidth. with g.711a and plc+jb you can sustain quite high packet loss and jitter without getting audio problems |
| 10:47.21 | fourcheeze | ahh ok |
| 10:48.13 | RoyK | not useful for trunking lots of calls over narrow bandwidth, though |
| 10:48.16 | fourcheeze | anyone use ilbc ? |
| 10:48.44 | RoyK | nah |
| 10:48.59 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:49.12 | dudes | ilbc is good |
| 10:49.30 | fourcheeze | what's the bandwidth use on it? |
| 10:50.23 | dudes | Not sure, never looked |
| 10:50.48 | dudes | I was able to manage 2 g729's and 1 ilbc on a 256k/bit cable NP |
| 10:51.21 | fourcheeze | I don't think it's supported on many phones |
| 10:51.33 | fourcheeze | at least sipuras and snoms don't seem to have it |
| 10:51.36 | dudes | Then don't use it |
| 10:51.37 | dudes | heh |
| 10:51.48 | RoyK | dudes: you should be able to run three G.711 connections over taht |
| 10:51.53 | dudes | Yea |
| 10:52.10 | dudes | I've had 4 g726's on it |
| 10:52.11 | RoyK | G.729A + overhead with 20ms packetization == 24kbps |
| 10:52.28 | RoyK | g.726-32 == 48kbps |
| 10:52.33 | fourcheeze | aha |
| 10:52.35 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 10:52.47 | fourcheeze | RoyK: are these figures documented somewhere? |
| 10:53.00 | RoyK | ~bandwidth |
| 10:53.02 | jbot | [bandwidth] This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 10:53.04 | x86 | anyone in UK want to give me a test call? |
| 10:53.04 | RoyK | ~codecs |
| 10:53.06 | jbot | somebody said codecs was http://snipurl.com/wiki_codecs |
| 10:53.52 | RoyK | generally, the RTP+UDP+IP overhead with 20ms packetization is 16kbps |
| 10:53.53 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 10:53.54 | RoyK | flat |
| 10:54.20 | RoyK | and since asterisk only supports 20ms packetization, it's simple :) |
| 10:54.28 | fourcheeze | :-) |
| 10:55.35 | fourcheeze | while we're on codecs |
| 10:55.44 | fourcheeze | I see that * supports g723.1 in passthru mode |
| 10:55.55 | RoyK | no |
| 10:56.01 | RoyK | it does't support transcoding |
| 10:56.05 | fourcheeze | http://voip-info.org/wiki/index.php?page=Asterisk+G.723+pass-thru |
| 10:56.08 | RoyK | no, it never will |
| 10:56.11 | fourcheeze | well that's what I mean |
| 10:56.14 | RoyK | yes, it supports passthrough |
| 10:56.29 | RoyK | but g.723.1 licensing is horribly expensive |
| 10:56.44 | fourcheeze | is it possible to configure * such taht it will use g723 in passthru mode |
| 10:56.47 | RoyK | it's FAR better to rewrite asterisk to support variable packetization |
| 10:56.50 | fourcheeze | and fall back to something else? |
| 10:57.00 | fourcheeze | if it needs to? |
| 10:57.04 | RoyK | forget it |
| 10:57.07 | fourcheeze | ok |
| 10:57.09 | RoyK | rewrite packetization |
| 10:57.12 | fourcheeze | :-) |
| 10:57.13 | RoyK | in comparison |
| 10:57.43 | RoyK | changing from g.729a to g.723.1 will alter the total bandwidth from 24kbps to 21.5kbps |
| 10:57.47 | fourcheeze | is it only transcoding that we pay for with g729? |
| 10:58.02 | fourcheeze | so if all our customers used g729 |
| 10:58.07 | RoyK | changing packetization from 20ms to 40ms with g.729a will alter the total bandwidth from 24kbps to 16kbps |
| 10:58.14 | RoyK | yes |
| 10:58.14 | fourcheeze | we would only have to pay for concurrent calls to voicemail etc? |
| 10:58.16 | RoyK | only transcoding |
| 10:58.23 | fourcheeze | hmmmm |
| 10:58.24 | RoyK | voicemail is also transcoded |
| 10:58.29 | fourcheeze | yes |
| 10:58.33 | fourcheeze | ahh |
| 10:58.37 | fourcheeze | yes |
| 10:58.45 | fourcheeze | so a channel to voicemail counts as a call |
| 10:58.56 | fourcheeze | but a channel to another g729 user doesn't? |
| 10:59.14 | fourcheeze | who is keeping score BTW? |
| 10:59.26 | fourcheeze | does asterisk stop working after so many concurrent channels? |
| 10:59.35 | RoyK | transcoding stops working |
| 10:59.37 | RoyK | show g729 |
| 11:00.39 | fourcheeze | hmm? |
| 11:01.46 | RoyK | hm. |
| 11:01.57 | fourcheeze | show g729 doesn't do anything for me |
| 11:01.58 | RoyK | sipgw1*CLI> show g729 |
| 11:01.58 | RoyK | 26/26 encoders/decoders of 75 licensed channels are currently in use |
| 11:02.05 | fourcheeze | ahh ok |
| 11:02.15 | fourcheeze | I guess I need to install a module first |
| 11:02.33 | fourcheeze | so those 75 channels cost $750 per month |
| 11:03.02 | RoyK | codec_g729a.so |
| 11:03.02 | fourcheeze | are there any mainstream clients that do not support g729? |
| 11:03.07 | RoyK | not per month |
| 11:03.13 | fourcheeze | one off? |
| 11:03.17 | RoyK | just one fee, yes |
| 11:03.19 | fourcheeze | ok |
| 11:03.24 | RoyK | so not bad, really |
| 11:03.28 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 11:03.36 | fourcheeze | so those 26 calls are all to voicemail or else transcoding to non g729 ? |
| 11:04.18 | RoyK | this box does't do voicemail |
| 11:04.20 | *** join/#asterisk zotz (n=zotz@24.244.133.10) |
| 11:04.30 | RoyK | ~seen zoa |
| 11:04.36 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 7d 16h 57m 11s ago, saying: '1000 interrupts per second are ok'. |
| 11:06.40 | *** join/#asterisk n3c8 (n=mat@84.19.35.10) |
| 11:08.05 | fourcheeze | RoyK: so some people are using g711? |
| 11:08.17 | fourcheeze | I'm wondering if I shold just insist on everyone using g729 |
| 11:08.36 | fourcheeze | since we have a g729 outbound feed I believe |
| 11:08.45 | fourcheeze | and inbound |
| 11:09.22 | RoyK | we're prolly going for g.711a only when the new jb stabilises |
| 11:11.24 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 11:11.42 | Chotaire | oh yeah, using g711 everywhere. |
| 11:12.24 | Chotaire | g729 doesn't do inband dtmf right? ;) |
| 11:13.03 | Chotaire | and some of my carriers only allow g711 |
| 11:13.42 | Chotaire | also I found out meetme is having least echoeing when using only one codec. |
| 11:14.19 | x86 | anyone in UK? |
| 11:17.18 | TallAndy | ye |
| 11:18.54 | RoyK | inband dtmf works with g.711 only |
| 11:19.11 | RoyK | theoretically it's possible over g.726-32 too, but don't count on it |
| 11:24.52 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 11:26.29 | *** part/#asterisk vmedrano (n=vmedrano@196.32.128.206) |
| 11:32.42 | vgster | im in uk |
| 11:33.55 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 11:33.59 | RoyK | .no way :) |
| 11:34.18 | *** join/#asterisk clive-- (n=pirch@dsl-145-33-235.telkomadsl.co.za) |
| 11:34.35 | clive-- | is anyone here using chan_capi ? |
| 11:37.24 | puzzled | clive--: I have used it and will again when I have rebuilt this box |
| 11:38.10 | puzzled | RoyK: interested in playing with a iax2 multithreaded patch for 1.2 branch? |
| 11:38.23 | x86 | how do i force an IAX2 registration from the CLI? |
| 11:38.47 | puzzled | dafaik you can't |
| 11:38.52 | puzzled | -d |
| 11:39.03 | x86 | 3 iax2 peers [0 online, 3 offline, 0 unmonitored] |
| 11:39.51 | puzzled | I guess you could reload chan_iax2.so |
| 11:40.30 | RoyK | puzzled: i may... where is it? |
| 11:40.55 | x86 | puzzled: how? |
| 11:41.05 | puzzled | x86: reload chan_iax2.so |
| 11:41.28 | puzzled | RoyK: I can email it to. msg me your addy |
| 11:41.47 | x86 | 3 iax2 peers [3 online, 0 offline, 0 unmonitored] |
| 11:41.52 | x86 | that worked, thanks man :) |
| 11:41.57 | puzzled | :) |
| 11:43.58 | RoyK | roy@karlsbakk.net |
| 11:44.58 | *** join/#asterisk sl16 (n=blah@tv.neterra.net) |
| 11:46.17 | *** join/#asterisk a5t3r15k (n=noguess@81.168.89.146) |
| 11:46.23 | clive-- | puzzled I am having a strange behavouiur with only 1 of the 4 ports becomming active |
| 11:47.43 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 11:47.49 | a5t3r15k | Hi, we've just had ISDN30E installed and have hooked up to our asterisk box using Digium TE110P card but I get a red alert. How do i check this? Thanks |
| 11:51.08 | fourcheeze | RoyK: any idea if asterisk is going to support 40ms packets? |
| 11:51.27 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:51.41 | clive-- | fourcheeze there is a way to set that in the source |
| 11:51.57 | fourcheeze | well it would be nice to support both |
| 11:53.27 | RoyK | fourcheeze: search the archives. |
| 11:54.37 | puzzled | clive--: I have a BRI card so don't know about multiport stuff but if you send an email to the asterisk-users list you will usually get a quick reply from Armin, the author of chan_capi-cm |
| 11:54.58 | puzzled | clive--: it helps if you mention chan_capi in the subject |
| 11:56.11 | clive-- | puzzled, thanks, I have sent an email to Armin |
| 11:57.01 | *** join/#asterisk voip470 (n=A_mail@pool-71-123-48-33.phlapa.fios.verizon.net) |
| 11:59.09 | *** join/#asterisk agkhram (n=arheil@yasski.sam-isp.net) |
| 11:59.25 | agkhram | hello |
| 12:00.14 | *** join/#asterisk vgster (n=vg@host217-45-221-53.in-addr.btopenworld.com) |
| 12:04.37 | agkhram | we have 15 external numbers. all outgoing calls are initiated by asterisk. asterisk uses OH323/xxx@remoteCISCOmediaGW. i've written alias=xxxxxxx in the /etc/asterisk/oh323.conf and cisco has seen all calls are initiated from xxxxxxx number. i want to split default context to 15 groups (15 numbers). What is the right way? |
| 12:05.24 | *** join/#asterisk Inkubot (n=inkubot@200.74.170.218) |
| 12:06.06 | agkhram | cisco makes accounting so i have to set "right" number (one from 15). |
| 12:07.30 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 12:13.00 | *** join/#asterisk ibob63 (n=hp@bb-87-82-22-110.ukonline.co.uk) |
| 12:13.11 | agkhram | i've tried exten => _9XXXXXXX,1,SetCIDNum(one_from_15) /// exten => _9XXXXXXX,2,Dial(OH323/${EXTEN:1}@mediaGW). It doesn't work. |
| 12:13.17 | ibob63 | morning all |
| 12:14.54 | agkhram | cisco has seen [voip-h323] /// alias=YYYYYYY from /etc/asterisk/oh323.conf only. |
| 12:15.06 | agkhram | any clue? |
| 12:15.19 | agkhram | hello ibob63 |
| 12:15.30 | ibob63 | hey agkhram |
| 12:15.51 | ibob63 | sorry, I can't help - i am at the start of my asterisk adventures |
| 12:26.37 | ibob63 | I can't seem to make outgoing calls. It was working and then suddenly stoped. Can anyone tell me how to start debugging - or point me to a good guide? |
| 12:27.47 | agkhram | ibob63: what channel type do you use? |
| 12:28.12 | ibob63 | I am sip based. |
| 12:28.17 | *** join/#asterisk astar` (n=astar@ANantes-154-1-1-69.w81-53.abo.wanadoo.fr) |
| 12:28.26 | agkhram | asterisk>sip debug |
| 12:29.22 | ibob63 | thanks - now I get loads of line - rather aload of information |
| 12:30.31 | ibob63 | the final line says destorying call foo_bar@ip_address. Is there an explaination of this information somewhere? |
| 12:32.05 | agkhram | www.cisco.com. use the google +SIP +something_specific |
| 12:33.09 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:34.23 | ibob63 | if I put the output in a paste bin will it reveal any of my passwords / secrets used for making connections? |
| 12:35.36 | RoyK | if it does, remove them first |
| 12:35.47 | RoyK | it shouldn't unless you've hardcoded them in the dialplan |
| 12:36.24 | ibob63 | I don't think I have hardcoded them into the dailplan - so heres the debug output: http://pastebin.com/576502 |
| 12:38.52 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 12:43.25 | *** join/#asterisk NDT (n=noone@cpe-24-195-218-134.nycap.res.rr.com) |
| 12:44.58 | *** join/#asterisk Rhizome (n=rhizome@tor/session/x-fdb0a949e8c9dca0) |
| 12:45.20 | ibob63 | are there any other debugging tools that I can use to keep that asterisk is connecting correctly with the trunk? |
| 12:47.49 | RoyK | http://www.ascotel.com/office25.htm |
| 12:48.02 | RoyK | anyoen here that can help me finding out what sort of phone this is? |
| 12:48.14 | RoyK | analog? isdn? proprietary? |
| 12:49.48 | mut | what causes far end echo? |
| 12:49.58 | mut | i got this tellab canceller in and far end has echo |
| 12:53.13 | *** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca) |
| 12:53.28 | a5t3r15k | using ddi, if i needed to split 1 long number into two variables for use is there an easy way to do it? |
| 12:54.09 | queuetue | Hi. I have an asterisk@home installation that sends 100 empty voicemails to a specific extension every once in a while - maybe once a month, twice in the last few days. Is this a known issue? |
| 12:54.33 | queuetue | It does not seem to be tied to any logged user activity. |
| 12:54.46 | puzzled | RoyK: looks like a proprietary key system phone |
| 12:55.06 | *** join/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 12:57.53 | NotFreak | lalalala |
| 12:59.01 | NotFreak | more more more |
| 12:59.31 | NotFreak | i think i'm near the root of evil now |
| 13:00.34 | mut | i can't get these tellab cancellers to get rid of the echo |
| 13:01.11 | ibob63 | what does a warning about "noisy feedback" mean? |
| 13:02.09 | r_evolution | anyone have experience connected * with mysql? |
| 13:02.25 | r_evolution | interesting issue... v1.2.4... can read/write to CDR... |
| 13:02.34 | r_evolution | asterisk formats the sql statement correctly... |
| 13:02.34 | *** part/#asterisk madounet (n=mad_net@juv34-2-82-226-155-19.fbx.proxad.net) |
| 13:02.38 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 13:02.40 | r_evolution | but cannot read from the sip database |
| 13:02.42 | dpryo | I'm using mysql for CDRs, in 1.2.4 |
| 13:03.01 | r_evolution | I've got the CDR half working great... but when it comes to pulling users... it flops |
| 13:03.20 | r_evolution | therein lies my confusion |
| 13:03.25 | dpryo | I don't use that.. I have a config-file-writer that writes SIP-users to the configuration |
| 13:03.42 | dpryo | Too much bad experiences around with storing directly to mysql |
| 13:04.04 | r_evolution | See I wouldn't object to writing SIP/Exten/Vmail to a flat file |
| 13:04.04 | dpryo | and I never update anything special in the configuration that needs to be realtime |
| 13:04.22 | r_evolution | but we're porting current 300+ users from an emergent switch to a * solution |
| 13:04.41 | r_evolution | b/c the Emergent switch is a miserable failure at handling our customers |
| 13:04.48 | dpryo | :) |
| 13:05.08 | r_evolution | sooo... dual 3.8 w/ 2GBs of RAM and a RAID1 165GB SCSI |
| 13:05.48 | r_evolution | given the whole industry standard of 10/1 |
| 13:06.06 | r_evolution | (i.e. for every 10 customers... there are supposed to be 1 concurrent connections) |
| 13:06.46 | r_evolution | SO this box should hold up for a minute |
| 13:06.52 | r_evolution | but they want database configuration :-\ |
| 13:07.25 | mut | anyone around dealt with tellab echo cans? |
| 13:11.23 | *** join/#asterisk occam23 (n=seb@extgw.carmunity.de) |
| 13:12.03 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:14.00 | occam23 | hi, testing asterisk with a german isdn line using capi_chan-cm 0.6.4 / asterisk 1.2.4 build from source, dial in works just fine. Now testing the Dial Command i get the problem that only the first call works, the second one aborts with a non-zero short time afer the destination is rinning, even befor you can pick up the call |
| 13:14.17 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 13:14.52 | occam23 | its just this one liner... exten => 223xxxx,1,Dial(CAPI/g1/01xxxxxx) |
| 13:15.13 | occam23 | same if i use CAPI/ISDN1 or CAPI/contr1 |
| 13:17.10 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 13:19.25 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:19.34 | RoyK | occam23: ?223xxxx |
| 13:19.35 | RoyK | eh |
| 13:19.39 | RoyK | _223xxx |
| 13:19.51 | RoyK | prefix it with underscore |
| 13:21.06 | Rhizome | What do you guys think? lowest specs for an asterisk, mysql and apache box with about 50 users? :P |
| 13:21.45 | RoyK | Rhizome: how many concurrent calls? |
| 13:21.54 | RoyK | how large a database? |
| 13:22.01 | RoyK | what traffic on apache? |
| 13:23.21 | queuetue | I have an asterisk@home installation that sends 100 empty voicemails to a specific extension every once in a while - maybe once a month, twice in the last few days. Is this a known issue? I'm slightly concerned it's some kind of intrusion attempt. |
| 13:23.50 | RoyK | queuetue: upgrade asterisk to latest version and try again |
| 13:24.06 | tuxinator_linux | queuetue: I second RoyK |
| 13:24.21 | queuetue | RoyK: Is this a known issue? |
| 13:24.51 | tuxinator_linux | queuetue: don't your logs tell you whats going on? |
| 13:24.53 | Rhizome | I hear the rule is 10%, so 10 concurrent calls, not very big database, just for asterisk realtime and some other things, and probably the occasinal apache hit :P |
| 13:25.22 | queuetue | tuxinator_linux: No, no user activity seems to be causing it - nothing in the logs I know to check anyway. |
| 13:25.32 | queuetue | A "Random reinstall" fix sounds a little windows-ish to me. :) |
| 13:25.48 | occam23 | RoyK, hehe, i just replaced the nummber with xxx |
| 13:25.55 | tuxinator_linux | reinstall, no, upgrade |
| 13:26.10 | occam23 | its just this one liner... exten => 123456,1,Dial(CAPI/g1/1234567) |
| 13:26.18 | Rhizome | I'll probably just buy the cheapest box I can find with 512mb ram :P |
| 13:26.43 | tuxinator_linux | occam23: new underscore before number _123456,1,... |
| 13:27.01 | tuxinator_linux | occam23: or I need sleep |
| 13:27.07 | queuetue | What's the process of converting an asterisk@home install to a "standard" * install going to be like? |
| 13:27.23 | tuxinator_linux | queuetue: mmm, good luck |
| 13:28.06 | kll | queuetue: better to do it right from the beginning ;) |
| 13:28.13 | occam23 | same, the first call works fine, the second one just dies after the destination starts rinnging |
| 13:28.19 | tuxinator_linux | queuetue: you are better off installing centos 4.2 and then subersion the lastest trunk |
| 13:28.20 | r_evolution | still looking for anyone w/ realtime experience ;) |
| 13:28.40 | r_evolution | seeing as how * refuses to read from a mysql database... yet can write to the same db |
| 13:29.02 | queuetue | tuxinator_linux: If I'm starting from scratch, it will NOT be with a rpm-based distro. Certainly not one based off of RHEL. :) |
| 13:29.08 | tuxinator_linux | occam23: are you sure the dest is ringing? |
| 13:29.44 | tuxinator_linux | queuetue: I use both debian and centos, both work fine for asterisk |
| 13:29.48 | occam23 | yes, its a mobile phone right beside me, and it also show a "non ansfer call" |
| 13:30.00 | tuxinator_linux | queuetue: I'm not a fan of redhat by any means |
| 13:30.38 | ibob63 | I keep getting an error "noisy feedback" and the incoming / outgoing call dont' work - but internal do. Can anyone help me with this? |
| 13:31.29 | tuxinator_linux | queuetue: Is there such a think as 'capi debug' like 'sip debug' |
| 13:31.43 | diLLec | a question about sip realtime: |
| 13:31.47 | tuxinator_linux | queuetue: that was for occam23 |
| 13:31.48 | queuetue | tuxinator_linux: I'm sorry, I'm a newbie here. :) |
| 13:31.49 | diLLec | is musiconhold working ? |
| 13:31.54 | occam23 | capi debug shows some more infos yeah |
| 13:31.56 | occam23 | wait |
| 13:33.18 | jbenson | Hi everyone. I have UK CID working on incoming ZAP channels, but when I set outgoing CID on my FXS port it is not being sent correctly. Does anyone have any ideas please? |
| 13:33.42 | tuxinator_linux | can you set CID on a POTS line? |
| 13:34.15 | jbenson | I am not setting it on a POTS line, I have a FXS port on my TDM400P which has a phone plugged in to it |
| 13:34.35 | jbenson | The CID come in from the POTS line and I need to pass it to the FXS port |
| 13:34.41 | tuxinator_linux | oh, sending CID to the phone |
| 13:34.45 | jbenson | yes |
| 13:34.49 | jbenson | :-) |
| 13:34.54 | luke-jr_ | wtf... iConnectHere is sending me the wrong CID #... |
| 13:35.00 | occam23 | k show only one diff |
| 13:35.13 | tuxinator_linux | occam23: do tell |
| 13:35.29 | occam23 | ISDN1: received CONNECT_CONF PLCI = 0x201 |
| 13:35.30 | luke-jr_ | queuetue: AEL might help |
| 13:35.45 | occam23 | that shows up if the call works |
| 13:35.54 | occam23 | it doesnt if not.. |
| 13:36.29 | occam23 | uh, no , bullshit, got ost on the logs, same output.. |
| 13:36.40 | luke-jr_ | Unfortunately, I think I might be the only person here using AEL seriously XD |
| 13:36.53 | occam23 | no diff between broken and ok call |
| 13:36.54 | tuxinator_linux | AEL? |
| 13:37.07 | luke-jr_ | tuxinator_linux: programming language syntax alternative for extensions.conf |
| 13:37.34 | tuxinator_linux | haven't looked at it yet.. someday |
| 13:37.44 | agkhram | sounds good. |
| 13:37.44 | luke-jr_ | macro prepcall(context, exten) { &prepcall_beginmonitor(${context},${exten}); SetMusicOnHold(default); }; |
| 13:38.04 | luke-jr_ | tuxinator_linux: it's somewhat buggy, tho, so be careful |
| 13:38.13 | tuxinator_linux | occam23: you though you something scroll by on the console? |
| 13:38.20 | luke-jr_ | tuxinator_linux: hard to find syntax bugs, and for-loops are broken |
| 13:38.54 | tuxinator_linux | luke-jr_: no fun... when you work out all the bugs... I'll take a look |
| 13:39.11 | agkhram | luke-jr: is something like "goto" work? |
| 13:39.19 | luke-jr_ | tuxinator_linux: heh, I haven't done much w/ the implementation code yet |
| 13:39.29 | luke-jr_ | agkhram: I think it does. I prefer 'jump' tho |
| 13:39.54 | luke-jr_ | tuxinator_linux: for all I know, the CVS code has them fixed |
| 13:39.54 | agkhram | jump... thanks for the hint |
| 13:40.07 | luke-jr_ | agkhram: yeah, it was undocumented on voip-info, but I added it when I looked at the code |
| 13:40.25 | luke-jr_ | agkhram: seems to be 'jump extention@context' |
| 13:40.26 | tuxinator_linux | occam23: I setup mine, sometimes to a verbosity of 9999... not sure how verbose it can get, doubt it would be any higher than 5 |
| 13:42.50 | occam23 | hmmm.. -> CAPI INFO 0x34e6: Recovery on timer expiry (broken call) ---> CAPI INFO 0x3490: Normal call clearing (normal call) |
| 13:44.11 | *** join/#asterisk qzxcd (n=heroine@gut75-1-82-67-176-238.fbx.proxad.net) |
| 13:47.28 | *** part/#asterisk a5t3r15k (n=noguess@81.168.89.146) |
| 13:47.28 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:48.23 | tuxinator_linux | occam23: well... there is your start, good luck.. my eyes are too heavy |
| 13:48.27 | tuxinator_linux | night guys |
| 13:48.40 | occam23 | sleep well |
| 13:48.47 | agkhram | sweet dreams |
| 13:49.04 | tuxinator_linux | thanks |
| 13:49.49 | Druken | morning peoples |
| 13:50.11 | agkhram | good evening |
| 13:50.43 | TallAndy | Just discovered $agi->get_variable in phpagi does not work with global variables e.g ${DIALSTATUS} Is there any way of getting this variable from within an AGI script? |
| 13:51.39 | *** join/#asterisk SpooForBrains (n=wolf@82-36-140-168.cable.ubr02.perr.blueyonder.co.uk) |
| 13:51.42 | Druken | set it to a seperate non global variable before calling the agi ? |
| 13:51.43 | Druken | :) |
| 13:51.44 | SpooForBrains | hey all |
| 13:51.49 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 13:52.02 | SpooForBrains | anyone here successfully used a Grandstream phone with Asterisk? |
| 13:52.15 | Druken | probably lots... |
| 13:52.17 | Druken | hehe |
| 13:52.28 | Druken | i have a handytone working on mine... |
| 13:53.39 | TallAndy | Druken: the variable is set within the AGI, ${DIALSTATUS} is set when you execute Dial |
| 13:55.00 | Druken | TallAndy: ahh.. i see... well i've never had much luck with agi... :( |
| 13:55.20 | TallAndy | Ok ;) |
| 13:56.42 | occam23 | hmm, seems to be a problem with interconnection / direct isdn bridge, if i let asterisk asnwer the call first, so asterisk has to simulate the ring, it works every time |
| 13:56.45 | Druken | SpooForBrains: wut problems are you having? |
| 13:58.00 | SpooForBrains | Druken: can't seem to get the bastard thing to register |
| 13:58.44 | SpooForBrains | we have an asterisk server in the office which is SUPPOSED to be open to the net, and I'm at home behind an iptables firewall |
| 13:59.23 | Druken | nat ? |
| 13:59.29 | *** join/#asterisk heroine (n=heroine@m37.net81-66-58.noos.fr) |
| 13:59.56 | SpooForBrains | NAT on my end. Not supposedly on the office end |
| 14:00.16 | Druken | is the account your trying to register to set to nat=yes ? |
| 14:01.07 | SpooForBrains | on the asterisk server? I don't know |
| 14:01.48 | Druken | well, that's probably your problem right there... |
| 14:04.00 | Druken | SpooForBrains: if your on ipchains, why not send the voip ports to the phone? |
| 14:04.21 | Druken | then there's no need for nat.... unless you have more than one phone behind it |
| 14:06.49 | SpooForBrains | no, just the one |
| 14:06.51 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 14:07.35 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 14:07.47 | *** join/#asterisk fugitivo (n=ajf@201.255.176.38) |
| 14:07.53 | fugitivo | hello |
| 14:08.21 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:08.43 | fu3 | hiu |
| 14:08.58 | fugitivo | if i have a tdm400p with 2 fxo modules, and one module is accepting calls ok, but the other one is busy, is that module fu..ed? |
| 14:09.01 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-231.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 14:10.00 | fugitivo | the channel is onhook |
| 14:10.13 | fugitivo | sorry |
| 14:10.15 | fugitivo | offhook |
| 14:10.51 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 14:11.10 | DeeJay[2] | With zaptel's zttest, is it normal that I get sometimes a 99.975% ? |
| 14:11.12 | DeeJay[2] | Is it bad? |
| 14:11.22 | DeeJay[2] | On a test, I even had 99.84% once.. |
| 14:11.45 | [TK]D-Fender | DeeJay[2] : thats just fine |
| 14:12.15 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
| 14:13.40 | clive- | deejay i think thats normal for zydummy |
| 14:13.45 | clive- | ztdummy |
| 14:13.51 | DeeJay[2] | ztdummy? |
| 14:14.02 | clive- | are you using a zaptel card ? |
| 14:14.04 | DeeJay[2] | yes |
| 14:14.16 | DeeJay[2] | TE410P |
| 14:14.18 | clive- | if it works fine then ...its cool |
| 14:14.42 | DeeJay[2] | at this stage it's mostly fine tuning... |
| 14:14.44 | Druken | fugitivo: i would say it's either fu..ed or... the wires are crossed in the pinout |
| 14:15.17 | DeeJay[2] | Sometimes we hear "clicks" with SIP thru a Zap channel on the Zap side... |
| 14:16.18 | fugitivo | Druken: well, is that covered by the warranty? |
| 14:16.25 | Druken | no idea |
| 14:16.35 | Druken | where'd ya buy it? |
| 14:16.42 | fugitivo | local distributor |
| 14:16.53 | Druken | recently? |
| 14:16.55 | fugitivo | digium authorized |
| 14:17.05 | fugitivo | 5 or 6 months ago |
| 14:17.24 | Druken | it's worked fine till now? |
| 14:17.32 | fugitivo | yes |
| 14:17.33 | Druken | did you reboot recently?? |
| 14:18.05 | fugitivo | i did when it stopped working |
| 14:18.23 | Druken | i know with my cards, if i don't do a POWER CYCLE, not all my shit comes back online |
| 14:18.44 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 14:18.57 | fugitivo | the strange thing, is that the other module is working fine |
| 14:19.36 | [TK]D-Fender | fugitivo : Have you tried moving the module to another slot on the card? Tha works sometimes... |
| 14:20.08 | fugitivo | [TK]D-Fender: no, i'm not near the server, i'll try that |
| 14:20.13 | fugitivo | any other idea i sould try? |
| 14:20.16 | fugitivo | should |
| 14:20.35 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 14:21.08 | Druken | fugitivo: a good boot to the side of the cpu?? :) |
| 14:21.17 | DeeJay[2] | Best: 100.000000 -- Worst: 99.462891 -- Average: 99.981154 |
| 14:21.19 | DeeJay[2] | hmm... |
| 14:21.23 | DeeJay[2] | 99.46 once... |
| 14:21.24 | fugitivo | now, look at this |
| 14:21.26 | DeeJay[2] | that's bad |
| 14:21.33 | fugitivo | if I reboot the server, asterisk starts fine |
| 14:21.33 | DeeJay[2] | (134 passes) |
| 14:22.23 | fugitivo | if I stop asterisk, then unload zaptel, load zaptel again (it detects the modules ok), when i start asterisk it gives an error with channel 4 |
| 14:22.32 | [TK]D-Fender | DeeJay[2] : are you sue the card is on its own IRQ and that you're not running anything that may overload the CPU from processing it? |
| 14:22.47 | DeeJay[2] | [TK]D-Fender: pretty sure... |
| 14:22.52 | [TK]D-Fender | fugitivo : pastebin your zapata & zaptel files. |
| 14:23.02 | DeeJay[2] | cat /proc/interrupts shows everything has its own IRQ |
| 14:23.04 | DeeJay[2] | no sharing |
| 14:23.43 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 14:23.46 | DeeJay[2] | I added this to my kernel parameters: "pci=noacpi noapic nolapic" |
| 14:24.03 | DeeJay[2] | It's a P4-3Ghz with HT disabled |
| 14:24.13 | DeeJay[2] | it has only one PCI card and it's the TE410P |
| 14:24.38 | Druken | that alot of machine for an asterisk only system... |
| 14:24.41 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 14:25.02 | Hmmhesays | depends on how many calls and how much transcoding you are doing |
| 14:25.05 | [TK]D-Fender | DeeJay[2] : pastebin your cat /proc/interrupts... |
| 14:25.08 | *** part/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
| 14:25.24 | Druken | Hmmhesays: true |
| 14:25.38 | Katty | Hmmhesays: mew. |
| 14:25.45 | DeeJay[2] | http://pastebin.ca/43593 |
| 14:25.58 | Katty | Hmmhesays: i spent 96 minutes on the phone last night. |
| 14:26.01 | Hmmhesays | I'm using a p4 3.0ghz machine for a 96 seat sip pbx |
| 14:26.05 | Hmmhesays | Katty: oh yeah? |
| 14:26.09 | Katty | Hmmhesays: yeah |
| 14:26.12 | Katty | Hmmhesays: with a girly. |
| 14:26.17 | Hmmhesays | cool |
| 14:26.31 | Katty | yubyub |
| 14:26.33 | Hmmhesays | my girly was doing up her fiance last night, so I played unreal tournament |
| 14:26.38 | Hmmhesays | lol |
| 14:26.43 | [TK]D-Fender | DeeJay[2] : Dear God.. thats it? dump DMESG too then... |
| 14:26.53 | Katty | sounds fun ;) |
| 14:27.05 | Hmmhesays | yeah I like unreal tournament |
| 14:27.28 | Katty | me too...it's addictive. |
| 14:27.49 | Druken | i've never played it... |
| 14:28.41 | [TK]D-Fender | I miss the days of Action:Half-Life :( |
| 14:28.44 | Hmmhesays | Unreal tournament was probably one of the first of the higher rez 1st persion shooters |
| 14:28.52 | Hmmhesays | blood guts gore swearing |
| 14:29.03 | Hmmhesays | "headshot" |
| 14:29.42 | Hmmhesays | give me a sniper rifle and a perch and hmmhesays is a happy unreal'er |
| 14:29.48 | fugitivo | [TK]D-Fender: http://pastebin.ca/43595 |
| 14:29.59 | DeeJay[2] | http://pastebin.ca/43596 |
| 14:31.09 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:33.39 | [TK]D-Fender | DeeJay[2] : You are running an Intel E1000 NIC apparently which is a BAD thing which is documented as being incompatible with T4XX cards. Disable the on-board NIC direct from BIOS, and replace it with a PCI card. |
| 14:34.59 | DeeJay[2] | I'm using the 100 NIC |
| 14:35.00 | fugitivo | really? |
| 14:35.01 | DeeJay[2] | not the 1000 |
| 14:35.02 | [TK]D-Fender | fugitivo : And that config was working previously? |
| 14:35.08 | fugitivo | [TK]D-Fender: yes |
| 14:35.18 | [TK]D-Fender | DeeJay[2] : I see the module looks like its loading... give it a shot... |
| 14:35.37 | DeeJay[2] | [TK]D-Fender If i'm just not loading the kernel module |
| 14:35.38 | [TK]D-Fender | fugitivo : Move the modules to ports 1 & 2, change your config and see what happens... |
| 14:35.39 | DeeJay[2] | is it correct? |
| 14:36.23 | [TK]D-Fender | DeeJay[2] : Not 100% sure. I'm not a "real" Linux tech, I just know that it caused problems for me as well.... you're best to keep as far away from Intel NICs as possible |
| 14:37.04 | fugitivo | [TK]D-Fender: ok, i'll try that. what if that doesn't work? maybe the module is broken? |
| 14:39.19 | Druken | fugitivo: when you rebooted, did you do a power cycle or just a reboot ? |
| 14:39.48 | Druken | i'd try the power cycle first, before cracking the system open.. :) |
| 14:40.19 | fugitivo | Druken: shutdown -r now |
| 14:40.25 | [TK]D-Fender | fugitivo : First try SWAPPING the 2 modules. if the defect follows the swap I think that answers it.. |
| 14:40.52 | mut | anyone around that has used a tellab echo can? |
| 14:41.03 | fugitivo | [TK]D-Fender: ok, thanks |
| 14:41.35 | *** join/#asterisk heka (n=Horror@82.114.68.74) |
| 14:41.43 | heka | Hello, |
| 14:41.47 | fugitivo | Druken: are you sure about that? |
| 14:41.57 | heka | Im trying to call and call exits with the following message: Invalid timeout specified: 'L(60010000:60000:20000)' |
| 14:42.06 | heka | any idea whats wrong with my configuration? |
| 14:42.16 | Druken | fugitivo: -r is only a reboot |
| 14:42.21 | *** join/#asterisk devin (i=devin@203.141.139.231.user.ad.il24.net) |
| 14:42.23 | Druken | shutdown -h now |
| 14:42.32 | Druken | wait for everything to power OFF, then start her back up |
| 14:42.44 | fugitivo | Druken: i know, but i'm not near the machine, hold on, i'll try to call a person near the server |
| 14:43.12 | Druken | no foreign power control ? |
| 14:43.14 | *** join/#asterisk linville (n=linville@nat-pool-rdu.redhat.com) |
| 14:43.28 | fugitivo | no, it's a small crappy server |
| 14:43.42 | Druken | ahh, like mine :) |
| 14:43.49 | fugitivo | lol |
| 14:47.21 | *** join/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 14:47.59 | Katty | :> |
| 14:48.19 | iCEBrkr | yo yo yo |
| 14:48.23 | Hmmhesays | I really hate visio |
| 14:48.31 | fugitivo | use kivio! |
| 14:48.52 | Hmmhesays | what is that? |
| 14:48.59 | fugitivo | a visio for kde |
| 14:50.43 | Hmmhesays | this workstation is windows |
| 14:50.51 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:53.06 | *** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 14:53.55 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 14:53.59 | PakiPenguin | evening |
| 14:56.17 | sl16 | evening |
| 14:56.58 | heka | anybody can help me on setting timeout? |
| 14:57.14 | Seyr | heka: whats the question? |
| 14:57.57 | Hmmhesays | go to your room heka |
| 14:58.20 | fugitivo | anyone is using asterisk as a recording device for another pbx_ |
| 14:59.57 | heka | Seyr: I got Invalid timeout specified: 'L(60010000:60000:20000)' |
| 15:00.54 | Katty | Hmmhesays: be nice, boy. |
| 15:01.02 | Katty | Hmmhesays: before i send you to my room. |
| 15:01.08 | heka | Seyr: there must be some missconfiguration here! |
| 15:02.55 | iCEBrkr | I need a jobby job.. |
| 15:03.15 | heroine | i'm trying to use app_rxfax .. i've got an audiocode which forward calls to asterisk. my first test failed so i assume that i should use G.711 fax signaling method instead of T.38 .. |
| 15:04.35 | Curus | T.38 won't work yet |
| 15:04.38 | puzzled | heroine: you can rebuild the srpm at http://www.laimbock.com/asterisk/ It has t.38 passthrough. Rebuild it with the jitterbuffer patch disabled since it has some issues |
| 15:04.45 | Goral | Hmmhesays you around? |
| 15:05.01 | Curus | T.38 passthrough is fairly useless for rxfax, isn't it? |
| 15:05.29 | heroine | Curus: it's what i'm thinking but my boss thinks the contrary :) |
| 15:05.39 | puzzled | your endpoint needs to support t.38/t.30 iirc |
| 15:05.51 | puzzled | otherwise t.38 passthrough is useless |
| 15:05.55 | *** part/#asterisk Seyr (n=Seyr_@cpe-67-10-139-141.houston.res.rr.com) |
| 15:06.01 | Curus | rxfax is an end point |
| 15:06.06 | Curus | And it doesn't support t.38 |
| 15:06.13 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:06.13 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:06.20 | heroine | ok, thanx Curus |
| 15:07.13 | Katty | yay, anthm! |
| 15:07.17 | Hmmhesays | Yeah somewhat around |
| 15:07.23 | anthm | hi |
| 15:07.27 | Katty | hihi |
| 15:07.42 | anthm | the answer to your long awaited question is Aug 2-4 |
| 15:07.43 | fugitivo | i'm wondering, if there's any way to plug asterisk to a pbx, and tell the pbx to send audio stream to asterisk to record it, anyone with experience on this? |
| 15:07.48 | Katty | anthm: yay! |
| 15:08.13 | Hmmhesays | there are many ways to do that |
| 15:08.42 | fugitivo | Hmmhesays: for example? |
| 15:09.10 | Hmmhesays | depends on your pbx setup |
| 15:10.03 | fugitivo | Hmmhesays: for example, is it possible to use fxo cards or E1 connected to the regular pbx, and tell the pbx to send audio stream through that channels? |
| 15:10.12 | iCEBrkr | voip-info.org is slow today |
| 15:10.18 | fugitivo | Hmmhesays: nortel or avaya pbx |
| 15:10.59 | Katty | nortel sounds like a name for a turtle |
| 15:11.05 | Hmmhesays | northern telecom |
| 15:11.20 | Katty | file: if you ever get a pet turtle, you must name him nortel. |
| 15:11.21 | fugitivo | Hmmhesays: i don't want to put asterisk before the other pbx |
| 15:11.28 | *** join/#asterisk agkhram (n=arheil@yasski.sam-isp.net) |
| 15:11.40 | iCEBrkr | Katty: I named my cat Skr1pt |
| 15:11.40 | fugitivo | Hmmhesays: i know that way is easier, but asterisk should handle all calls and i don't want that |
| 15:11.49 | fugitivo | iCEBrkr: kiddie? |
| 15:11.54 | iCEBrkr | fugitivo: right on |
| 15:11.56 | fugitivo | lol |
| 15:12.05 | *** join/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl) |
| 15:12.12 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:12.15 | Katty | anthm: have anything that's printable? |
| 15:12.19 | Katty | anthm: accounting loves paperwork |
| 15:12.37 | anthm | ask bkw |
| 15:12.38 | anthm | brb |
| 15:12.51 | Katty | bkw__: i need paperwork, deary! |
| 15:13.19 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:13.40 | fugitivo | Hmmhesays: any idea? |
| 15:13.45 | ScaredyCat | weak... |
| 15:13.49 | *** part/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl) |
| 15:14.10 | Katty | fugitivo: you don't want to know what's going on in Hmmhesays's head. |
| 15:14.18 | Katty | fugitivo: it's a skeery place up there. |
| 15:14.48 | fugitivo | :) |
| 15:15.12 | Meaty | [10:12] <iCEBrkr> voip-info.org is slow today || Use Google Cache ! |
| 15:17.08 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 15:19.01 | PakiPenguin | ~jbot help monitor |
| 15:19.30 | Druken | fugitivo: did ya get ahold of someone? |
| 15:20.14 | fugitivo | not yet |
| 15:20.17 | kpettit | Anybody using a Sangoma A102 pri card? |
| 15:20.45 | Hmmhesays | fugitivo: its not really a question you can answer specifically, just think of any way you would normally get a call from your pbx to asterisk bud |
| 15:21.28 | kpettit | I'm tring to figure out how to use the ztdummy driver with wanpipe |
| 15:21.30 | fugitivo | Hmmhesays: yeah, problem is i don't know much about regular pbx :/ |
| 15:21.34 | [TK]D-Fender | kpettit : Whats your question on it? |
| 15:21.54 | fugitivo | Hmmhesays: i don't know what the pbx is capable of |
| 15:21.55 | kpettit | using "zaptel" gives me too low of a score on zttest. So it was suggested that I use the ztdummy driver |
| 15:22.01 | Hmmhesays | fugitivo: then we can't answer your quesiton |
| 15:22.03 | Hmmhesays | *question |
| 15:22.15 | kpettit | but I'm not sure how to do that. When i load the ztdummyh module i get a span error from my zaptel.conf |
| 15:22.35 | kpettit | Any idea how I can define a span in zaptel.conf and still load the ztdummy driver? |
| 15:23.08 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:23.44 | Hmmhesays | haha dual proc 3.2ghz p4 for a 30 simultaneous sip call machine.. OVERKILL |
| 15:24.35 | agkhram | really? |
| 15:25.00 | Hmmhesays | yeah i'd say so |
| 15:25.44 | Hmmhesays | no tdm hardware |
| 15:25.47 | agkhram | it's a pity. |
| 15:25.58 | Hmmhesays | what is? |
| 15:26.02 | agkhram | i have no h/w too |
| 15:26.24 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 15:26.34 | Hmmhesays | i don't use any pci tdm cards on asterisk boxes |
| 15:27.05 | agkhram | 15 extern lines via h323... ~100 internal sip... |
| 15:27.36 | agkhram | BTW have anybody h323 stats? |
| 15:28.16 | Druken | h323 == evil |
| 15:29.23 | agkhram | i have no choices |
| 15:29.33 | x86 | ok guys, i need a cheap, black IP phone that supports SIP, G.711u, G.711a, call forwarding, DND, call transfer, and caller ID |
| 15:29.37 | x86 | any suggestions? |
| 15:30.01 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 15:30.15 | Darwin35 | need input on cdr issue |
| 15:30.57 | [TK]D-Fender | kpettit : Are you clocking to the telco? |
| 15:31.06 | [TK]D-Fender | kpettit : Any frame slips, etc? |
| 15:31.15 | kpettit | no |
| 15:31.31 | kpettit | But I'm still somewhat a newbie with the PRI stuff |
| 15:31.41 | Druken | ya know what? snow sucks.... |
| 15:31.48 | Darwin35 | my cdr are only showing the base of a call the call coming in and and going bak out if forwarded but it does not reflect in the cdr the diff parts of the call |
| 15:31.53 | Druken | i'm getting entirly tired of it..... |
| 15:31.56 | kpettit | [TK]D-Fender, we have a Cisco from the Telco, and i'm just going PRI from my box to the Cisco. I'm not sure how to mess with the clocking |
| 15:32.07 | [TK]D-Fender | kpettit : pastebin your zaptel, zapata, "ifconfig |
| 15:32.29 | [TK]D-Fender | ", and "pri show span [port your PRI is on]" |
| 15:32.48 | kpettit | [TK]D-Fender, Voice works fine btw. Put on my Digium PRI card I was getting enough HDLC errors to bugger up fax, and my zttest scores were bad |
| 15:33.00 | [TK]D-Fender | kpettit : You should have the Cisco taking clock from telco, then sending clock to * |
| 15:33.23 | kpettit | [TK]D-Fender, so I bought a Sangoma card to test out sense I heard they might be able to get riid of some of those issues. I was screwing with the clocking last night, but wasn't sure how to work it into the span definition |
| 15:33.44 | kpettit | [TK]D-Fender, so I need to get clock form the Cisco |
| 15:34.20 | [TK]D-Fender | kpettit : You *should* be.... |
| 15:34.37 | *** join/#asterisk fulgas (n=fulgas@207.226.175.10) |
| 15:34.52 | kpettit | [TK]D-Fender, any docs on how I do that with wanpipe? |
| 15:35.07 | kpettit | don't see anything obvious in wancfg to pull from another device |
| 15:36.17 | bkw__ | Katty, dear |
| 15:36.18 | bkw__ | soon |
| 15:36.27 | bkw__ | waiting on MikeJ to finish up the hotel bidness |
| 15:36.55 | kpettit | [TK]D-Fender, I have a option for "TDM PRI HW-HDLC Timeslot" which I can put in a single number |
| 15:37.43 | *** join/#asterisk heison (n=heison@gw-yyz.somanetworks.com) |
| 15:38.52 | [TK]D-Fender | kpettit : the timeslot is the D-Channel. Not the problem you're looking for... |
| 15:39.05 | kpettit | ah ok |
| 15:39.07 | [TK]D-Fender | pastebin the bits I asked for and also your wanpipe config. |
| 15:39.12 | kpettit | ok 1 sec... |
| 15:40.04 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 15:41.10 | x86 | anyone with suggestions on cheap IP phones? |
| 15:41.52 | Druken | ebay ? |
| 15:41.57 | [TK]D-Fender | x86 : not if you want the work GOOD attached to it either. |
| 15:42.53 | kpettit | [TK]D-Fender, http://pastebin.com/576779 |
| 15:42.53 | kpettit | thanks for taking a look. |
| 15:42.53 | Druken | he's looking for a wal-mart voip phone :) |
| 15:43.08 | x86 | [TK]D-Fender: no, the "work" GOOD doesnt have to be attached :) |
| 15:43.17 | heison | does anyone here have experience with asterfax? I'm running 1.0.7 and facing two problems: it would take me a few tries to get the software successfully launched (java keeps dying) which was not the case with 1.0.6; once asterfax is running, i can send fax via email, but the output on the fax machine seem to have the pagesize set incorrectly (the images looks huge). I have tried playing with gs -sPAPERSIZE=a4|letter|legal and I get the same resul |
| 15:44.55 | [TK]D-Fender | x86 : Go buy a BT-101 then :p |
| 15:45.09 | RoyK | wiki down again :( |
| 15:47.09 | Hmmhesays | the wiki is never down with google cache |
| 15:47.21 | [TK]D-Fender | kpettit : You have D-Chan defined in your wanpipe config.... |
| 15:47.27 | x86 | [TK]D-Fender: does it have speakerphone? |
| 15:47.37 | kpettit | [TK]D-Fender, ?? |
| 15:47.38 | RoyK | Hmmhesays: that's a workaround. the wiki is down all the fscking time |
| 15:47.45 | [TK]D-Fender | x86 : Fine... get a GXP-2000 instead..... |
| 15:48.00 | Skid | hmm |
| 15:48.03 | [TK]D-Fender | kpettit : You seem to be using PRI but with no functioning D-Channel... |
| 15:48.04 | Skid | I'd mirror it |
| 15:48.04 | Darwin35 | how do you make a cdr record show all stages of a call including the forward |
| 15:48.14 | Skid | if it's possible :-) |
| 15:48.18 | Darwin35 | it is hairpining |
| 15:48.27 | x86 | [TK]D-Fender: hah... that's like 3 times the price of a BT-101 ;) |
| 15:48.29 | Darwin35 | or drop cut |
| 15:48.34 | [TK]D-Fender | kpettit : line 40 in you post should say "TDMV_DCHAN = 24" according to "normal" rules. |
| 15:48.43 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 15:48.44 | [TK]D-Fender | x86 : Hardly... more like 50% more. |
| 15:48.53 | [TK]D-Fender | x86 : but you get a lot for that 50% |
| 15:49.06 | kpettit | [TK]D-Fender, That's not a feature that's enabled in 1.2.4 of Zaptel |
| 15:49.08 | Hmmhesays | omg, why would you put an untested box into prodution |
| 15:49.15 | Mimmus | hi, does anyone know if I can set CallerIDNum on a PRI line with DID in Italy? |
| 15:49.31 | kpettit | I tried using TDMV_DCHAN=24 but that option dosen't exist. I get a error if I try to use it. |
| 15:49.40 | kpettit | I was told that's in CVS not in the 1.2.4 release |
| 15:49.57 | [TK]D-Fender | kpettit : I'm running ZAptel 1.2.4 here... change the value, and reload wanrouter |
| 15:50.18 | [TK]D-Fender | kpettit : I'm running release here... |
| 15:50.30 | kpettit | ok cool |
| 15:50.30 | [TK]D-Fender | kpettit : and just finished doing a full upgrade |
| 15:50.45 | [TK]D-Fender | Wanpipe, *, zap, AFT firmaware, the works.... |
| 15:50.54 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-70.dialup.tiscali.it) |
| 15:50.59 | Gennaro | hi |
| 15:51.01 | kpettit | I've been doing that alot this week |
| 15:51.05 | Skid | I wonder what sort of transfer that wiki eats though |
| 15:51.16 | kpettit | normally I do sip connect for voice, these PRI's are more painfull |
| 15:51.20 | Gennaro | some one know where is possible download front end of asterisk |
| 15:51.20 | Gennaro | ? |
| 15:51.35 | x86 | [TK]D-Fender: $84 for a GXP-2000 a good deal? |
| 15:51.46 | Gennaro | to create apparate, internal, route, and config ? |
| 15:51.47 | Mimmus | Gennaro: hi (I'm italian) |
| 15:52.01 | Mimmus | Gennaro: front-end? a GUI? |
| 15:52.06 | Gennaro | si |
| 15:52.13 | *** join/#asterisk obsidian-studios (n=obsidian@c-24-129-16-31.hsd1.fl.comcast.net) |
| 15:52.14 | Gennaro | na'ggeggio |
| 15:52.15 | Mimmus | Gennaro: AMP |
| 15:52.27 | Skid | nop |
| 15:52.28 | Skid | oops |
| 15:52.30 | Gennaro | i'm studing.. |
| 15:52.43 | Gennaro | and i'dlike to use a gui.. |
| 15:52.50 | [TK]D-Fender | x86 : The best price out there.... but remember as usual, you get what you pay for... |
| 15:52.52 | Mimmus | Gennaro: AMP e' una interfaccia web |
| 15:52.58 | kpettit | [TK]D-Fender, Notice: Configuration file is /etc/zaptel.conf |
| 15:52.58 | kpettit | line 5: Unknown keyword 'TDMV_DCHAN' |
| 15:53.10 | Gennaro | non gli piace che si parli in italiano dove lo trovo? |
| 15:53.19 | Mimmus | Gennaro: funziona ma sei un po' vincolato |
| 15:53.25 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:53.25 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:53.26 | obsidian-studios | greetings, recently updated a few * boxes to 1.2.4, so far so good, but for some reason on one box, no main_menu? the .mp3 file is where it always was, anyone got any suggestions or ideas where to start, here is the output http://pastebin.ca/43608 |
| 15:54.07 | [TK]D-Fender | kpettit : Your MTU is also wrong.. its showing as 192 instead of 8.... |
| 15:54.36 | Mimmus | Gennaro: AMP works well but it is slightly restraining |
| 15:55.00 | kpettit | [TK]D-Fender, do you knwo why I don't have that option? I'm using 1.2.4 of asterisk, and zaptel |
| 15:55.06 | Mimmus | Gennaro: you can find it here: http://amp.coalescentsystems.ca/ |
| 15:55.19 | Goral | hello all, i get no audio from incomming calls they can hear me but i cannot hear them any sugestions |
| 15:55.20 | Gennaro | ok tanx |
| 15:55.32 | heroine | grep 14 /var/log/* n'etait pas une bonne idée |
| 15:55.33 | *** join/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 15:55.37 | [TK]D-Fender | kpettit : pastebin the CLI showing that error |
| 15:55.41 | oej | ~seen lusyn |
| 15:55.45 | jbot | oej: i haven't seen 'lusyn' |
| 15:56.33 | Mimmus | Gennaro: you can also download AsteriskAtHome, a pre-packaged installation of Asterisk featuring also AMP |
| 15:57.43 | *** join/#asterisk austinnichols101 (n=austinni@70.46.69.131) |
| 15:58.11 | Gennaro | http://sourceforge.net/search/?words=asterisk+gui&type_of_search=soft |
| 15:58.15 | Gennaro | try here |
| 15:58.31 | Gennaro | i want to study how.. |
| 15:58.34 | Gennaro | tanx.. |
| 15:58.41 | *** join/#asterisk Fedoracore6 (n=FC$@218.111.176.138) |
| 15:58.43 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfi30.dialup.mindspring.com) |
| 15:58.43 | Gennaro | are u using it? |
| 15:58.46 | x86 | [TK]D-Fender: what's the diff between the BT-101 and BT-102? |
| 15:58.48 | Mimmus | can I set CallerIDNum on a PRI line with DID? does telco allow it? |
| 15:58.55 | Fedoracore6 | hai all |
| 15:59.02 | Fedoracore6 | X-Rob are you there |
| 15:59.12 | [TK]D-Fender | x86 : 2nd ethernet port. |
| 15:59.14 | Fedoracore6 | where i can paste my code |
| 15:59.19 | Fedoracore6 | asterisk |
| 15:59.23 | iCEBrkr | ~pb |
| 15:59.24 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:59.31 | kpettit | [TK]D-Fender, http://pastebin.com/576803 |
| 15:59.51 | [TK]D-Fender | x86, but let me tell you that they are FUGLY, and cheap shit phones... you really should go no lower than a GXP-2000, and IMO not even that... |
| 16:00.44 | [TK]D-Fender | kpettit : that error doesn't match the zaptel.conf you first pastebin'd for me... |
| 16:01.04 | [TK]D-Fender | kpettit : Show me what you did to it... |
| 16:02.05 | Hmmhesays | ahh the eternal question, sata or scsi |
| 16:02.06 | Fedoracore6 | http://pastebin.com/576808 |
| 16:02.17 | Gennaro | minnus |
| 16:02.19 | kpettit | TDMV_DCHAN=24 |
| 16:02.28 | Gennaro | costa na mazzata... |
| 16:02.32 | Fedoracore6 | some budy plase help me check what my false |
| 16:02.39 | *** join/#asterisk Trevor_b (n=FreePBX3@office1.ct.vpn.cleartunnel.net) |
| 16:02.42 | Fedoracore6 | hemm X-Rob u have there |
| 16:02.55 | Fedoracore6 | any alse .... wanna help me |
| 16:03.00 | [TK]D-Fender | kpettit : that line does NOT belong in zaptel.conf.. thats for wanpipe1.conf |
| 16:03.25 | Fedoracore6 | http://pastebin.com/576808 |
| 16:03.31 | Fedoracore6 | what my false |
| 16:03.54 | Fedoracore6 | i already setting at cdr_mysql.conf |
| 16:03.57 | Mimmus | Gennaro: sorry, I was away... |
| 16:04.01 | Mimmus | Gennaro: AMP is free |
| 16:04.05 | Katty | Hmmhesays: i'm looking for a song you might have. |
| 16:04.07 | Hmmhesays | amp is a bitch to install |
| 16:04.15 | Abydos313 | no its not |
| 16:04.15 | Hmmhesays | Katty: what might that be? |
| 16:04.16 | Katty | Hmmhesays: it goes something like sara beth's scared to death..something something |
| 16:04.18 | Fedoracore6 | but didint work |
| 16:04.21 | Katty | Hmmhesays: i think th ename of the song is skin |
| 16:04.28 | Trevor_b | if you dont like to do the work Asterisk@home gives you AMP |
| 16:04.28 | Katty | Hmmhesays: by rascal flatts |
| 16:04.47 | x86 | [TK]D-Fender: why are they shit? sound quality bad? |
| 16:04.52 | Hmmhesays | yeah I think i have that |
| 16:04.55 | Hmmhesays | hold on a sec |
| 16:04.59 | Abydos313 | asterisk@home does the whole thing for you, mysql/amp/asterisk |
| 16:05.19 | [TK]D-Fender | Katty : Thats from "Rascal Flatts" - Skin |
| 16:05.31 | x86 | [TK]D-Fender: i cant really afford a GXP-2000 right now... i could if it wasnt $15 for shipping... but i really need the phone under $75, including shipping |
| 16:05.53 | Hmmhesays | Abydos313: read what I said, amp is a b1@tch to install |
| 16:05.58 | [TK]D-Fender | x86, flakey firemwares, poor use of display, cheap feel, rfeaking bad echo (comes and goes with "firmware"), and other such non-sense. |
| 16:05.58 | Fedoracore6 | some budy plase help me check my false |
| 16:06.00 | Hmmhesays | you don't install amp when you install a@h |
| 16:06.14 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfi30.dialup.mindspring.com) |
| 16:06.37 | x86 | [TK]D-Fender: well it's what i can afford right now so I guess it will have to work :) |
| 16:06.37 | *** join/#asterisk Weezey (n=weezey@206.210.109.233) |
| 16:06.43 | Weezey | voip-info down? |
| 16:06.55 | Trevor_b | Roby's, but they dont give multiple lines unless you just use flash/call waiting style. |
| 16:06.58 | Trevor_b | they are cheap |
| 16:07.00 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 16:07.01 | Mimmus | Trevor_b: yes |
| 16:07.10 | Weezey | Hmmhesays: takin' too long |
| 16:07.22 | [TK]D-Fender | x86 : Careful.. you don't want to find you hate what you've got when it arrives.... nothing worse than wasted money when better doesn't cost that much more.... |
| 16:07.22 | Katty | [TK]D-Fender: indeed, just as i said above. |
| 16:07.48 | [TK]D-Fender | Katty: So I'm a little "slow" this morning! |
| 16:08.02 | Gennaro | if you have a prob... and u will have a prob. you must pay a loTT |
| 16:08.12 | heroine | has somebody the documentation of audiocodes mediant 2000 gateway ? |
| 16:08.30 | Darwin35 | walmart makes voip phones |
| 16:08.32 | x86 | [TK]D-Fender: it's only a temporary fix... |
| 16:08.37 | Katty | [TK]D-Fender: ;) |
| 16:08.42 | Katty | Hmmhesays: you gmailing it? |
| 16:08.48 | Abydos313 | Darwin35 how much are they selling them for? |
| 16:09.01 | x86 | [TK]D-Fender: I'm going to buy myself a cisco 7970G and offload this BT-101 to my wife ;) |
| 16:09.16 | x86 | s/wife/accounting dept/ hehe |
| 16:09.32 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-133-198-176.prod-infinitum.com.mx) |
| 16:09.33 | Darwin35 | I was asking |
| 16:09.51 | Darwin35 | bt101 have 1 goog uds |
| 16:09.54 | Darwin35 | use |
| 16:10.04 | x86 | whats that? |
| 16:10.04 | Goral | i need some sugestions on why i cannot hear incomming parties but they can hear me |
| 16:10.05 | Darwin35 | connect it to pa system |
| 16:10.16 | [TK]D-Fender | x86: How sad... the Cisco doesn't even support SIP, and you're spending a fortune for it I'm sure, and looking to waste money on a real POS for your SO? I don't think I want to know what drugs you are on.... |
| 16:10.31 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
| 16:10.31 | [TK]D-Fender | Goral : My guess.... NAT issues... |
| 16:10.35 | wiseguy_ | hellow |
| 16:10.40 | x86 | [TK]D-Fender: eht 7970 doesnt support SIP? |
| 16:10.49 | [TK]D-Fender | Darwin35 : There are other ways I'd sooner use... |
| 16:10.53 | [TK]D-Fender | x86 : Nope.... |
| 16:10.55 | wiseguy_ | anyone can tell me how to set callerid to the first 9 numbers of callerid? |
| 16:11.11 | Darwin35 | the best cheap 69 dollar phone I have found is chineese made Yughen |
| 16:11.24 | ibob63 | how do I swtich off the sip debugger? |
| 16:11.33 | wiseguy_ | ibob63: sip no debug |
| 16:11.35 | Darwin35 | it can be flashed for sip/iax/mgcp |
| 16:11.55 | Mimmus | wiseguy_: SetCIDNum(${CALLERIDNUM:-9:9}) |
| 16:11.56 | ibob63 | wiseguy_ thanks that was driving me crazy :) |
| 16:11.56 | brad_mssw | Darwin35: if teliax is still looking for a place to put a east coast server, my ISP would like to talk to you |
| 16:11.58 | [TK]D-Fender | wiseguy_ : It can be set with "exten 1234,1,Set(CALLERID(number)=${CALLERID(number):0:9}) |
| 16:12.12 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 16:12.13 | [TK]D-Fender | Mimmus : close, but the first is offset, THEN length. |
| 16:12.25 | Mimmus | [TK]D-Fender: OK |
| 16:12.25 | x86 | [TK]D-Fender: oh weak... i just wanted it because it had a very nice, high-color, graphical display and looks sexy... |
| 16:12.31 | [TK]D-Fender | Mimmus : he wanted the FIRST 9, not the LAST nine |
| 16:12.36 | x86 | [TK]D-Fender: but Asterisk runs SCCP fairly well too, no? |
| 16:12.36 | Goral | well i gave up on calling out now i'm only going to call within the local network |
| 16:12.42 | wiseguy_ | [TK]D-Fender: thanks |
| 16:12.51 | [TK]D-Fender | goral : I just gave you a hint for it... |
| 16:13.12 | [TK]D-Fender | Goral : Describe where the phone and * are relative to each other and your network scenario.... |
| 16:13.32 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:13.33 | Mimmus | [TK]D-Fender: well. And what's the difference between Set(CALLERID(number)... and SetCIDNum()? |
| 16:14.07 | Mimmus | Darwin35: where can I find a picture of this phone? |
| 16:14.20 | [TK]D-Fender | Mimmus : SetCIDnum is deprecated in 1.2.x |
| 16:14.53 | Mimmus | [TK]D-Fender: well... so AAH/AMP are not 1.2 compliant... |
| 16:14.54 | [TK]D-Fender | x86 : There is no point to a phone like that with * right now. Better off redistributing your purchase... |
| 16:15.11 | [TK]D-Fender | Mimmus : Depends which.... I don't use AMP..... |
| 16:15.20 | kpettit | [TK]D-Fender, so do I need dchan in zaptel.conf and in wanpipe1.conf? |
| 16:15.45 | Goral | [TK]D-Fender i call from local computer using softphone through asterisk routing to voipbuster to my landline |
| 16:15.47 | kpettit | [TK]D-Fender, I mean TDMV_DCHAN in wanpipe1.conf |
| 16:16.00 | [TK]D-Fender | kpettit : dchan=24 in zaptel, TDMV_DCHAN = 0 in wanpipe1.conf |
| 16:16.01 | Mimmus | [TK]D-Fender: I use it and it is filled of SetCIDnum and other not-1.2 compliant stuffs |
| 16:16.19 | kpettit | [TK]D-Fender, thanks I'll give that a try. |
| 16:16.33 | [TK]D-Fender | Mimmus : AMP = Shit-on-a-stick for which I'd sooner not hear anything else about.... |
| 16:16.56 | Mimmus | [TK]D-Fender: Ok, I will avoid any discussion about it! |
| 16:17.32 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 16:17.58 | Goral | was that the info you needed [TK]D-Fender? |
| 16:18.33 | Katty | [TK]D-Fender: do you have that song? |
| 16:18.38 | Mimmus | [TK]D-Fender: but first I need to convert AMP dialplan by hand. It's not so simple write a moderately complicate dialplan from scratch |
| 16:18.47 | Fedoracore6 | http://pastebin.com/576808 |
| 16:18.54 | Hmmhesays | what was the name of hte song again? |
| 16:18.59 | x86 | [TK]D-Fender: i think for my needs the BT-101 will work for a while anyway :) |
| 16:19.01 | Trevor_b | [TK]D-Fender: What do you use instead of AMP, or do you use conf files directly only? |
| 16:19.09 | Fedoracore6 | i run this script but have some error |
| 16:19.39 | Hmmhesays | using config files is not hard if you know how to plan |
| 16:20.00 | Hmmhesays | amp has its place |
| 16:20.04 | Mimmus | it is slightly harder coding a dialplan from scratch |
| 16:20.08 | Katty | Hmmhesays: skin, by rascal flatts. |
| 16:20.19 | Hmmhesays | hmm I can probably grab that |
| 16:20.23 | Katty | thanks. |
| 16:20.26 | Hmmhesays | Mimmus: depends on what you're doing |
| 16:20.36 | Mimmus | Hmmhesays: I think that AMP has its place too |
| 16:21.05 | Hmmhesays | I use a@h for small installs, its great |
| 16:21.32 | Mimmus | Hmmhesays: if you want to replace all AMP features, it is hard |
| 16:22.32 | [TK]D-Fender | Goral : is your * behind a NAT? |
| 16:22.49 | [TK]D-Fender | Katty: not yet, but I might be able to find it.... |
| 16:23.05 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 16:23.07 | [TK]D-Fender | x86: I seriously wouldn't waste your money on that POS.... |
| 16:23.13 | Fedoracore6 | lor no budy wanna help me lor :( |
| 16:23.45 | [TK]D-Fender | x86: And I mean the Cisco too ;) Just get her a Polycom IP 301 and you an IP 601. Collectively would cost less than the Cisco you were looking at by far... |
| 16:23.53 | iCEBrkr | [TK]D-Fender: hey a BT101 isn't all that bad for home use. |
| 16:23.59 | *** part/#asterisk obsidian-studios (n=obsidian@c-24-129-16-31.hsd1.fl.comcast.net) |
| 16:24.06 | iCEBrkr | [TK]D-Fender: Who wants to spend $200-$300 on an IP phone for home? |
| 16:24.07 | [TK]D-Fender | iCEBrkr : Its not even a worthy PAPERWEIGHT. |
| 16:24.16 | iCEBrkr | [TK]D-Fender: Oh please, my BT100 works just fine. |
| 16:24.23 | [TK]D-Fender | iCEBrkr: He's looking to buy a 7970G for himself! |
| 16:25.14 | [TK]D-Fender | Trevor_b : I do straight *. GUI's have a very limited audience as far as * goes... the point of * is control which is what every GUI takes away from you. |
| 16:25.20 | Katty | [TK]D-Fender: i think Hmmhesays is going to get it |
| 16:25.30 | Mimmus | in this channel, all uses only >300$ phones! |
| 16:25.31 | Goral | [TK]D-Fender : yes |
| 16:25.53 | Goral | cable modem to linksys router to network |
| 16:26.08 | iCEBrkr | I got better things to spend $300 on.. Like a new set of Pilot Pro street tires for my motorcycle |
| 16:26.11 | [TK]D-Fender | Goral : You'll neewd to set one of "EXTERNIP" or "EXTERNHOST", as well as "LOCALNET" in the [general] section of SIP.CONF. |
| 16:26.32 | [TK]D-Fender | Goral : And forward UDP 5060, 10000-20000 to * |
| 16:26.40 | x86 | [TK]D-Fender: heh |
| 16:26.52 | x86 | which is better... G.723 or GSM? |
| 16:27.03 | *** join/#asterisk pagefault (n=pagefaul@pdpc/supporter/active/pagefault) |
| 16:27.08 | RoyK | which is better? blue or green? |
| 16:27.12 | x86 | voice-quality-wise |
| 16:27.16 | x86 | they are about the same? |
| 16:28.25 | [TK]D-Fender | x86 : Depends I guess.... I like GSM better for most general purose low bandwidth. most ofther codes mangle music and other things to wierdly. |
| 16:28.46 | [TK]D-Fender | RoyK : Green obviously! Don't you know anything?!?!?! |
| 16:29.05 | RoyK | x86: does it matter? you can't use g.723.1 anyway :) |
| 16:29.07 | [TK]D-Fender | x86 : its subjective... thy and see... |
| 16:29.37 | fu3 | wow |
| 16:29.40 | fu3 | that takes me back :) |
| 16:29.40 | fu3 | thanks |
| 16:30.04 | Gennaro | mimmus are u?!? |
| 16:30.45 | Mimmus | Gennaro: yes |
| 16:30.56 | Gennaro | i downloaded amp |
| 16:31.04 | Gennaro | and installation is a mass! |
| 16:31.08 | [TK]D-Fender | Mimmus : I never found it hard to do a dial-plan from scratch, and all you really need is to keep a good STDEXTEN macro handy and the rest builds itself.... |
| 16:31.19 | Goral | [TK]D-Fender so under general i would put somthing like externip=xxx.xxx.xxx.xxx |
| 16:31.19 | Gennaro | i need to download a lot of software.. |
| 16:31.28 | Gennaro | arghh... |
| 16:31.36 | Gennaro | i'm surfing @56 |
| 16:31.37 | [TK]D-Fender | Goral : correct, and you should also have "nat=yes" in there as well. |
| 16:31.50 | *** join/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl) |
| 16:32.04 | Hmmhesays | asterisk behind nat doesn't like to play nice |
| 16:32.09 | Gennaro | is it a good way?!? |
| 16:32.11 | Mimmus | [TK]D-Fender: thanks for your suggestion, I think that I will be forced to do this because I cannot keep 60 users with AMP |
| 16:32.27 | Mimmus | Gennaro: I cannot help |
| 16:32.42 | Hmmhesays | why can't you keep 60 users with amp? |
| 16:32.53 | Gennaro | oh sorry.. good.. tanx |
| 16:32.56 | [TK]D-Fender | Hmmhesays : Always has for me... and I've gone double-NAT before as well.... |
| 16:32.59 | jbalcomb | Mimmus whats the trouble with AMP? I was thinking about implementing it but i have 100+ people here. |
| 16:33.16 | Hmmhesays | i had a helluva time with a 1:1 nat awhile back, never did solve it |
| 16:34.00 | Trevor_b | Hmmhesays: did you have iptables rules filtering with NEW, ESTABLISHED? |
| 16:34.08 | Hmmhesays | no firewalls |
| 16:34.21 | Trevor_b | what was doing your 1:1 NAT? |
| 16:34.23 | Hmmhesays | you can hanlde 100+ users with amp |
| 16:34.47 | Mimmus | Hmmhesays: here all gurus say that is shit-on-a-stick! |
| 16:35.02 | Hmmhesays | i think that's just [TK]D-Fender |
| 16:35.30 | Trevor_b | Mimmus: bah, its preference. I have the FreePBX (AMP 2 beta) running, not too shabby, its the online support module i am using to get into IRC here.... |
| 16:35.34 | jbalcomb | "we don't talk like that here" |
| 16:35.35 | Hmmhesays | the whole thing with amp is, you need to know what you're doing in the first place, otherwise you might as well rip your hair out and piss everyone in here off |
| 16:36.22 | heison | anyone here using spandsp, in particular txfax? |
| 16:36.27 | jbalcomb | i was thinking about trying out that asterPBX from fonality too. |
| 16:36.34 | austinnichols101 | amp shouldn't have any significant impact on the number of users. It's a limiting factor only in that it limits what you can do... |
| 16:36.51 | Fedoracore6 | http://pastebin.com/576808 |
| 16:37.00 | jbalcomb | heison I just removed it because I couldn't get it working right. I'm a Nub though so it doesn't mean much. |
| 16:37.02 | Hmmhesays | but it doesn't limit what you can do |
| 16:37.04 | Fedoracore6 | have any budy help me |
| 16:37.07 | *** join/#asterisk areski (n=areski@62.235.228.133) |
| 16:37.23 | Hmmhesays | because you can still use a dialplan written from scratch |
| 16:37.34 | Trevor_b | AMP limits what you can do in the interface, but there are *_custom.conf files you can add just about anything you would put into the main files, as long as their scripts dont conflict... |
| 16:37.34 | Mimmus | bah, I'm pretty satisfied with AMP but sometime I find some limitation |
| 16:37.38 | [TK]D-Fender | jbalcomb : besides your needs are not compatible with AMP's methodologies.... Quite custom. |
| 16:37.41 | digime | looking for * tech in san diego.. anyone? |
| 16:37.44 | heison | ~seen coppice |
| 16:37.51 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 7h 39m 38s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 16:37.51 | Hmmhesays | my installs with a@h using are all hyrbids |
| 16:37.51 | jbalcomb | Hmmhesays i thought AMP was in the habit of mangling custom configs |
| 16:38.07 | Hmmhesays | jbalcomb: yeah if you don't read the documentation |
| 16:38.14 | Hmmhesays | and you don't know how amp works in the first place |
| 16:38.18 | Trevor_b | heheh |
| 16:38.21 | jbalcomb | [TK]D-Fender yeah, i think thats the case too. i have a side job i might try it on though |
| 16:38.26 | Trevor_b | AMP mangles your DEFAULT configs |
| 16:38.26 | Mimmus | Hmmhesays: I'm using *_custom.conf files quite extensively |
| 16:38.35 | austinnichols101 | hmmhesays: I would agree that you can implement your own scripts, etc. but there are specific areas where you can't extend in a way that would allow you to upgrade amp and not break your own stuff. |
| 16:38.43 | Trevor_b | *_custom are for NON mangled configs, they stay every time AMP reloads. |
| 16:38.43 | Hmmhesays | i've never had amp mangle a config where it wasn't user error |
| 16:38.56 | Goral | [TK]D-Fender localnet=xxx.xxx.xxx.xxx(ip of server)/255.255.255.0(local sub net) is this correct |
| 16:39.08 | [TK]D-Fender | Goral : looks good. |
| 16:39.09 | jbalcomb | well, I'm prone to User Error so I guess AMP will not be my buddy. |
| 16:39.23 | Mimmus | I had to put some customization in extension.conf and they will not survive! |
| 16:39.31 | austinnichols101 | hmmhesays: otoh, I use amp myself on one of my systems because it saves me time and keeps me from making simple editing mistakes. |
| 16:39.32 | Hmmhesays | Mimmus: duh |
| 16:39.56 | Goral | [TK]D-Fender : do i have to do externhost |
| 16:40.05 | Mimmus | Hmmhesays: I needed to replace CallerID logic to lookup an LDAP server |
| 16:40.05 | Hmmhesays | if you edit extensions.conf itself, then you use gui to modify it, yes it will mangle your config |
| 16:40.11 | Trevor_b | jbalcomb: look at a demo, if your not a great conf file editor and such, it helps alot. Although yes it is confusing as well (i learned the conf's first, so it was tough to change to a gui) but it made other features much simpler to implement, like IVR, and such. |
| 16:40.25 | austinnichols101 | jbalcomb: you should try it out enough so that you can form your own opinion. |
| 16:40.33 | *** join/#asterisk MatsK (n=mk@84-217-8-79.tn.glocalnet.net) |
| 16:40.40 | [TK]D-Fender | Goral : not if you're using externip. |
| 16:40.45 | iCEBrkr | AMP is tehgay |
| 16:40.50 | MatsK | Is there a way to fake a 503 in asterisk ? |
| 16:41.08 | Goral | ok kool |
| 16:41.08 | Mimmus | Hmmhesays: then I needed to replace fax logic, to lookup destination mail address from LDAP |
| 16:41.11 | jbalcomb | Trevor_b austinnichols101 I think I may. I have a side with just 13 phones so that will be my test bed for it. Thanks for the infos. |
| 16:41.29 | Hmmhesays | Mimmus: you should have been able to do that no problem |
| 16:41.35 | *** part/#asterisk ScaredyCat (n=ScaredyC@net-pbx.demon.nl) |
| 16:42.06 | Mimmus | Hmmhesays: yes but they are all in extension.conf |
| 16:42.23 | Hmmhesays | then you need to learn more about how amp uses its conf files |
| 16:43.11 | Trevor_b | jbalcomb: yeah 13 is simple, i have a site with 8 polycom601's with multiple line monitoring, and 3 remote sites running accross VPN connections, all works great. |
| 16:43.15 | jbalcomb | iCEBrkr!! |
| 16:43.18 | austinnichols101 | jbalcomb: I really wish there was a better thought out and implemented alternative out there. AMP has plenty of issues for sure. |
| 16:43.24 | [TK]D-Fender | Coalescent's site is down... eek |
| 16:43.47 | Mimmus | Hmmhesays: uh? I understand that 'main' logic is in extensions.conf and you can add your stuff to extensions_custom.conf if they don't conflict with others |
| 16:43.55 | Hmmhesays | I'm not saying amp is for every install, but it has its place and if you understand how it works, then you can do just about anything |
| 16:44.04 | austinnichols101 | jbalcomb: there are plenty of times that I feel like I'm fighting against AMP to do what I want. In those cases I either implement something custom or just get rid of AMP completely |
| 16:44.17 | Trevor_b | [TK]D-Fender: I am browsing their site right now. |
| 16:44.38 | Mimmus | austinnichols101: same for me, useful but too restricting |
| 16:44.40 | Hmmhesays | Mimmus: when a fax comes in you would send it to a custom-context to do your ldap stuff, and then send it back into the fax logic |
| 16:44.48 | Hmmhesays | same for callerid |
| 16:44.48 | [TK]D-Fender | Hmmhesays : I'd rephrase that as "can do all the common stuff a "toaster" PBX can do and a few more things in an "accessory" fashion, but not "core". |
| 16:45.01 | Mimmus | Hmmhesays: OK, you are right |
| 16:45.17 | jbalcomb | Mimmus well said. |
| 16:45.21 | [TK]D-Fender | Trevor_b : Link please? I'm following http://amp.coalescentsystems.ca/ and getting NOWHERE... |
| 16:45.29 | Mimmus | Hmmhesays: but you cannot change regular-hours logic or callerid logic in a similar way |
| 16:45.36 | [TK]D-Fender | Trevor_b : Which is where every other reference points to. |
| 16:45.41 | austinnichols101 | [TD]D-Fender: http://sourceforge.net/forum/?group_id=121515 |
| 16:46.01 | Hmmhesays | [TK]D-Fender: I"m modifying your rephrase. You can do all the toaster pbx stuff with amp, and you can integrate your custom dialplan in to do most anything you need it to |
| 16:46.15 | [TK]D-Fender | austinnichols101 : try going to HOME PAGE from there.... |
| 16:46.32 | austinnichols101 | [TK]D-Fender: I'm there |
| 16:46.41 | austinnichols101 | http://coalescentsystems.ca/index.php?option=com_content&task=view&id=31&Itemid=57 |
| 16:46.45 | Hmmhesays | Mimmus: why not? |
| 16:47.14 | [TK]D-Fender | austinnichols101 : must be some sort of internet outage in the way.... |
| 16:47.17 | mut | anyone around that has used a tellab echo can? |
| 16:47.25 | Hmmhesays | you can send any incoming outgoing call to a custom part of your dialplan |
| 16:47.27 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:47.40 | Mimmus | Hmmhesays: how can I set callerid from ldap in a _custom.conf file? |
| 16:47.54 | Trevor_b | can anyone say traceroute. |
| 16:47.55 | Katty | you can make muffins with asterisk if you really wanted to. |
| 16:47.57 | Hmmhesays | from there you can do anything you want with it, then plug it back into the amp dialplan where you want it |
| 16:47.59 | fu3 | is it reasonable to do echo cancellation work between VoIP and a POTS phone, with the analog phone plugged directly into the asterisk box? |
| 16:48.09 | fu3 | I'm concerned that I may only fix echo on that short of distance, which isnt reasonable. |
| 16:48.20 | Katty | or dial extension 40, and have asterisk feed the kittis. |
| 16:48.24 | Katty | s/kittis/kitties/ |
| 16:48.40 | Katty | jbot: thank you captian obvious. |
| 16:48.40 | jbot | no problem, Katty |
| 16:48.50 | Hmmhesays | Mimmus: ldapget my friend |
| 16:48.55 | Hmmhesays | handy little app |
| 16:49.10 | Katty | man, i'm starved. |
| 16:49.16 | [TK]D-Fender | Katty : Yes... I could make muffins with mine.... I have the X-10 stuff long since finished... now all I need is the Barbie EZ-Bake over and I'm set! |
| 16:49.23 | Fedoracore6 | <PROTECTED> |
| 16:49.24 | Fedoracore6 | Feb 28 11:48:41 ERROR[4913]: cdr_addon_mysql.c:437 my_load_module: Failed to connect to mysql database ivr on localhost. |
| 16:49.29 | Katty | [TK]D-Fender: whoo! |
| 16:49.30 | Fedoracore6 | hello my bro |
| 16:49.31 | austinnichols101 | <PROTECTED> |
| 16:49.36 | Fedoracore6 | waht mean this error |
| 16:49.37 | Mimmus | Hmmhesays: I use LDAPget app and it works well |
| 16:49.39 | Katty | file: meet me for lunch! |
| 16:49.43 | file | Katty: where? |
| 16:49.45 | Katty | file: i want indian |
| 16:49.57 | Katty | file: who has biryani? |
| 16:49.59 | file | I want chicken! |
| 16:50.04 | file | are you a chicken? |
| 16:50.08 | Katty | amn't. |
| 16:50.12 | Katty | maybe next life. |
| 16:50.17 | file | k then |
| 16:50.19 | [TK]D-Fender | Katty : I suppose I COULD use a real toaster oven with a locked down timer too.... Oven would require too much modding for me to feel "comforatable" with..... |
| 16:50.20 | Katty | are you an indian? |
| 16:50.21 | Hmmhesays | ok, so you sending an outgoing call to a custom_context where you ldapget the callerid, set the callerid variable and plug it back into the amp dialplan right after callerid is set |
| 16:50.23 | Mimmus | Hmmhesays: but I had to insert it in [macro-user-callerid] |
| 16:50.24 | Hmmhesays | BAM |
| 16:50.25 | file | last I checked I wasn't |
| 16:50.28 | Katty | k |
| 16:50.53 | Fedoracore6 | some budy ... plase help me |
| 16:51.05 | Hmmhesays | get that man some mountain dew and pizza |
| 16:51.08 | iDunno | (erm, *cram*) |
| 16:51.08 | Hmmhesays | STAT! |
| 16:51.36 | Mimmus | Hmmhesays: anyway you put a flea in my ear... I will try to clean my setup... |
| 16:51.50 | Hmmhesays | Mimmus: not saying it is easy.. just saying its possible |
| 16:51.53 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:52.14 | Hmmhesays | You seem reasonably intelligent, you'll figure it out |
| 16:52.38 | Mimmus | Hmmhesays: another problem is that many people say that AMP dialplan is not fully Asterisk 1.2 compliant |
| 16:53.04 | [TK]D-Fender | Katty: Originally I wanted to come back as a matress; so I'd never have to get out of bed. But then a friend of mine told me that with my luck I'd have a bed-ridden 400 pound incontinent person as punishment :) |
| 16:53.07 | Hmmhesays | aren't most of the deprecated funtions still there? |
| 16:53.09 | austinnichols101 | Mimmus: I haven't run into anything that's failed yet with 1.2, but they're definitely using deprecated stuff |
| 16:53.32 | austinnichols101 | Not sure if it's still in that 2.0 version or not... |
| 16:53.57 | Mimmus | ok, great discussion about AMP this evening! |
| 16:54.01 | [TK]D-Fender | I want to see the AMP 2 Beta screenshots and code samples..... if there is any hope for its viability.... |
| 16:54.13 | austinnichols101 | Mimmus: this morning |
| 16:54.21 | Mimmus | austinnichols101: eh eh, I'm in Italy |
| 16:54.32 | Mimmus | (sorry for my bad english) |
| 16:55.09 | austinnichols101 | [TK]D-Fender: screenshots seem to be more-or-less the same |
| 16:55.15 | Fedoracore6 | <PROTECTED> |
| 16:55.15 | Fedoracore6 | Feb 28 11:48:41 ERROR[4913]: cdr_addon_mysql.c:437 my_load_module: Failed to connect to mysql database ivr on localhost. |
| 16:55.22 | Fedoracore6 | wha mean this error |
| 16:55.46 | Mimmus | austinnichols101: where are screenshots? |
| 16:55.47 | [TK]D-Fender | Fedoracore6 : it means that your info for connecting to MySQL is wrong or that MySQL is just not working. |
| 16:55.53 | austinnichols101 | [TK]D-Fender: I love this statement from the 2.0 change list "New module system allows for drop-in functionality" |
| 16:56.08 | [TK]D-Fender | <PROTECTED> |
| 16:57.04 | Mimmus | I will give a look at #amportal someday |
| 16:57.07 | austinnichols101 | Mimmus: I haven't found anything yet that describes what they're trying to accomplish with 2.0 or with the screenshots. The only one I've seen is basically the AMP UI rebranded with "FreePBX". Lemmie see if I can find the link |
| 16:58.09 | austinnichols101 | Mimmus: found this: wiki.freepbx.org |
| 16:58.19 | x86 | anyone here use Stanaphone? |
| 16:58.23 | austinnichols101 | describes the modules system |
| 16:58.32 | Mimmus | yes, I'm seeing that sourceforge is about freepbx |
| 16:59.32 | Hmmhesays | i might have to take a look at that |
| 16:59.36 | Hmmhesays | i hate installing amp though |
| 16:59.51 | Mimmus | a 2.0beta 1 is already here |
| 17:00.01 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 17:00.15 | austinnichols101 | Just read that 'modules' should allow 3rd party modules to be uploaded through the UI. |
| 17:00.35 | Fedoracore6 | [Tk]-Fender : what should i do ... i try setting add cdr_mysql_conf but this didint work |
| 17:00.40 | MRH2 | anyone using sip canreinvite=yes |
| 17:00.44 | Fedoracore6 | any sugestion |
| 17:01.19 | Hmmhesays | Fedoracore6 what is the issue? |
| 17:01.56 | Fedoracore6 | hemm i try to connect to database |
| 17:02.03 | Fedoracore6 | using mysql from asterisk |
| 17:02.17 | Fedoracore6 | but when i try run the code |
| 17:02.29 | Fedoracore6 | the system say like that: |
| 17:02.48 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 17:03.01 | Mimmus | another topic now: how can I share a dialplan between 2 sites? |
| 17:03.05 | heison | ~seen coppice |
| 17:03.09 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 8h 4m 56s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 17:03.13 | Hmmhesays | nfs |
| 17:03.34 | Fedoracore6 | Feb 28 10:54:48 WARNING[4191]: app_addon_sql_mysql.c:365 aMYSQL_disconnect: Invalid connection identifier 0 passed in aMYSQL_disconnect |
| 17:03.34 | Fedoracore6 | <PROTECTED> |
| 17:03.34 | Fedoracore6 | <PROTECTED> |
| 17:03.34 | Fedoracore6 | Feb 28 10:54:48 ERROR[4191]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server localhost. |
| 17:03.34 | Fedoracore6 | <PROTECTED> |
| 17:03.41 | Hmmhesays | whoa, pastebin dude |
| 17:04.00 | Fedoracore6 | k sorry |
| 17:04.04 | Mimmus | no, I have two asterisk servers and users can dial any extension |
| 17:04.06 | file | hot'n'sexy Hmmhesays! |
| 17:04.08 | Trevor_b | AMP 2 or FreePBX is more modulare, you actually choose in the admin module to install ring groups, queue's, IVR, etc. etc. plus you can just click the uninstall and remove options from the admin module as well to remove things you dont need. |
| 17:04.28 | Mimmus | no dialplan separation is possible |
| 17:04.54 | Fedoracore6 | http://pastebin.com/576808 |
| 17:04.57 | Mimmus | switch statement or DUNDI? |
| 17:04.59 | Fedoracore6 | ok i already paste |
| 17:05.10 | fu3 | does pasting five lines really call for pastebin? |
| 17:05.12 | fu3 | i mean, its five lines. |
| 17:05.15 | Hmmhesays | yes |
| 17:05.18 | fu3 | cool |
| 17:05.23 | Qwell | links are 1 line |
| 17:05.30 | fu3 | haha, true! |
| 17:05.33 | Hmmhesays | think if 10 of us in here pasted 5 lines at once |
| 17:05.41 | fu3 | yeah, but how often has that ever happened? |
| 17:05.47 | Qwell | even a "~pb", followed by jbot bitching, followed by a link...is only 3 lines |
| 17:06.03 | fu3 | oh well.. pastebin's pro's outweigh the cons. |
| 17:06.05 | fu3 | no question |
| 17:06.55 | mut | who uses what for e911 here? |
| 17:07.00 | Goral | Hmmhesays since i have added externip = 72.136.246.243 and localnet=192.168.1.201/255.255.255.0externip = 72.136.246.243 and localnet=192.168.1.201/255.255.255.0 to general i cannot make out going calls now. |
| 17:07.02 | mut | i've called a few places and none got back to me yet |
| 17:07.06 | mut | so i'm not sure what to go with |
| 17:08.01 | Hmmhesays | localnet=192.168.1.0/255.255.255.0 |
| 17:08.17 | Goral | ok |
| 17:08.58 | Goral | so a setting like that stays general |
| 17:09.13 | Hmmhesays | yes because you are defining your localnetwork |
| 17:09.25 | *** join/#asterisk flashnet (i=flashnet@Darkstar.AceShells.com) |
| 17:10.35 | [TK]D-Fender | Goral : Your localnet is a single host... not good. |
| 17:10.45 | [TK]D-Fender | yeah.. like Hmmhesays said :) |
| 17:11.05 | [TK]D-Fender | Goral : And you still need "nat=yes" in there... |
| 17:11.44 | Goral | okay |
| 17:13.05 | FuriousGeorge | anyone usin asterisk-xmpp |
| 17:13.48 | Goral | i now cannot make out going calls |
| 17:13.59 | Goral | 404 not found |
| 17:14.03 | RoyK | hwat is xmpp? |
| 17:14.28 | austinnichols101 | RoyK: jabber IM |
| 17:14.31 | RoyK | k |
| 17:14.52 | x86 | who can i use for LNP DID service? |
| 17:15.00 | x86 | just inbound-only would be ideal |
| 17:15.08 | heroine | hmm .. |
| 17:15.18 | x86 | i have outbound PSTN already... |
| 17:15.26 | austinnichols101 | x86: telasip.com is who I'm using |
| 17:15.35 | x86 | austinnichols101: inbound-only? |
| 17:15.54 | austinnichols101 | x86: both ways, but they have a plan that's basically inbound only |
| 17:16.05 | fu3 | ok l;ads |
| 17:16.07 | austinnichols101 | x86: and I'm porting over a number now |
| 17:16.09 | fu3 | i need advice |
| 17:16.14 | fu3 | i just got off the phone with a qwest rep |
| 17:16.27 | fu3 | he says that lines coming across a copper connection from the CO should be around -47VDC. |
| 17:16.41 | fu3 | he also said that lines coming across a fiber, into a CB, and then to a demarc, will be at a lower voltage. |
| 17:16.50 | fu3 | he says that my lines operating at -38VDC is correct and normal |
| 17:16.57 | fu3 | that doesnt make sense to me.. |
| 17:17.00 | heroine | hmm what is the difference between g711u and g711a ? |
| 17:17.19 | x86 | austinnichols101: do they charge you for LNP? |
| 17:17.28 | austinnichols101 | x86: 10 bucks |
| 17:17.35 | austinnichols101 | x86: one time |
| 17:17.59 | RoyK | heroine: minor details. |
| 17:18.19 | RoyK | heroine: g711u is used in the US and g711a is used in most of the rest of the world |
| 17:18.25 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:19.17 | Hmmhesays | i'm going to have to try out this new amp now |
| 17:19.26 | Hmmhesays | is it a biatch to install like amp 1.X? |
| 17:19.41 | heroine | RoyK: hmm ok thanx |
| 17:19.42 | x86 | austinnichols101: and then $6/mo |
| 17:19.50 | x86 | seems to be their lowest plan... |
| 17:19.51 | austinnichols101 | x86: right |
| 17:19.53 | Hmmhesays | Goral your stuff working now? |
| 17:19.54 | x86 | including outbound :( |
| 17:20.06 | austinnichols101 | x86: just don't use the outbound |
| 17:20.38 | x86 | austinnichols101: if i'm paying for it i sure as hell will! :) |
| 17:20.43 | RoyK | heroine: http://lists.digium.com/pipermail/asterisk-users/2004-December/078326.html |
| 17:20.47 | x86 | austinnichols101: they are SIP-only, correct? |
| 17:20.57 | x86 | they dont support IAX2? |
| 17:21.05 | *** join/#asterisk _MartinCabrera_ (n=_MartinC@litigaractivos1.att.net.co) |
| 17:21.19 | austinnichols101 | x86: figure that you can get a single DID (wholesale) from didx.org for around $3-5 a month (inbound only). The same line is around $7/month with virtualphoneline.com. |
| 17:21.46 | austinnichols101 | x86: no - I read that they do also support iax2 on request |
| 17:22.03 | austinnichols101 | x86: haven't tried it out myself |
| 17:22.06 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:22.11 | austinnichols101 | (iax2 + telasip) |
| 17:22.17 | x86 | SIP would work, but I'd prefer IAX2 |
| 17:22.50 | _MartinCabrera_ | Dear sirs! |
| 17:22.59 | austinnichols101 | x86: the $6 plan doesn't have any outbound minutes included in it |
| 17:22.59 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 17:23.08 | x86 | austinnichols101: ah, i see now :) |
| 17:23.10 | x86 | austinnichols101: cool |
| 17:23.31 | austinnichols101 | x86: I found them from the stuff written up over on mundy.org/blog |
| 17:23.43 | x86 | austinnichols101: are they decent? |
| 17:24.01 | x86 | how is quality / reliability? |
| 17:24.20 | x86 | judging by their website i'm not impressed ;) |
| 17:24.27 | austinnichols101 | x86: so far I haven't had any significant issues, but I haven't run a ton of calls through them either |
| 17:24.30 | x86 | the self-signed SSL cert is really the kicker ;) |
| 17:24.40 | x86 | austinnichols101: ever do fax over it? |
| 17:24.48 | austinnichols101 | x86: nope - never tried |
| 17:24.54 | *** join/#asterisk jskcr (n=zaphod@1-pool1.ras14.floca.alerondial.net) |
| 17:25.09 | jskcr | hy all |
| 17:25.19 | Trevor_b | anyone here deal with "loader.c: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_scaling_dbm0f" while booting after setting up spandsp-0.0.2? |
| 17:25.49 | _MartinCabrera_ | Dear Sirs, Please give me any clue about this: After some time. some sip extensions stops receiving inbound calls because of "WARNING[20836]: app_dial.c:1020 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)". The problem goes away if the extension is re-logged. Thanks |
| 17:26.00 | x86 | austinnichols101: does the activation fee cover LNP? |
| 17:26.28 | austinnichols101 | x86: I don't think so. Shoot them an email... |
| 17:26.49 | austinnichols101 | x86: who are you using for outbound? |
| 17:27.12 | tzafrir | Trevor_b, which version of spandsp? IIRC binary compatibility was broken at around pre2x |
| 17:27.18 | austinnichols101 | fu3: you still there? |
| 17:29.26 | Mimmus | bye |
| 17:29.47 | *** join/#asterisk stoffell (n=stoffell@d5153FA4F.access.telenet.be) |
| 17:30.08 | x86 | austinnichols101: teliax |
| 17:30.36 | x86 | austinnichols101: a little over a penny a minute anywhere in the US and Canada... I couldn't find anyone cheaper ;) |
| 17:30.56 | austinnichols101 | x86: I'm using voxee.com |
| 17:31.17 | austinnichols101 | x86: more-or-less the same (0.011/min for US) |
| 17:31.31 | x86 | pay-as-you-go? |
| 17:31.33 | Trevor_b | tzafrir: current spandsp-0.0.2 but the errors appears to be mpg123 about writing audio? or at least the console shows that, asterisk/full shows actual DSP error i pasted. |
| 17:31.38 | x86 | no contracts? |
| 17:31.40 | austinnichols101 | x86: prepaid |
| 17:31.51 | austinnichols101 | x86: no contracts |
| 17:32.05 | Trevor_b | tzafrir: should i move up to one of the pre builds? |
| 17:32.20 | x86 | austinnichols101: mine is pay-as-you-go |
| 17:32.29 | austinnichols101 | x86: how does that work? |
| 17:32.34 | x86 | setup fee is $10, but that's all usable... |
| 17:32.58 | x86 | austinnichols101: when the account reaches $2 left in it, they send you an email to add money to it... so i guess it's kind of pre-pay like yours... |
| 17:33.08 | austinnichols101 | x86: yeah, but without the hastle |
| 17:33.15 | austinnichols101 | x86: hassle |
| 17:33.21 | *** join/#asterisk goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 17:33.33 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 17:33.43 | *** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net) |
| 17:33.50 | r_evolution | k... 1.2.4 not recognizing PSTN DTMF... any help? |
| 17:34.10 | RoyK | it's not? |
| 17:34.16 | RoyK | isdn or pots? |
| 17:34.18 | Trevor_b | r_evolution: like not getting the DID for routing? |
| 17:34.19 | r_evolution | pots |
| 17:34.21 | *** part/#asterisk RoyK (n=roy@80.239.107.70) |
| 17:34.25 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 17:34.31 | r_evolution | nah nah... basically i'm trying to allow remote vmail access |
| 17:34.40 | r_evolution | at first I thought it simply wasn |
| 17:34.55 | r_evolution | was* ignoring the 'a' in the dialplan |
| 17:35.06 | r_evolution | then it occured that when I left a vmail I couldn't press # to end the call |
| 17:35.36 | r_evolution | dtmfmode=auto |
| 17:35.46 | r_evolution | but that's only to do with the SIP calls... nothing from PSTN |
| 17:35.48 | r_evolution | right? |
| 17:36.42 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 17:36.50 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 17:39.00 | r_evolution | sudden silence. |
| 17:39.31 | *** join/#asterisk ToTo (n=ToTo@host254-91.pool8260.interbusiness.it) |
| 17:39.46 | *** join/#asterisk razu_ (n=razu@ip228.cab74.mus.starman.ee) |
| 17:41.03 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:42.06 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 17:43.28 | *** join/#asterisk jwest- (n=jwest@217.164.245.235) |
| 17:43.57 | *** part/#asterisk jwest- (n=jwest@217.164.245.235) |
| 17:44.40 | goral | how do i add extensions to asterisk for existing members |
| 17:45.11 | Nugget | goral, your question is dangerously close to |
| 17:45.17 | Nugget | "just tell me what to type!" |
| 17:45.39 | trixter | so |
| 17:45.49 | Rhizome | hehe |
| 17:46.01 | goral | lol |
| 17:46.14 | goral | well i would like an ez to use guide |
| 17:46.19 | salviadud | if i want to register iax via RSA |
| 17:46.29 | Nugget | I guarantee that it will be simpler for you to read the documentation in your web browser than it would be if you want us to load the documentation and paste it to you over irc. |
| 17:46.30 | salviadud | what do i have to add? |
| 17:46.36 | salviadud | i already got the keys |
| 17:47.27 | salviadud | auth=rsa |
| 17:47.30 | salviadud | is that all? |
| 17:47.40 | salviadud | init keys and zeez boom ba? |
| 17:47.52 | salviadud | somebody throw me a bone! |
| 17:48.13 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 17:48.22 | _MartinCabrera_ | Ok, second try: Why after some time, a registered sip extension can change their Addr->IP from a static to unespecified IP? |
| 17:50.30 | _MartinCabrera_ | Wrong question again :(( |
| 17:50.45 | *** join/#asterisk Preytell (n=bubba@216.63.37.117) |
| 17:50.50 | justinu | that may happen for a number of reasons |
| 17:50.57 | Preytell | afternoon all |
| 17:51.08 | r_evolution | O_O |
| 17:51.10 | r_evolution | justin. |
| 17:51.12 | justinu | a few things to check: NAT Bindings, Registration Timeouts, or other network connectivity issues |
| 17:51.12 | r_evolution | my usual savior. |
| 17:51.19 | justinu | yo yo |
| 17:51.32 | Preytell | First time using this IRC channel. Any rules I should know of? |
| 17:51.38 | justinu | don't flood the channel |
| 17:51.40 | r_evolution | So here's one... why is it that * won't recognize DTMF from PSTN? |
| 17:51.54 | [TK]D-Fender | r_evolution : wHAT IS YOU CALL COMING IN ON? |
| 17:51.59 | justinu | preytell: don't expect people to do your work for you, unless you're paying |
| 17:52.02 | r_evolution | the first rule of #asterisk... don't tell anyone about #asterisk. |
| 17:52.10 | Preytell | fair enough. Are the any 1.2.4 perl AGI people in the room? |
| 17:52.10 | justinu | r_evolution: describe how the call flows into your ast box |
| 17:52.13 | r_evolution | coming in over SIP |
| 17:52.20 | r_evolution | basically... we've got a termination provider |
| 17:52.25 | r_evolution | sends the IP over IP |
| 17:52.27 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 17:52.29 | r_evolution | erm |
| 17:52.31 | r_evolution | info over IP |
| 17:52.32 | justinu | ok |
| 17:52.36 | r_evolution | comes into *... rings... etc. |
| 17:52.43 | r_evolution | however... I'm trying to allow remote voicemail |
| 17:52.43 | justinu | but they're inbound calls to you right? that's called origination, fyi |
| 17:52.55 | justinu | r_evolution: play with the dtmfmode settings, it should be rfc2833 |
| 17:53.03 | justinu | you might need inband, depending on your provider/codec |
| 17:53.04 | *** join/#asterisk bcnl (n=mike@S01060002e30bf999.vs.shawcable.net) |
| 17:53.14 | *** join/#asterisk seele2 (n=_seele@200.124.172.72) |
| 17:53.15 | r_evolution | I had it set to auto... |
| 17:53.16 | r_evolution | one moment |
| 17:53.35 | Preytell | Ok, I'll just ask. What's with the fact the in a GET DATA command, hitting "*" exits the AGI script? |
| 17:53.52 | RoyK | ~seen zoa |
| 17:53.55 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 7d 23h 46m 30s ago, saying: '1000 interrupts per second are ok'. |
| 17:53.55 | _MartinCabrera_ | thans justinu |
| 17:53.56 | justinu | Preytell: personally I have no experience with AGI, sorry |
| 17:54.31 | justinu | r_evolution: if you continue to have problems making it work, please pastebin an invite message from your ITSP |
| 17:54.37 | seele2 | Hey i need some advice here.. i'm not hearing all of my voice recordings,(i know they're installed) are not being listened by any extension.. Could it be that my Asterisk PC has no Sound card??? |
| 17:54.42 | justinu | that way I can see how they want to send DTMF to you. |
| 17:55.18 | r_evolution | negative on the rfc2833 |
| 17:55.24 | _MartinCabrera_ | Work around to may problem: use static ip instead of dynamic registration (Sucks!) |
| 17:55.55 | Preytell | is there an Asterisk developers irc channel? |
| 17:56.05 | file | #asterisk-dev |
| 17:56.15 | r_evolution | hey justin |
| 17:56.18 | r_evolution | ^ |
| 17:56.25 | Preytell | thanks |
| 17:56.30 | *** part/#asterisk Preytell (n=bubba@216.63.37.117) |
| 17:56.30 | goral | what would the setting be to allow a person outside my local network access to my asterisk service? |
| 17:56.55 | seele2 | Hey i need some advice here.. i'm not hearing all of my voice recordings,(i know they're installed) are not being listened by any extension.. Could it be that my Asterisk PC has no Sound card??? |
| 17:58.16 | goral | seele2 no |
| 17:58.18 | [TK]D-Fender | goral : They should each have an account in your sip.conf for them to register to you with |
| 17:58.28 | *** join/#asterisk nain (n=nain@202.59.73.36) |
| 17:58.30 | [TK]D-Fender | seele2 : NO |
| 17:58.48 | *** join/#asterisk trelane_ (n=trelane@mail.allthingsit.com) |
| 17:58.54 | seele2 | goral, So.. why can't hear the messages like timing and those.. |
| 17:59.09 | seele2 | [TK]D-Fender, So.. why can't hear the messages like timing and those.. |
| 17:59.15 | goral | seele2 you may have to set nat settings, that solved my problem |
| 17:59.45 | seele2 | goral, Im not behind a nat... i'm in a switch. |
| 18:00.16 | goral | well then its more than i know.. |
| 18:00.37 | seele2 | goral, what nat settings do you specifically refer to? |
| 18:00.38 | goral | [TK]D-Fender i have my brother setup but its not allowwing him to register |
| 18:01.43 | [TK]D-Fender | goral : Did you forward the UDP ports I told you you needed? |
| 18:02.00 | goral | natnat=yes |
| 18:02.00 | goral | externip = 72.136.246.243 |
| 18:02.00 | goral | localnet=192.168.1.0/255.255.255.0 |
| 18:02.00 | goral | canreinvite=no |
| 18:02.13 | goral | yes |
| 18:02.21 | [TK]D-Fender | goral : You also need to forward 5060, 10000-20000 UDP to your * box. |
| 18:02.23 | goral | [TK]D-Fender : yes i did |
| 18:02.31 | [TK]D-Fender | goral : is your brother behind NAT as well? |
| 18:02.35 | goral | yes |
| 18:03.04 | goral | [TK]D-Fender : he has to do the same correct? |
| 18:03.24 | seele2 | look at http://pastebin.com/576977 , it is clear that the festival conf it's OK and it shows there is a message playing.. the problem is, i cant hear it form any extension. |
| 18:04.50 | salviadud | could somene please explain http://pastebin.ca/43628. |
| 18:05.19 | [TK]D-Fender | goral: what is he using? |
| 18:05.46 | goral | [TK]D-Fender : dlink or linksys router |
| 18:06.01 | [TK]D-Fender | goral : And as a SIP device? |
| 18:06.03 | *** join/#asterisk saftsack (n=saftsack@p54A7D0A5.dip.t-dialin.net) |
| 18:06.18 | goral | hes using sjphone |
| 18:06.46 | seele2 | seele2, why.. why the i cant hear the message???? http://pastebin.com/576977 |
| 18:07.07 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:07.16 | *** join/#asterisk gbodemantv (n=gbodeman@mail.televerde.com) |
| 18:07.23 | gbodemantv | hi |
| 18:07.39 | gbodemantv | so what linux IAX phones do you all recommend? |
| 18:07.48 | goral | [TK]D-Fender : hes using sjphone |
| 18:08.56 | [TK]D-Fender | goral : You need to put "nat=yes" and "qualify=yes" in his sip.conf entry. |
| 18:09.45 | goral | under his heading only or under general |
| 18:09.54 | cpm | I use kiax, but it sucks, like all softphones. |
| 18:10.17 | cpm | that's not fair, it does what it does, and it does work, so |
| 18:10.18 | goral | [TK]D-Fender : under his heading only or under general |
| 18:10.39 | goral | sorry all just geting use to directing messages |
| 18:10.44 | gbodemantv | kix huh? |
| 18:11.04 | seele2 | Please Help me with this... i need some clue on figuring how to hear my Festival recordings... the * shows them playing in console, but i hear nothing from extensions |
| 18:12.15 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 18:12.42 | goral | [TK]D-Fender : how do i assign extensions to each user so we can call eachother |
| 18:13.00 | SwK[Work] | is the iaxmodem guy around? |
| 18:13.01 | [TK]D-Fender | goral : under his heading |
| 18:13.29 | [TK]D-Fender | goral : thats all part of extensions.conf. WOuld I be right in assuming you are extremely new to *? |
| 18:13.49 | goral | [TK]D-Fender : yes |
| 18:13.56 | *** part/#asterisk steveaj (n=steve@82-71-15-37.dsl.in-addr.zen.co.uk) |
| 18:14.10 | Druken | anyone here know how to get ahold of astnorm ? |
| 18:14.21 | Trevor_b | Does anyone have asterisk 1.2.4 tied together with SpanDSP for inbound fax? |
| 18:14.41 | [TK]D-Fender | goral : you need to do some more reading... check out the WIKI, and download the TFOT book |
| 18:14.43 | [TK]D-Fender | ~thebook |
| 18:14.45 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 18:14.47 | [TK]D-Fender | ~docs |
| 18:14.49 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 18:15.22 | heroine | Trevor_b: yup |
| 18:15.35 | heroine | i got that here but got trouble receiving faxes :/ |
| 18:15.40 | Trevor_b | heroine: mind letting me know your spandsp version? |
| 18:15.46 | Trevor_b | damn |
| 18:15.50 | Trevor_b | so its not funcitonal then? |
| 18:16.06 | goral | [TK]D-Fender : thank you |
| 18:16.11 | heroine | the latest spandsp-0.0.2pre8 |
| 18:16.25 | Trevor_b | but you say your having problems receiving? |
| 18:16.29 | heroine | yes |
| 18:16.31 | Trevor_b | damn |
| 18:16.48 | heroine | but i'm not sure problem is linked to spandsp or to the audiocodes configuration |
| 18:17.21 | Trevor_b | had an asterisk@home build and had the other fax machine list OK as the transmit code, but never had the AGI conversion mail it to me. Tried upgrading and broke the spandsp ;) |
| 18:17.23 | heroine | i think it's more a problem in the audiocodes codec configuration |
| 18:17.45 | Trevor_b | you get an error during asterisk loading? |
| 18:17.57 | heroine | no error during the loading |
| 18:18.12 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 18:18.12 | heroine | app_rxfax is launched , doesn't report any error |
| 18:18.19 | Hmmhesays | ok evne the new amp sucks to install |
| 18:18.20 | Trevor_b | lucky you |
| 18:18.31 | heroine | but the destination file is not created, an the sending fax machine report a problem |
| 18:18.40 | Trevor_b | guess ill try pre8 then, since my non pre errors during boot on the libspandsp |
| 18:18.42 | heroine | Trevor_b: what's your error ? |
| 18:18.59 | heroine | while loading asterisk, you get an error message ? |
| 18:19.06 | Trevor_b | Hmmhesays: Yeah i hate the full load, thats why i play with asterisk@home to do all the steps, then i just upgrade the tarballs as needed. |
| 18:19.29 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 18:19.33 | heroine | could be /etc/ld.so.conf not containing the directory where spandsl lib is |
| 18:19.35 | Hmmhesays | can you got up to beta2 on a@h? |
| 18:19.42 | Trevor_b | yeah, erors about the lib, but in the CLI asterisk acts like its mpg123 broken audio pipes. |
| 18:19.54 | Trevor_b | freepbx beta2? |
| 18:19.58 | Hmmhesays | yeah |
| 18:20.28 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
| 18:20.43 | Trevor_b | i have one running freepbx beta1 (different box) but figured since i cant even get asterisk to load without the fax error why bother hitting beta 2 until i fix the startup. |
| 18:21.03 | Trevor_b | ohh |
| 18:21.05 | Trevor_b | your asking |
| 18:21.06 | Trevor_b | yeah |
| 18:21.06 | Trevor_b | DUH |
| 18:21.16 | Trevor_b | i upgraded a ASterisk@home 2.5 to beta 1 |
| 18:22.05 | Trevor_b | had an issue with a script not working properly (deprecated functions.php) that was not in the new functions.inc.php so i uncommented the require_once field in config.php and it worked./ |
| 18:22.30 | Trevor_b | think it was function get_aas if i remember right. |
| 18:23.01 | Trevor_b | heroine: nope, ld.so.conf has it, and i ran ldconfig already |
| 18:23.03 | *** join/#asterisk jbenson (n=jbenson@87.194.2.120) |
| 18:23.07 | heroine | hmm ok |
| 18:23.12 | Trevor_b | it finds the /usr/local/lib file, just errors, here ill paste it |
| 18:23.36 | CRON_AWAY | whats good replacement for cisco ata186/188 |
| 18:23.40 | Trevor_b | heroine: VERBOSE[25867] logger.c: [app_rxfax.so]Feb 28 09:28:03 WARNING[25867] loader.c: /usr/local/lib/libspandsp.so.0: undefined symbol: dds_scaling_dbm0f |
| 18:24.12 | heroine | hmm |
| 18:24.22 | Trevor_b | but im not running preX's |
| 18:24.27 | Trevor_b | just stock 0.0.2 |
| 18:24.45 | *** join/#asterisk vgster (n=vg@spc1-ledn1-3-0-cust136.seac.broadband.ntl.com) |
| 18:25.23 | *** join/#asterisk sack (n=sack@127.Red-81-38-35.dynamicIP.rima-tde.net) |
| 18:25.53 | Trevor_b | tzafrir mentioned something about the pre's, but wasnt sure if he meant to use the pre10-19's and not the pre2x's or what. |
| 18:26.48 | heroine | hmm |
| 18:26.57 | heroine | weird error .. |
| 18:27.10 | Trevor_b | rebuilding asterisk again, see if spandsp didnt patch properly or something |
| 18:27.35 | heroine | spandsp doesn't patch asterisk |
| 18:30.09 | [av]bani | where is Qwell hehe |
| 18:30.16 | _Sam-- | hiding |
| 18:30.57 | [av]bani | :D |
| 18:31.01 | [av]bani | he better be |
| 18:31.09 | _Sam-- | not down with the sccp eh? |
| 18:31.14 | Trevor_b | hmm, could have sword i read something that said that, probably getting lost. |
| 18:32.14 | *** join/#asterisk pzwahlen (n=pz@130.21.202.62.fix.bluewin.ch) |
| 18:33.58 | *** join/#asterisk Seldon1975 (n=someone@199.243.101.131) |
| 18:34.08 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 18:34.34 | gbodemantv | ok I tried Kix as an iax phone any others you would recommend |
| 18:34.42 | websae | does anyone here own a residential voip company? |
| 18:35.10 | rayvd | we do residential voip... |
| 18:35.13 | rayvd | not on a huge scale though |
| 18:35.24 | Seldon1975 | hey guys; I have a bunch of Polycom501s on my Asterisk network, when Brian transfers a call to Max, can the Max's handset display indicate that the incoming call is Transferred, not Internal? |
| 18:38.06 | justinu | ~seen _sam-- |
| 18:38.18 | jbot | _sam-- <n=sam@dca.kneedraggers.com> was last seen on IRC in channel #asterisk, 7m 9s ago, saying: 'not down with the sccp eh?'. |
| 18:38.22 | *** join/#asterisk jstr5707 (n=jstrebel@70.100.62.81.cust.bluewin.ch) |
| 18:38.25 | justinu | hey sam |
| 18:39.51 | nain | Hi,,,Is there any other predictive dialer for asterisk except vicidial and gnudialer ?? |
| 18:40.31 | [av]bani | ~seen qwell |
| 18:40.34 | jbot | qwell <n=north@unaffiliated/qwell> was last seen on IRC in channel #asterisk, 1h 34m 46s ago, saying: 'even a "~pb", followed by jbot bitching, followed by a link...is only 3 lines'. |
| 18:40.42 | Druken | websae: i do residential voip... |
| 18:41.19 | Seldon1975 | ~seen Seldon1975 |
| 18:41.21 | jbot | seldon1975 is currently on #asterisk (7m 23s). Has said a total of 2 messages. Is idling for 2s, last said: '~seen Seldon1975'. |
| 18:42.12 | [TK]D-Fender | [av]bani : Snom pricing jsut went up... update your tables.... |
| 18:42.48 | [av]bani | \o/ |
| 18:47.08 | _Sam-- | <PROTECTED> |
| 18:47.31 | _Sam-- | whats going on justin |
| 18:48.08 | Seldon1975 | whats the best alternative to Comedian mail? |
| 18:48.23 | mog_work | clown mail? |
| 18:48.34 | mog_work | maybe jester mail... |
| 18:48.57 | Seldon1975 | ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha ha |
| 18:49.12 | mog_work | sorry |
| 18:49.15 | mog_work | couldnt resist |
| 18:49.25 | AndyCap | Meridian mail? :> |
| 18:49.30 | Nivex | Crustacean mail? |
| 18:49.41 | Seldon1975 | LameJokes mail? |
| 18:49.42 | Nivex | Communist mail? |
| 18:49.51 | jstr5707 | crich1999 are you there? |
| 18:49.59 | justinu | sam: interesting in buying 4 GXP2000s @ $70? |
| 18:50.03 | justinu | (i know, i'm late) |
| 18:50.16 | justinu | includes box, powersupply, patch cord |
| 18:50.39 | justinu | s/interesting/interested/ |
| 18:50.44 | [TK]D-Fender | [av]bani : $225 vs IP 601 @ $250. Getting to be less of a desirable option.... |
| 18:51.03 | [TK]D-Fender | justinu : EA or collective? ;) |
| 18:51.23 | justinu | each |
| 18:51.30 | [TK]D-Fender | DAMN YOU TO HECK! |
| 18:51.47 | [TK]D-Fender | I might be tempted to pick on up off of you.... |
| 18:51.52 | justinu | k |
| 18:51.57 | [TK]D-Fender | Just to say "I've had one of those" |
| 18:52.10 | justinu | then again, I have a feeling when the customer sees this estimate for 18 IP301s, he might want to keep the GXPs |
| 18:52.14 | justinu | dumbass |
| 18:52.36 | goral | [TK]D-Fender : was that port 5060 and 10000 to 20000 |
| 18:52.38 | *** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 18:52.55 | [TK]D-Fender | goral : Correct. |
| 18:53.23 | [TK]D-Fender | justinu : The thing holding me back is I want to get my Polycom's first.... if I'm going to specialize and be able to demo them I should do that first. |
| 18:54.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:54.20 | [TK]D-Fender | And I've got to see what kind of special I can get them for.... |
| 18:54.37 | justinu | i know I can move the GXPs |
| 18:54.57 | justinu | they also have the latest firmware |
| 18:55.02 | Trevor_b | [TK]D-Fender: there is a decent shop on ebay that dutch auctions NIB models. its where i get my 601's from. |
| 18:55.03 | justinu | not that you guys can't do that |
| 18:55.05 | justinu | but it saves time |
| 18:55.35 | [TK]D-Fender | justinu : I'm sure it won't take long for you to liquidate them at that price |
| 18:56.29 | [TK]D-Fender | Too far.... and risk factor... |
| 18:56.31 | _Sam-- | i already ordered another 11 last week |
| 18:56.48 | _Sam-- | i told ya i needed em badly |
| 18:56.53 | [TK]D-Fender | Trevor_b : link? |
| 18:57.00 | justinu | i know, just checking |
| 18:57.16 | _Sam-- | i may still be able to use them, have another meeting tomorrow |
| 18:57.18 | *** join/#asterisk clive- (n=pirch@dsl-145-33-235.telkomadsl.co.za) |
| 18:57.50 | justinu | this guy can't make up his mind and it irritates me |
| 18:57.53 | Seldon1975 | are there any alternatives to Comedian mail? |
| 18:58.02 | _Sam-- | justinu : sounds the same as the mortgage place i worked with :) |
| 18:58.32 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 18:58.33 | [TK]D-Fender | Seldon1975 : You could always use another PBX VM system or create your own... |
| 18:59.18 | justinu | they tried to blame the system for the fact that the GXPs sounded kinda hollow |
| 18:59.40 | justinu | i had to bring them an IP301 and show them it wasn't the system, but the fact that they bought the cheapest phone, and expected lawyer office quality. |
| 18:59.43 | justinu | grr |
| 18:59.56 | asterisk99 | anyone know why CUT returns 'non-numeric argument' when I use CUT(foobar||1) but not CUT(foobar||2) This is w-e-i-r-d |
| 19:00.01 | Seldon1975 | [TK]D-Fender: dont have time to make my own. How would another PBX's system know to write user-password changes to voicemail.conf? |
| 19:00.34 | websae | anyone own a residentail VoIP service/company in here? |
| 19:00.45 | [TK]D-Fender | Seldon1975 : You're talking about NOT using comedian Mail, so what does Voicemail.conf have to do with anything? |
| 19:00.50 | _Sam-- | justinu : they were able to notice an audible difference between the two phones? |
| 19:00.58 | justinu | yes - they felt it was night and day |
| 19:01.00 | [TK]D-Fender | websae : I moonlight * services... |
| 19:01.20 | websae | how many minutes do you estimate that your endusers use? |
| 19:01.34 | asterisk99 | justinu: Do the IP301 have that infuriating speed dial? Assuming the IP300 is similar to the IP301, if I pick up the handset and dial 1234, asterisk sees 12 and the 34 is thrown away |
| 19:02.06 | MstlyHrmls | asterisk99: sounds like a dialplan issue |
| 19:02.13 | asterisk99 | justinu: But if I dont use the handset, punch in 1234, then dial, it works beautifully |
| 19:02.21 | justinu | i haven't had that problem 99 |
| 19:02.35 | [TK]D-Fender | asterisk99 : that depends on how you set up your dial plans... |
| 19:02.37 | justinu | but I also upgraded the 301's to SIP 1.6.x |
| 19:02.54 | asterisk99 | justinu: I think it may be a firmware issue ... there's not a whole lotta info on the web interface either |
| 19:03.05 | justinu | give up on the web interface |
| 19:03.11 | justinu | just edit the XML files |
| 19:03.16 | Katty | intarweb interface! |
| 19:03.30 | [TK]D-Fender | asterisk99 : Its your Polycom dial-plan, all but certainly... nothing to "fix", just tweak. |
| 19:03.46 | asterisk99 | justinu: Polycom makes a great piece of hardware, but they seem to hire morons to write their software |
| 19:04.00 | nestar | lol |
| 19:04.05 | justinu | yeah - sipura is the exact opposite |
| 19:04.10 | nestar | and man their support department |
| 19:04.11 | justinu | cheap hardware, and brilliant software |
| 19:04.54 | [TK]D-Fender | asterisk99 : As much as I guess I might have liked the Polycom provisioning to be more "flat", I see NO problem using the format they offer and its quite well documented... |
| 19:04.58 | Trevor_b | [TK]D-Fender yeah sorry, give you it in a minute. |
| 19:05.14 | asterisk99 | What kind of idiot presents two radio buttons to the user and no text to say which is which!!!! Is one off and the other on? Is the left one 'Yes' or is it the right one??? Grrrrr |
| 19:05.36 | justinu | i'm not really upset with the polycoms either, i know how to edit the xml files to make them do what I want |
| 19:05.42 | justinu | and it works |
| 19:06.01 | asterisk99 | I LOVE the hardware... It's a REAL MAN'S phone!!!!! ;) |
| 19:06.42 | asterisk99 | anyone know why CUT returns 'non-numeric argument' when I use CUT(foobar||1) but not CUT(foobar||2) |
| 19:06.58 | trelane_ | screw that give me a snom any day |
| 19:06.59 | Trevor_b | [TK]D-Fender: http://feedback.ebay.com/ws/eBayISAPI.dll?ViewFeedback&userid=ntegratedconsulting&iid=5869887763&frm=284 |
| 19:07.03 | trelane_ | based on linux, designed in germany. |
| 19:07.07 | Trevor_b | has 301's for $114 |
| 19:07.29 | asterisk99 | Trevor_b: wowser!!! xcellent $ |
| 19:07.37 | websae | im curious---does anyone know what a residential VoIP company would estimate their customers minutes being to cover the cost on a unlimited plan---anyone???much appreciate :) |
| 19:07.47 | mog_work | based on linux? is linux is what you mean |
| 19:08.25 | *** join/#asterisk pawal (n=pawal@c-65fee253.203-1-64736c11.cust.bredbandsbolaget.se) |
| 19:08.35 | [av]bani | [TK]D-Fender: what, snom 320 price increase? |
| 19:08.42 | heison | ~seen coppice |
| 19:08.46 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 10h 10m 33s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 19:08.46 | asterisk99 | websae: That is the $64,000,000,000,000 question! Sprint Canada at first had an unlimited plan and lost tons of $$$$ on it --- they eventually limited it to 800 minutes |
| 19:09.00 | justinu | asterisk99: you can buy them from atacomm.com for $249 |
| 19:09.16 | [TK]D-Fender | Trevor_b : Poor pricing.... |
| 19:09.16 | mog_work | man i hate the "unlimited plan" is such bs |
| 19:09.18 | websae | 800 minutes is average? |
| 19:09.22 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 19:09.25 | mog_work | does anyone really have unlimited plan? |
| 19:09.30 | [TK]D-Fender | [av]bani : Look everywhere... Atacomm..... |
| 19:09.34 | websae | vonage has unlimited |
| 19:09.40 | mog_work | no its 5000 minutes |
| 19:09.45 | mog_work | thats not unlimited |
| 19:09.48 | [av]bani | [TK]D-Fender: ata front page says $199.95 |
| 19:09.58 | [TK]D-Fender | [av]bani : Click on the link.... |
| 19:10.00 | *** join/#asterisk AAH_USER (n=none@198.60.73.230) |
| 19:10.02 | asterisk99 | websae: It really depends on your users. I did a marketing plan last year and found that my target users (teenagers) hardly used long distance at all |
| 19:10.04 | websae | they softcap on 5000minutes? |
| 19:10.27 | mog_work | their tos says if you go over 5000 minutes they can bill you for those minutes and drop you as a customer |
| 19:10.40 | Trevor_b | [TK]D-Fender: What do you pay for 601's? |
| 19:10.44 | websae | what's a good way to provide local service---without buy a PRI? |
| 19:10.53 | websae | is that in the business or residential? |
| 19:10.58 | asterisk99 | websae: The only way to guess is to survey your intended users.... it really depeneds on age group, ethnic origin, whether or not you have a lot of immigrants, etc, etc, etc |
| 19:11.03 | [TK]D-Fender | Trevor_b : IP 601 = $249 US. |
| 19:11.42 | asterisk99 | websae: That was residential |
| 19:11.46 | Trevor_b | hahaha |
| 19:11.55 | Trevor_b | $249 to $251 is that big a deal? |
| 19:12.04 | *** join/#asterisk Assid (n=assid@59.183.40.9) |
| 19:12.11 | asterisk99 | Two buckazoids is a coffee |
| 19:12.13 | mog_work | Trevor_b, can i have 2 dollars |
| 19:12.17 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:12.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:12.22 | mog_work | as you obviously dont need it ^_^ |
| 19:12.24 | stas_ | question, does anybody knows if it is possible to do telebanking with Analog modem --> Analog/Voip converter --> asterisk --> Zaptel Callout to telebanking ?? |
| 19:12.30 | justinu | i need about tree fiddy |
| 19:12.33 | stas_ | if this works?:-) |
| 19:12.45 | justinu | i ain't giving you no tree fiddy you GODDAMN lochness monster! |
| 19:12.47 | asterisk99 | Anyone here using the new CUT() function? |
| 19:12.54 | Seldon1975 | justinu: you give the Loch Ness Monster $3.50..... |
| 19:13.03 | Seldon1975 | hehehe |
| 19:13.04 | Trevor_b | hey, 2 dollars for a phone from a dealer who ships ASAP, has brand new units, and does good business, ill pay the $2 for the ability to talk to the guy if i have a problem, rather that then save $2 for a vendor whi is a PITA when you have an issue. |
| 19:13.06 | [TK]D-Fender | Trevor_b : VS buying from a certified reseller... |
| 19:13.07 | mog_work | you gave him a dollar! |
| 19:13.31 | trelane_ | <3 Snom. |
| 19:13.43 | Trevor_b | [TK]D-Fender: Who's a certed reseller for 2 bucks less? |
| 19:13.55 | justinu | atacomm.com |
| 19:14.03 | trelane_ | atacomm is cool |
| 19:14.08 | justinu | yep |
| 19:14.16 | Seldon1975 | I said, "Dammit monster! Get off my lawn! I ain't giving you no tree-fitty!" It said, "how about just two-fitty?" I said, "Oh, now it's only two-fitty!! What?! Is there a sale on Loch Ness munchies or something?!" |
| 19:14.17 | trelane_ | owner hangs out here relatively frequently |
| 19:14.23 | asterisk99 | websae: btw, PRI is better anyhoo --- you can provide custom DNIS routing |
| 19:14.47 | stas_ | no one? ;-( |
| 19:14.56 | websae | isn't there a cheaper solution than a PRI? |
| 19:15.07 | trelane_ | stas_, try it and let us know |
| 19:15.27 | asterisk99 | websae: you might be able to buy a fractional PRI instead of the whole sheee-bang (23 channels) |
| 19:15.45 | *** part/#asterisk pzwahlen (n=pz@130.21.202.62.fix.bluewin.ch) |
| 19:15.46 | Trevor_b | hmm, yeah they are only strategic partners with commpartners, cisco, 3com, and nortel, no Polycom. |
| 19:15.46 | stas_ | hmm, thanks.. must i first buy a analog/voip converter but i wil try it thanks for good support. |
| 19:15.54 | Druken | Seldon1975: stay off the juice man :) |
| 19:16.08 | justinu | seldon: that episode was hilarious |
| 19:16.27 | Seldon1975 | heehee |
| 19:16.28 | Seldon1975 | Now this alien had a big head and big black eyes, and it was all bent over me. I said, "What do you want from me, alien?!" and do you know what he said? |
| 19:16.28 | Seldon1975 | Nellie: Tree-fitty. |
| 19:16.46 | Seldon1975 | ok ill stop now |
| 19:16.58 | Seldon1975 | i almost laughed my spleen out |
| 19:17.02 | kpettit | I'm trying to turn a voicmail recording into some on hold music. |
| 19:17.19 | [av]bani | [TK]D-Fender: hmm. tried calling atacomm's 800 # to verify pricing.. i get a weird message and hangup |
| 19:17.25 | [av]bani | [TK]D-Fender: you wanna try? |
| 19:17.39 | kpettit | But it's very garbled, bad quality. I've tried using mpg123 and madplay, as well as different wav2mp3 encoding bit rates and can't seem to make it sound any better. Any suggestions? |
| 19:18.17 | Druken | kpettit: does it sound find on comedian mail ? |
| 19:18.24 | kpettit | yes |
| 19:18.33 | justinu | kpettit: asterisk can play your VM message without conversion to MP3 |
| 19:18.34 | kpettit | sounds great if I listen to it as a wav or mp3 on my system |
| 19:18.38 | Druken | then just copy the damn wav file to your moh directory |
| 19:18.43 | justinu | yep, listen to druken |
| 19:18.44 | kpettit | but as a mp3 played through music on hold it's bad |
| 19:19.00 | justinu | you're making it harder than it has to be |
| 19:19.08 | [av]bani | kpettit: sounds like gsm |
| 19:19.17 | kpettit | musiconhold won't play .wav unless I'm missing something simple |
| 19:19.26 | kpettit | I tried using the .wav and it didn't pick it up |
| 19:19.33 | Druken | it'll play wav, mp3, gsm, etc |
| 19:19.35 | kpettit | it'll pickup mp3's but now wav |
| 19:19.39 | Druken | or at least it SHOULD |
| 19:19.51 | [TK]D-Fender | [av]bani : They are just God awful slow at updating their site, but its clean in that others have gone up as well... |
| 19:19.54 | justinu | set the type=native in moh.conf |
| 19:20.06 | kpettit | justinu, ok |
| 19:20.38 | kpettit | that will let me play .wav? |
| 19:20.45 | justinu | it should |
| 19:21.01 | kpettit | thanks I'll give it a whirl. |
| 19:21.04 | justinu | make sure you do a reload moh |
| 19:21.09 | justinu | or moh reload, whatever it is |
| 19:21.27 | *** join/#asterisk rezzen (n=pkn@ottawa-dial-209-217-80-49.d-ip.magma.ca) |
| 19:21.31 | websae | anyone know what a good FXO gateway appliance is....without have to buy fxo card and setup asterisk server....i want my remote asterisk server to be able to use the PSTN lines at a location to make calls out on |
| 19:21.33 | Druken | just restart the whole damn thing |
| 19:21.36 | Druken | restart now |
| 19:21.37 | Druken | :) |
| 19:21.49 | justinu | not really necessary |
| 19:22.22 | [TK]D-Fender | websae : What kind of port density? |
| 19:22.25 | Druken | justinu: not really... but it'll save time for him :) and less confusion |
| 19:22.34 | AAH_USER | But restarting the whole damn thing is fun |
| 19:22.37 | justinu | heh |
| 19:22.42 | kpettit | webmind, http://www.x100p.com/products_2.htm |
| 19:22.43 | justinu | says the AAH_USER |
| 19:23.00 | kpettit | that'll let you do IAX and PSTN and create a dialplan on the box on which to use. |
| 19:23.11 | kpettit | it's a pretty kick ass box. |
| 19:23.17 | AAH_USER | kpettit, create box, I have one and love it |
| 19:23.22 | Druken | can't ya use a SPA-3000 for a foreign FXO ? |
| 19:23.33 | justinu | Druken: haven't seen you around here in a while |
| 19:23.36 | justinu | what up? |
| 19:23.44 | AAH_USER | Yes for spa-300, but not a cheap, and not as good |
| 19:23.47 | Druken | been busy... :) |
| 19:23.50 | *** join/#asterisk SimplTon (n=bushah@user-108756k.cable.mindspring.com) |
| 19:24.03 | Druken | wuts up with you justinu ? |
| 19:24.03 | kpettit | AAH_USER, Had to walk somebody through config'ing the IP address through there dialpad on the phone today, that was sooo cool. |
| 19:24.03 | AAH_USER | The x100p box is only $70 |
| 19:24.10 | justinu | Druken: ditto... |
| 19:24.21 | justinu | busy invoicing customers right now |
| 19:24.29 | AAH_USER | Easiest thing in the world, almost |
| 19:24.55 | kpettit | I got some freaky good echo canceling out of it |
| 19:24.58 | rezzen | websae: www.mediatrix.com or www.sipura.com |
| 19:25.03 | Druken | i wish i was busy invoicing customers.... |
| 19:25.10 | [TK]D-Fender | kpettit : and thats FXS, not FXO...... |
| 19:25.26 | AAH_USER | Customers, wish I had some to invoice |
| 19:25.47 | kpettit | AAH_USER, haha |
| 19:25.57 | Druken | :) |
| 19:26.04 | justinu | aah_user: customers are jagoffs |
| 19:26.11 | kpettit | [TK]D-Fender, Yeah, but it sounded like he needed FXS for what he wanted to do |
| 19:26.27 | [TK]D-Fender | [14:21] <websae> anyone know what a good FXO gateway appliance is....without have to buy fxo card and setup asterisk server....i want my remote asterisk server to be able to use the PSTN lines at a location to make calls out on |
| 19:26.36 | AAH_USER | But invoicing them and getting $ is fun, ok maybe only the $ part |
| 19:26.40 | [TK]D-Fender | kpettit : does that SOUNDS like FXO to you? I don't think so.... |
| 19:26.47 | [TK]D-Fender | FXS that is ;) |
| 19:26.51 | justinu | aah_user: getting them to fork over the money is a fucking hassle |
| 19:26.55 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 19:26.59 | [av]bani | spa3k :< |
| 19:27.02 | [av]bani | ugh |
| 19:27.18 | AAH_USER | I use a small asterisk box as a remote fxo with a x100p fxo card in it |
| 19:27.36 | AAH_USER | justinu, too true |
| 19:27.39 | websae | just a FXO gateway |
| 19:28.05 | Trevor_b | aah_user: you paid $70 for an x100p!?!? |
| 19:28.20 | justinu | aah_user: these guys think i'm a fucking moron, trying to play me all the time... |
| 19:28.31 | kpettit | asterisk still not playing .wav files. I get Found no files in '/home/tcs/sounds/queuethree with the .wav's in there |
| 19:28.47 | Trevor_b | kpettit: file permisions? |
| 19:28.53 | AAH_USER | Sorry, $30 for x100p fxo, $60 for x100p.com's fxs ATA |
| 19:29.00 | iCEBrkr | ~amp |
| 19:29.03 | jbot | [amp] NOT supported here! people using it should join #amportal |
| 19:29.07 | AAH_USER | Or maybe $70 for the ATA |
| 19:29.16 | jaiger | websae, we use spa3000 for low port density fxo |
| 19:29.21 | kpettit | Trevor_b, 755 |
| 19:29.32 | Trevor_b | yeah the x100p remarked ones you can pickup as the IA92's for like $9-15 |
| 19:29.34 | jaiger | couldn't get a mediatrix to work |
| 19:29.55 | AAH_USER | Spa 3000 Usable as remote fxo, if in the dmz |
| 19:30.33 | [TK]D-Fender | AAH_USER : Doesn't need ANY port forwarding..... |
| 19:30.45 | websae | low port density? |
| 19:30.45 | AAH_USER | Trevor_B, maybe so, but I like to play it a little safe, $30 ain't bad |
| 19:30.48 | websae | what do you mean by that? |
| 19:30.58 | [TK]D-Fender | AAH_USER : You can have it as "nat=yes" and not have to forward anything at all to it... |
| 19:31.40 | AAH_USER | But the spa3000 it sip, you can lose rtp using it as a remote fxo if you don't forward udp ports |
| 19:31.44 | znoG | anyone know if the TDM2400P RJ11 panel is a standard thing or Digium specific? |
| 19:31.59 | Seldon1975 | justinu: how they play you |
| 19:32.13 | Trevor_b | aah_user: not 100% positive, but 99%, the x100P are relabeled intel softmodems. IA92 is the model, supposedly they just scrathed the chip so you couldnt tell when you purchased x100p's. I use the IA92's for basic FXO testing all the time, work fine for single line softmodems.... |
| 19:32.23 | Seldon1975 | znoG: it's standard |
| 19:32.25 | [TK]D-Fender | AAH_USER : Not true... it registers the FXO port just it would an FXS ports and we all know you don't need to forward anything for NAT keep-alives to work just fine... |
| 19:32.26 | Trevor_b | znoG: specific?? |
| 19:32.27 | trixter | the rj11 panel should be standard |
| 19:32.37 | trixter | its an amphenol connector right? that is standard and has been for decades |
| 19:32.38 | justinu | Seldon1975: just trying to say that money they paid was for stuff they haven't gotten yet. |
| 19:32.49 | justinu | Seldon1975: so I have to reprint the invoices and show them exactly what they've paid for, etc. |
| 19:32.59 | justinu | Seldon1975: they don't play me, but they try. |
| 19:33.00 | Seldon1975 | justinu: asses |
| 19:33.05 | znoG | Trevor_b: i thought it may be a digium specific connector or something, but seeing as how it's not i'll buy it from a local shop |
| 19:33.05 | Trevor_b | Seldon1975: have one? I do, its just RJ11 running across a 25 pair, maybe a 50pair. Stock way punch blocks have been for years...... |
| 19:33.21 | Seldon1975 | Trevor_b: yep |
| 19:33.31 | znoG | another question: can the hardware echo cancellation be added on later (ie. as a module) or is it soldered to the board? |
| 19:33.33 | jaiger | Trevor_b, they glue a heatsink over the Intel chip so you can't read it |
| 19:33.34 | Seldon1975 | Trevor_b: white/blue blue/white etc, etc |
| 19:33.34 | AAH_USER | I have seen sip rtp issues resulting from nat without appropriate forwarding, just my own experience |
| 19:33.36 | Trevor_b | so why would you say its not standard? |
| 19:33.55 | Seldon1975 | Trevor_b: I didn't.... |
| 19:34.01 | Seldon1975 | Trevor_b: read again |
| 19:34.28 | Trevor_b | misread znoG's name flying by ;) |
| 19:34.36 | Seldon1975 | Trevor_b: k |
| 19:34.37 | Trevor_b | though i say No standard. sorry |
| 19:34.38 | [TK]D-Fender | AAH_USER : Ok well I've done plenty of the reverse. Depends on how shitting the router is I guess. Typically the SPAs work jsut fine behint NAT as they were the first units used by all the big players like Vonage who knew that'd be the case with the majority of their customers... |
| 19:34.38 | *** join/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 19:34.58 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:34.59 | Chotaire | ~callback |
| 19:35.13 | [av]bani | i wouldnt use spa for fxo... |
| 19:35.14 | Seldon1975 | Trevor_b: no probs, 'znoG' does look like 'not' when you speed read |
| 19:35.34 | [TK]D-Fender | znoG : RJ11 (amphenol) is an ancianet standard and you can pick up the cross connect stuff anywhere... |
| 19:35.47 | AAH_USER | True, reverse (normal ata usage) is more nat friendly than using the spa3000 as a remote fxo |
| 19:36.08 | kpettit | what "mode" do you use in musiconhold.conf to be able to play plain old .wav files? |
| 19:36.10 | Trevor_b | yeah we almost used the port B on my breakout to connect to the punch block directly instead of RJ11, but the wiring didnt match ports 1-8..... |
| 19:36.21 | jaiger | [av]bani, what do you recommend instead? I've been looking for alternatives |
| 19:36.46 | Trevor_b | jaiger: how many FXO lines? |
| 19:36.55 | jaiger | one to four |
| 19:36.58 | Seldon1975 | Trevor_b: how long have you been using your 2400 |
| 19:37.07 | Trevor_b | personally i like the 4dm4xxB |
| 19:37.16 | Trevor_b | 4 lines max, either FXO or FX |
| 19:37.19 | Trevor_b | FXS |
| 19:37.45 | [av]bani | tdm are problematic. some people get lucky with the software based echo canceller, but most have problems |
| 19:37.48 | znoG | [TK]D-Fender: aha, what they're suggesting here is that we can plug the TDM2400B card using a RJ11 to RJ11 cable straight into a Lucent Definity PBX |
| 19:37.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.76.246) |
| 19:38.00 | znoG | [TK]D-Fender: not sure if it'll work, but it kinda makes sense. |
| 19:38.03 | AAH_USER | The sangoma fxo/fxs cards are nice |
| 19:38.05 | Trevor_b | Seldon1975: about 2-3 weeks now, having an issue with DID's, but not sure if thats asterisk 1.2.4, zaptel 1.2.4, or the wctdm24xxp driver... OH it could also be the centrex... |
| 19:38.17 | kpettit | AAH_USER, just bought a few of them |
| 19:38.19 | [av]bani | ~gateways |
| 19:38.20 | jbot | rumour has it, gateways is at http://bani.anime.net/gateways/ |
| 19:38.22 | kpettit | and a PRI as well |
| 19:38.39 | [TK]D-Fender | znoG : Sorry.. it isn't RJ11, but rather RJ21, and yes it could be that you have an amphenol direct-connect option to it... |
| 19:38.40 | jaiger | Trevor_b, got a link? |
| 19:38.41 | kpettit | still trying to figure out wanpipe, but i love the options the cards have |
| 19:38.43 | AAH_USER | kpettit, Hard core |
| 19:38.53 | FuriousGeorge | my sata drives are making a different noise that they used too |
| 19:38.57 | [av]bani | the sangoma cards are expensive :> |
| 19:38.58 | FuriousGeorge | its kinda disconcerting |
| 19:39.08 | Trevor_b | jaiger: digium.com or voipsupply.com is where i usually look. |
| 19:39.09 | znoG | [TK]D-Fender: i actually wrote RJ21 but then changed to RJ11 since you wrote RJ11 hehe, i thought i had it wrong |
| 19:39.11 | AAH_USER | FuriousGeorge, sounds not so good |
| 19:39.22 | jaiger | Trevor_b, oh you mean the tdm4xxb? |
| 19:39.30 | znoG | [TK]D-Fender: yep we have a few lucent RJ21 to RJ21 cables lying around, so it's worth a shot |
| 19:39.30 | FuriousGeorge | AAH_USER: yeah, like a rattle |
| 19:39.37 | Trevor_b | jaiger: yeah what were you asking about :)? |
| 19:39.45 | jaiger | I have some, they're crap. The ecan is aweful |
| 19:39.49 | Druken | rj11 or rj12? |
| 19:39.53 | Seldon1975 | ppl here using Polycoms?; I have a bunch of Polycom501s on my Asterisk network, when Brian transfers a call to Max, can the Max's handset display indicate that the incoming call is Transferred, not Internal? |
| 19:39.55 | AAH_USER | FuriousGeorge, hope you have them in a raid 1 or 5 |
| 19:40.02 | Druken | 11 os 1 pair, 12 is 3 |
| 19:40.05 | DarKnesS_WolF | i have a question .. i'm trying to make voicemail, now i added the mailbox => password,owner, in voicemail.cfg and the extstion has the voicemail enabled .. and i added into it exten => 9999,1,VoiceMailMain(user@relm) but when i diall 999 9it askes for the password i put it but it's sayes login incorrect like it's not feeling the DTMF tone .. any idea? |
| 19:40.09 | jaiger | Trevor_b, you said "4dm4xxb" which I couldn't find in google |
| 19:40.10 | AAH_USER | Otherwise I recommend a speedy backup |
| 19:40.15 | FuriousGeorge | AAH_USER: its my home pc, you know its 0:) |
| 19:40.57 | [av]bani | Seldon1975: thats because it IS transferred |
| 19:40.57 | AAH_USER | FuriousGeorge, time for a quick backup |
| 19:40.57 | [av]bani | Seldon1975: err, youd have to modify cid |
| 19:40.58 | Trevor_b | jaiger: ahh sorry, too fast typing not paying attention. If you have more then 4 lines, i like the tdm24xxE (e model has hardware echo cancelation) |
| 19:40.58 | Seldon1975 | [av]bani: ok ist that possible? |
| 19:41.02 | [av]bani | Seldon1975: probably |
| 19:41.03 | AAH_USER | FuriousGeorge, you will need a new sata drive shortly me thinks |
| 19:41.17 | Seldon1975 | [av]bani: any idea how? |
| 19:41.19 | [av]bani | Trevor_b: looks like the hardware EC in the 24xx is problematic |
| 19:41.23 | AAH_USER | Furious george, Check techbargains.com for good deals |
| 19:41.32 | jaiger | Trevor_b, I recommend T1 + channel bank with more than 4 lines already but I'm looking for a decent 1 to 4 port analog solution |
| 19:41.33 | [av]bani | Seldon1975: transfer via phone or via asterisk function? |
| 19:41.39 | FuriousGeorge | AAH_USER: should be under warranty |
| 19:41.51 | [av]bani | Seldon1975: eg are you using transfer of dial() function or transfer button on phone? |
| 19:41.55 | jaiger | currently use sipura ATAs which work ok for fxs but I have mixed luck with fxo |
| 19:41.57 | AAH_USER | Good stuff those warranties, still make sure you backup |
| 19:42.02 | Seldon1975 | [av]bani: on the phone |
| 19:42.20 | FuriousGeorge | AAH_USER: then again i accidentally ripped the data cable and screwed up the pins and the socket area |
| 19:42.25 | [av]bani | Seldon1975: probably not if done from the phone |
| 19:42.44 | Seldon1975 | [av]bani: ok thx, |
| 19:42.52 | AAH_USER | FuriousGeorge, might not be covered by warranty anymore me thinks |
| 19:43.03 | Seldon1975 | [av]bani: out of curiosity could you do it in the dialplan following transfer from Dial? |
| 19:43.05 | [av]bani | Seldon1975: you could make a special dialplan which would announce it |
| 19:43.15 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 19:43.15 | Seldon1975 | [av]bani: aah I see |
| 19:43.20 | diclophis | howdy all |
| 19:43.22 | [av]bani | Seldon1975: transfer to extension *9705 instead of 9705 |
| 19:43.24 | AAH_USER | FuriousGeorge Lets take it out of the channel |
| 19:43.27 | diclophis | anyone want to email me the PDF of ISO13714 |
| 19:43.31 | [av]bani | Seldon1975: which would prepend "Transfer:" to the CID |
| 19:43.38 | Seldon1975 | [av]bani: that might work actually |
| 19:43.42 | Seldon1975 | [av]bani: thanks! |
| 19:43.50 | [av]bani | Seldon1975: its the easiest way imo, but a little more work for users |
| 19:44.01 | FuriousGeorge | AAH_USER: we'll see |
| 19:44.03 | Seldon1975 | [av]bani: yeah, I think they'll go for it though |
| 19:44.18 | [av]bani | Seldon1975: though really, attended transfer would make it obvious :) |
| 19:44.21 | jaiger | Seldon1975, aren't the polycom transfers attended? the caller can tell the receiver it's transfered |
| 19:44.28 | Seldon1975 | [av]bani: if they think it's too much effort they can still transfer direct to the ext. |
| 19:44.30 | *** part/#asterisk JerJer (n=jj@pdpc/supporter/bronze/jerjer) |
| 19:44.40 | [TK]D-Fender | jaiger : Polycom can do attended AND blind transfers |
| 19:44.43 | Seldon1975 | jaiger: you can do blind t/f |
| 19:44.47 | [av]bani | Seldon1975: you can just tell them a * before the extension will announce it |
| 19:44.59 | Seldon1975 | [av]bani: yeah, sweet |
| 19:45.07 | Seldon1975 | [av]bani: good thinking 99 |
| 19:45.10 | [av]bani | Seldon1975: so they can yell at each other for not announcing it |
| 19:45.38 | Seldon1975 | [av]bani: well only 4 or 5 of the users care |
| 19:45.54 | [av]bani | they will make sure everyone else cares :) |
| 19:47.02 | Seldon1975 | [av]bani: usually... |
| 19:47.14 | Seldon1975 | [av]bani: esp since one is the CEO |
| 19:47.42 | diclophis | come on.. someone must have the pdf of the iso |
| 19:48.02 | [av]bani | Seldon1975: :) |
| 19:48.04 | bsdfreak | the problem is |
| 19:48.06 | jaiger | Seldon1975, always get buy-in from the top |
| 19:48.06 | bsdfreak | lewl is not a mclure |
| 19:48.36 | heison | ~seen coppice |
| 19:48.38 | jbot | coppice <n=chatzill@212.197.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 2d 10h 50m 25s ago, saying: 'you mean packets are delivered by a cron job? :-\'. |
| 19:49.35 | [av]bani | whee, buggy cisco |
| 19:49.44 | AAH_USER | Buggy cisco? |
| 19:49.53 | AAH_USER | I like my 12sp with asterisk |
| 19:50.04 | AAH_USER | I have also used 7960 |
| 19:50.08 | [av]bani | 7970 |
| 19:50.22 | AAH_USER | The color phone, no |
| 19:50.47 | [av]bani | keeps losing the ability to route through default gateway |
| 19:50.52 | [av]bani | after about 30min |
| 19:50.58 | Druken | i go into a place with hundreds of 7960's... i keep telling the guy i'm gonna rob him of them |
| 19:51.13 | [av]bani | meh |
| 19:51.13 | AAH_USER | I understand that one isn't officially supported yet, which driver are you using, sccp or skinny |
| 19:51.20 | [av]bani | they arent _that_ nice |
| 19:51.42 | Druken | [av]bani: they would be if they were free.... :) |
| 19:51.46 | [av]bani | AAH_USER: this has nothing to do with the driver... the phone itself loses the ability to route through default gateway |
| 19:51.48 | [TK]D-Fender | [av]bani : What's not "_that_ nice"? |
| 19:51.50 | *** join/#asterisk frankj (n=frankj@38.136.42.2) |
| 19:51.57 | AAH_USER | Are you using skinny (sccp) firmware , or the sip firmware |
| 19:52.13 | [av]bani | [TK]D-Fender: not nicer enough over the ip601 to justify the exorbitant prices |
| 19:52.27 | [av]bani | AAH_USER: there is no sip for 7970 |
| 19:52.40 | AAH_USER | Gotcha, skinny only |
| 19:52.58 | AAH_USER | Are you using chan_sccp.so or chan_skinny.so as the driver? |
| 19:53.12 | jaiger | [av]bani, you using DHCP to assign addresses? |
| 19:53.15 | [av]bani | AAH_USER: why would chan_*.so have anything to do with the phone becoming unpingable after 30min? |
| 19:53.18 | [av]bani | jaiger: yes |
| 19:53.27 | jaiger | is that the lease time? |
| 19:53.31 | [av]bani | jaiger: no |
| 19:53.37 | jaiger | or half the lease time? |
| 19:53.40 | [av]bani | no |
| 19:53.42 | AAH_USER | 2 different drivers, it could be a driver issue |
| 19:53.50 | [TK]D-Fender | [av]bani : At the 797x or all Ciscos? |
| 19:54.01 | [av]bani | AAH_USER: the phone becomes _unpingable_... this has nothing to do with asterisk |
| 19:54.16 | AAH_USER | Gotcha |
| 19:54.17 | [av]bani | jaiger: the lease time is 4 days |
| 19:54.28 | frankj | can anyone tell me how to configure the polycom phone to dial 9 + the user's extension to reach his voicemail when the "message" button is pressed? By default it dials just the user's extension. |
| 19:54.37 | AAH_USER | Have you tried a static IP on the phone? |
| 19:54.48 | [av]bani | jaiger: also... it can ping local net, but not route through default gw |
| 19:55.00 | [av]bani | jaiger: it works for about 30 min then stops being able to route |
| 19:56.31 | AAH_USER | I have seen that alot with cisco phone with the sccp/skinny firmware. |
| 19:57.15 | Hmmhesays | cpan you are pissing me off today |
| 19:57.24 | znoG | cool it was pissing me off yesterday |
| 19:57.34 | delmar | dudes.. donmt get me started. |
| 19:57.37 | delmar | cpan = teh sux |
| 19:57.38 | [av]bani | it pissed me off last week |
| 19:57.52 | jaiger | [av]bani, I'm happy to say I only use Polycom |
| 19:57.54 | znoG | 28 retries before it successfully downloads something |
| 19:58.01 | [av]bani | jaiger: yay no backlight |
| 19:58.08 | delmar | znoG, thats nuts |
| 19:58.13 | jaiger | [av]bani, I know. I've been wanting that lately |
| 19:58.21 | [av]bani | jaiger: and 3 minute boots |
| 19:58.46 | [TK]D-Fender | [av]bani : Yeah, but lack of multiple spontaneous reboots like other phones! |
| 19:58.47 | [av]bani | i dont understand why polycom doenst support minibrowser in the 500/501 |
| 19:58.57 | [av]bani | [TK]D-Fender: ah you mean like spa-941 |
| 19:59.13 | [TK]D-Fender | [av]bani : Really? Haven't heard that bout it yet... |
| 19:59.15 | jaiger | [av]bani, 500 probably doesnt' have enough memory. 501 should tho |
| 19:59.28 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 19:59.49 | digime | <PROTECTED> |
| 19:59.57 | [av]bani | jaiger: yea but the 501 doesnt either :) |
| 20:00.04 | jaiger | would be nice though. I only need a couple lines so the 500 works for me mostly but a minibrowser would be cool |
| 20:00.07 | jaiger | [av]bani, I know |
| 20:00.36 | [av]bani | jaiger: but then polycom has not been terribly respnsive to their customers. eg 7 blf limit |
| 20:00.39 | AAH_USER | Dgime, a ring group? Or a queue |
| 20:00.43 | [TK]D-Fender | digime : Nope... in SIP a call is a call is a call... it can differentiate.... |
| 20:00.48 | [TK]D-Fender | can't* |
| 20:01.45 | [TK]D-Fender | [av]bani : How many IP-PBX's use the subscribe method for BLF instead of SIP-B SLA? I think its a quesiton about "nudging" towards the "norm". |
| 20:01.48 | [av]bani | jaiger: watch polycom release their next phone. color touchscreen, without backlight! |
| 20:01.54 | digime | aah: queue |
| 20:02.09 | [av]bani | [TK]D-Fender: most of them? sla is new |
| 20:02.17 | [TK]D-Fender | [av]bani : Actually they're next phone is a model to fit between the 301 & 501 with features of both. |
| 20:02.32 | [TK]D-Fender | [av]bani : SLA has been on Sipura and plenty of others for a while.... |
| 20:02.39 | [av]bani | [TK]D-Fender: most ip-pbx's are single-vendor proprietary |
| 20:02.47 | [TK]D-Fender | [av]bani : so not that new... its the more "proper" way |
| 20:02.58 | [av]bani | eg linksys 9000, ccm, mitel |
| 20:03.33 | AAH_USER | digime, what is your queue ring strategy, I assume ringall |
| 20:03.39 | digime | yes |
| 20:03.39 | *** part/#asterisk frankj (n=frankj@38.136.42.2) |
| 20:03.52 | [av]bani | [TK]D-Fender: well the fact is it's a boneheaded hardcoded limit that nobody else has... |
| 20:03.56 | [TK]D-Fender | [av]bani : I don't think the 9000 is a stretch from basic SIP as we know it.... what does it do thats "off the record" as far as standards goes? |
| 20:04.23 | [TK]D-Fender | [av]bani : Boneheaded? Yeah... I'll agree with you there..... assuming its for the reasons some suspect... |
| 20:04.38 | AAH_USER | That is the problem, if you used something like "least recent" it wouldn't ring everyone resulting in all the missed calls |
| 20:05.54 | [av]bani | [TK]D-Fender: there are some useful points for hints though... they are independent of lines, bla will allow something closer to key systems, but not everyone will want that |
| 20:06.09 | [av]bani | [TK]D-Fender: eg some will want the line monitoring divorced from the actual line functionality |
| 20:06.25 | digime | aah: ahh ok |
| 20:06.30 | [TK]D-Fender | AAH_USER : Least recent has its own curse if the guy walks away from his desk.... NOONE gets calls... |
| 20:06.40 | AAH_USER | true |
| 20:06.41 | [TK]D-Fender | [av]bani : Yeah, there is that... |
| 20:06.52 | [TK]D-Fender | [av]bani : They'll get around to it I guess... |
| 20:06.56 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:06.58 | [av]bani | [TK]D-Fender: like, how the heck are you going to light up a button for meetme room with bla? |
| 20:07.04 | digime | aah: but we need to have multiple people hear the ring at once :( |
| 20:07.21 | AAH_USER | Maybe rrmemory would be better |
| 20:07.31 | [TK]D-Fender | rrmemory is good... |
| 20:07.37 | [av]bani | [TK]D-Fender: you can do it with bristuff and the hinting tools... but that wont work for bla ... |
| 20:08.26 | [av]bani | will be interesting to see what happens |
| 20:08.31 | [TK]D-Fender | [av]bani : I know.... its a question of target audience. SLA is more for lines/phones in a traditional PBX style. |
| 20:08.46 | [TK]D-Fender | [av]bani : we DO deserve it all though.... |
| 20:08.52 | [av]bani | [TK]D-Fender: but some people will still be cramped by polycom's boneheaded limit |
| 20:09.03 | [av]bani | [TK]D-Fender: well... backlight anyone? hellooooooooooo polycom |
| 20:09.20 | jaiger | ~bla |
| 20:09.21 | jbot | foo |
| 20:09.25 | jaiger | ~blf |
| 20:09.26 | jbot | blf is, like, Busy Lamp Field, aka little lights next to speed dials that light up when the person is on the phone and blink when that line is ringing. hint extensions are static mapped to SIP or other channels. |
| 20:09.30 | [av]bani | heh |
| 20:09.56 | [TK]D-Fender | [av]bani : only a small portion of the markey offers a backlight, and the only decent phone withit would be Snom right now. Frankly I let a few things slide... |
| 20:10.14 | [av]bani | [TK]D-Fender: eh... about 1/2 of the market has backlight |
| 20:10.15 | jaiger | ~sla |
| 20:10.17 | jbot | from memory, sla is service level agreement. if they're down for more then XX minutes, they pay *YOU* |
| 20:10.18 | [TK]D-Fender | market* |
| 20:10.38 | *** join/#asterisk Riedel (i=riedel@darktower.versahell.net) |
| 20:10.47 | [av]bani | [TK]D-Fender: 7970 is 'decent' and has backlight :)) actually the aastra 480i is nice i guess |
| 20:11.34 | saftsack | are some isdn experienced people here? |
| 20:11.37 | AAH_USER | The 7970s have a few asterisk issues though. Not that they can't be worked around |
| 20:12.00 | Riedel | It seems that I'm unable to obtain information about the number dialed to reach my PBX. Is there any way to obtain this information ? |
| 20:12.11 | [av]bani | AAH_USER: what asterisk issues? |
| 20:12.36 | [av]bani | Riedel: what are you using for incoming lines? |
| 20:12.57 | Riedel | [av]bani: solely SIP calls |
| 20:12.57 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:12.58 | AAH_USER | Some issue relating to the sccp driver, I understand the 7970s don't have a sip firmware like the 7960s |
| 20:13.16 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:13.25 | [av]bani | AAH_USER: yeah, they dont. but i have a 7970 on my desk and it seems to work :)) |
| 20:14.08 | Riedel | [av]bani: if I sip debug/go very verbose I don't see anything related to the number I called as well. But I may have missed it / it may not show |
| 20:14.13 | [av]bani | the 7970 has other issues, talking to asterisk isnot one of them |
| 20:14.31 | [av]bani | Riedel: how are calls coming into your asterisk? pri? bri? pstn? |
| 20:14.39 | Riedel | pstn |
| 20:14.48 | [av]bani | you cant do it then |
| 20:15.19 | [av]bani | you want DNIS but thats only available from pri/bri or voip trunks |
| 20:15.39 | AAH_USER | 7970s are good phones (as all cisco phones I have tried) nothing is without its "hickups" |
| 20:15.47 | Riedel | [av]bani: I have a VOIP trunk |
| 20:15.53 | AAH_USER | I use a 12sp at the moment with * |
| 20:16.06 | Riedel | DNIS remains unknown |
| 20:16.13 | [av]bani | Riedel: DNID is what * calls it |
| 20:16.25 | [av]bani | Riedel: but your voip supplier has to send it |
| 20:16.47 | Riedel | [av]bani: it should show up in the headers on sip debug / extremely verbose mode, right ? |
| 20:16.50 | [av]bani | Riedel: some will send it as the actual extension dialed |
| 20:16.57 | Riedel | aka, I should stop wasting my time investigating |
| 20:17.00 | AAH_USER | What kind of voip trunk? Sip or iax |
| 20:17.07 | Riedel | SIP |
| 20:17.27 | [av]bani | Riedel: some voip providers will send it to you as the actual extension dialed into your pbx |
| 20:17.27 | AAH_USER | Most iax send dnid, most sip don't (my experience) |
| 20:17.39 | [av]bani | Riedel: but you have to ask them to |
| 20:18.02 | AAH_USER | You can cheat on some sip though, if it only has one did |
| 20:18.03 | [av]bani | Riedel: then it would just be ${EXTEN} |
| 20:18.07 | Riedel | [av]bani: They said I could retrieve it from the headers. But I reckon they ust don't send it. |
| 20:18.13 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 20:18.21 | [av]bani | Riedel: it would be in the sip debug if they did send it yes |
| 20:18.23 | AAH_USER | Does the dnid show in your cli? |
| 20:18.37 | *** join/#asterisk lalito (n=erg@201.154.26.12) |
| 20:18.40 | digime | wanted: * tech, must live in san diego. http://sandiego.craigslist.org/eng/137738499.html |
| 20:18.47 | AAH_USER | If not, they aren't sending it |
| 20:18.52 | [av]bani | Riedel: ask them what header |
| 20:19.02 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 20:19.03 | Riedel | [av]bani : ${EXTEN} is either s, or what I specified in the registerline |
| 20:19.14 | [av]bani | Riedel: if they dont say, then find a new provider :) |
| 20:19.57 | [av]bani | Riedel: if they are sending it in a special header, it would show up in sip debug |
| 20:20.05 | Riedel | [av]bani: Is IAX configuration very different from SIP ? |
| 20:20.10 | [av]bani | Riedel: no |
| 20:20.15 | Hmmhesays | weee try to install zaptel with no kernel headers |
| 20:20.19 | asterisk99 | anyone try the new AEL language? |
| 20:20.20 | iCEBrkr | (03:19)Ignoring NOTICES from *!*@* |
| 20:20.29 | [av]bani | iCEBrkr: :) |
| 20:20.30 | *** join/#asterisk cthompson (n=ct@66.98.136.16) |
| 20:20.48 | iCEBrkr | Hi, my name is Lilo, I got nothing better to do than pat myself on the back to make myself feel better.. K/THX |
| 20:20.50 | Riedel | [av]bani: Lemme see if iax works better for me, then |
| 20:21.02 | Riedel | Thanks for answering my Q's. |
| 20:21.44 | asterisk99 | What is this??? Daily Affirmation with Stuart Smalley?? ... and gosh-darn it, I'm good enough!!!! |
| 20:21.50 | cthompson | * config is, um, daunting the first time |
| 20:22.15 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-26-222.dsl.pltn13.pacbell.net) |
| 20:22.23 | cthompson | chan_iax2.c:7398 socket_read: Registration of '751725' rejected: 'Registration Refused' from: '192.246.69.186' |
| 20:22.26 | iCEBrkr | cthompson: Fear not, grasshopper |
| 20:22.27 | cthompson | heh |
| 20:22.56 | justinu | 30,000 users on a deprecated protocol |
| 20:23.00 | justinu | bfd |
| 20:23.30 | [av]bani | o_O |
| 20:23.53 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 20:24.00 | cthompson | is iaxtel dead? |
| 20:24.06 | cthompson | the website isn't responding |
| 20:24.58 | trelane_ | Riedel, should I know you? |
| 20:25.07 | dijit0 | does anyone else have nufone and when someone calls their toll free number with *67, it shows up as Unknown?? |
| 20:25.37 | dijit0 | the call records show the real number though |
| 20:27.22 | rayvd | nufone = sip? |
| 20:27.41 | dijit0 | iax or sip |
| 20:27.56 | dijit0 | but im connecting directly TO them with xlite at the moment |
| 20:28.24 | dijit0 | but when i use iax.cc and receive a call on my 888 numver, it shows the real number.. so i dont understand why nufone doesnt |
| 20:28.34 | cthompson | is the Oreilly * book good? |
| 20:28.36 | dijit0 | if they *67d me that is |
| 20:28.41 | cthompson | I may have to pick that up on the way home |
| 20:28.42 | rayvd | ah, if it's sip you could take a look at the sip headers |
| 20:28.49 | rayvd | see if the From: address shows the correct number |
| 20:28.51 | trelane_ | cthompson, extremely useful |
| 20:29.05 | rayvd | or if using rpid, take a look at the rpid line... asterisk doesn't parse some rpid lines correctly |
| 20:29.05 | cthompson | there is SO MUCH in this conf |
| 20:29.26 | cthompson | trelane, your nick looks familiar. did you use to hang on #linux on linuxpro? |
| 20:29.29 | cthompson | er |
| 20:29.30 | rayvd | (ie some sip devices put a party=called entry first) |
| 20:29.32 | cthompson | linuxnet |
| 20:30.37 | trelane_ | cthompson, no, I'm well known around IRC though |
| 20:30.47 | cthompson | trelane_: hmm |
| 20:30.53 | cthompson | you on irc.perl.org? |
| 20:30.59 | trelane_ | no |
| 20:31.02 | cthompson | odd |
| 20:31.06 | trelane_ | efnet, freenode,oftc mainly |
| 20:31.16 | trelane_ | occasionally undernet |
| 20:31.44 | cthompson | I'm just hallucinating in my older years |
| 20:32.02 | trelane_ | it happens |
| 20:32.04 | RaYmAn-Bx | trelane_: and rarely roxnet? :P (or are you a different trelane..) |
| 20:33.04 | trelane_ | possibly a different trelane |
| 20:33.07 | trelane_ | I've been on IRC since 94 |
| 20:33.12 | cthompson | trelane_: you've got a doppelganger |
| 20:33.12 | cthompson | 15:28 [perl] -!- Trelane [~mark@gan.2shortplanks.com] |
| 20:33.12 | cthompson | 15:28 [perl] -!- was : Mark Fowler |
| 20:33.12 | cthompson | 15:28 [perl] -!- server : london.irc.perl.org [Tue Feb 28 05:53:04 2006] |
| 20:33.21 | cthompson | I knew I knew that nick :) |
| 20:33.23 | trelane_ | yup |
| 20:33.40 | cthompson | so I'm not crazy, no matter what the voices in my head say |
| 20:33.42 | trelane_ | apparently my little ray of sunshine has been stolen |
| 20:33.44 | [TK]D-Fender | Trelane is a common Trekkie nick :) |
| 20:33.54 | trelane_ | [TK]D-Fender, which episode? |
| 20:33.56 | cthompson | ah, is it? I didn't catch the reference |
| 20:34.18 | cthompson | forty quatloos on my asterisk conf never being able to connect to FWD |
| 20:34.21 | [TK]D-Fender | trelane_ : Can't remember the title....he's supposedly a "Q". |
| 20:34.55 | trelane_ | [TK]D-Fender, common error, he dates back to the original series season one "The Squire of Gothos" he was not in fact a Q but was in fact very similar |
| 20:36.36 | *** join/#asterisk [Outcast] (n=bill@222-152-110-218.jetstream.xtra.co.nz) |
| 20:36.37 | RaYmAn-Bx | trelane_: it was actual your long irc visiting that makes me think you're the trelane...#unrealircd, if you've ever been there. |
| 20:36.52 | trelane_ | of course |
| 20:36.53 | [Outcast] | can anyone tell what this error message means? > Insufficient information for SDP (m = '', c = '') |
| 20:36.57 | trelane_ | I know Stskeeps quite well |
| 20:37.03 | RaYmAn-Bx | ah, it is you then :) |
| 20:37.09 | cthompson | so, if I have a register => in iax.conf to FWD, and I get registration refused, how do I start troubleshooting that? |
| 20:37.16 | trelane_ | right, I'm the real trelane, that one is an imposter! |
| 20:37.33 | [TK]D-Fender | trelane_ : No, what I'm referring to comes from the books which are taken as "official". |
| 20:37.49 | cthompson | register => 751725:password@iax.fwdnet.net |
| 20:37.53 | trelane_ | [TK]D-Fender, the Q-Squared series is publised but not Canonical |
| 20:38.33 | justinu | outcast: it means you have a BROKEN sip client. |
| 20:38.49 | cthompson | chan_iax2.c:7398 socket_read: Registration of '751725' rejected: 'Registration Refused' from: '192.246.69.186' |
| 20:38.54 | FuriousGeorge | i get 30 of these at a time sometimes: -- Attempting native bridge of Zap/3-1 and Zap/7-1 |
| 20:39.11 | cthompson | I see long long hours of .conf madness ahead |
| 20:39.36 | FuriousGeorge | cthompson: sounds like you just gotta get your authentication right |
| 20:39.38 | [Outcast] | justinu: explain broken |
| 20:39.47 | Katty | [Outcast]: not working. |
| 20:39.56 | Katty | [Outcast]: screwy. |
| 20:39.57 | cthompson | that's odd |
| 20:40.01 | Katty | [Outcast]: OUT OF MUFFINS> |
| 20:40.07 | cthompson | iax.fwdnet.net doesn't exist |
| 20:40.19 | [Outcast] | katty: why is broken? |
| 20:40.24 | justinu | [Outcast]: katty is right. |
| 20:40.31 | cthompson | ing: unknown host iax.fwdnet.com |
| 20:40.31 | justinu | it's broken because it's not sending SDP |
| 20:40.40 | cthompson | bah, sorry, putty paste |
| 20:40.42 | Katty | i hate when it stops sending SDP :< |
| 20:40.43 | Katty | i mean muffins :< |
| 20:40.54 | [Outcast] | i like muffins |
| 20:40.59 | justinu | maybe we should design a muffin descriptor protocol |
| 20:41.03 | cthompson | OK, I'm not bright |
| 20:41.10 | Katty | yay! |
| 20:41.16 | Katty | file: i have muffins again! |
| 20:41.17 | justinu | c=chocolate chips |
| 20:41.21 | justinu | m=bran |
| 20:41.32 | *** join/#asterisk jskcr (n=zaphod@26-pool1.ras14.floca.alerondial.net) |
| 20:41.41 | cthompson | changed from iax.fwd to iax2 |
| 20:41.43 | cthompson | woot |
| 20:41.53 | [TK]D-Fender | in CLI how can I see if there are agent call-backs registered? |
| 20:42.08 | Riedel | trelene :) |
| 20:42.31 | Riedel | [av]bani, still here ? |
| 20:42.33 | Riedel | :-) |
| 20:42.35 | [TK]D-Fender | agents.conf has been flushed, but I still see targeted "LOCALXXXX@context" as the agent in queue_log but nothing to support it... |
| 20:43.25 | cthompson | nope still rejected, just took longer |
| 20:44.06 | [av]bani | :< |
| 20:45.38 | AAH_USER | I am using iax2.fwdnet.net and it works great |
| 20:47.02 | bsdfreak | heh |
| 20:47.12 | AAH_USER | cthompson, I am using iax2.fwdnet.net and it works great |
| 20:47.31 | cthompson | AAH_USER: what's your register=> look like, minus password, of course |
| 20:47.49 | CePe | Any1 have a link to a list of Asterisk startup warnings and errors and solutions to them? |
| 20:48.03 | AAH_USER | cthompson, Just a sec |
| 20:48.27 | cthompson | register => 751725:password@iax2.fwdnet.net/751275 |
| 20:48.30 | [Outcast] | justinu, Katty: believe it or not that information you gave was very helpful. |
| 20:48.40 | cthompson | I've tried that with and without the /751275 |
| 20:48.42 | AAH_USER | fwdnumber:password@iax2.fwdnet.net |
| 20:48.48 | cthompson | actually, with it, I get no error |
| 20:48.49 | justinu | [Outcast]: i don't give out unhelpful info |
| 20:48.54 | justinu | not unless you're a deal dick |
| 20:49.01 | Katty | [Outcast]: i'm glad my muffiny outbursts came in handy. |
| 20:49.02 | justinu | which most people here aren't |
| 20:49.11 | justinu | s/deal/real |
| 20:49.18 | [Outcast] | justinu: cool thanks |
| 20:49.19 | AAH_USER | cthompson, try without the /751275 |
| 20:49.24 | cthompson | but iax2 oh |
| 20:49.26 | cthompson | hey |
| 20:49.29 | cthompson | I'm in |
| 20:49.38 | Katty | iDunno: :< |
| 20:49.42 | iDunno | people really shouldn't put "place order" buttons on things ;) |
| 20:49.44 | cthompson | er, maybe |
| 20:49.47 | [Outcast] | Katty: next astricon muffins are on me |
| 20:49.53 | iDunno | Katty: is shiny new toy - though :) |
| 20:49.57 | AAH_USER | cthompson, glad I could help (if I did) |
| 20:50.06 | Katty | iDunno: :> |
| 20:50.09 | AAH_USER | Maybe? |
| 20:50.11 | cthompson | no, I'm not |
| 20:50.15 | tainted- | is it possible to listen for DTMF during a call with asterisk? |
| 20:50.30 | iDunno | (a toshiba portƩgƩ R200 :) |
| 20:50.30 | AAH_USER | Pastebin your config |
| 20:50.40 | justinu | iDunno: did you buy anything cool? |
| 20:50.54 | justinu | ah |
| 20:50.55 | cthompson | I'm new to freenode, how do I pastebin here? |
| 20:50.57 | tainted- | like MeetMe() but with an IVR |
| 20:50.57 | iDunno | *grin* |
| 20:50.59 | justinu | i want a macbook pro |
| 20:51.05 | justinu | but I'm considering buying a used T42p |
| 20:51.24 | AAH_USER | Pastebin.com |
| 20:51.27 | iCEBrkr | tainted-: WTF are you talking about, chico? |
| 20:51.31 | iDunno | ahh - I have lost most respect for apple since they're dropping powerpc, which was the only thing that I liked about apple to start off with ;) |
| 20:52.03 | tainted- | iCEBrkr say a person is in a call w/ another person.. is it possible to still read DTMF from the call? |
| 20:52.09 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 20:52.15 | iCEBrkr | tainted-: Not really. |
| 20:52.20 | justinu | it is |
| 20:52.21 | tainted- | why not? |
| 20:52.21 | iDunno | justinu: T42p? IBM Thinkpad? they're apparently quite nice |
| 20:52.33 | justinu | that's how you can enable the recording on deman |
| 20:52.43 | justinu | iDunno: yep, it's got a 15" display at 1600x1200 |
| 20:52.50 | justinu | 1.8ghz + 80gig disk |
| 20:52.52 | iDunno | hmm - shiny :) |
| 20:52.52 | iCEBrkr | justinu: Whooptieshit, it listens for * and # |
| 20:52.54 | justinu | builtin wifi |
| 20:53.03 | tainted- | say i want to press 9 to end the call and place nother call |
| 20:53.03 | iDunno | what's the battery life like? :) |
| 20:53.15 | tainted- | iCEBrkr is that possible? instead of hanging up |
| 20:53.23 | justinu | tainted: that's done in the options to Dial |
| 20:53.29 | justinu | tainted-: show application dial |
| 20:53.31 | iCEBrkr | tainted-: I'm going to say no. |
| 20:53.37 | iCEBrkr | justinu: You're smoke'n |
| 20:53.58 | iCEBrkr | justinu: you can slam hH on there so you can use * to hang up |
| 20:54.15 | iCEBrkr | justinu: you can slam wW on there so you can hit *1 to record. |
| 20:54.19 | clive- | tainted, why dont you use the "h" context |
| 20:54.23 | justinu | <PROTECTED> |
| 20:54.23 | justinu | <PROTECTED> |
| 20:54.23 | justinu | <PROTECTED> |
| 20:54.23 | justinu | <PROTECTED> |
| 20:54.42 | justinu | iCEBrkr: so how does that work then, if asterisk can't decode DTMF in a call? |
| 20:54.54 | iCEBrkr | justinu: You're not hearing him correctly.. |
| 20:54.59 | iCEBrkr | justinu: Sure, Asterisk can do DTMF in a call. |
| 20:55.11 | tainted- | but i want to place another call or execute a command on DTMF during a call |
| 20:55.12 | iCEBrkr | justinu: But I'm pretty sure he wants to do more than what's listed in features. |
| 20:55.25 | *** join/#asterisk LostFrog (n=reallyno@dsl093-100-093.wdc2.dsl.speakeasy.net) |
| 20:55.35 | justinu | <PROTECTED> |
| 20:55.35 | justinu | <PROTECTED> |
| 20:55.38 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 20:55.38 | iCEBrkr | tainted-: Go read Dial() on the wiki and see if it works the way you want it to. |
| 20:55.54 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 20:56.23 | tainted- | the dial command has features to enable hangup, recording, and transfer |
| 20:56.32 | iCEBrkr | wow, Features.conf is actually used and works now? |
| 20:56.32 | tainted- | but say i want to execute sendmail or something on '9' |
| 20:56.45 | iCEBrkr | justinu: See what I mean? |
| 20:56.49 | justinu | hmm |
| 20:56.59 | cthompson | sorry, back, phone had rang |
| 20:57.24 | justinu | i didn't see sendmail |
| 20:57.38 | cthompson | AAH_USER: I cleansed and copied to my website before you mentioned pastebin.com |
| 20:57.39 | iCEBrkr | hahahaahahaha |
| 20:57.44 | tainted- | or just any kind of external app |
| 20:57.45 | cthompson | http://cthompson.com/asterisk/ |
| 20:57.59 | tainted- | i've tried all sorts of stuff with Dial() and agi.. but to no avail |
| 20:58.02 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 20:58.03 | cthompson | I'm thinking I need to get rid of the sample confs and start over :) |
| 20:58.19 | fifer | <PROTECTED> |
| 20:58.28 | fifer | <PROTECTED> |
| 20:59.16 | tainted- | if it's not currently available, is it possible to add it as a feature? |
| 20:59.32 | justinu | iDunno: battery life is not as good as your portege |
| 20:59.43 | justinu | iDunno: 15" screen with that rez sucks up the juice |
| 20:59.50 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 20:59.52 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 21:00.00 | iCEBrkr | tainted-: Well, first of all why do you need such an option, it's possible you're going about things the wrong way |
| 21:00.30 | FuriousGeorge | i dunno what happened but suddenly eyebeams all stopped working in this one location |
| 21:00.37 | Tagor | I'm trying to setup music on hold. However I just don't hear anything if I call the number. Has anybody an idea how to fix this? |
| 21:00.39 | *** join/#asterisk gbodemantv (n=gbodeman@mail.televerde.com) |
| 21:00.42 | iDunno | justinu: uh huh :/ |
| 21:00.48 | fifer | anyone using ARI? |
| 21:00.53 | saftsack | is it possible to connect more than one TE card to an isdn "anlagenanschluss"? |
| 21:01.07 | iCEBrkr | Tagor: dial the extension and then put the call on hold. |
| 21:01.23 | justinu | iDunno: 3 hours w/ the extended battery, i guess |
| 21:01.43 | Tagor | Ehu, that's not what I want, iCEBrkr. I want the music on hold when someone calls my number |
| 21:02.21 | iCEBrkr | Tagor: Then use MusicOnHold() in your dialplan for that extension |
| 21:02.25 | iDunno | justinu: that's not bad, that's more than my current laptop has - which has 2 hours under 'normal' use (or 20 mins if you cane it with a 3D game ;) |
| 21:02.38 | Tagor | I have, but it doesn't work, iCEBrkr |
| 21:02.52 | iCEBrkr | Tagor: |
| 21:02.53 | iCEBrkr | exten => 6000,1,Answer |
| 21:02.53 | iCEBrkr | <PROTECTED> |
| 21:02.55 | iCEBrkr | like that? |
| 21:03.00 | justinu | iDunno: i'm currently using a powerbook g4 667 which has crap life also |
| 21:03.06 | Tagor | When I do ps -aux it also doesn't show me mpg123 |
| 21:03.09 | Tagor | Yes, iCEBrkr |
| 21:03.53 | iCEBrkr | Tagor: you got the correct version of mpg123 and it's in /usr/local/bin? |
| 21:04.06 | Tagor | Yes, iCEBrkr |
| 21:04.19 | Tagor | And I have one mp3 file in the folder /var/lib/asterisk/mohmp3 |
| 21:04.47 | iCEBrkr | Tagor: and you have.... ok |
| 21:04.51 | fourcheeze | In an IVR I'm using Dial() with with m option to play MOH. However the music fades in and out. Normal MOH works fine. |
| 21:04.52 | iCEBrkr | I was gonna say. /var/lib/asterisk/mohmp3/ |
| 21:05.12 | Tagor | Is it possible that it's not a correct mp3 file |
| 21:05.17 | Tagor | In example the bitrate is not ok |
| 21:05.17 | fourcheeze | also r option on Dial to ring doesn't work either |
| 21:05.25 | iCEBrkr | fourcheeze: thats the new Macrovision copy protection they put on MP3s :P |
| 21:05.27 | Tagor | Are there example mp3s available? |
| 21:05.36 | fourcheeze | iCEBrkr: :P |
| 21:05.48 | iCEBrkr | Tagor: the fpm- examples are fine. |
| 21:06.07 | cpm | I don't know what I am doing. |
| 21:06.13 | Tagor | Ok, I will try one, iCEBrkr |
| 21:06.26 | iCEBrkr | fourcheeze: Just like when you watch a DVD through your VCR's RCA jacks.. The video fades in and out. MP3s with it will do the same thing. LOL |
| 21:06.39 | fourcheeze | it certainly seems like that |
| 21:06.53 | iCEBrkr | fourcheeze: If you're testing with your cellphone, MoH sucks ass. |
| 21:07.07 | fourcheeze | no, it's an ordinary phone |
| 21:07.11 | iCEBrkr | ok |
| 21:07.16 | fourcheeze | MOH is fine too |
| 21:07.20 | Tagor | Ok, I still don't hear anything, iCEBrkr |
| 21:07.24 | fourcheeze | only fading when adding to Dial |
| 21:08.05 | justinu | fading? as in low and high volume? |
| 21:08.10 | fourcheeze | yes |
| 21:08.20 | fourcheeze | starts fairly quiet and goes to 0 |
| 21:08.20 | justinu | i can't see how that's possible |
| 21:08.27 | fourcheeze | then comes back up again |
| 21:08.32 | justinu | i could understand it if completely cut out |
| 21:08.32 | fourcheeze | it's certainly odd |
| 21:08.39 | justinu | but not fading away |
| 21:08.53 | tainted- | iCEBrkr basically i just need the functionality of someone in a MeetMe conference to be able to Dial() another person into the MeetMe() |
| 21:08.55 | iCEBrkr | I've heard MoH get really staticy when the music hits a quiet spot |
| 21:09.12 | iCEBrkr | tainted-: That doesn't tell me much, really |
| 21:09.19 | fourcheeze | it's fairly quiet so hard to tell if it cuts out or fades out |
| 21:09.22 | cthompson | is there a bare minimum * config? |
| 21:09.25 | iCEBrkr | tainted-: you want 3 way dialing or something? |
| 21:09.45 | tainted- | so u have two people in a MeetMe(), one dials out and the person who answers the phone is automatically in the MeetMe() |
| 21:09.55 | fourcheeze | justinu: if it cut in and out how would you explain that? |
| 21:09.55 | justinu | fourcheeze: when you're playing this MoH, have you answered the call? |
| 21:10.01 | fourcheeze | yes |
| 21:10.10 | justinu | fourcheeze: is VAD off on the phone? |
| 21:10.14 | [TK]D-Fender | cthompson : nope, they give a buch of bloated nearly worhtly psychotic samples that don't add up to anything and you're better off starting from scratch :) |
| 21:10.16 | iCEBrkr | tainted-: I don't think you'll be able to start conferencing people in on a call, that's what MeetMe is for |
| 21:10.24 | fourcheeze | justinu: it's a dial-in via pstn |
| 21:10.27 | cthompson | that's what I figured |
| 21:10.33 | justinu | fourcheeze: zap channel? |
| 21:10.41 | fourcheeze | no |
| 21:10.55 | fourcheeze | pstn via SIP provider |
| 21:11.09 | tainted- | iCEBrkr no, i mean two people (A & B) already in a MeetMe conference. A calls C from within the conference using DTMF and C is automatically joined to the conference when C picks up the phone |
| 21:11.15 | justinu | fourcheeze: g711? |
| 21:11.24 | fourcheeze | justinu: yep |
| 21:11.32 | iCEBrkr | tainted-: Sounds like MeetMe would have to be modified to do that. |
| 21:11.41 | fourcheeze | justinu: also the the r option to Dial doesn't work either |
| 21:11.49 | cthompson | so |
| 21:11.57 | justinu | fourcheeze: got ztdummy loaded? |
| 21:11.58 | cthompson | if I create JUST the stuff listed at http://www.freeworlddialup.com/help/?p=knowledgebase&c=18&a=76 |
| 21:12.02 | fourcheeze | justinu: yep |
| 21:12.05 | cthompson | is that enough of a conf to load *? |
| 21:12.09 | justinu | fourcheeze: not sure what's going on them |
| 21:12.14 | fourcheeze | really odd |
| 21:12.35 | tainted- | iCEBrkr is that something you could do? |
| 21:12.52 | fifer | Anyone using ARI standalone? (without AMP)? |
| 21:13.00 | *** join/#asterisk knutvi (n=knutvi@vindheim.ta-a.ntnu.no) |
| 21:13.01 | tainted- | iCEBrkr i would be willing to pay for play for a feature like that |
| 21:13.11 | fourcheeze | justinu: am I supposed to do an Answer() after someone dials an extension? |
| 21:13.28 | justinu | fourcheeze: not if you've answered the call already. |
| 21:13.30 | iCEBrkr | tainted-: I haven't looked into MeetMe's code.. I dunno |
| 21:13.51 | fourcheeze | justinu: yeah I do that almost as soon as the call comes in |
| 21:14.01 | iCEBrkr | tainted-: It seems as if ripping some code out of Page() and stuffing it into MeetMe() might work |
| 21:14.05 | fourcheeze | it's a really simple IVR |
| 21:14.09 | fourcheeze | I can't beleive it's not working |
| 21:14.12 | [TK]D-Fender | cthompson : Somewhat.. there is plenty of other setups you'll need for your phones, etc. I may be available later to help you get a head start. |
| 21:14.13 | justinu | fourcheeze... your situation is pretty simple... you looked thru the logs, etc.? |
| 21:14.17 | justinu | full log |
| 21:14.30 | justinu | fourcheeze: try older versions of ast/zaptel, perhaps |
| 21:14.34 | justinu | maybe it's a bug |
| 21:14.41 | fourcheeze | it certainly seems like a bug |
| 21:14.53 | cthompson | [TK]D-Fender: right now my "phones" are, um, x-lite |
| 21:15.01 | [TK]D-Fender | cthompson : no problem there... |
| 21:15.03 | cthompson | until payday comes and I can afford my Sipura |
| 21:15.09 | fourcheeze | justinu: ringing would be fine too - it's just silence that's not an option |
| 21:15.28 | justinu | fourcheeze: yeah - the fact that ringing isn't working is odd... |
| 21:15.45 | cthompson | right now I'd like to see * connect to FWD over IAX2, and show something on console when I dial into my one FWD account from the wife's with xten |
| 21:16.06 | fourcheeze | justinu: when you do Wait() it seems you can't accept dtmf |
| 21:16.11 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 21:16.14 | fourcheeze | is there a way to wait and accept dtmf? |
| 21:16.16 | [TK]D-Fender | cthompson : What are you looking at buying? |
| 21:16.29 | justinu | fourcheeze: Background(silence/10) ? |
| 21:16.33 | fourcheeze | ahhh |
| 21:16.40 | fourcheeze | that's cheating ;-) |
| 21:16.46 | justinu | sometimes you gotta cheat |
| 21:16.46 | cthompson | [TK]D-Fender: Sipura SP-3000 sitting in between my POTS and our house wiring |
| 21:16.47 | cthompson | for now |
| 21:17.06 | trixter | wouldnt setting the timeout appropriately be better? |
| 21:17.19 | cthompson | though part of me wants to go PAP2 and saying to hell with POTS altogether |
| 21:17.24 | cthompson | er |
| 21:17.25 | cthompson | PSTN |
| 21:17.26 | [TK]D-Fender | cthompson : A good first choice for FXS / FXO. |
| 21:17.26 | cthompson | sorry |
| 21:17.40 | cthompson | my FXO right now is Time Warner |
| 21:17.47 | [TK]D-Fender | cthompson : No, ditching POTS altogether is a great plan :) |
| 21:17.51 | justinu | voip? |
| 21:17.53 | cthompson | no |
| 21:18.04 | cthompson | it's some freaky RF thing they piggy back on your cable |
| 21:18.15 | [av]bani | ugh |
| 21:18.24 | cthompson | my cable modem has a FXO type port on the back that my home wiring plugs into |
| 21:18.34 | [av]bani | [TK]D-Fender still recommending the spa-3000 pos |
| 21:18.42 | cthompson | pos? |
| 21:18.57 | cthompson | I was looking at the PAP2 |
| 21:19.05 | cthompson | and getting a two line cordless phone |
| 21:19.07 | [av]bani | pap2 has no fxo |
| 21:19.10 | cthompson | right |
| 21:19.24 | heison | does anyone here use gs & txfax (spandsp)? |
| 21:19.33 | cthompson | I could get a $15 fx100 until I can ditch FXO completely |
| 21:19.48 | [av]bani | fx100? |
| 21:19.52 | justinu | cthompson: FDM |
| 21:19.59 | cthompson | teliax and broadvoice both say they can portablize my number |
| 21:20.01 | trixter | looking at using chan_fax becuase that appears to be btter in terms of scalability than iaxmodem and more error tolerant than rx/txfax |
| 21:20.08 | cthompson | justinu: er, x100p |
| 21:20.15 | cthompson | fx100 is one of our internal codes here :) |
| 21:20.20 | justinu | i mean your freaky RF think is FDM |
| 21:20.23 | justinu | ~fdm |
| 21:20.28 | cthompson | oh |
| 21:20.31 | [TK]D-Fender | [av]bani : For those wanting to take in a line and have a FXS port, its worth it.... |
| 21:20.34 | cthompson | probably |
| 21:20.41 | justinu | jbot, fdm is frequency division multiplexing |
| 21:20.42 | jbot | okay, justinu |
| 21:20.42 | trixter | http://www.sofaswitch.org/chan_fax/ |
| 21:20.43 | [TK]D-Fender | x100p is pretty crappy.... |
| 21:20.51 | [av]bani | [TK]D-Fender: for testing sure, not for anything production level |
| 21:21.06 | cthompson | I probably don't need FXO at all |
| 21:21.10 | [TK]D-Fender | [av]bani : for HOME. |
| 21:21.26 | cthompson | any opinions on PAP2? |
| 21:21.48 | [av]bani | [TK]D-Fender: its barely acceptable for home, unless you like lots of echo |
| 21:22.14 | [TK]D-Fender | [av]bani : I did OK with... you're pretty jilted by the entire industry... you should seek medication... |
| 21:22.18 | justinu | the x100p's supposedly can be tuned ok |
| 21:22.41 | [TK]D-Fender | cthompson : if its unlocked the PAP2 is ok. Better just getting the SPA-2002 though as its pretty mcuh teh same price. |
| 21:22.54 | cthompson | of course, I should mention a wrinkle |
| 21:23.02 | cthompson | my asterisk is running inside vserver |
| 21:23.07 | [TK]D-Fender | however any PCI card in your server can lead to all sorts of other problems as well.... |
| 21:23.10 | cthompson | following the telephreak article |
| 21:23.30 | iCEBrkr | My X100P worked PERFECTLY |
| 21:23.39 | iCEBrkr | No echo. No problems what so ever. |
| 21:23.50 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 21:24.19 | [TK]D-Fender | iCEBrkr : Lucky you... now get to the back of the bus like the rest of the minority! ;) |
| 21:24.23 | iCEBrkr | LOL |
| 21:24.27 | Katty | goodness. |
| 21:24.37 | Katty | [TK]D-Fender: let me run him through the papershredder :> |
| 21:25.23 | iCEBrkr | Katty: I'm impervious to paper shredders. >:) |
| 21:25.33 | AAH_USER | cthompson, Vserver can cause issues with * |
| 21:25.40 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-190-62.red.bezeqint.net) |
| 21:25.41 | cthompson | AAH_USER: yes |
| 21:26.03 | cthompson | but with no hardware, I didn't think that was an issue |
| 21:26.33 | cthompson | ztdummy was sufficient |
| 21:26.42 | cthompson | I don't have ztdummy loaded |
| 21:26.47 | cthompson | heh |
| 21:27.53 | justinu | katty: maybe just cut him to ribbons with your claws |
| 21:27.57 | AAH_USER | I went from a vserver and all voip trunks (my phones had issues registering among other issues) to a wimpy 450mhz processor with 512 ram on a dedicated machine and it solved every one of my problems |
| 21:28.37 | [TK]D-Fender | ok, I'm off.... later all |
| 21:28.40 | cthompson | I may still do that |
| 21:28.49 | Katty | justinu: oh,there's an idea. |
| 21:28.51 | cthompson | vserver is what allowed me to play, as I don't have a spare machine |
| 21:29.03 | cthompson | a friend of mine has an old p2 |
| 21:29.03 | cthompson | [fromiaxfwd] |
| 21:29.03 | cthompson | exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r) |
| 21:29.03 | cthompson | exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX} |
| 21:29.08 | AAH_USER | I know the feeling |
| 21:29.10 | cthompson | damn putty |
| 21:29.13 | cthompson | [fromiaxfwd] |
| 21:29.13 | cthompson | exten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r) |
| 21:29.13 | cthompson | exten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX} |
| 21:29.14 | justinu | more than a feeling! |
| 21:29.16 | cthompson | gah |
| 21:29.29 | justinu | cthompson: don't blame putty for luser error :P |
| 21:29.41 | cthompson | I need to turn off right-click paste |
| 21:29.48 | cthompson | and autocopy |
| 21:29.58 | tainted- | cthompson could u turn off right-click paste? k thx |
| 21:30.05 | cthompson | tainted-: will do |
| 21:30.19 | tainted- | need you to go ahead and come in on saturday too |
| 21:30.40 | cthompson | I don't think I'm gonna go anymore |
| 21:30.41 | justinu | and i'm going to have to ask you to go ahead and uhh, come in on sunday also |
| 21:31.02 | *** join/#asterisk _DAW (n=_DAW@adsl-6-66-81.msy.bellsouth.net) |
| 21:31.04 | justinu | i have PEOPLE skills! |
| 21:31.24 | tainted- | hey so i heard you've been missing work lately |
| 21:31.36 | justinu | you know I have 8 bosses? |
| 21:32.13 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 21:34.59 | _MartinCabrera_ | Non technical question: May i have here some paid asterisk support? |
| 21:35.20 | AAH_USER | Sure, just ask |
| 21:35.42 | iCEBrkr | _MartinCabrera_: Please direct payments to my paypal account, K/THX |
| 21:35.48 | iCEBrkr | :P |
| 21:35.49 | fourcheeze | justinu: to have 2 bosses could be thought of as careless - to have 8 is surely reckless |
| 21:36.24 | *** join/#asterisk crich1999 (n=crich@port-212-202-198-154.dynamic.qsc.de) |
| 21:36.29 | justinu | _MartinCabrera_: people offer paid support here, but it's not sanctioned by digium |
| 21:36.38 | justinu | i offer paid support |
| 21:36.53 | *** part/#asterisk jstr5707 (n=jstrebel@70.100.62.81.cust.bluewin.ch) |
| 21:37.21 | AAH_USER | I also offer paid support (pay when you are happy with my solution) |
| 21:37.41 | cthompson | I will take your money but not provide any useful answers |
| 21:37.49 | denon | sounds like efnet |
| 21:37.54 | cthompson | keep me in mind when you need a tax write off |
| 21:37.59 | AAH_USER | Thank you cthompson |
| 21:38.06 | justinu | heh |
| 21:41.53 | Hmmhesays | wow freepbx 2.0 beta is damn nice |
| 21:42.16 | justinu | was dat? |
| 21:42.49 | Hmmhesays | the lastest incarnation of amp |
| 21:42.51 | *** join/#asterisk sch19 (n=sch19@adsl-11-74-66.mia.bellsouth.net) |
| 21:43.23 | jbalcomb | woot! almost done moving all the fax machines to the analog Key-BX. |
| 21:44.41 | [av]bani | :o |
| 21:45.05 | *** join/#asterisk pixolex (n=chatzill@87-196-158-168.net.novis.pt) |
| 21:45.24 | justinu | any of you guys know how someone in colombia can pay me for consulting services? |
| 21:45.31 | justinu | they say they can't use paypal |
| 21:45.35 | _Sam-- | wire |
| 21:45.35 | LostFrog | I didn't like AAH, should I even bother trying hat, Hmmhesays? |
| 21:45.41 | justinu | wire, like western union? |
| 21:45.50 | _Sam-- | nah like go to the bank and have it wired into your account |
| 21:45.52 | LostFrog | Send packets of cocaine? |
| 21:45.53 | justinu | fuck dat |
| 21:46.12 | _Sam-- | short of paypal there is no other way |
| 21:46.14 | justinu | i'd work for coke, that shit sells |
| 21:46.16 | _Sam-- | unless you want to charge his credit card |
| 21:46.21 | justinu | i can't do that |
| 21:46.21 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:46.29 | [av]bani | colombia? |
| 21:46.31 | [av]bani | hah |
| 21:46.32 | _Sam-- | we have customers wire us money all the time |
| 21:46.38 | _Sam-- | its pretty painless actually |
| 21:46.40 | Hmmhesays | LostFrog: its very prety |
| 21:46.49 | [av]bani | western union is the only other way, short of bank transfer |
| 21:47.15 | justinu | can he wire me money to my local western union? |
| 21:47.20 | [av]bani | yes |
| 21:47.31 | justinu | that might work |
| 21:47.39 | _Sam-- | it only works if its convenient for him down there |
| 21:47.51 | _Sam-- | might be easier for him to hit a bank, than a western union in bogota |
| 21:48.09 | [av]bani | http://www.westernunion.com/info/homePage.asp?country=CO&origination=global |
| 21:49.42 | _Sam-- | the global economy, in reality, isnt so easy to do globally. |
| 21:49.59 | _Sam-- | payments across borders is still a hassle for our biz. |
| 21:50.02 | *** join/#asterisk casterman (n=casterma@kasadsyed.net8.nerim.net) |
| 21:50.18 | AAH_USER | I like paypal, provided they can get a paypal account |
| 21:50.45 | _Sam-- | paypal, for internet merchants, isnt a real solution...because they dont provide any address or identity verification if you dont live in the USA or UK |
| 21:50.55 | _Sam-- | ok, maybe canada and australia are verifiable now too |
| 21:51.00 | nestar | move |
| 21:51.01 | nestar | :) |
| 21:51.25 | _Sam-- | im in the US...it is difficult for US based merchants to receive international payments with any degree of identity verification. |
| 21:51.47 | AAH_USER | But if they give me money, I could care less who they are. |
| 21:52.06 | justinu | no paypal for colombia |
| 21:52.09 | _Sam-- | you would care if the funds are transferred to you fraudently, and then taken back after you ship 1000s of dollars of goods |
| 21:52.10 | justinu | i don't care who they are either |
| 21:53.07 | justinu | is there a known issue with asterisk 1.2.4 dropping sip regs? |
| 21:53.29 | justinu | seems a bunch of people are having that issue lately |
| 21:53.33 | AAH_USER | I haven't had a problem, been running it 2 weeks |
| 21:54.12 | AAH_USER | Iax registration on the other hand get lost, often requires a total restart to get it working again |
| 21:56.04 | GerbilNut | there seems to be an issue with 1.2.4 not announcing queue position |
| 21:56.52 | AAH_USER | As well as some other audio-related issues |
| 21:56.55 | jbalcomb | justinu: travelers checks, western union, or AAA might be able to help with those international payments |
| 21:57.06 | justinu | jbalcomb: thanks for those suggestions |
| 21:57.54 | jbalcomb | AAH_USER if you take illegal money you are breaking the law |
| 21:58.14 | iCEBrkr | o/~ Break'n the law, break'n the law! o/~ |
| 21:58.23 | jbalcomb | tard |
| 21:58.26 | AAH_USER | I know, I was being sarcastic |
| 21:58.27 | iCEBrkr | :P |
| 21:58.46 | jbalcomb | I know, I have no sense of sarcasm |
| 21:58.55 | iCEBrkr | jbalcomb: Would you have prefered o/~ wash'n the dog! wash'n the dog!! o/~ ???? |
| 21:59.03 | AAH_USER | I realize that |
| 21:59.09 | *** join/#asterisk goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 21:59.17 | jbalcomb | how do you do that slap with a fish thing? cause icebrkr needs it |
| 21:59.28 | robin_sz | oh great! .. the latest version of the GXP2000 firmware is out to finally fix the GUI problems |
| 21:59.29 | iCEBrkr | jbalcomb: /me does whatever |
| 21:59.49 | jbalcomb | robin_sz: a new new one or the one released last week? |
| 21:59.57 | justinu | <PROTECTED> |
| 22:00.00 | robin_sz | jbalcomb: errm . |
| 22:00.04 | justinu | it's some kinda mirc macro |
| 22:00.15 | justinu | ouch |
| 22:00.17 | robin_sz | jbalcomb: just keep working on the sarcasm thing |
| 22:00.20 | justinu | flimsy plastic |
| 22:00.24 | iCEBrkr | haha |
| 22:00.35 | iCEBrkr | justinu: I wouldn't want to waste a good Polycom or Cisco :) |
| 22:00.36 | justinu | might explore into dangerous shards of flimsy plastic |
| 22:00.41 | justinu | s/explore/explode |
| 22:00.55 | justinu | a polycom/cisco is more of a bludgeon |
| 22:01.00 | iCEBrkr | BT101's are expendable :) |
| 22:01.09 | robin_sz | I really wish my GXP2000 was a lot thinner near the fron th of the phone ... that would make it much more usefu; |
| 22:01.23 | jbalcomb | robin_sz hahaha! |
| 22:01.37 | justinu | ok, serious question... my turn for help |
| 22:01.40 | De_Mon | hrm, just installed a X100P in a PowerEdge 350 but the wcfxo module fails to load saying "Failed to initilize DAA, givup up..." |
| 22:01.45 | robin_sz | ideally, if it tapered down to say, 5mm that would be very very good. |
| 22:01.50 | justinu | got a box w/ 4 (four, yes) X100Ps in it. |
| 22:02.02 | justinu | one of the cards is sharing an IRQ w/ eth0 |
| 22:02.06 | jbalcomb | justinu sounds like interupt problem |
| 22:02.10 | casterman | i have my asterisk that won't start Ouch ... error while writing audio data |
| 22:02.11 | _DAW | Kelly Robinson!! |
| 22:02.12 | robin_sz | then I could use it to wedge my door open with |
| 22:02.13 | justinu | we've tried everything to get it to sit on it's own IRQ |
| 22:02.25 | justinu | the question is: is it possible? |
| 22:02.26 | AAH_USER | justinu, and it works fine? |
| 22:02.35 | jbalcomb | justinu i think its up to the MoBo |
| 22:02.39 | justinu | no, we haven't tested it yet, we're aware that it's going to be problematic. |
| 22:02.42 | AAH_USER | I didn't think so |
| 22:02.49 | justinu | the eth0 is onboard. |
| 22:02.49 | jbalcomb | justinu it might not be really |
| 22:03.08 | jbalcomb | justinu yeah, check in the bios for an option to assign IRQs |
| 22:03.19 | jbalcomb | iCEBrkr goodbye good buddy |
| 22:03.22 | justinu | apparently, we've tried everything in bios |
| 22:03.56 | jbalcomb | justinu i wonder if you can specify the IRQ for the cards when it loads the module? |
| 22:04.10 | jbalcomb | justinu I used to do that for firewall/router PCs |
| 22:04.12 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:04.19 | justinu | i'll search the wiki |
| 22:04.24 | jbalcomb | [TK]D-Fender has arrived yall!! |
| 22:04.27 | justinu | if you guys think/know of anything, please msg me. |
| 22:04.53 | *** join/#asterisk Homer99 (n=homer@67.128.26.42) |
| 22:04.59 | jbalcomb | [TK]D-Fender i be submitten' my PO for 4 SNOM 360 Business phones now |
| 22:05.04 | Hmmhesays | "I can't find the software on this page", "me: don't click on the cms link","them: I clicked the cms link" |
| 22:05.20 | jbalcomb | [av]bani whats your phone page address? |
| 22:05.44 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool138-231.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:05.48 | goral | Hmmhesays i almost got it all worked out thanks for the push.. |
| 22:06.45 | Chingetas | Does anyone know something about DTMF using sip, I got a lot of problems with it... |
| 22:06.52 | Hmmhesays | yup |
| 22:06.59 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:07.10 | websae | is 5 REN enough to ring all the phones in my house????? |
| 22:07.17 | [TK]D-Fender | jbalcomb : neato. So far its looking like they are a losing deal more and more... the price just went up.... |
| 22:07.29 | AAH_USER | How big of a house? |
| 22:07.40 | websae | has about 10 phones |
| 22:07.45 | justinu | most new phones are REN 0.1 to 0.5 |
| 22:07.45 | Chingetas | anyone?? |
| 22:07.48 | justinu | so probably. |
| 22:07.48 | AAH_USER | Should work |
| 22:07.49 | websae | 4 cordless.....6 on the walls |
| 22:08.06 | jbalcomb | [TK]D-Fender the SNOM or GS? |
| 22:08.08 | AAH_USER | I have about the same, works fine |
| 22:08.13 | [TK]D-Fender | jbalcomb : Snom |
| 22:08.32 | [TK]D-Fender | the 360 just hit $225, which really makes Polycom more appealing |
| 22:08.39 | jbalcomb | [TK]D-Fender that sucks. if polycom had PoE builtin and a backlight I'd get them. |
| 22:08.52 | AAH_USER | Same here |
| 22:08.58 | Nivex | websae: Get a Xorcom Astribank and patch each phone to its own port ;) |
| 22:09.12 | [TK]D-Fender | jbalcomb : 301/501 just need an adapter cable (not TOO bad), and the 601 has it by default. |
| 22:09.40 | _DAW | Polycom is the best phone for the money even with the POE cable |
| 22:09.47 | jbalcomb | [TK]D-Fender hrmm.. 501 just needs a cable eh? I thought it said a $40 adapter |
| 22:11.49 | [TK]D-Fender | jbalcomb : it splices the PoE into it. not an injector. |
| 22:11.58 | Homer99 | Greetings good people! Problem -- Just installed asterisk server with sangoma 4 port fxo card. Can dial out, but when dialing in, it sounds like it rings once and something picks up to the caller (silence), but phone does not ring and nothing happens -- help? |
| 22:12.05 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:12.13 | *** part/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net) |
| 22:12.23 | [TK]D-Fender | homer : pastebin you CLI of an incoming call. |
| 22:12.25 | [TK]D-Fender | ~pb |
| 22:12.26 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:13.21 | *** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com) |
| 22:13.50 | Egonis | I just switched a number of phones to IAX, but they can't transfer calls anymore or have a message waiting indicator, what am I missing? It's probably something obvious |
| 22:14.58 | jbalcomb | [TK]D-Fender i see. well, maybe I'll get two of each then.. seems like a chore to keep track of all these custom cables though. |
| 22:15.15 | *** join/#asterisk Eitch (i=[U2FsdGV@unaffiliated/eitch) |
| 22:15.54 | SkramX | Anyone ever setup call forwarding using chan_sccp? Please help. |
| 22:16.46 | jbalcomb | [TK]D-Fender how many ring tones on a IP501 and Biz360? |
| 22:16.48 | goral | The only thing i have a problem with now is allowing users outside my local network access to asterisk |
| 22:17.14 | goral | won't register the user |
| 22:17.16 | SkramX | ? |
| 22:17.40 | jbalcomb | [TK]D-Fender also, does the IP501 have RJ headset connector? |
| 22:18.03 | goral | is there more settings i need to do for users outside my network access? |
| 22:18.23 | _DAW | goral - firewall? |
| 22:18.41 | SkramX | how does one create call forwarding in general for Asterisk? |
| 22:19.02 | goral | yeah, i guess but i have all ports forwarded to the asterisk box |
| 22:19.08 | Egonis | how do I allow a vm indicator on an iax phone? mailbox= has been set on all phones, but no indications |
| 22:19.11 | goral | 1-65535 |
| 22:19.15 | _DAW | jbalcomb - the 501 does have an RJ headset adapter.. |
| 22:19.36 | [av]bani | [TK]D-Fender: polycom isnt more appealing if you need backlight... |
| 22:19.50 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:20.11 | [av]bani | someone should make a backlight mod for ip501/601 |
| 22:20.19 | justinu | hey guys... i pulled all the X100 cards except one out of my box |
| 22:20.26 | justinu | and the single card always binds to the same interrupt as eth0 |
| 22:20.34 | justinu | any suggestions on how to stop that from happening? |
| 22:20.37 | goral | would this stop them? found under users sip.conf title defaultip=192.168.1.101 |
| 22:20.42 | [av]bani | justinu: sledgehammer |
| 22:20.59 | goral | [av]bani : i might be using it soon |
| 22:21.10 | [av]bani | goral: let me know the results |
| 22:21.25 | _DAW | goral - also remember that sip and nat are not usually friendly. |
| 22:21.37 | _DAW | especially when on both ends |
| 22:22.02 | Hmmhesays | then they are especially unfriendly |
| 22:22.03 | goral | _DAW : the user at the other end would be using a softphone |
| 22:22.47 | _DAW | goral : are they behind nat? |
| 22:22.55 | goral | i had him forward ports 5060 and 10000 to 20000 to thier computer that was using the softphone |
| 22:23.08 | goral | _DAW : yes |
| 22:23.18 | _DAW | are you behind nat? |
| 22:23.24 | goral | _DAW : yes |
| 22:26.14 | _DAW | Goral : here is an interesting article http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 22:26.45 | Goral | kool |
| 22:26.56 | Goral | _DAW : ty |
| 22:27.11 | _DAW | your welcome |
| 22:27.55 | *** join/#asterisk rene- (n=rene-@201.127.24.218) |
| 22:28.19 | rene- | hello, i wonder if i can have two trunks in zaptel that belong to the same group but belong to different contexts...? |
| 22:29.32 | [TK]D-Fender | Goral : You do not need for him to forward ports. He just needs to let his client know that he's behi NAT |
| 22:29.36 | wasim | rene-: but of course |
| 22:29.45 | rene- | cool |
| 22:29.57 | rene- | wasim; thx |
| 22:30.43 | [TK]D-Fender | jbalcomb : You can use as many different ringtones as you like. I use the Cisco "24" one personally. |
| 22:31.08 | Goral | [TK]D-Fender : okay now i guess i need to know how to do that with sjphone |
| 22:33.16 | Goral | I just figured it out |
| 22:33.45 | Goral | wow why the heck would they put things together like that? |
| 22:34.22 | Goral | sombody should kick me in the nuts for using this crap.. NAT... |
| 22:35.23 | Goral | Ty all, for the push now to take it to the next steps... more reading... |
| 22:35.38 | justinu | SIP was conceived before NAT was everywhere |
| 22:36.20 | [TK]D-Fender | ummmm.... Feels like the reverse.... |
| 22:36.42 | Goral | tell me my ass still hurts |
| 22:36.45 | *** join/#asterisk af_ (n=af@ip-172-156.sn1.eutelia.it) |
| 22:36.52 | De_Mon | your ass still hurts? |
| 22:37.08 | Goral | yes it does with that learning curve |
| 22:37.09 | _DAW | thats kind of scarry :/ |
| 22:37.38 | justinu | anyways, SIP works thru nat pretty well now... double nat is a different story tho |
| 22:38.15 | _DAW | when it comes to double nat you have to love IAX :) |
| 22:41.28 | Nivex | How many gnats could a NAT gnat NAT if a NAT gnat could NAT gnats? |
| 22:42.14 | Goral | thats a mouth full |
| 22:43.05 | SpooForBrains | About as many gnats as a NAT gnat could NAT is a NAT gnat could NAT gnats ... |
| 22:43.11 | SpooForBrains | s is/if |
| 22:43.13 | _DAW | mmmmmm.... gnats.. |
| 22:43.38 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-67.cybersurf.com) |
| 22:43.59 | Goral | _DAW : extensions how can i assign one statically to each sip account |
| 22:44.09 | *** join/#asterisk pixolex (n=chatzill@87-196-158-168.net.novis.pt) |
| 22:45.06 | _DAW | in your extensions.conf have something like exten => 1234,1,Dial(Sip/???,20,r) |
| 22:45.22 | _DAW | where 1234 is your extension |
| 22:47.14 | Goral | eg. exten => 1000,1,Dial(SIP/goral,20,t,r) |
| 22:47.50 | _DAW | at the end just ,tr not t,r |
| 22:48.06 | Goral | but i'm not logged in as goral i'm logged in as someone else and i though sice i was not logged in it would just end |
| 22:48.28 | Goral | but it sends the request as if goral was on |
| 22:48.42 | Goral | right back to the account i'm logged into |
| 22:49.01 | _DAW | can I pm? |
| 22:49.07 | Goral | by all means |
| 22:49.41 | Tagor | I'm trying to get music on hold working under 1.2.4. But I just don't hear anything |
| 22:49.45 | Tagor | Playback works fine |
| 22:49.54 | Tagor | But I want the sound file while a phone is dialed |
| 22:50.00 | Tagor | Anybody an idea how to get this working? |
| 22:50.38 | SkramX | what does your exten=> lines look like? |
| 22:50.52 | Tagor | exten => s,3,Answer() |
| 22:50.52 | Tagor | exten => s,4,SetMusicOnHold(default) |
| 22:50.52 | Tagor | exten => s,5,Dial(SIP/${ARG1},20,m) |
| 22:51.44 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 22:51.53 | franck | Hi |
| 22:52.02 | Tagor | SkramX ^ |
| 22:52.06 | SkramX | Tagor: yeah |
| 22:52.12 | SkramX | im having network issues, one sec. |
| 22:52.17 | Tagor | Ok, sure :) |
| 22:52.25 | franck | can somebody tell me if asterisk business edition has silence suppression capability? |
| 22:52.50 | SkramX | Tagor: what does it say in console? |
| 22:53.20 | Tagor | Just that it sets the musiconhold to default, SkramX |
| 22:53.29 | Tagor | No errors, or something like that |
| 22:53.54 | SkramX | crap, I have to jet-- be back within an hour |
| 22:53.55 | SkramX | sorry. |
| 22:54.21 | Tagor | Ok, no problem. Maybe there's someone else who knows how to get this working :) |
| 22:56.50 | *** join/#asterisk Vyeperman (n=Vye@ip68-8-174-154.sd.sd.cox.net) |
| 22:57.54 | FuriousGeorge | whats that softphone for linux something ***fsck |
| 22:57.59 | FuriousGeorge | or ***fsc |
| 22:58.11 | SpooForBrains | well, there's twinkle |
| 22:58.15 | SpooForBrains | or kphone |
| 22:58.32 | jbalcomb | [TK]D-Fender ok on the ringtones. the grandstream only has 3 custom ringtones. |
| 22:59.17 | jbalcomb | [av]bani whats the URL of your phones page? |
| 23:00.12 | *** join/#asterisk Cool_One (n=bclinton@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 23:00.58 | Cool_One | does anyone here use cisco 7960 phones on their asterisk network? |
| 23:01.12 | Cool_One | if so... anyone have a clue how to set them up? |
| 23:01.24 | [TK]D-Fender | http://bani.anime.net/phonez/ |
| 23:01.37 | tuxinator_linux | Cool_One: A lot of us do |
| 23:01.53 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 23:02.00 | tuxinator_linux | Cool_One: You need to ask your question |
| 23:02.05 | jbalcomb | sure seems like the IP501 is more comparable to the SNOM 320, cept for the LCD. |
| 23:02.55 | [TK]D-Fender | jbalcomb : Better audio quality & manufacturing, but otherwise similar. |
| 23:03.12 | tuxinator_linux | FuriousGeorge: I think it is over at asteriskguru |
| 23:03.37 | [av]bani | http://www.engadget.com/2006/02/28/ipod-hi-fi-brings-music-to-the-masses/1#comments |
| 23:03.47 | [av]bani | ~phones |
| 23:03.48 | jbot | methinks phones is at http://bani.anime.net/phones/ |
| 23:03.48 | FuriousGeorge | idefisk |
| 23:04.27 | tuxinator_linux | FuriousGeorge: there you can.. I still don't understand the name |
| 23:04.36 | tuxinator_linux | s/can/go |
| 23:04.45 | jbalcomb | [TK]D-Fender ok, im going to request 2 SNOM 360 and 2 IP 501. Thanks again for the info. |
| 23:04.53 | FuriousGeorge | tuxinator_linux: thanks again |
| 23:04.53 | *** join/#asterisk mikefoo (n=mikefoo@64.124.169.2) |
| 23:05.06 | tuxinator_linux | FuriousGeorge: was I right on the site? |
| 23:05.14 | mikefoo | Anyone interested in a 7960? have a spare one I can sell. |
| 23:05.26 | mikefoo | works fine. |
| 23:05.32 | tuxinator_linux | mikefoo: I bit $10 US |
| 23:05.42 | mikefoo | I will pass. |
| 23:05.55 | tuxinator_linux | 15? |
| 23:05.58 | tuxinator_linux | ;-) |
| 23:06.03 | mikefoo | ... |
| 23:06.33 | tuxinator_linux | seriously, about all I can afford today is $25 for it |
| 23:06.45 | FuriousGeorge | tuxinator_linux: yeah it was listed |
| 23:06.49 | FuriousGeorge | there |
| 23:06.54 | tuxinator_linux | cool |
| 23:06.54 | jbalcomb | mikefoo: eBay iT! |
| 23:07.02 | *** part/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 23:07.27 | jbalcomb | [TK]D-Fender ok, I'm out tomorrow so Thrusday. thanks. |
| 23:07.28 | *** part/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 23:07.38 | *** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net) |
| 23:08.05 | Goral | do i need this in sip.conf under the user? defaultip=192.168.1.100 |
| 23:10.05 | tuxinator_linux | Goral: try it out either way, and learn |
| 23:10.29 | *** join/#asterisk mzo (n=moz@ool-435193b3.dyn.optonline.net) |
| 23:10.31 | Goral | just doing that |
| 23:10.42 | mzo | bleh fwd is still broken for me. |
| 23:11.34 | mzo | anyone know what i broke because it's still not registering me after 11 days |
| 23:13.44 | shido6 | ? |
| 23:14.00 | *** join/#asterisk file (n=joshnet@mctnnbsa24w-142167032200.pppoe-dynamic.nb.aliant.net) |
| 23:14.05 | mzo | doing an iax connection to fwd just says code 29, registration refused for the last 11 days |
| 23:16.16 | glm2k | mzo: heh. that's familiar, did you doublecheck your login and password? |
| 23:16.25 | mzo | about 300 times now |
| 23:16.41 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.235) |
| 23:16.48 | mzo | unless they save passwords differently on the website as what I would use to connect |
| 23:17.00 | glm2k | no, it's pretty much copy and paste |
| 23:18.41 | mzo | I'm lsot then, and no one seems to know what 'code 29' is on a failure |
| 23:22.37 | jarrod | im getting "Smoother was working on 1077940456 format frames, now trying to feed 4" on a call thats causing asterisk to spew out 100meg of data |
| 23:22.39 | jarrod | heh |
| 23:22.43 | jarrod | what could that problem be |
| 23:23.32 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 23:23.48 | justinu | jarrod, you win the bizzare error message of the day prize |
| 23:23.57 | mzo | mine is better :P |
| 23:24.06 | jarrod | haha |
| 23:24.10 | jarrod | what the heck is 'smoother' |
| 23:24.18 | Abydos313 | afternoon everyone |
| 23:24.20 | jarrod | <PROTECTED> |
| 23:24.21 | jarrod | 00148 ast_log(LOG_WARNING, "Smoother was working on %d format frames, now trying to feed %d?\n", s->format, f->subclass); |
| 23:24.34 | justinu | jarrod: that's gotta be a bug |
| 23:24.38 | justinu | report it |
| 23:24.52 | justinu | find a bug marshall in #asterisk-dev perhaps |
| 23:24.54 | justinu | or here |
| 23:25.00 | x86 | jarrod: "Smoother" sounds like a nickname of someone ;) |
| 23:25.15 | jarrod | well apparently its a crucial part of the code ha |
| 23:25.29 | justinu | the fact that s->format is 1077940456 can't be intentional |
| 23:25.40 | justinu | sounds like an uninitialized value to me |
| 23:27.35 | *** part/#asterisk casterman (n=casterma@kasadsyed.net8.nerim.net) |
| 23:29.38 | *** join/#asterisk veepster (n=veepster@67.130.38.2) |
| 23:29.49 | jarrod | diznang |
| 23:30.39 | veepster | basic question. what kind of hardware converts phone line to pc friendly? by pc friendly do we mean ethernet? |
| 23:30.59 | veepster | cant be a basic modem right? |
| 23:31.00 | rene- | and ATA device |
| 23:31.41 | rene- | an ATA device with an FXO port will take your phone line and make it into a voip line |
| 23:31.52 | veepster | ok |
| 23:32.12 | veepster | in addition to this device will my linux box need an ethernet port? |
| 23:32.30 | rene- | well yes if you want to run asterisk and connect to the ATA |
| 23:32.37 | fifer | Any ARI users? |
| 23:32.48 | rene- | you can install a phone line port in asterisk with an specific modem |
| 23:32.51 | fifer | Having issues getting ARI working with out AMP |
| 23:33.58 | veepster | makes sense |
| 23:33.59 | Goral | now i have to set up voice mail |
| 23:34.24 | veepster | is zaptel neccessary if Im using ata? or is it only for specific ata devices? |
| 23:34.40 | rene- | zaptel is only for cards that go into your coomputer |
| 23:34.42 | shido6 | dont need zaptel, veepster... |
| 23:35.00 | Abydos313 | you need a timing device and you can use ztdummy in the zaptel package |
| 23:36.28 | veepster | so this ata device wont go into a computer? |
| 23:36.41 | veepster | I just need to get started. in addition to a pc, what hardware do you guys recommend? |
| 23:37.12 | rene- | i have a centos 4.2 install, i installed asterisk and want to use asterisk voicemail to email feature, tail /var/log/maillog shows my message as queued and with a relay direction as root@.localhost, this machine is using sendmail for smtp what do i need to chekc before i can send an voicemail to email |
| 23:37.21 | rene- | ata is an external device |
| 23:37.32 | Cool_One | does anyone know how to reset a 7960g cisco phone to it's original settings? And also... do you setup the cisco as a sip phone or a skinny |
| 23:37.44 | joe | rene-: how did you install * ? |
| 23:37.44 | veepster | like sipura? |
| 23:37.48 | Cool_One | in asterisk |
| 23:37.58 | rene- | joe: from rpm i think |
| 23:38.41 | rene- | veepster: yes |
| 23:38.45 | rene- | sipura is an AAT |
| 23:38.45 | *** join/#asterisk Gennaro (n=Gennaro@ppp-62-10-136-236.dialup.tiscali.it) |
| 23:38.47 | rene- | ATA |
| 23:38.53 | Gennaro | hi |
| 23:39.04 | veepster | where can I buy these? circuitcity? or do I have to buy them online? |
| 23:39.15 | Gennaro | some one can help me with extension / context / queue ?! |
| 23:39.37 | rene- | buy them online, you can buy the more commercial linksys pap2 but you have to make sure that it is ann unlocked version, that is not locked to a vonage of sorts |
| 23:40.02 | veepster | ok |
| 23:40.04 | rene- | the vonage locked version will bring you little fun |
| 23:40.17 | veepster | last question. if my input phone line is already a pbx line is that a problem? |
| 23:40.18 | syzygybsd | has anyone had issues with the times on polycom phones being gmt no matter what the settings are? |
| 23:40.55 | joe | rene-: which? re your mail issue: in your voicemail.conf you have the 5734 => 5734,Foo Name,foo@bar.com? |
| 23:41.11 | rene- | joe: i am using signate 1.2.4 asterisk |
| 23:41.21 | rene- | joe: i do |
| 23:41.57 | rene- | this is mail log; Feb 28 17:14:20 mexico sendmail[6530]: k1SNAKKj006530: from=root, size=20066, class=0, nrcpts=1, msgid=<Asterisk-0-1059585951-709-4172@mexico>, relay=root@localhost |
| 23:41.59 | rene- | Feb 28 17:14:20 mexico sendmail[6530]: k1SNAKKj006530: to=Joselo Fernandez <rmendoza@bluebottle.com>, delay=00:04:00, mailer=esmtp, pri=50066, dsn=4.4.3, stat=queued |
| 23:42.03 | rene- | sorry for that |
| 23:42.24 | *** join/#asterisk Cool_One (n=bclinton@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 23:43.43 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:43.53 | russellb | rene-: what does "signate asterisk" provide you that you couldn't get yourself? |
| 23:44.35 | rene- | nothing at all, since i have done gui asterisk dev in the past |
| 23:44.44 | rene- | the costumer just wants it |
| 23:46.00 | rene- | i could flame them but it does get a few things right, i couldnt get auto provisioning to work, and i ve done it by myself |
| 23:46.03 | Abydos313 | what is signate asterisk? haven't heard of it |
| 23:46.13 | rene- | signate sigman is an asterisk gui |
| 23:46.20 | rene- | they have an asterisk rpm |
| 23:46.29 | Abydos313 | ahh. any better then amp? |
| 23:46.32 | rene- | back on topic: any helpers? |
| 23:46.37 | rene- | yes way better than amp |
| 23:46.42 | rene- | but still broken install procedure |
| 23:46.49 | rene- | so i will take the way in way better |
| 23:46.50 | rene- | heh |
| 23:46.52 | Abydos313 | :) something worth trying then |
| 23:47.05 | Abydos313 | i can suffer thru an in stall if gui is good |
| 23:47.06 | rene- | yes but its ~800 usd per machine |
| 23:47.22 | Abydos313 | doesn't really matter, after it's all setup how often will you really make changes |
| 23:47.28 | rene- | not really |
| 23:47.31 | rene- | many times |
| 23:47.53 | Abydos313 | what's 800 per machine? signate? |
| 23:47.56 | rene- | yes |
| 23:48.12 | Abydos313 | fuck that , that's more than the cost of the server running it |
| 23:48.40 | rene- | that is my point |
| 23:48.48 | Abydos313 | maybe it's free for home use? |
| 23:49.14 | syzygybsd | lol.. just put the company's server in your home... |
| 23:49.29 | joe | rene-: from the cli can you email? |
| 23:49.36 | Abydos313 | i guess i should say non-commercial use |
| 23:50.31 | asterisk99 | Does anyone here use the Perl AGI Interface? |
| 23:50.45 | Grizzy | Is there an asterisk driver for an FXO port using a "winmodem", for FreeBSD? |
| 23:51.02 | robin_sz | asterisk99: sure .. |
| 23:51.12 | robin_sz | asterisk99: what do you want to know? |
| 23:51.35 | asterisk99 | robin_sz: I can't find doc as to what directory to put the .PL file and how to link to it |
| 23:52.01 | robin_sz | well, usalluy /etc/asterisk/agi/ |
| 23:52.44 | robin_sz | and then just call it like you would any other agi script |
| 23:52.47 | russellb | by default, it's /var/lib/asterisk/agi-bin/ |
| 23:53.00 | robin_sz | russellb: yeah, you are right |
| 23:53.25 | russellb | ha |
| 23:53.34 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 23:53.45 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 23:53.46 | *** join/#asterisk Goral (n=needsand@CPE0012172e9c9f-CM014080205433.cpe.net.cable.rogers.com) |
| 23:53.52 | terrapen | hey, has anybody used the Redfone foneBridge? |
| 23:53.56 | terrapen | http://www.red-fone.com/fonebridge.html |
| 23:54.08 | asterisk99 | russellb: thanks... let's see is this will work |
| 23:54.57 | robin_sz | exten => h,1,deadagi,fax.pl for example |
| 23:55.01 | *** join/#asterisk redax (n=redax@r6.hu) |
| 23:55.03 | redax | hi |
| 23:55.54 | asterisk99 | russellb: I found astagidir => /var/lib/asterisk/agi-bin in /etc/aterisk/asterisk.conf ... I should have known better!!! |
| 23:56.07 | russellb | :) |
| 23:56.13 | redax | I'm tring to use chan_capi. the 0.3.5 (or 0.4.0pre) is working (just noisy voice) but the chan_capi-cm-0.6.4 wont |
| 23:56.20 | terrapen | wow, i thought people would be all over this thing |
| 23:56.35 | Abydos313 | terrapen kinda pricey |
| 23:56.36 | terrapen | no more TDM cards in your asterisk servers |
| 23:56.44 | justinu | some people here use them apparently |
| 23:56.47 | terrapen | abydos, its not much more than a 4-port sangoma card |
| 23:56.51 | terrapen | if at all |
| 23:56.54 | redax | the call coming and not routed to the given context, instead I got this message before hangup: 'DISCONNECT_IND on incoming without pbx, doing hangup' |
| 23:57.16 | Abydos313 | how much is a 4port sangoma? |
| 23:57.23 | russellb | no more tdm cards in your asterisk box, you just pay a lot to have someone put it in a different box for you! |
| 23:57.29 | terrapen | redfone box: $2495 sangoma 4-port: $2495 |
| 23:57.32 | terrapen | heh, same same |
| 23:57.41 | Abydos313 | ok |
| 23:57.49 | terrapen | russel, yes, but you save all of the CPU overhead of trying to run TDM in your asterisk box |
| 23:57.51 | Goral | if i'm running an asterisk box at home and i have two people outside my local network calling each other do i need the bandwidth to support the conversation or does asterisk point it to each other like a peer2peer network? |
| 23:58.09 | terrapen | and all of the associated hardware incompatabilities between the PCI cards and finicky motherboards |
| 23:58.46 | Abydos313 | it also provides a single point of failure for all your asterisk servers |
| 23:59.06 | terrapen | no more so than the PCI card does |
| 23:59.23 | terrapen | and if your Asterisk server goes down with the PCI solution, you have to move your cables around |
| 23:59.42 | terrapen | with this little box, it appears that you can just use Linux-HA or whatnot to provide instant failover |
| 23:59.59 | Abydos313 | i'd like to see it in action |
