| 00:01.24 | *** join/#asterisk tlyng (n=torkel@84-52-246.62.3p.ntebredband.no) |
| 00:02.03 | tlyng | does anyone here have a skeleton plugin for asterisk using autotools? |
| 00:03.13 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 00:09.58 | *** join/#asterisk dude7064 (n=dude7064@78-86-79-212.zone2.bethere.co.uk) |
| 00:10.27 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 00:12.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:14.45 | dan__t | pdfhacker, thanks again for the tips |
| 00:19.20 | dude7064 | How can I have too many calls routed to one number without it being busy all the time ? |
| 00:21.08 | k-man | how well does SIP work over a wireless network? |
| 00:23.09 | jblack | fine. |
| 00:23.54 | k-man | when i tried putting my asterisk server on the wireless network, it seemed to cause a bit of dropping out in the audio |
| 00:24.06 | jblack | maybe you have a piss poor wireless network. |
| 00:26.24 | k-man | jblack: maybe |
| 00:39.00 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:44.14 | mattb | hi, I'm looking for some pointers on how I can send an outbound call I initiated to a context to await user DTMF input |
| 00:44.23 | mattb | eg. I want to dial someone and then present them with a menu |
| 00:44.41 | mattb | can someone point me at where I should go RTFM ? :p |
| 00:48.06 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-237-246.lns10.mel6.internode.on.net) |
| 00:49.30 | *** join/#asterisk nargon (n=nargon@217.194.139.4) |
| 01:01.11 | *** join/#asterisk DarkRift (n=dark@65.92.249.176) |
| 01:09.55 | jaytee | ~book |
| 01:09.56 | jbot | rumour has it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 01:10.04 | jaytee | mattb, ^^^^^^^ |
| 01:10.32 | nargon | sorry can anyone tell me how to contcatenat a string onto the end of an existing string in a dialplan |
| 01:11.06 | nargon | exten => _7XXXXXXX,10,Set(peerconcat = ${peerconcat}${peername}) |
| 01:11.12 | nargon | something like this |
| 01:11.58 | nargon | I'm trying to build up a string of sip/peers to dial() |
| 01:12.57 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:13.03 | *** join/#asterisk friendly12345 (n=friendly@ppp118-208-145-44.lns10.mel4.internode.on.net) |
| 01:14.43 | jaytee | nargon, try this instead Set(peerconcat = $["${peerconcat}${peername}"]) |
| 01:15.02 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 01:16.12 | *** join/#asterisk BeeBuu (n=beebuu@61.145.77.61) |
| 01:18.56 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d412c52ac0a5356d) |
| 01:19.27 | dwery | 'morning. just to let you all know.. it took me a almost week to workaround a series of bugs in the latest wanpipe drivers for the usbfxo. the usb driver in particular has some very nasty bugs and I wonder how such code could work at all. I'll see if I can make sangoma fix them in a saner way. |
| 01:20.52 | nargon | jaytee I notice when I set(peerconcat = somthing) and then NoOp ${peerconcat} on the nextline the NoOp variable echo's empty |
| 01:20.56 | dwery | don't eve try it, you'e been warned :) |
| 01:22.33 | k-man | is there any linksys ATA's with more than 1 fxo? (to plug analogue phone into) |
| 01:22.45 | k-man | or is that fxs? i get confused |
| 01:23.01 | k-man | its FXS |
| 01:23.09 | dwery | k-man: fxs is to plug analog phones, fxo to plug analog land lines from the telco |
| 01:23.24 | k-man | yeah, so i meant FXS |
| 01:23.30 | k-man | i want to plug 2 analogue phones into it |
| 01:23.54 | dwery | k-man: there must be some.. haven't checked.. I'm pretty sure ZyXEL has one |
| 01:24.04 | k-man | are they a good brand? |
| 01:24.37 | dwery | k-man: not worst than any other brand :) |
| 01:25.00 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 01:28.22 | k-man | the SPA2102 seems to have dual FXS |
| 01:30.47 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 01:31.10 | Ritzerisk | how would i tell what codec im using if i have a inbound sip trunk |
| 01:32.36 | dwery | Ritzerisk: try iax2 show channels |
| 01:32.52 | dwery | or iax2 show perr <peername> to see all the available codecs |
| 01:32.56 | dwery | peer* |
| 01:37.26 | Ritzerisk | even though its a sip trunk from the carrier |
| 01:37.53 | dwery | Ritzerisk: sorry, it's late. I confused sip with iax :( |
| 01:38.51 | dwery | sip show channels will probably work |
| 01:39.34 | Ritzerisk | i did a call while it was in session with that before u said hehee :) it was a ulaw |
| 01:40.00 | Ritzerisk | now my issue is im not getting any dtmf nor any audio of the ivr |
| 01:40.33 | Ritzerisk | but if i direct a call to a ext it works (xlite) and if i make a call it works carrier is les.net and i have g.711 checked |
| 01:42.32 | Ritzerisk | oxc |
| 01:42.43 | dwery | sip has three different ways to send/receive dtmf |
| 01:43.01 | dwery | but the audio is sent via RTP |
| 01:43.32 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 01:43.42 | *** join/#asterisk op3r (n=op3r@114.108.201.205) |
| 01:43.45 | dwery | gtg... see you |
| 01:45.21 | Ritzerisk | eeeeek |
| 01:45.25 | Ritzerisk | kk thanks ;) |
| 01:46.09 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 01:47.37 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
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| 01:54.49 | *** mode/#asterisk [+o denon] by ChanServ |
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| 02:01.10 | ZX81 | hey all, anyone know if there is a bug open discussing file descriptor leaks? We're getting too many open files on a system with about 5 concurrent calls - 3 crashes in last week |
| 02:03.48 | ZX81 | hmmm |
| 02:05.58 | ZX81 | looks like gsm files left open |
| 02:06.25 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:09.12 | Ritzerisk | now my issue is im not getting any dtmf nor any audio of the ivr im at ulaw too |
| 02:10.59 | jaytee | sounds like a NAT issue |
| 02:13.43 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 02:15.09 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 02:17.35 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 02:22.44 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:23.31 | [T]ank | what version what the first to introduce the dahdi package? I am looking for the latest version that did not include dahdi |
| 02:24.29 | thehar | i'm using 1.4.22 with latest zaptel |
| 02:24.39 | [TK]D-Fender | [T]ank: 1.4.21 was the last pre-DAHDI IIRC |
| 02:24.47 | [T]ank | thank you |
| 02:24.55 | thehar | [TK]D-Fender: ^_^ |
| 02:25.08 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 02:25.19 | thehar | it does use chan_dahdi, tho |
| 02:25.59 | *** part/#asterisk dlynes (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 02:29.31 | Ritzerisk | im open up all the way dmz |
| 02:29.53 | drmessano | DMZ is your problem |
| 02:29.59 | drmessano | Open the proper ports |
| 02:30.03 | drmessano | Set NAT correctly |
| 02:30.13 | drmessano | The DMZ in your router is screwing you |
| 02:30.27 | Ritzerisk | is there a quick port guide eeeeek |
| 02:30.37 | drmessano | ~sipnat |
| 02:30.38 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:32.47 | *** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar) |
| 02:33.52 | Ritzerisk | doesnt a dmz forward all ports to an ip |
| 02:35.02 | drmessano | In theory, but theres more to it, and DMZ just doesnt work |
| 02:35.09 | drmessano | Open the correct ports |
| 02:35.12 | drmessano | Take DMZ off |
| 02:36.24 | Ritzerisk | hmm k k |
| 02:36.41 | Ritzerisk | let me just try an open them all real quick haha ;) |
| 02:39.03 | [TK]D-Fender | DMZ is fine. Overkill, but fine |
| 02:39.19 | jblack | anyone know of a voice chat network? |
| 02:39.20 | [TK]D-Fender | But isn't enough by itself to configure * to operate behind NAT. |
| 02:39.34 | [TK]D-Fender | A point thats been beaten ove Ritzerisk's head COUNTLESS times. |
| 02:40.12 | jblack | ritzerisk: The sip protocol embeds the ip, so port forwarding alone doesn't handle it. |
| 02:41.10 | Ritzerisk | well bam bam :) |
| 02:42.16 | Ritzerisk | i have ulaw outbound works fine just the inbound with the ivr i followed les.net to a T on the setup |
| 02:42.40 | jblack | there must be some free conference calling Out There |
| 02:45.48 | Ritzerisk | you miss me :) |
| 02:46.08 | Ritzerisk | this ones gonna keep me up all night .... |
| 02:46.40 | [TK]D-Fender | Ritzerisk: Yes.. but our aim is improving... |
| 02:46.54 | [TK]D-Fender | trims the main battery another 2 degrees |
| 02:47.31 | Ritzerisk | haha :) |
| 02:47.35 | jblack | just go with iax for your provider and be done with it. ;) |
| 02:47.38 | Ritzerisk | wooot wooot |
| 02:47.50 | Ritzerisk | really i can change that quick with les |
| 02:58.51 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 03:00.53 | drmessano | LES's IAX is horrid |
| 03:01.04 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:01.09 | drmessano | Works about 20% of the time |
| 03:01.18 | drmessano | Option 2 is correct setup |
| 03:01.27 | Ritzerisk | i cant even get it to pass to my system hehe |
| 03:01.40 | Ritzerisk | so i reverted back to sip trunk |
| 03:01.41 | [TK]D-Fender | drmessano: That's greater than Ritzerisk's average success rate, so I wouldn't balk at that! |
| 03:01.54 | [TK]D-Fender | plays the odds |
| 03:01.55 | Ritzerisk | wooot wooot |
| 03:01.57 | drmessano | I dont see why this is so difficult |
| 03:02.06 | drmessano | Open ports, follow the guide, move on |
| 03:02.15 | [TK]D-Fender | drmessano: It's not the odds.. its that playa y0! |
| 03:02.25 | drmessano | lol |
| 03:02.30 | Ritzerisk | they are all open or i did the dmz route |
| 03:02.53 | [TK]D-Fender | Ritzerisk: What part of "you need to CONFIGURE *" is still not getting in your head? |
| 03:03.05 | [TK]D-Fender | Ritzerisk: Pile'o'settings without which you are screwed/. |
| 03:03.15 | [TK]D-Fender | Ritzerisk: Follow the damn guide already |
| 03:04.13 | Ritzerisk | the part of not the codec but maybe on why dtmf is not even passing do i need some sort of proxy from the pox to the carrier |
| 03:04.34 | [TK]D-Fender | Ritzerisk: No, you just need to set your mode right |
| 03:05.08 | Ritzerisk | i did a dtmfmode=rfc2833 even though i saw taht in the setup of the les trunk |
| 03:05.49 | Ritzerisk | no audio so its somethign when it hits inbound not coming back to the carrier but im googling more stuff as we speak :) |
| 03:07.32 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 03:07.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 03:13.52 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:14.00 | Ritzerisk | ohhh poop on a stick |
| 03:15.34 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:15.53 | [TK]D-Fender | Ritzerisk: Get yours while quantities last... |
| 03:17.00 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 03:17.15 | jchonig | Anyone familiar with configuring an Aastra 480i with *? |
| 03:18.24 | *** join/#asterisk CrazyTux (n=brandon@c-98-196-6-54.hsd1.tx.comcast.net) |
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| 03:29.50 | *** mode/#asterisk [+o denon] by ChanServ |
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| 03:47.37 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 03:50.26 | Ritzerisk | hah |
| 04:05.53 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 04:07.29 | drmessano | Googling |
| 04:08.04 | drmessano | Still a good excuse for not RTFMing |
| 04:09.51 | [TK]D-Fender | sharpens the edges of his "spoonfeeding" cutlery |
| 04:10.44 | Octothorpe | That's such a hot expression. Consider it stolen. :) |
| 04:11.15 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:11.46 | Octothorpe | [TK]D-Fender, my previous statement was directed at you. |
| 04:12.35 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 04:12.50 | [TK]D-Fender | Octothorpe: My works are all CC licensed :) |
| 04:13.05 | Octothorpe | Fantastic, thanks muchly :) |
| 04:13.20 | Octothorpe | Wait, which license? |
| 04:13.22 | Octothorpe | :P |
| 04:13.49 | Octothorpe | speeds up his plans for commercialization and derivation before [TK]D-Fender can answer |
| 04:14.04 | [TK]D-Fender | Octothorpe: Creative Commons |
| 04:14.14 | Octothorpe | There are several of them. |
| 04:14.31 | Octothorpe | Hence the question |
| 04:15.23 | Octothorpe | http://creativecommons.org/licenses/ |
| 04:15.44 | [TK]D-Fender | Octothorpe: Derivation of a joke tends to create substantially different results thus qualifying as a "different work". Go right ahead :) |
| 04:16.14 | Octothorpe | Yes, but commercialization doesn't. |
| 04:16.22 | Octothorpe | speeds up his T-shirt screenprinting equipment |
| 04:16.36 | Octothorpe | :) |
| 04:17.17 | [TK]D-Fender | Octothorpe: /me will make sure Octothorpe is wearing one of them when the body is found... |
| 04:17.38 | [TK]D-Fender | heh |
| 04:17.56 | Octothorpe | Heh, niiiiiice. :) |
| 04:18.25 | [TK]D-Fender | Octothorpe: Where poetic licence meets poetic justice. |
| 04:18.54 | Octothorpe | So true! |
| 04:19.08 | [TK]D-Fender | "Man killed by meteorite : Why so Sirius?!" |
| 04:19.09 | Octothorpe | nonchalantly hides another pilfered line up his shirt and walks out the door |
| 04:21.13 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:21.14 | *** mode/#asterisk [+o Mog] by ChanServ |
| 04:24.33 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 04:32.04 | *** join/#asterisk tlyng (n=torkel@84-52-246.62.3p.ntebredband.no) |
| 04:34.27 | tlyng | i've created an extension which i can call with my softphone. When I call this extensions the call is answered, then i do Dial(Mobile/my_phone/somenumber) and then WaitForSilence.... |
| 04:35.18 | tlyng | my intension is that WaitForSilence should stop when it's silent in the cellphone... but the dialplan seems to stop execution after the Dial command |
| 04:35.27 | tlyng | how should i do this? |
| 04:40.26 | [TK]D-Fender | tlyng: if you continue past dial that means you are not in that Dial'd call |
| 04:40.53 | *** join/#asterisk richardlynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 04:43.05 | richardlynch | I'm looking for somebody with experience in writing EAGI scripts, preferably in PHP, for contract work. Can I post here? If not, where? |
| 04:43.26 | tlyng | [TK]D-Fender: ok, do you know how I'm supposed to do that kind of functionallity? |
| 04:45.04 | [TK]D-Fender | tlyng: What functionality? You haven't described any real action to take. |
| 04:45.32 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:45.33 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:47.35 | tlyng | [TK]D-Fender: I want to call an extension using my softphone. When that call is answered I want asterisk to dial another number using my cellphone. The number which is dialed is external and I want to have the possibility to control that channel with dialplan applications. |
| 04:48.56 | [TK]D-Fender | tlyng: You want * to call out and dump the caller into the dialplan basically? So far I don't see any purpose being filled by your softphone in this |
| 04:51.23 | drmessano | Do I need to do anything special to enable T.38 for a call, or on a specific peer? |
| 04:52.03 | drmessano | For a hot date |
| 04:52.04 | drmessano | T38 |
| 04:52.07 | drmessano | Burma shave |
| 04:52.15 | drmessano | :( |
| 04:52.50 | drmessano | I have an ATA with T38 enabled.. T38 passthru all nice and set up in my configs |
| 04:54.11 | tlyng | [TK]D-Fender: actually i only use the softphone to trigger the outgoing Dial. I was thinking about using the softphone while i create the dialplan, so that I can see what asterisk does at the same time hearing the conversation. Better ways to do this? |
| 04:54.51 | [TK]D-Fender | tlyng: Yes, espcially since your callee isn't IN the dialplan when you call him with "Dial" |
| 04:55.01 | tlyng | :) |
| 04:55.02 | [TK]D-Fender | tlyng: Lookup "call files" and "AMI Originate" on the WIKI |
| 04:55.12 | tlyng | ok, thanks |
| 04:58.53 | *** join/#asterisk pcrack (n=pcrack@121.58.195.10) |
| 04:59.37 | pcrack | hi is alcatel omniPCX is based on asterisk? |
| 04:59.43 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:59.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:04.53 | [TK]D-Fender | pcrack: wHAT GIVES YOU THAT IMPRESSION? |
| 05:05.35 | pcrack | just asking... |
| 05:05.50 | pcrack | i just wanted to know... |
| 05:05.58 | pcrack | is it? |
| 05:06.22 | [TK]D-Fender | pcrack: a proprietary PBX by a proprietary PBX maker.... I'd say **NO** |
| 05:06.39 | pcrack | ic... |
| 05:06.53 | pcrack | have you tried PBX? |
| 05:07.12 | [TK]D-Fender | pcrack: I've tried all sorts of PBX's |
| 05:07.37 | pcrack | really cool |
| 05:08.34 | drmessano | I've tried PBX too |
| 05:08.38 | drmessano | and network |
| 05:08.58 | pcrack | maybe you could help me on this...i put the alcatel omniPBX behind the firewall and i already configured the port forwarding for the SIP and RtP ports..it can call but it cannot heard the recepient nor no ringing sound on the IP phone |
| 05:09.45 | drmessano | This is #asterisk |
| 05:10.05 | drmessano | Try #alcatel, #omnipbx, or #pbx |
| 05:10.43 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 05:11.20 | voxter | unreal |
| 05:11.31 | drmessano | Thats an IRCd |
| 05:11.35 | drmessano | Try #unrealircd |
| 05:11.47 | voxter | hahaha |
| 05:11.54 | drmessano | Umm |
| 05:11.59 | drmessano | #humor for that |
| 05:12.01 | voxter | yeah, nothin there. |
| 05:12.12 | voxter | had i said lol, you could have pointed me to aol |
| 05:12.14 | drmessano | #0 if you need nothin |
| 05:12.43 | [TK]D-Fender | pcrack: Your PBX is not supported here. Go read its documentation or call up a consultant who specializes in them. |
| 05:13.10 | drmessano | Hmm |
| 05:13.19 | drmessano | Seems like I have an old copy of internet around here too |
| 05:13.22 | drmessano | Maybe in the same box with pbx |
| 05:14.07 | drmessano | I stopped buying internet since I could get cracks for it on bittorrent |
| 05:14.28 | drmessano | I need a good keygen for pbx though, if anyone has it |
| 05:14.46 | [TK]D-Fender | ok, checkout time, later all |
| 05:14.50 | drmessano | Later |
| 05:15.25 | *** join/#asterisk toddejohnson (n=toddejoh@69.220.214.70) |
| 05:15.28 | drmessano | PBX-Corporate-Edition-FINAL-1.0-Z0Mg.rar |
| 05:18.26 | *** part/#asterisk Magicblaze007 (n=sony@fl-67-233-204-182.dhcp.embarqhsd.net) |
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| 05:29.13 | [T]ank | getting the following when I create a call file. http://pastebin.ca/1356325 havent got a clue where to start looking to fix it. regular dialed calls work ok |
| 05:33.02 | [T]ank | is there anybody out there.... |
| 05:33.06 | [T]ank | just nod if you can hear me |
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| 05:42.03 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:51.05 | richardlynch | Do any of the Playback() Background() etc functions take an ongoing stream input, rather than a fully-formed audio file? I want to pipe real-time audio output to a channel somehow... |
| 05:56.19 | ikevin_ | richardlynch, i don't know if you can, so, you can use moh function for that |
| 05:56.40 | ikevin_ | going at work |
| 05:56.41 | ikevin_ | cya |
| 05:57.06 | richardlynch | Thanks! |
| 05:57.46 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 05:58.04 | [T]ank | anyone here know anything about call files? |
| 05:58.24 | BeeBuu | [T]ank: i got a little bit |
| 05:58.51 | BeeBuu | [T]ank: maybe i can help,if you like |
| 05:59.29 | [T]ank | i can dial out from my system just fine... but when I try to do a callfile, this is the output I get on the cli: http://pastebin.ca/1356325 |
| 05:59.37 | [T]ank | BeeBuu: can I pm you? |
| 06:00.00 | BeeBuu | privite chat? |
| 06:00.05 | [T]ank | yep |
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| 06:10.20 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 06:10.43 | Micc | I'm unable to get color with asteirsk -vvvcr, but asterisk -vvvc works fine. |
| 06:10.55 | Micc | I set my TERM variable to xterm-color, but still no color. |
| 06:11.16 | Micc | I've read everything I can find on google, and nothing seems to help. |
| 06:12.35 | *** part/#asterisk richardlynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
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| 06:15.44 | Micc | drmessano, I'm also working on getting customers setup with a T.38 ATA. |
| 06:16.12 | Micc | drmessano, I haven't found a provider that does t.38 to pstn yet that will return my calls. |
| 06:26.49 | drmessano | Flowroute does |
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| 06:28.41 | *** mode/#asterisk [+o denon] by ChanServ |
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| 06:39.15 | Micc | drmessano, you like flowroute? pretty good service? |
| 06:39.38 | drmessano | So far |
| 06:39.40 | drmessano | Yep |
| 06:40.06 | Micc | how long have you been using them? |
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| 06:49.52 | Micc | I like the look of their site, very nice layout and I like the colors. |
| 06:50.18 | bobsaccamano | hi..how do i configure asterisk to forward anonymous calling from a sip channel?? |
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| 07:00.45 | mmlj4 | NOTICE[32564]: chan_sip.c:18390 handle_request_register: Registration from '<sip:0004f2144c31-a@192.168.2.1>' failed for '192.168.2.40' - No matching peer found | my sip.conf has an entry for [0004f201079d-a] # this is a polycom phone on * 1.6.0.3 |
| 07:14.15 | mmlj4 | ok, I'm an idiot |
| 07:20.09 | mmlj4 | I'm up apparently way too late, trying to think, and it's not working |
| 07:29.48 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.ut.comcast.net) |
| 07:31.07 | mmlj4 | all better now, and I'm off to bed |
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| 07:55.13 | k-man | murio? |
| 07:55.21 | k-man | are you around? |
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| 09:12.07 | tamiel | hello, bugs.digium.com is out ? I tried to browse it from two differents sites and browser timeout ... |
| 09:13.18 | tamiel | now bugs.digium.com is responding slowly ... |
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| 10:46.34 | xmlsoap | hi all. I have put wW in both DIAL_OPTIONS and TRUNK_OPTIONS. I have enabled the recordings with record_out=Adhoc,record_in=Adhoc |
| 10:46.40 | xmlsoap | but i can not record the calls |
| 10:46.44 | xmlsoap | Mar 9 11:41:41 VERBOSE[3022] logger.c: [app_record.so]Mar 9 11:41:41 VERBOSE[3022] logger.c: [app_record.so] => (Trivial Record Application) |
| 10:46.44 | xmlsoap | Mar 9 11:43:02 VERBOSE[3091] logger.c: recordingcheck|20090309-114302|1236595382.0: Outbound recording not enabled |
| 10:46.49 | xmlsoap | i can see this in the logs |
| 10:46.54 | xmlsoap | any help please? |
| 10:47.16 | xmlsoap | (yes, i;m calling *1 in middle of a call to record stuff) |
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| 11:20.24 | double_cheesburg | Where are the asterisk playback sounds located? Does anyone know? i.e. transfer, tt-weasels |
| 11:20.34 | double_cheesburg | The directory? |
| 11:20.50 | Nobbie | asterisk-addons |
| 11:20.55 | double_cheesburg | thx |
| 11:20.58 | Nobbie | <PROTECTED> |
| 11:21.26 | Nobbie | i still extract the old discontinued asterisk-sounds packages |
| 11:22.02 | double_cheesburg | sweet |
| 11:23.25 | double_cheesburg | Looks like the audio format is .gsm, right? Anyone know of a good converter for MP3 or WAV to .GSM ? |
| 11:24.12 | mort_gib | sox |
| 11:26.11 | double_cheesburg | mort_gib : Thank you |
| 11:30.22 | Nobbie | asterisk shows incoming calls from a SIP peer as SIP/x.x.x.x where x.x.x.x is it's IP Address. How can i make it show as SIP/Name-Of-Peer-$id instead ? |
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| 12:36.31 | richardlynch | I would like to dynamically add a class to the musiconhold.conf file. I'm sure it's in the docs somewhere, but I'm having a tough time figuring out which bits to re-read. Any pointers? |
| 12:43.42 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 12:46.09 | [TK]D-Fender | richardlynch: nothing "dynamic". change the config file, issue a reload |
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| 12:56.09 | *** join/#asterisk rashed2020_ (n=shabati@67.205.245.208) |
| 12:56.19 | rashed2020_ | Hello everyone |
| 12:56.34 | rashed2020_ | Is there a way to route an FXO port over the internet |
| 12:56.35 | *** join/#asterisk stevetotaro (n=Steve@c-69-243-124-5.hsd1.md.comcast.net) |
| 12:57.10 | rashed2020_ | Actually nevermind. I realized that's not really relevant here |
| 12:57.43 | [TK]D-Fender | rashed2020_: You can provide ACCESS to it using a VoIP protocol |
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| 13:10.16 | *** join/#asterisk matrix1233 (n=Administ@196.203.44.3) |
| 13:10.36 | matrix1233 | hello |
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| 13:11.04 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
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| 13:11.29 | rashed2020_ | [TK]D-Fender: I'm trying to get a DID in a country that none of the VOIP providers I looked at offer. Is that the way to go? |
| 13:11.46 | rashed2020_ | The Asterisk box is going to be half way around the world |
| 13:11.56 | matrix1233 | i have a problem with installing my B410P |
| 13:12.04 | matrix1233 | any one can help |
| 13:12.29 | beek | ~ask |
| 13:12.30 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:13.20 | Gido-E | matrix1233 isn't that a isdn card, running on mISDN? |
| 13:13.45 | GreyFoxx | Can anyone recommend any open source SIP load testing software? I'm looking for something that I could use to simulate 500-1000 or more sip clioent connections and then start placing calls between each other ? |
| 13:14.20 | Gido-E | GreyFoxx sipp |
| 13:20.39 | *** join/#asterisk dr_gogeta86 (n=fisgro@81-208-88-100.ip.fastwebnet.it) |
| 13:21.10 | *** join/#asterisk pcrack (n=pcrack@122.53.131.237) |
| 13:21.55 | pcrack | can anyone recommend what model of HP server asterisk installed can be use for 50 users? |
| 13:23.00 | Gido-E | pcrack i think, anny new hardware wil do the trick for 50 uers. |
| 13:23.17 | MaliutaLap | just ensure sufficient RAM |
| 13:23.45 | Gido-E | i think 1 gig is enough. |
| 13:24.14 | pcrack | im thinking using this http://h20341.www2.hp.com/integrity/cache/452196-0-0-101-121.html |
| 13:24.46 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 13:24.47 | pcrack | and 4 gig of ram so for further expansion, and featured to be used like call recordings etc... |
| 13:24.58 | brad_mssw | err, i don't know if I'd go itanium |
| 13:25.37 | MaliutaLap | is that arch still alive? |
| 13:25.50 | tzafrir_laptop | Yes |
| 13:26.07 | brad_mssw | just go x86 (or x86_64, amd64), you're less likely to have issues as that is what most people use |
| 13:26.15 | tzafrir_laptop | In fact, HP still pushes it. And SGI's monster machines use it |
| 13:26.29 | tzafrir_laptop | The rest of the world has mostly abandoned it |
| 13:26.33 | MaliutaLap | I thought it died when x86_64 (or AMD64 ... whatever you want to call it) became the defacto standard |
| 13:26.46 | pcrack | nop ill go to dual core |
| 13:27.00 | MaliutaLap | dual core on which arch? |
| 13:27.17 | MaliutaLap | you are likely to have issues with build and support on and ia64 arch |
| 13:27.34 | pcrack | sorry i didnt see it.. |
| 13:27.45 | pcrack | maybe ill go on IBM..xSeries... |
| 13:27.53 | MaliutaLap | it's like going and buying a digium card and hoping to run it on openbsd ... the OS is great but has no support for those cards |
| 13:28.03 | pcrack | HP uses itanium |
| 13:28.18 | MaliutaLap | HP kit is still fine, you can get x86_64 machines from them |
| 13:28.19 | brad_mssw | hp has their proliant line which uses x86_64 |
| 13:28.43 | brad_mssw | if you want to stick with HP |
| 13:28.52 | MaliutaLap | I would recommend HP kit ... ILO is much better than say the Dell offering |
| 13:29.01 | pcrack | any other server recommendations? |
| 13:29.07 | MaliutaLap | !Dell |
| 13:29.12 | MaliutaLap | hates on Dell |
| 13:29.19 | brad_mssw | pcrack: have you determined which cards you might need, or are you just needing a pure SIP/IAX solution with no need for bringing in a PRI, or POTS lines? |
| 13:29.45 | pcrack | im just going to use TDM cards |
| 13:29.50 | pcrack | 800P to be exact |
| 13:29.54 | brad_mssw | pcrack: as I'd be more concerned if the cards will physically fit in the machine (enough slots), rather than the manufacturer of said machine |
| 13:29.56 | pcrack | 8 fxo port |
| 13:30.10 | pcrack | ic... |
| 13:30.19 | MaliutaLap | pcrack: you don't want some PRI/BRI solution? |
| 13:30.55 | brad_mssw | pcrack: you might be better off with the AEX800, rather than the TDM800 ... as most servers these days provide PCI-e risers, not PCI risers |
| 13:31.17 | pcrack | how about both...coz maybe it the future we will use PRi/BRI currently we are just using standard POTS |
| 13:32.33 | mort_gib | MaliutaLap: Why do you hate Dell?? |
| 13:32.38 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 13:33.45 | brad_mssw | pcrack: as long as you make sure the server can take 2 full-height PCI-e cards, you should be fine ... |
| 13:34.23 | MaliutaLap | mort_gib: I have had to work with their servers in the past and getting some of the DRAC stuff and the monitoring stuff working under linux wasn't easy ... and I just didn't like the way they did things. I much prefer working with HP kit, even if it's not perfect |
| 13:34.25 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:34.27 | brad_mssw | pcrack: other than that, you may want at least a raid 1 for reliability, maybe dual redundant power supplies ... can't really suggest much else ... minimum specs from the manufacturer would be sufficient |
| 13:35.11 | pcrack | can u give me a good server link on HP that we can use based on my requirements.. |
| 13:35.24 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:35.27 | mort_gib | MaliutaLap: Hmm, I'm stuffed here, HP support is done by the biggest crooks in town |
| 13:35.49 | tzafrir_laptop | pcrack, besides the analog trunk, what else does the server need to do? |
| 13:36.12 | brad_mssw | pcrack: you'd probably need to call them, HP gives basically 0 information on their expansion slots ... like the ProLiant DL160 G5 Server lists 2 expansion slots ... doesn't say PCI-e, doesn't say full or half height, etc ... useless info |
| 13:36.32 | [TK]D-Fender | rashed2020_: Sure you can put your own server in there. Plenty of other gateway devices you could use as well without a full server to manage |
| 13:37.21 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 13:37.31 | pcrack | its just for small business use..the usual office type features...call recordings,queue,sip/iax trunking and analog trunks |
| 13:37.43 | MaliutaLap | mort_gib: I tend to work in areas where there is more than one choice |
| 13:38.04 | Katty | hmm |
| 13:38.22 | Katty | Obama is lifting the restrictions on federal funding of stem cell research today. |
| 13:38.38 | MaliutaLap | that's a good thing |
| 13:38.53 | Katty | very. |
| 13:38.56 | MaliutaLap | stem cell research holds promise for a number of areas |
| 13:39.17 | mort_gib | MaliutaLap: Uhm, don't much like HD servers, seems slow and heavy and expensive |
| 13:39.20 | pcrack | how about IBM servers? any model recommendations on my requirements |
| 13:39.22 | MaliutaLap | as the recipient of a Bone Marrow Transplant I appreciate that more than most |
| 13:39.24 | mort_gib | s/HD/HP |
| 13:40.14 | Katty | I wish people would do a little homework on stem cell research before bad-mouthing it |
| 13:40.19 | jaytee | hugs Katty |
| 13:40.23 | jaytee | good mornin! |
| 13:40.42 | beek | morning jaytee , [TK]D-Fender , et al |
| 13:40.48 | jaytee | mornin beek |
| 13:40.49 | MaliutaLap | Katty: I wish some of the US states would lift bans on giving GCSF to healthy people, it makes it more likely to get donors for BMT (given that most transplants these days are adult stem cell transplants) and better for the recipients |
| 13:40.56 | Katty | hugs jaytee |
| 13:41.44 | MaliutaLap | my donor was a native american woman, she had to have a dual asperate (hence I got actual marrow not adult stem cells) because of where she lived |
| 13:41.47 | Katty | MaliutaLap: i'm just tired of people calling it an ethical debate. |
| 13:41.50 | *** part/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 13:42.01 | Katty | MaliutaLap: it is NOT an ethical debate, we've found ways to reprogramming skin cells. |
| 13:42.10 | *** join/#asterisk LuisTorres (n=chatzill@a213-22-94-93.cpe.netcabo.pt) |
| 13:42.17 | Katty | MaliutaLap: but southern missouri wants nothing of that. they want to turn this into a Killing Babies debate. |
| 13:42.43 | brad_mssw | pcrack: doesn't seem like IBM discloses full vs half height in their specs either |
| 13:42.44 | jaytee | without things like stem cell research we won't be able to overcome the genetic diseases that we've only encouraged through science letting people with chronic illness survive to adulthood and reproduce passing on those genetic traits, i.e. diabetes |
| 13:42.49 | MaliutaLap | the christian right is the christian right, no matter where they are |
| 13:42.50 | brad_mssw | pcrack: otherwise the x3350 looks fine |
| 13:43.13 | Katty | the christian's can stuff it "where the sun don't shine" for all i care (= |
| 13:43.35 | MaliutaLap | Katty: you might need to explain that one to them |
| 13:43.56 | MaliutaLap | Katty: I suggest showing them a replica of goatse in explanation ;) |
| 13:43.56 | jaytee | Jesus is just all right with me but most of his followers are a stuffy, self-righteous bunch I have no use for. |
| 13:44.15 | mmlj4 | I get off on insulting folks I don't know and don't understand |
| 13:44.23 | MaliutaLap | I used to argue with the evangelicals for fun |
| 13:44.44 | Katty | religion is fine as long as you keep it a personal thing. |
| 13:44.54 | Katty | i don't appreciate hearing about it. lol |
| 13:44.56 | beek | Amen to that! |
| 13:45.21 | Katty | and i REALLY don't appreciate it when it gets in the way of curing diseases |
| 13:45.21 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 13:45.57 | MaliutaLap | there were these brothers that I did history with at uni ... they understood that in history you need more than one source so things can be verified, but still argued the bible was the literal word of god and all the proof that was needed |
| 13:46.00 | Katty | how do you feel now missouri?! you just banned the cure of HIV |
| 13:46.31 | MaliutaLap | I don't live in US, so I don't give a rats what they think they won or didn't |
| 13:46.40 | Katty | well i can understand how a christian could sorta kinda believe the bible is still in tact. |
| 13:46.51 | Katty | but any reasonable person would have to agree it's been altered. |
| 13:47.01 | MaliutaLap | CSIRO can still do it's research ... that's the same CSIRO that bought us 802.11G |
| 13:47.06 | Katty | whether by accident in translation, or on purpose by the church |
| 13:47.28 | Katty | you know in Denmark, it's more embarassing to talk about God than it is to go running naked through the streets? |
| 13:47.35 | coppice | jaytee: diabetes has more to do with diet than genes |
| 13:47.43 | mmlj4 | well, I'm not embarrassed |
| 13:47.53 | Katty | just some perspective ;) |
| 13:48.08 | Katty | coppice: it really depends on the type. |
| 13:48.19 | Katty | coppice: your'e right, a lot of diabetics are over weight and don't watch what they eat... |
| 13:48.24 | jaytee | I have alot of the "literal word of God" types around me. I point out that in Mark when Jesus came before Pilate he was wearing a scarlet robe and in Matthew it was a purple robe. How can an omniscient being make a mistake like that? |
| 13:48.26 | Katty | coppice: but there are still children born diabetic. |
| 13:48.38 | Katty | coppice: and that's a different level of severity |
| 13:48.58 | mmlj4 | what embarasses me is when newbies who think they know something technical try to tell a guru how things ought to be |
| 13:49.02 | coppice | true, but until countries become affluent enough to eat badly, diabetes is a very rare disease |
| 13:49.05 | mmlj4 | like you guys |
| 13:49.27 | Katty | jaytee: i'm still trying to figure out that All Loving god bit. |
| 13:49.28 | jaytee | coppice, the genetic trait that allows for early onset diabetes is passed through generations via the genes. Although the exact cause is unknown and suspected to be viral, it's a genetic susceptibilty. |
| 13:49.45 | coppice | the market for blood glucose meters in the west is huge. with india and china becoming more affluent, markets are expanding quickly :-) |
| 13:49.54 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:50.06 | fcois93 | how can I use fastAGI ? |
| 13:50.30 | Katty | that's a very vague question |
| 13:50.32 | jaytee | coppice, I will agree that western diets exacerbate the situation but that is adult onset diabetes, which is a different form of the disease. |
| 13:51.02 | Katty | coppice: i think what jaytee's trying to say is it's a REAL disease. not an eating disorder :P |
| 13:51.28 | fcois93 | I saw how to send an agi to another server. how another server listen for that? it need asterisk ? |
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| 13:52.59 | jaytee | Katty, right and there are two distinct forms of the disease, one that strikes during childhood and is due to damage of the Islets of Langerhass section of the pancreas that produces insulin, the other is adult onset diabetes and is generally due to obesity, poor diet high in glucose and carbohydrates, i.e. American food in general. |
| 13:53.09 | [TK]D-Fender | fcois93: http://www.voip-info.org/wiki-Asterisk+FastAGI |
| 13:53.19 | tamiel | fcois93: fastagi : you need a socket server listening |
| 13:53.46 | coppice | katty: that's like saying stupidity is a REAL disease, because it runs in families |
| 13:53.47 | fcois93 | [TK]D-Fender: I saw it, but how to listen for ? |
| 13:54.05 | [TK]D-Fender | fcois93: As you were told, YOU write a program that listens on a socket |
| 13:54.10 | Katty | coppice: i think stupidity is a disease that runs in families :P |
| 13:54.32 | Katty | hugs [TK]D-Fender |
| 13:54.48 | [TK]D-Fender | hugz teh Katty |
| 13:54.51 | Katty | [TK]D-Fender: obama lifts the stem cell funding ban today! horay!!! |
| 13:55.23 | [TK]D-Fender | Katty: thats only so he can resume the creation of his Socialist clone Army! the Empire begins! |
| 13:55.39 | Katty | [TK]D-Fender: that's an HIV FREE empire, thankyouverymuch! |
| 13:55.42 | mmlj4 | if that weren't true, it would be funny |
| 13:56.12 | coppice | few people susceptible to blood sugar problems don't need medical care unless they abuse themselves. their susceptibility is just like being genetically tall. that's something relatively harmless, as long as you watch out for low beams. |
| 13:57.19 | pcrack | on the forum someone used AEX800 on a HP proliant DL360 Server? is that fine |
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| 14:01.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:02.57 | jaytee | coppice, I know you're a very intelligent person, especially with regard to DSP, voip, et.al but go read up on diabetes a little more. Your claim that's ALL due to diet is sadly misinformed and misunderstood. There are people on this planet that regardless of their diet would die without daily injections of insulin. 99.999999% of doctors would agree diabetes is a disease. Show me your P.H.D. in medicine. |
| 14:03.41 | Katty | show me your phd |
| 14:03.44 | *** join/#asterisk Chex (i=chex@random.supermario.org) |
| 14:03.44 | Katty | or the teddy gets it |
| 14:03.48 | Katty | WITH A BANANA |
| 14:03.49 | jaytee | hehehe |
| 14:04.28 | Katty | teddy goes on to have an abortion, and gets excommunicated from the catholic church. |
| 14:04.42 | Katty | banana is supported by the vatican. |
| 14:05.11 | coppice | jaytee: I didn't say it was all diet, but the majority are. that's what diabetes rates closely match bad diet patterns. Its something I happen to follow for slightly obtuse reasons. |
| 14:05.45 | Katty | http://www.gearfuse.com/wp-content/uploads/andrew/2_jan07/154735364_c175cda85b_1.jpg <- coppice |
| 14:06.51 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 14:07.02 | jaytee | coppice, I would agree that in the U.S. most cases of adult onset diabetes (still one form of the disease) ARE caused by bad diet and especially because of obesity. (BIGGIE SIZE ME!!! I'll take the double half-pounder and 3 extra orders of cheese fries please! Oh, and a 96oz size soda full of high-fructose corn syrup!") |
| 14:07.47 | jaytee | hahahaha, a headless teddy USB drive? priceless. |
| 14:07.56 | Katty | isn't it cute? |
| 14:07.56 | jaytee | I still like my TARDIS usb hub |
| 14:08.07 | [TK]D-Fender | jaytee: Actually its more like the bacon-triple-cheezeburger, uber cheeze fries, apple turnover, oh... and a Diet Coke... I'm trying to watch my weight! |
| 14:08.20 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 14:08.36 | coppice | jaytee: do you realise the bonanza in blood glucose meters and insulin the pharma companies are expecting from China, as they eat worse and worse? glucose meters are all in English right now, but everyone is preparing Chinese display models :-\ |
| 14:08.50 | jaytee | [TK]D-Fender, ummm, considering poutine I'd say you guys are just one step behind us if it weren't for all the red wine you consume. :-) |
| 14:08.55 | [TK]D-Fender | jaytee: And you're right it is a disease, and there are people who are going to have it regardless. Its the other 99.999% that give them a bad name :p |
| 14:09.02 | jaytee | coppice, so now you know which stocks to buy :-) |
| 14:11.31 | Katty | drop kicks polycom501 through window |
| 14:11.53 | jaytee | I am not a WoW addicted couch potato, I'm an office chair, internet Zynga Mafia Wars addict. I think that's worse due to the lower back strain and the poor circulation in my legs. |
| 14:12.00 | Katty | hey now |
| 14:12.11 | Katty | let's not stereotype wow addicts. |
| 14:12.28 | jaytee | Has anyone else here ever had their ass fall asleep? It's an interesting sensation.... or lack thereof. |
| 14:12.44 | Katty | cant' say i've had that happen |
| 14:13.08 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:13.44 | Katty | not even during a 5 hour nax raid :P |
| 14:13.53 | jaytee | Katty, oh c'mon! "everyone" knows ALL WoW players are magenta mohawked, multiple face peircings tribal tattooed pot smokers. |
| 14:14.02 | jaytee | :-) |
| 14:14.05 | Katty | looks at her hair |
| 14:14.17 | Katty | hmm. i still seem to be brunette |
| 14:14.25 | jaytee | I was kidding!!! |
| 14:14.28 | Katty | ;) |
| 14:15.17 | jaytee | The only reason I'm not a WoW addict is because I deliberately avoid playing the game. It rocks too much that I know that after the "second toke" I'm hooked for life. |
| 14:15.43 | jaytee | and God forbid I ever buy a console with Guitar Hero |
| 14:15.56 | Katty | never could get into guitar hero |
| 14:16.34 | coppice | WoW is a true killer application. it has killed several in korea, china and here |
| 14:19.41 | [TK]D-Fender | Katty: I tried it twice, my hand cramped really bad.... never again :p |
| 14:20.16 | [TK]D-Fender | is doing fine with 19 years playing the real thing :) |
| 14:20.17 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 14:20.57 | coppice | haven't you seen South Park? real guitars are gay |
| 14:21.08 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:22.08 | [TK]D-Fender | coppice: No.... I haven't :) |
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| 14:56.03 | pcrack | which is more good TDM800P or AEX800? |
| 14:56.19 | [TK]D-Fender | pcrack: YES |
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| 14:59.09 | pcrack | i mean which more you recommend TDM800P or AEX800? |
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| 15:00.45 | [TK]D-Fender | pcrack: Depends what kind of slot you have. |
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| 15:03.01 | pcrack | im going to use this server? |
| 15:03.54 | pcrack | http://www-03.ibm.com/systems/x/hardware/rack/x3250m2/specs.html |
| 15:05.19 | [TK]D-Fender | pcrack: Well what do YOU think? |
| 15:05.44 | *** join/#asterisk _BBV_ (n=boris@213.138.71.254) |
| 15:06.50 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:07.23 | pcrack | i think so..im asking just to make sure |
| 15:07.48 | pcrack | is it? |
| 15:08.01 | [TK]D-Fender | pcrack: What slots does it have? |
| 15:09.22 | djMax | is it possible to transfer an inbound call using Asterisk with a T1 and NOT take up any channels afterwards? |
| 15:09.46 | pcrack | its said this Riser assembly PCI-E (4911) |
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| 15:11.29 | [TK]D-Fender | pcrack: So what do you think about the TDM800P then? |
| 15:11.52 | [TK]D-Fender | djMax: With PRI and a carrier that supports 2BCT, yes. |
| 15:12.15 | pcrack | it will work also? |
| 15:12.43 | [TK]D-Fender | pcrack: its &#^$ing PCI. there are no PCI slots in that damn server. |
| 15:13.18 | [TK]D-Fender | grumbles that for some even giant flashing neon signs aren't enough. |
| 15:13.43 | pcrack | what do you recommend? |
| 15:14.03 | pcrack | sorry im very noob on hardware server |
| 15:14.29 | djMax | so in theory if I ask XO (our T1 provider) if they support 2BCT they'll not think I'm from Mars? |
| 15:14.53 | angryuser | djMax, they will if you ask the wrong person ;) |
| 15:14.54 | pcrack | so ill go on AEX800P |
| 15:15.03 | pcrack | AEX800 I mean |
| 15:15.15 | [TK]D-Fender | pcrack: Server? This is a stupid basic slot on any PC. PCI is NOT PCI-E |
| 15:15.27 | djMax | I think they will even if I do, but basically was saying is there a "higher level question" that will get me to the right person while still being in Telco speak. |
| 15:15.52 | [TK]D-Fender | djMax: Prepare to say "get me a level 2+ tech" |
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| 15:20.46 | Katty | hugs anthm |
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| 15:21.24 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 15:21.33 | anthm | hi |
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| 15:23.25 | qp | afternoon |
| 15:23.34 | *** part/#asterisk matrix1233 (n=Administ@196.203.44.3) |
| 15:25.34 | qp | has anyone heard of conference room issues with garbled sound but not during normal calls? |
| 15:25.55 | mort_gib | qp:Timing issue?? |
| 15:26.29 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 15:26.32 | qp | worried about this. normal calls are fine. but I am running it in vmware, and have been told its fine these days for * |
| 15:26.38 | *** join/#asterisk obdz (n=bdz@tlg.net54.hu) |
| 15:26.41 | obdz | hi |
| 15:27.34 | mort_gib | qp: You need to get some kind of timing device up and running, like ztdummy or similar |
| 15:27.56 | qp | how can I test if there is one or not? |
| 15:28.24 | obdz | asterisk 1.4.18. i have a h323 trunk to a definity but when i call TO the definity i got a 80secs DNS SRV lookup timeout. do you know how to resolv this ? |
| 15:28.51 | obdz | DNS server ofcourse is set and from the os i have no problem with it |
| 15:28.51 | mort_gib | qp: You just need to set it up.... |
| 15:29.19 | qp | modprobe ztdummy gives me nothing |
| 15:29.57 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 15:30.16 | mort_gib | qp: There you go! Go set it up :-) |
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| 15:30.42 | qp | :) |
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| 15:35.48 | qp | mort, dmesg seems to show its running: |
| 15:35.55 | qp | Zaptel Version: 1.4.10.1 |
| 15:35.55 | qp | ztdummy: RTC rate is 1024 |
| 15:36.18 | angryuser | qp, it's rather goot if modprobe ztdummy gives you no output, to be sure if ztdummy is loaded type "zap show status" in cli or "dahdi show status" |
| 15:36.23 | angryuser | good* |
| 15:37.51 | mort_gib | qp: Then I don't know, it would have been a classic scenario for "missing timing" that normal calls were fine... |
| 15:38.04 | qp | yeah :/ |
| 15:38.32 | *** join/#asterisk siera08 (n=chatzill@218.207.141.90) |
| 15:39.30 | angryuser | qp, some kernels work bad with zaptel :( i had some situation where the only solution was to add a souce of timing |
| 15:39.45 | angryuser | zaptel dummy i mean* |
| 15:39.52 | qp | odd that this is only in conf rooms? |
| 15:40.11 | angryuser | qp, not at all, what do you call a normal call ? voip ? |
| 15:41.15 | qp | yeah, we use conference rooms to connect 2 voip calls, like a managed transfer from our softphone. |
| 15:43.28 | qp | asterisk01*CLI> zap show status |
| 15:43.28 | qp | Description Alarms IRQ bpviol CRC4 |
| 15:43.28 | qp | ZTDUMMY/1 (source: RTC) 1 UNCONFIGUR 0 0 0 |
| 15:44.17 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 15:44.32 | qp | not sure if thats a good thing angryuser :) |
| 15:44.38 | angryuser | qp, yes it's classic, you have a timing issue, try to execute zttest |
| 15:44.46 | qp | ok |
| 15:44.49 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 15:45.51 | qp | loads of 99.x % |
| 15:46.08 | angryuser | qp, paste the average min and max |
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| 15:47.47 | angryuser | qp, ctrl +c to complete |
| 15:48.16 | qp | heh one result went mad |
| 15:48.17 | qp | --- Results after 47 passes --- |
| 15:48.18 | qp | Best: 99.998 -- Worst: -6359.631 -- Average: -37.562894, Difference: 237.414602 |
| 15:48.30 | qp | most are 99.9x |
| 15:49.09 | qp | i'll do another 20 |
| 15:49.20 | angryuser | qp, something is definatly wrong , normally values dont ever go under 99. |
| 15:49.31 | angryuser | definetely |
| 15:50.03 | angryuser | qp, pastebin all output |
| 15:50.54 | qp | http://pastebin.com/d152fc152 |
| 15:50.57 | qp | 2 runs |
| 15:54.11 | angryuser | qp, second one is not so bad , but not great either , do you use g729 ? |
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| 15:57.30 | murraytm | is there any trick to getting the i option for the Dial app to work on a PRI? i'm using asterisk 1.4.22 with sangoma hardware. |
| 15:58.32 | qp | hmm, I think so angry |
| 15:59.13 | angryuser | qp, try to use meetme without transcoding in local environement |
| 15:59.30 | brunner | is there a valid way to comment something out in extensions.conf? |
| 15:59.38 | Qwell | brunner: ; |
| 15:59.41 | brunner | thanks |
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| 16:01.00 | qp | thanks angry, will look into that. would timer issues cause calls to not come in properly too? ie sometimes get number unavaialble? or is that down to the voip provider |
| 16:02.32 | angryuser | qp, not a time issue, voip provider or your config |
| 16:02.37 | angryuser | timer* |
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| 16:04.30 | qp | g711 we use |
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| 16:50.38 | dwery | hi. I need to make a Dahdi channel to go off-hook, dial a Flash and then a number. Is that possible? |
| 16:51.19 | [TK]D-Fender | dwery: Yes |
| 16:51.51 | dwery | [TK]D-Fender: nice to know. now let's hope the usbfxo supports that! |
| 16:53.22 | *** join/#asterisk mintos (n=mvaliyav@123.237.78.9) |
| 16:55.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:57.29 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 17:01.41 | dwery | [TK]D-Fender: should I use Flash() after Dial() ? |
| 17:01.48 | thehar | is SetCallerPres() from 1.0 now Set(CALLERPRES()=) in 1.4.22? |
| 17:02.53 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:04.12 | *** join/#asterisk Magicblaze007 (n=sony@garpc.cs.fsu.edu) |
| 17:05.11 | [TK]D-Fender | dwery: No, because after Dial is too late. |
| 17:05.38 | dwery | [TK]D-Fender: mm then I'm missing something.. :( |
| 17:05.47 | *** join/#asterisk mitchGaffigan (n=mitch@c-24-12-44-14.hsd1.in.comcast.net) |
| 17:05.52 | [TK]D-Fender | dwery: call-file -> Dial(zap/1), dump into dialplan, then call flash, SendDTMF, then bridge to wherever |
| 17:06.31 | dwery | [TK]D-Fender: it seems that Dial() waits for the other party to answer |
| 17:06.53 | [TK]D-Fender | dwery: Depends |
| 17:07.34 | dwery | [TK]D-Fender: should I try the timeout option? |
| 17:08.06 | [TK]D-Fender | dwery: I have jsut told you the process for this. |
| 17:08.07 | mitchGaffigan | Were there any changes made to the MailboxExists function in 1.6? |
| 17:08.13 | [TK]D-Fender | dwery: Please actually read it |
| 17:08.26 | dwery | [TK]D-Fender: I've read it but I'm probably missing the critical part. |
| 17:08.28 | [TK]D-Fender | mitchGaffigan: What does the ChangeLog say? |
| 17:08.33 | [TK]D-Fender | dwery: CALL-FILE |
| 17:09.20 | eppigy | hello |
| 17:09.22 | eppigy | i am dave |
| 17:10.30 | dwery | [TK]D-Fender: ty, I'll read docs on call-file |
| 17:12.38 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 17:12.48 | mitchGaffigan | [TK]D-Fender: only that it was converted to a dialplan function |
| 17:13.12 | [TK]D-Fender | mitchGaffigan: Then there you ahve it |
| 17:13.28 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:14.29 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 17:14.55 | mitchGaffigan | would that cause it to not respond to the j argument? |
| 17:15.37 | mitchGaffigan | or require any chagnes to the dialplan for it to work correctly? |
| 17:15.52 | mitchGaffigan | (relative to it's usage in 1.4) |
| 17:17.20 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:19.20 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 17:20.38 | [TK]D-Fender | mitchGaffigan: Show us what you're doing. |
| 17:23.19 | mitchGaffigan | http://pastebin.com/m565687d |
| 17:24.23 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 17:25.05 | [TK]D-Fender | mitchGaffigan: And this is 1.6? |
| 17:25.14 | mitchGaffigan | yes |
| 17:25.23 | [TK]D-Fender | mitchGaffigan: go read that apps instructions again. |
| 17:26.24 | mitchGaffigan | where would I find that (other than on a site like voip-info)? |
| 17:26.36 | [TK]D-Fender | mitchGaffigan: "core show application mailboxexists" |
| 17:28.19 | mitchGaffigan | "Options: (none)" meaning do your own jump? |
| 17:29.59 | [TK]D-Fender | mitchGaffigan: meaning there are no "options" any more and you should be suing the channel variable for this. Also that "|" is not a valid delimiter in 1.6 which you seemed to have missed |
| 17:30.00 | *** join/#asterisk Zeryl (n=Zeryl@97-87-122-210.dhcp.stls.mo.charter.com) |
| 17:30.26 | mitchGaffigan | thanks |
| 17:30.33 | Zeryl | Hey all, not sure how to phrase my question for google, or even the correct terminology, so I'm curious if someone could assist me in lettingme know if what I want to do is possible |
| 17:30.38 | *** part/#asterisk Magicblaze007 (n=sony@garpc.cs.fsu.edu) |
| 17:30.59 | Zeryl | the company i work for (6 of us) all work from home, and currently outsource our PBX to a company, so that when someone calls for support, it round robins the support staff |
| 17:31.23 | Zeryl | is that something asteisk could do i.e. each person logs in w/ a sip phone, and when someone calls the main support number, it'd go out to one of us? |
| 17:31.33 | kerx | any suggestions on a good method to integrate an Order Entry system and call recording w/ Monitor() so that call recordings from agents who place entries into the Order Entry web-interface have an attached call recording in the database for that person who they were speaking to? |
| 17:31.49 | mitchGaffigan | Zeryl: look at http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue |
| 17:32.08 | Katty | i hurt my hand swapping motherboards )= |
| 17:32.10 | Zeryl | mitchGaffigan: ty! I'll take a look now |
| 17:32.14 | Katty | shakes fist |
| 17:32.19 | Katty | damn you manual labor! |
| 17:32.48 | eppigy | D: |
| 17:32.56 | Talkradio | heh |
| 17:33.02 | Talkradio | you go girl |
| 17:33.16 | Talkradio | nice to see someone not afraid of alittle hard work |
| 17:33.50 | [TK]D-Fender | Zeryl: Quick answer : Yes |
| 17:34.08 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 17:34.35 | mitchGaffigan | Kerx: Record the call, then move the file to a defined location and have your entry system attach that file once the call is completed. |
| 17:34.38 | Zeryl | ty [TK]D-Fender, going to look into it then, as it's gotta be cheaper than what we/they are paying currently (already have servers sitting around doing nothing) |
| 17:35.09 | Katty | Talkradio: heh ;) |
| 17:35.15 | mitchGaffigan | Zeryl: if you are trying to do something quickly, take a look at Trixbox |
| 17:35.22 | [TK]D-Fender | ... |
| 17:35.24 | [TK]D-Fender | EW |
| 17:35.29 | mitchGaffigan | sorry... |
| 17:35.40 | mitchGaffigan | It works to a point. |
| 17:36.01 | [TK]D-Fender | mitchGaffigan: Sure thing... Vlad |
| 17:36.04 | Zeryl | we have pretty basic requirements, so i'm sure almost anything would work |
| 17:36.08 | mitchGaffigan | lol |
| 17:36.24 | *** part/#asterisk strk (n=strk@ip-123-78.static.adsl.cheapnet.it) |
| 17:37.12 | Talkradio | i have 4 trixbox setups out there :) all working fine but watchout for phones not working when internet goes down |
| 17:39.03 | kerx | mitchGaffigan, how would I know if the order through the web-based order entry has been completed to know to attach it to example order id #1243 |
| 17:40.33 | Zeryl | based on my limited knowledge, i'd say have asterisk save the file out to a drive, with the telephone number + timestamp, and when you click "complete order" or something, it checks for the presnece of that file, and attaches it |
| 17:40.37 | Zeryl | imo |
| 17:40.59 | kerx | Zeryl, i like this suggestion, i'm only a bit worried about people who call in w/ Blocked Numbers |
| 17:40.59 | Zeryl | from a programming perspective at least (don't know the full capabilities of asterisk tho, so it may not be possible) |
| 17:41.20 | kerx | yeah, i'm trying to figure out that |
| 17:41.22 | Zeryl | depending on your situation, you may be able to get ANI delivery (especially if it's an 800 number) |
| 17:42.27 | kerx | hrm. i'm not sure i know what that is, also quickly reading the Wikipedia page. Is it some sort of caller id un-block? ;-) |
| 17:43.03 | Zeryl | it's automatic number identification (essentially), it's the REAL number that calls. Since the 800 numbers are paying for you to call them, they apparently have the right to know who's calling them, so they use ANI |
| 17:43.34 | kerx | awesome |
| 17:43.42 | kerx | unfortunately 800 numbers are expensive |
| 17:43.43 | Zeryl | i would imagine that some sip providers also allow you to see ANI, but I can't guarentee that, or have any resoning to back that up |
| 17:43.55 | kerx | yeah, i hear ya. i'll check that out |
| 17:44.08 | Zeryl | i'd be willing to bet your telephone provider offers it (probably at a price though) |
| 17:44.11 | kerx | i read a bit on AGI's, and it seemed promising.... |
| 17:44.37 | mitchGaffigan | I would have the exension of the user attached to their user account and use manager to get the call id when the user clicks a button on the UI to retrive the filename. |
| 17:44.44 | kerx | i was thinking of the possibility of having agent's sit in a Meet-me room, and when a inbound call comes in then through some AJAX pop there web-interface w/ the call |
| 17:44.49 | [TK]D-Fender | kerx: Expensive? Never seen them cost anything more than normal LD... |
| 17:45.07 | kerx | [TK]D-Fender, AFAIK 800,866,877, etc. are per-minute basis |
| 17:45.24 | [TK]D-Fender | kerx: Yes, just like normal LD |
| 17:45.47 | Zeryl | mitchGaffigan's idea is better. if asterisk supports querying current calls based on extension, seems like a much safer method |
| 17:46.00 | kerx | Also, I can't be bound to just a 800 number for the inbound calls. |
| 17:46.08 | kerx | I'm going to be routing them lots of SIP and IAX2 calls |
| 17:46.28 | kerx | Yeah, I like mitch's Idea also |
| 17:46.33 | kerx | I think that's going to be the route |
| 17:46.35 | mitchGaffigan | so, the complete thing is: call comes in, user clicks button to say they have a call, order entry server logs in to AMI to get the call id, the call finishes recording and is saved to a web accessible directory with the call id as the filename, the filename is stored in the database for later retreval |
| 17:47.03 | [TK]D-Fender | kerx: Just 800 number? HUH? And then SIP & IAX2 are PROTOCOLS. this says nothing of the kind of calls being passed over it. |
| 17:47.08 | kerx | Seems promising |
| 17:47.23 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 17:47.45 | kerx | [TK]D-Fender, What I mean is that the call center will be receiving inbound calls through various methods. They do have 800 number but also they have other forms of receiving calls. |
| 17:47.45 | *** join/#asterisk socram (i=c86c8936@gateway/web/ajax/mibbit.com/x-c1829a15676f4885) |
| 17:48.02 | kerx | I'm trying to find an all-around solution if one exists |
| 17:48.05 | [TK]D-Fender | kerx: "800 #" is not a way of receiving calls. |
| 17:48.13 | kerx | And so far I like this idea |
| 17:48.16 | [TK]D-Fender | kerx: Stop mixing this up. |
| 17:48.21 | *** join/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca) |
| 17:48.37 | [TK]D-Fender | kerx: 800 # is a BILLING concern |
| 17:48.42 | mitchGaffigan | I don't think the call protocol or sourse is an issue |
| 17:48.48 | mitchGaffigan | *source |
| 17:48.53 | kerx | Ok, I may have mixed it up sorry |
| 17:49.16 | kerx | I like the above idea though :) |
| 17:49.24 | ]technophreak[ | Is there a way to have registration keep trying when handle_response_register gets a 404 ? |
| 17:49.29 | mitchGaffigan | thanks... it is similar to what we use |
| 17:49.47 | dwery | [TK]D-Fender: it seems to work, I had to play a bit with the timings. I now need to understand how to open a SIP channel toward a registered user and bridge to it |
| 17:50.04 | [TK]D-Fender | dwery: ... DIAL |
| 17:50.26 | dwery | [TK]D-Fender: d'oh... |
| 17:50.35 | kerx | Agent 123 receives call, customer seems interested, he begins a order-entry. The web script logs in through AMI to receive the call ID being recorded. Then keeps that recorded in the database. When the call is completed and the order is completed, some daemon that monitors the directory comes in and grabs the recording and attaches it into the database |
| 17:50.53 | kerx | mitchGaffigan, What do you use in AMI to see which call ID the Rep is on? |
| 17:50.58 | kerx | If you don't mind me asking |
| 17:51.45 | socram | i need to make an ivr for a bank-like org. People will call and make transactions over the phone, check account status, billing status, etc. how much is asterisk suitable for this? |
| 17:52.37 | mitchGaffigan | socram: AGI |
| 17:52.41 | dwery | [TK]D-Fender: but if I make an agi or something like that in order to generate the call-file on-the-fly, the SIP channel to the phone should already be established. So I need to find it and bridge to it, right? |
| 17:52.45 | kerx | socram, seems very suitable once you become pretty efficient and knowledgeable of the AGI |
| 17:53.12 | kerx | mitchGaffigan, maybe all of the IVR i'm doing should be an AGI script instead of the dialplan? |
| 17:53.42 | mitchGaffigan | you could... but why? |
| 17:54.23 | mitchGaffigan | kerx: Command with the command of core show channels |
| 17:54.51 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:54.52 | kerx | got it |
| 17:54.53 | mitchGaffigan | sorry, sip show channels |
| 17:54.58 | mitchGaffigan | that has call id |
| 17:55.02 | kerx | yea |
| 17:55.03 | kerx | i see now :0 |
| 17:55.05 | kerx | slaps himself |
| 17:55.10 | kerx | ok i'm going to go through w/ this |
| 17:55.19 | kerx | i would prefer not to make an AGI |
| 17:55.28 | mitchGaffigan | what is your order entry system written in? |
| 17:55.41 | kerx | it's PHP and Dojo javascript framework |
| 17:56.03 | guax | kerx, zend? |
| 17:56.10 | kerx | guax, no PHP framework |
| 17:56.21 | guax | humm |
| 17:56.23 | kerx | i'm not good at any of those unfortunately |
| 17:56.27 | mitchGaffigan | you're in luck, PHP has a full API already written |
| 17:56.27 | kerx | just basic PHP |
| 17:56.42 | Zeryl | i don't blame you there kerx, i've tried to get into the whole MVC w/ zend/cake/etc, and I can't |
| 17:56.51 | mitchGaffigan | see http://code.google.com/p/asterisk-php-api/ |
| 17:56.58 | guax | uses Zend Framework, PostgreSQL and PhpAgi |
| 17:57.20 | kerx | mitchGaffigan, the only reason i didn't want to make an AGI is because I like to continue to use asterisk's queue mechanism w/ rrmemory |
| 17:57.24 | Zeryl | don't get me wrong, the zend framework looks to be really strong, I just can't do it :( |
| 17:57.30 | guax | Zeryl, do you program in java or another OO language? |
| 17:57.31 | kerx | i'd hate to have to create my own ACD queue inside the agi |
| 17:58.04 | Zeryl | i'm a php dev, know a little bit for a couple other languages, but none as comfortably as php |
| 17:58.29 | mitchGaffigan | who said anything about AGI? |
| 17:58.41 | kerx | what are u talking about then? |
| 17:58.48 | kerx | OH! |
| 17:58.50 | kerx | slaps himself |
| 17:58.51 | kerx | you said API |
| 17:58.56 | mitchGaffigan | oh, that was in response to dwery about accessing a bank system |
| 17:58.58 | kerx | ok, it looks like my 4 hours of night sleep wasn't enough |
| 17:59.01 | Talkradio | wow for a sec i felt my wallet getting tugged on when i thought you typed aig heh |
| 17:59.03 | mitchGaffigan | same here |
| 17:59.05 | kerx | nevermind me |
| 17:59.06 | kerx | sorry |
| 17:59.56 | siera08 | Using cmd dial with 'm' or 'r' option, i have problem in asterisk 1.4.18. |
| 18:00.23 | siera08 | if i call internal phone and he cancel the call, i can hear some voice message. |
| 18:00.49 | siera08 | but when i call external phone through sip/zap/iax trunk, i can't hear voice message which other pbx(asterisk, too) plays. |
| 18:01.07 | siera08 | In this case, after i heared "connecting..." from other pbx, called party should be called. |
| 18:01.35 | siera08 | but i can't hear anything(if 'm' option is enabled, moh plays) before called party accepts or cancels the call. |
| 18:01.58 | mitchGaffigan | siera08: check that your codecs are compatible |
| 18:02.02 | siera08 | i want to hear sound message that other pbx plays before calling the phone. |
| 18:02.12 | Gido-E | in sip.conf is it limitonpeer or limitonpeers ? |
| 18:02.22 | siera08 | sorry, me is very poor english. |
| 18:02.45 | mitchGaffigan | Gido-E: plurarl |
| 18:02.56 | mitchGaffigan | *limitonpeers |
| 18:03.06 | RobH | Anyone offhand know what the entry is to increase the default wait time for entering a name in the directory? |
| 18:03.10 | siera08 | mitchGaffigan: thank u for your kind. codec has relation with it? |
| 18:03.20 | ]technophreak[ | Is there a way to have registration keep trying when handle_response_register gets a 404 ? |
| 18:03.25 | Gido-E | ok, i see a lot of examples with limitonpeer option. But that doesn't do annything for me, i thought. |
| 18:03.28 | siera08 | sorry [TK]D-Fender, yesterday. |
| 18:03.32 | mitchGaffigan | if asterisk cannot convert between two codecs, it will not give any sound |
| 18:03.50 | *** join/#asterisk |AbsyntH| (n=never@host216-187-dynamic.16-79-r.retail.telecomitalia.it) |
| 18:04.25 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:05.06 | siera08 | mitchGaffigan: en, but if the called party accepts the call, i can have conversation with him.. |
| 18:05.17 | mitchGaffigan | ah |
| 18:05.36 | |AbsyntH| | a little question from an asterisk n00b ...which linux distribution do you suggest ? |
| 18:05.46 | mitchGaffigan | most |
| 18:06.43 | *** join/#asterisk wbw (n=caboose@c-66-229-196-96.hsd1.fl.comcast.net) |
| 18:07.00 | kerx | |AbsyntH|, speaking on behalf of myself CentOS, Debian, Ubuntu |
| 18:07.04 | haps | out of curiosity, does anyone here use freebsd/asterisk? |
| 18:07.05 | |AbsyntH| | yep but for example i can manage gentoo,debian,ubuntu,opensuse or centoos but for this work a don't wont to spend a lot of time |
| 18:07.24 | |AbsyntH| | so i think about debian/ubuntu |
| 18:08.15 | kerx | yes, that's a good one |
| 18:08.24 | kerx | big community backing |
| 18:08.26 | kerx | go for it ! |
| 18:08.36 | mitchGaffigan | siera08: can you show the relevant dialplan |
| 18:08.44 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:08.57 | |AbsyntH| | right...and a good gui? |
| 18:09.44 | ]technophreak[ | Is there a way to have registration keep trying when handle_response_register gets a 404 ? |
| 18:10.05 | |AbsyntH| | for a quick manage...i know that the big work i've to do by shell,but for some quick change |
| 18:13.32 | [TK]D-Fender | |AbsyntH|: Generally no |
| 18:13.57 | [TK]D-Fender | |AbsyntH|: the GUI's out there are largely complete solutions and working outside the box is challenging. |
| 18:14.14 | [TK]D-Fender | |AbsyntH|: the more direct phrase would be "owns your sorry ass" |
| 18:14.21 | |AbsyntH| | ahah |
| 18:14.25 | |AbsyntH| | ok ok |
| 18:14.42 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 18:15.23 | |AbsyntH| | the shell is the way i understand ;) |
| 18:16.50 | |AbsyntH| | i've to go...tnx and goodbye |
| 18:18.30 | ]technophreak[ | Does anyone even know what I am talking about ? |
| 18:19.01 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 18:19.04 | ]technophreak[ | Is there a way to have registration keep trying when handle_response_register gets a bad password response or a 404 ? |
| 18:20.42 | siera08 | mitchGaffigan: http://pastebin.com/d45ec4941. |
| 18:21.18 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:21.27 | RobH | Hrmm, suddenly since I switched to .6 my voicemail dialing from phones is odd. It now always prompts the user to enter both the extension and password, when before it only asked for the password. I have the extension listed here: http://pastebin.com/m776111f |
| 18:21.29 | edoceo | is it possible to build a geo-redundant * setup? I want my 800 number to go to San Jose server if on/available or my Baltimore server if not |
| 18:21.32 | RobH | Anyone see what I am doing incorrectly? |
| 18:22.35 | [TK]D-Fender | RobH: Yes... you are not showing us the failed call or your VM config |
| 18:23.29 | ]technophreak[ | Is there anyway for asterisk registration to keep trying when it gets a 404 or bad password instead of giving up ? |
| 18:24.12 | RobH | [TK]D-Fender: http://pastebin.com/m1c4e518d now shows the CLI |
| 18:24.30 | RobH | but my question is can you still pass the callerid(num) into the voicemailmain application? |
| 18:25.20 | [TK]D-Fender | RobH: Try following my actual request in its ENTIRETY |
| 18:25.41 | RobH | You want the contents of my voicemail.conf? |
| 18:25.59 | [TK]D-Fender | RobH: CLEARLY |
| 18:26.02 | *** join/#asterisk Jabka (n=jabka@CBL217-132-73-4.bb.netvision.net.il) |
| 18:26.16 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 18:28.00 | *** join/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 18:28.39 | *** part/#asterisk Zeryl (n=Zeryl@97-87-122-210.dhcp.stls.mo.charter.com) |
| 18:28.41 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 18:29.22 | Jabka | what should i read to set asterisk between two as a relay (any incoming call transfer to some sip server (ekiga for exanple)) |
| 18:29.23 | docid | ok, so i need to strip all the leading digits from incoming calls whose last 4 digits begin with 68 or 69, haveing trouble figgureing out how to write that |
| 18:29.26 | RobH | [TK]D-Fender: http://pastebin.com/m59471db5 |
| 18:29.32 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 18:29.51 | RobH | I am not sure why the voicemail.conf is needed, when I am not sure why it is not passing the callerid number to the voicemailmain application is all =[ |
| 18:29.54 | RobH | but its there now |
| 18:30.30 | [TK]D-Fender | RobH: Its asking for the box #? |
| 18:30.34 | RobH | Yep |
| 18:30.52 | RobH | It prompts for user to enter voicemail box, then pass. Before it was just taking the callerid number, and prompting for the pass |
| 18:30.53 | [TK]D-Fender | RobH: Do another call and fully enter the box |
| 18:30.56 | mosty | Jabka, the book would be a good start |
| 18:30.58 | RobH | ok |
| 18:30.58 | mosty | ~thebook |
| 18:30.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:31.11 | Jabka | went ot amazon |
| 18:31.34 | [TK]D-Fender | Jabka: Dial(SIP/12345@ekiga.net) |
| 18:31.40 | [TK]D-Fender | Jabka: There. thats it. |
| 18:31.59 | RobH | bah, found the issue |
| 18:32.27 | Jabka | [TK]D-Fender that easy ? |
| 18:32.54 | [TK]D-Fender | Jabka: Yes |
| 18:33.07 | [TK]D-Fender | RobH: namely? Didn't reload your VM config? |
| 18:33.23 | RobH | did not reload the sip after fixing a users callerid |
| 18:33.26 | RobH | >_< |
| 18:33.33 | *** join/#asterisk op3r (n=op3r@114.108.201.205) |
| 18:33.56 | [TK]D-Fender | RobH: Yeah, I missed the 620/602 myself |
| 18:34.25 | *** part/#asterisk leanshen (n=bob@ool-43546a71.dyn.optonline.net) |
| 18:35.05 | RobH | if only all my problems were a result of my own inattention, I would be set. |
| 18:36.17 | Jabka | is reading thebook |
| 18:36.24 | *** join/#asterisk djin (n=djin@84-104-110-179.cable.quicknet.nl) |
| 18:37.07 | op3r | can anyone point me to the right direction why asterisk kept on playing default for music on hold while I only have a girl.raw on /var/lib/asterisk/mohmp3 which i also set on musiconhold.conf? |
| 18:37.34 | [TK]D-Fender | op3r: I don't see any backup for your problem.... |
| 18:37.45 | op3r | :( |
| 18:37.47 | op3r | lol |
| 18:38.24 | ]technophreak[ | Is there anyway for asterisk registration to keep trying when it gets a 404 or bad password instead of giving up ? |
| 18:38.43 | RobH | Anyone know off the top of their head what I can put into either my dialplan or elsewhere to increase the timeout a user has to enter someones name in the directory? I do not see any option for this kind of thing in the Directory application call itself. |
| 18:39.37 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 18:45.45 | mosty | ]technophreak[, registertimeout |
| 18:47.35 | ]technophreak[ | you sure ? |
| 18:47.55 | ]technophreak[ | it doesnt seem to be timing out, just abandoning trying to register |
| 18:51.11 | ]technophreak[ | I found registerattempts ... will try messing with that |
| 19:02.03 | *** join/#asterisk Jabka (n=jabka@CBL217-132-73-4.bb.netvision.net.il) |
| 19:02.08 | ]technophreak[ | It doesnt seem to change anything |
| 19:02.39 | ]technophreak[ | asterisk literally stops trying to register as soom as it gets a 403 |
| 19:04.29 | mosty | what combination of registertimeout and registerattempts did you try? |
| 19:05.14 | ]technophreak[ | registerattempts=0 |
| 19:07.35 | ]technophreak[ | I just tried with registerattempts=100 and registertimeout=60 |
| 19:07.38 | ]technophreak[ | same problem |
| 19:09.01 | *** join/#asterisk Shaun2222 (n=shaun@ip68-5-154-128.oc.oc.cox.net) |
| 19:09.11 | ]technophreak[ | I put up a trace, and iets just not retrying |
| 19:09.12 | Shaun2222 | how would i match what phone number sombody called in on? |
| 19:09.23 | Shaun2222 | i have some did's figured i should use them :) |
| 19:09.35 | mitchGaffigan | ${EXTEN} |
| 19:09.47 | Shaun2222 | lol, i guess that makes sense. |
| 19:09.48 | ]technophreak[ | shaun2222: is your carrier providing you multiple trunk or just one ? |
| 19:10.13 | Shaun2222 | ]technophreak[: it's a PRI T1, just one circuit. |
| 19:10.30 | ]technophreak[ | then you`ll need to match it at the dialplan |
| 19:10.36 | [TK]D-Fender | Shaun2222: It lands on a exten of the DID they dialed |
| 19:10.53 | [TK]D-Fender | Shaun2222: You already have this, you just weren't looking |
| 19:11.50 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 19:12.29 | *** join/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 19:12.49 | mintee | is there another way besides IAX2 and SIP to trunk 2 asterisk servers together? |
| 19:13.16 | mintee | or to fwd a number to another voip number without using the PRI? |
| 19:13.21 | jplank | pri |
| 19:13.31 | jplank | BRI? |
| 19:13.47 | mintee | O_o |
| 19:14.09 | pdmmm | mintee: what are you trying to do? |
| 19:14.13 | mintee | yeah, i'm not trying to use any of my zap channels |
| 19:14.28 | mintee | just route about 100 numbers over to another provider |
| 19:14.55 | jplank | So you dont want to use a ZAP channel, and you don't want to use SIP or IAX, do you have strings and a tin can? |
| 19:14.56 | Shaun2222 | [TK]D-Fender: yep, i just had a global match though _X. for that context. |
| 19:15.04 | pdmmm | use a laplink cable! |
| 19:15.05 | pdmmm | haha |
| 19:15.14 | mintee | I have a 10mB ethernet connection, and would prefer to use that. |
| 19:15.16 | mintee | pdmmm: hehe |
| 19:15.38 | jplank | but you don't want to use VOIP, so how are you going to use a ethernet connection? |
| 19:15.47 | mintee | jplank: no, I will use IAX or SIP, just making sure there are no other options |
| 19:15.52 | jplank | H323 |
| 19:15.54 | jplank | :) |
| 19:15.54 | mitchGaffigan | chan_console |
| 19:16.13 | mosty | mintee, do IAX over the ethernet connection |
| 19:16.17 | Shaun2222 | gtalk ;) |
| 19:16.32 | jplank | I'm sure you if you were a good enough coder, there's probably a million ways to do it |
| 19:16.37 | jplank | maybe write your own protocol? |
| 19:16.43 | mitchGaffigan | Speech Recognition -> serial |
| 19:16.43 | mintee | mosty: yeah, that's my main goal, but i don't know if they support IAX on the other side... |
| 19:16.52 | jplank | then why not SIP? |
| 19:17.03 | docid | any way to use gotoif to check incoming trunk and direct from there? im feelin a bit lost on this one |
| 19:17.08 | mosty | mintee, ask what they do support, then you know your options |
| 19:17.13 | jplank | lol |
| 19:17.17 | mintee | lol, yeah, i tried |
| 19:17.29 | Shaun2222 | docid: what are you trying to match? |
| 19:17.32 | mintee | they are indian, and i couldn't understand a goddamn word they were saying |
| 19:17.37 | mintee | so i'm composing an email to them |
| 19:17.37 | jplank | then use SIP |
| 19:17.52 | mosty | mintee, email and ask for authentication details |
| 19:18.00 | jplank | I'm sure they support SIP, I've never worked with a customer who had an office in India that didn't support SIP |
| 19:18.06 | mitchGaffigan | lol |
| 19:18.17 | docid | well, if a call comes in on g0 and doesnt match a sip extention of voicemailbox, then it needs to be dialed out on g1, and the reverse is also true |
| 19:18.18 | mintee | mosty: that's what i'm doing ;) |
| 19:18.31 | mintee | thanks kids... |
| 19:18.36 | jplank | kids? |
| 19:18.39 | mitchGaffigan | docid: what are you trying to do? |
| 19:18.46 | jplank | !kickban mintee |
| 19:18.49 | jplank | :) |
| 19:19.11 | mintee | zoooooo Nooooooooes |
| 19:19.19 | docid | mitchGaffigan, Shaun2222 well, if a call comes in on g0 and doesnt match a sip extention of voicemailbox, then it needs to be dialed out on g1, and the reverse is also true |
| 19:19.20 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 19:19.33 | docid | or not of |
| 19:19.55 | Shaun2222 | docid: so you want a call that comes in that doesnt have a voicemailbox setup to dial out on DAHDI/g1/${EXTEN})? |
| 19:20.20 | docid | yes, |
| 19:20.28 | docid | if it comes from g0 |
| 19:20.37 | docid | if it comes from g1 iut needs to dial out on g0 |
| 19:21.13 | Shaun2222 | sounds simple enough to do. |
| 19:21.17 | mitchGaffigan | so use _#. for the extension, use MailboxExists(${EXTEN}) and Dial(DAHDI/g1/${EXTEN}) otherwise |
| 19:21.40 | Shaun2222 | tis tis... MailboxExists is deprecated :) |
| 19:21.50 | docid | what is it replaced by? |
| 19:21.50 | mitchGaffigan | to be replaced with what? |
| 19:21.53 | jplank | I was just looking for the updated command |
| 19:22.05 | Shaun2222 | probably somthing like.... exten => _X.,1,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:1) |
| 19:22.10 | mitchGaffigan | I was under the impression that the j option was just depreciated |
| 19:22.19 | Shaun2222 | you would need to modify where it goes obviously |
| 19:22.28 | [TK]D-Fender | Shaun2222: Umm.... NO. |
| 19:22.29 | docid | ok, also in this i need to strip any leading number besides the last 4 if the last four match 68XX or 69XX |
| 19:22.44 | edoceo | Do I need to get a special carrier to get geo-redundant * servers? |
| 19:22.51 | jplank | I thought it was also, but it doesn't say so in the eiki |
| 19:22.57 | jplank | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MailboxExists |
| 19:23.00 | Shaun2222 | [TK]D-Fender: no? |
| 19:23.05 | [TK]D-Fender | docid: Fo read the CHANNELVARIABLES docs in your source tarball. You seem to lack the basics |
| 19:23.18 | mitchGaffigan | edoceo: just one that supports SRV records |
| 19:23.20 | [TK]D-Fender | 15:22]<Shaun2222>probably somthing like.... exten => _X.,1,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:1) <- tragic logic flaw |
| 19:23.21 | docid | okiez, will do..... |
| 19:23.30 | edoceo | mitchGaffigan: thx |
| 19:24.04 | Shaun2222 | [TK]D-Fender: well, ya i know that wont work as it.. he would have to change peices... |
| 19:24.05 | jplank | oh, that would just keep looping wouldn't it? |
| 19:24.22 | jplank | if not boxes exist |
| 19:24.29 | [TK]D-Fender | jplank: SHHHHHH!!!! |
| 19:24.32 | jplank | grrr |
| 19:24.34 | Shaun2222 | i use it like 'exten => _2XX,7,GotoIf(${MAILBOX_EXISTS(${EXTEN}@default)}?:9)' to jump over 8 which is voicemail() |
| 19:24.35 | [TK]D-Fender | :p |
| 19:24.54 | jplank | shaun look what you have and what you told him to do |
| 19:25.22 | mitchGaffigan | Shaun2222: if that is correct... i'm going to cry |
| 19:25.39 | Shaun2222 | jplank: yes i know, i see the loop, i just put _X. and 1 at the end, it's his task to make it work.. |
| 19:26.08 | [TK]D-Fender | Shaun2222: its not the "_X." and "1" that is the problem. It is the "1" and the "1" |
| 19:26.12 | docid | well, thanks for the bait, mabey can catch something with it :) |
| 19:26.19 | jplank | I've noticed, no matter how much you tell people to not take your suggestions verbatim, they will |
| 19:26.19 | Shaun2222 | mitchGaffigan: if whats correct? |
| 19:26.31 | mitchGaffigan | using Mailbox_exists like that |
| 19:27.01 | mitchGaffigan | it looks absolutely horrendus... yet much shorter than the three line construct I am using |
| 19:27.04 | Shaun2222 | mitchGaffigan: well the console yelled at me when i upgraded to 1.6 and my dialplan was using mailboxexists :) |
| 19:27.08 | mitchGaffigan | same here |
| 19:27.15 | mitchGaffigan | I was trying to fix that right now. |
| 19:27.47 | Shaun2222 | mitchGaffigan: for me it actually cut out some lines |
| 19:27.58 | Shaun2222 | mitchGaffigan: since i no longer had to check the mailbox and then check the var |
| 19:28.19 | mitchGaffigan | it would be the same number of lines as my 1.4 as I was using j to jump to n+101 on success |
| 19:28.21 | Shaun2222 | it does it in one task., |
| 19:30.28 | Shaun2222 | jplank: ya i know people copy/paste exactly what you give them... hell i'm guilty at times... either way, if i would have pasted him mine he would have just been trying to figure out why it keeps jumping to 9... probably would have had a bunch of exten => 2__,3,noop(filler);exten => 2__,4,noop(filler);exten => 2__,5,noop(filler);etc... lol |
| 19:31.02 | Shaun2222 | bah... s/2__/_2XX/g |
| 19:31.41 | Shaun2222 | mitchGaffigan: ah, well then it's a simple change for you... |
| 19:31.56 | Shaun2222 | i never liked the jump to n+101 though it was ghetto. |
| 19:31.58 | Shaun2222 | :) |
| 19:32.41 | mitchGaffigan | I didn't either and it makes sense to make it that way... more flexible too |
| 19:33.32 | *** join/#asterisk trippssss (n=tripps@140.239.207.58) |
| 19:33.57 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:39.19 | edoceo | Do I need to use a carrier or is their a way (via DIDx for example) to have my * servers directly handle inbound w/o an upstream SIP provider |
| 19:41.04 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 19:44.25 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:44.59 | mitchGaffigan | edoceo: set sip.conf's context to be the one where you want to process an anonymous sip call |
| 19:46.34 | mitchGaffigan | edoceo: or if you are referring to a group of * boxes that are geographically distributed, you can use DUNDI or just have a dialplan that routes over trunks forming a mesh to the correct box |
| 19:48.15 | dwery | I'm getting a bunch of "dahdi: Cannot start tone until a tone zone is loaded." I have both loadzone and defaultzone in system.conf . What am I missing? |
| 19:48.26 | [TK]D-Fender | edoceo: What does it mean to get SIP calls without an "upstream SIP provider"? Do packets appear out of thin air? SOMEONE is sending you them... |
| 19:48.53 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 19:53.07 | edoceo | [TK]D-Fender: Right - but instead of Caller->AT&T (their carrier) -> XO (my carrier) -> [sip] -> My Asterisk PBX in office -> my phone |
| 19:53.21 | *** part/#asterisk ]technophreak[ (n=]technop@modemcable048.23-81-70.mc.videotron.ca) |
| 19:53.23 | edoceo | I want to take XO out of the loop and have ATT know to SIP right to my machine |
| 19:53.52 | [TK]D-Fender | edoceo: You would need to arrange that with AT&T |
| 19:54.06 | [TK]D-Fender | edoceo: At which point... THEY become the unstream provider |
| 19:54.12 | edoceo | Oh - so I have to elevate myself to a Tier1 provider like status? |
| 19:54.37 | [TK]D-Fender | edoceo: Or have an arrangement directly with that telco, etc |
| 19:54.48 | [TK]D-Fender | edoceo: basically a waste of time mostly. |
| 19:54.50 | edoceo | Hmm - maybe more work than I expected |
| 19:56.07 | jplank | edoceo, whats wrong with XO? |
| 19:56.51 | edoceo | Nothing - just exploring how to move up the chain of telephony providing |
| 19:57.20 | jplank | theres nothing wrong with using XO over AT&T, hell, I'd personally rather XO over AT&T |
| 19:57.38 | Jabka | wonders if he can use his motorola e1000 (standart AT modem) as an FXO (maybe chan_mobile) |
| 19:58.25 | jplank | well, unless you have to deal with their support, then I'd rather neither :P |
| 19:58.42 | mitchGaffigan | or if you are the caller and the receiver, just make a direct sip connection |
| 20:08.02 | [TK]D-Fender | mitchGaffigan: If he's the caller & receiver he should speak to a psychologist. |
| 20:08.22 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 20:08.51 | mitchGaffigan | I suppose that would be an interesting occasion... |
| 20:09.33 | mitchGaffigan | perhaps I should rephrase that to be "If you are in charge of the dialplan of the caller and receiver..." |
| 20:10.16 | *** join/#asterisk shinao1 (n=shinao1@62.173.46.239) |
| 20:17.19 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 20:18.56 | Gido-E | is there a log facility to she the manager commands? |
| 20:19.09 | Gido-E | s/she/see |
| 20:23.17 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 20:26.24 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 20:29.47 | *** join/#asterisk Jabka (n=jabka@89-139-77-49.bb.netvision.net.il) |
| 20:31.10 | ecret | I have asterisk setup on my system and it can take sip calls. I had hoped to get a single PTSN connection. Can someone suggest a provider that preferably is instantly usable? |
| 20:32.27 | pdmmm | ecret: www.vitelity.com |
| 20:32.37 | ecret | pdmmm: thanks |
| 20:32.46 | pdmmm | welcome! |
| 20:33.34 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:37.44 | *** join/#asterisk horvath (n=horvath@74-51-45-109.telnetcommunications.com) |
| 20:38.50 | [T]ank | I am trying to learn how to use call files. I have entered in a callfile and had asterisk successful attempt to make a call, but I am running into some errors and could use some assistance. Please see pastebin for details. I can place calls from the T1s just fine. I am currently adding more details to another paste to show that. |
| 20:39.37 | Katty | ohai |
| 20:40.58 | *** part/#asterisk horvath (n=horvath@74-51-45-109.telnetcommunications.com) |
| 20:42.18 | [T]ank | I take that back... I can dial from the CLI, but when dialing from a softphone, I get the same error as the call file: http://pastebin.com/d37467c3b |
| 20:42.40 | [T]ank | just noticed my past in the first post was omitted... hang on |
| 20:42.53 | [T]ank | http://pastebin.com/d4519feb7 |
| 20:43.07 | [T]ank | those are the details of all my configs and the outcome of attempting a callfile |
| 20:44.07 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:48.45 | *** join/#asterisk ariel_ (n=ariel_@c-24-127-219-186.hsd1.fl.comcast.net) |
| 20:49.27 | Katty | jaytee: poke |
| 20:49.42 | *** join/#asterisk torr (n=sylar@bzq-79-183-140-138.red.bezeqint.net) |
| 20:49.45 | torr | hello |
| 20:49.49 | Katty | herro |
| 20:49.55 | [T]ank | so I guess the core issue is that the extension dialing out is failing: http://pastebin.com/d37467c3b what would cause this when I dial from the softphone? When I dial from the CLI it works: http://pastebin.com/d75e002b3 |
| 20:50.14 | jaytee | Katty, ???? |
| 20:50.38 | Katty | jaytee: incoming query |
| 20:50.42 | torr | Q: how do I connect asterisk to a regular phone? |
| 20:50.53 | Katty | torr: you stick a card in there. |
| 20:51.03 | beek | torr: or use an ATA |
| 20:51.03 | torr | is there something like for skype phone? |
| 20:51.18 | torr | what card? modem? |
| 20:51.59 | Katty | a card that support digital or analog phones. |
| 20:52.21 | *** join/#asterisk doolph (n=doolph@190.141.68.31) |
| 20:52.30 | [T]ank | torr: do you have a working server that can do sip phone? And now you are ready to tackle analog? It sounds like you might be in over your head |
| 20:53.18 | Gido-E | [T]ank if you see how the questions are formed. |
| 20:53.34 | torr | [T]ank, I have a linux server, without asterisk, and i think to install asterisk, and wonder if I can just exit a line from there to my operator |
| 20:54.21 | ariel_ | hello everyone |
| 20:54.27 | [T]ank | torr: over simplified, yes you can. There are many places with the documentation on how to do that. |
| 20:54.32 | doolph | hi ariel_ |
| 20:54.50 | [T]ank | Gido-E: Are you referring to my questions or torr's? |
| 20:54.54 | ariel_ | hello doolph |
| 20:54.59 | torr | [T]ank, I want to know if I can get the hardware for it for $50 |
| 20:55.20 | doolph | anyone got any problem with asterisk 1.6? it got disconnected from all ip phones without any reason, but from the server I can ping the phones, once I restart the asterisk all gets back |
| 20:55.31 | [T]ank | look for an ATA device. That will fall in your price range. They are however not as quality as using an FXO/FXS card |
| 20:56.14 | torr | how will this quality be manifested? |
| 20:56.25 | [T]ank | different for everyone |
| 20:56.39 | dude7064 | I want to have my own calling card-business , and I'm wondering if there is a service/application commercially available to facilitate this ? |
| 20:56.55 | [T]ank | So, regarding my questions above, is there anyone who may be able to identify what I have screwed up? |
| 20:58.17 | torr | 80$ Cisco ATA 188 - VoIP phone adapter |
| 20:58.20 | [T]ank | my softphone can dial other extensions just fine on the server, but if it tries to place a call to the t1, then it fails with a cause 99. However I can succesfully place calls from the console using the same context and t1 that the softphone is pointing to |
| 20:58.22 | torr | good |
| 20:58.32 | [T]ank | any ideas why it would do that to me? |
| 20:59.11 | jblack | Looks like verizon is having a problem via alternet. Just in case anyone happens to be on the same provider and is suddenly finding sip broken. |
| 21:00.36 | jblack | depending on hops, packet loss of 65-90% |
| 21:01.26 | torr | is this " Cisco ATA 186" good |
| 21:01.31 | torr | ? |
| 21:01.44 | torr | Do I need asterisk for it, or is it standalone? |
| 21:02.10 | jblack | This is #asterisk. You always need asterisk. ;) |
| 21:03.39 | torr | :/ |
| 21:07.11 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 21:11.32 | *** join/#asterisk af_ (n=getsmart@88-149-230-241.dynamic.ngi.it) |
| 21:13.20 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 21:13.34 | Gido-E | Tank it looks like you are dialing the wrong number |
| 21:15.06 | *** part/#asterisk Jabka (n=jabka@89-139-77-49.bb.netvision.net.il) |
| 21:15.34 | af_ | there any ip phone with an snmp agent? |
| 21:15.45 | [T]ank | Gido-E: you mean that the telephone number I am dialing is wrong? Its my cell phone. |
| 21:15.50 | Gido-E | [T]ank how long are the numbers you dial? 11 or 10 digits? |
| 21:16.01 | [T]ank | works when I dial the same number from the cli. |
| 21:16.08 | [T]ank | 1+10 digit number |
| 21:16.22 | Gido-E | 11 digits number. |
| 21:18.06 | Gido-E | you dont need to chop of the leading digit? |
| 21:18.17 | [T]ank | no. the carrier requires it |
| 21:19.46 | *** join/#asterisk andalou (n=chatzill@190.233.2.179) |
| 21:20.31 | *** part/#asterisk andalou (n=chatzill@190.233.2.179) |
| 21:21.58 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:22.31 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 21:23.17 | brunner | if my voip provider is sending calls to various extensions, how can I force the calls to go to extension s? |
| 21:24.22 | [TK]D-Fender | brunner: Got() |
| 21:24.25 | [TK]D-Fender | Goto() |
| 21:24.43 | brunner | okay, I wasn't sure if there was something I could do in sip.conf |
| 21:24.44 | brunner | thanks |
| 21:25.04 | [TK]D-Fender | brunner: Depends if you're the one telling them what extens to send |
| 21:25.26 | brunner | nope |
| 21:28.07 | *** join/#asterisk double_cheesburg (n=chatzill@ip68-98-36-177.ph.ph.cox.net) |
| 21:29.18 | [TK]D-Fender | brunner: Absolutely sure? Most people don't even know when they are. |
| 21:29.32 | brunner | I'm probably wrong, then |
| 21:29.40 | [TK]D-Fender | brunner: PB it up |
| 21:32.41 | brunner | [TK]D-Fender: http://pastebin.com/d3aad5d87 |
| 21:34.36 | [TK]D-Fender | brunner: Yup, its them. So make a catch-all and Goto "wherever" |
| 21:34.48 | brunner | okay, thanks |
| 21:35.02 | brunner | is it common for voip providers to do that? |
| 21:35.15 | brunner | seems like they would just pass the dnis |
| 21:38.46 | [TK]D-Fender | brunner: they are dialing in the DID that was dialed on their end which is entirely normal |
| 21:39.29 | [TK]D-Fender | brunner: This is supposed to be a GOOD thing. Why would you order 10 #'s from them only to have them all processed identically? |
| 21:42.59 | brunner | I wouldn't process them identically. I'd process them based on the dnis they pass |
| 21:43.13 | [TK]D-Fender | bruthat IS the extension. |
| 21:43.32 | brunner | I understand that, but how do things end up in CALLERID(dnis)? is that PRI-only or something? |
| 21:44.43 | dude7064 | when installing asteriskNow,, it's asking for username and password,,, which username should I choose ? |
| 21:44.44 | [TK]D-Fender | brunner: I'm not 100% on this, but I believe that because of ANI thats why that split exists. Multiple ways to pass varying levels of info. But Exten is the norm IIRC. |
| 21:44.57 | brunner | I see |
| 21:45.04 | dude7064 | I tried admin, but it's saying username/password is incorrect !!! |
| 21:45.13 | dude7064 | what's the username I'm supposed to be using ? |
| 21:45.31 | [TK]D-Fender | dude7064: Try asking in their channel. Check the topic |
| 21:46.25 | dude7064 | there are like 10 people over there !! Nobody answers anybody |
| 21:46.50 | jplank | did you ask google? |
| 21:47.18 | jplank | I find it hard to believe that the default login/password isn't a google search away |
| 21:48.00 | dude7064 | yes I did. It says "admin" is the username |
| 21:48.03 | [TK]D-Fender | jplank: I got the answer in the FIRST LINK of my own search just now |
| 21:48.05 | dude7064 | but it's not working for me. |
| 21:48.10 | dude7064 | root is working fine,, |
| 21:48.20 | dude7064 | but it's not taking me to the GUI |
| 21:48.34 | dude7064 | It's taking me to the command prompt,, how can I view the usernames from there ? |
| 21:49.03 | [TK]D-Fender | "The boot sequence stops at a welcome screen that tells you the default login is "admin," "password." Hit the return key to let it finish booting. When you see the Console Menu it's done. " |
| 21:49.46 | [TK]D-Fender | And for FreePBX ... Default username/password is freepbx/fpbx |
| 21:50.30 | dude7064 | just tried admin/password ,, still doesn't work |
| 21:50.33 | awk_r | dude7064, i'm offended by your 'Nobody answers anybody' comment |
| 21:50.34 | brunner | why bother with FreePBX? Just install ubuntu and apt-get install asterisk |
| 21:50.56 | brunner | or debian |
| 21:50.57 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 21:51.14 | jblack | brunner: That's what I do, but that's not necessarily a flawless solution. |
| 21:51.25 | bmoraca | some people like pretty things like freepbx |
| 21:51.26 | jaytee | this afternoon one of my coworkers was configuring a network printer and accidentally chose the same static IP address that our asterisk server uses. For 5 minutes until he disconnected the printer and reconfigured it's IP address our asterisk server could not respond to network requests. My messages log is full of messages like chan_sip.c: Maximum retries exceeded on transmission a13245f0-4e1a542a@128.68.116.145 for seqno 101 (Critical Response) |
| 21:51.26 | jaytee | and 4 calls that I know of were dropped at that time. Unfortunately one of those calls was our CEO talking to someone outside. |
| 21:51.27 | brunner | what's wrong with it? |
| 21:51.50 | jblack | current is 1.4.21.2 |
| 21:52.07 | brunner | then just compile your own |
| 21:52.20 | [TK]D-Fender | jaytee: maul his punk ass |
| 21:52.23 | bmoraca | jaytee: that's when you kill the idiot who statically set the IP at the printer and then show his decapitated corpse how to set up a DHCP reservation |
| 21:52.27 | awk_r | whats with the *-GUI bashing :-/ |
| 21:52.46 | [8none1] | Can anyone give me some insight into debugging this error : "Ring requested on channel 0/3 already in use or previously requested on span 1. Attempting to renegotiating channel." |
| 21:52.48 | dude7064 | any hints please ? what should I do to find out what the username is ? the password I remember it asked to input one,, but it didn't display any usernames !! |
| 21:52.59 | [8none1] | I'm having calls rejected on my PRI because of this. |
| 21:52.59 | Gido-E | awk_r the asterisk_gui sucks :) |
| 21:53.03 | [TK]D-Fender | jaytee: Better yet... new chew-toy for Kisa ;) |
| 21:53.16 | awk_r | Gido-E, you'll eat those words |
| 21:53.50 | jaytee | [TK]D-Fender, that won't solve the problem that user expectations coming from a Nortel environment is that the phone system almost never, ever goes down and yet in less than 1 week we've had two incidents where calls were dropped. Prior to that we had over 9 months of uninterrupted service with Asterisk in the equation. |
| 21:54.40 | jaytee | and there's only so much you can do with building in redundancy. No way can I make this 100% fail proof |
| 21:54.59 | [TK]D-Fender | jaytee: Its a NETWORK environment and network screwups = service failures. Couple this with an incopetant admin and lose the fake surprise |
| 21:57.01 | jaytee | [TK]D-Fender, agreed but then you and I understand the complexity of networks and the potential for failure. Others are not so well informed and have shall we say, "unreasonable expectations". It has my boss almost apoplectic this afternoon. After he went off on me he later apologized and when I told him he didn't need to apologize he shoved 20 bucks in my pocket and said, "Dinner's on me tonight" |
| 21:58.05 | jblack | "Here's free mcdonald's for abuse." How well he expresses regret. |
| 21:58.19 | jaytee | He's worried that the CEO will hold him accountable and that it will be held against him for going with VOIP/Asterisk despite all the cost savings, advances, convergence, etc. that we gain by it. |
| 21:58.58 | jblack | jaytee: You could put your phone system on a discrete network to avoid that sort of risk. |
| 21:59.04 | jaytee | jblack, he can be a pain but he's basically a decent guy for the most part. He just gets too stressed and is worried because the CEO can be a major prick. |
| 21:59.57 | jblack | a switch and a pile of monkey hours. |
| 22:00.25 | jaytee | jblack, all the phones are in one vlan and the server is in the primary vlan because it needs to talk to Exchange via sipX. I need to reconfigure to use both NICs in the server, 1 to talk to all the phones and the other as a route to Exchange is what I'm thinking. |
| 22:01.23 | jaytee | putting the entire VOIP system in it's own physical network is impractical and cost prohibitive in our environment. |
| 22:01.24 | jblack | I'm thnking if you go with physical seperation, mr. printer configuration guy can't cause an ip conflict. |
| 22:01.27 | Kobaz | [Mar 9 18:01:08] WARNING[24052]: chan_iax2.c:2169 __attempt_transmit: Max retries exceeded to host 192.168.24.31 on IAX2/2248-300 (type = 6, subclass = 11, ts=6409767, seqno=179) |
| 22:01.44 | Kobaz | i keep getting a bazillion of those when i place an iax2 channel on park |
| 22:02.14 | jblack | Perfectly reasonable. So, tell the boss that you could prevent that sort of problem, but it would cost (hold arms out) that much. |
| 22:05.00 | jblack | Then, when _his_ boss gets snooty, he can say "We can solve that type of problem, but it would cost (holds arms out) that much" |
| 22:05.25 | *** join/#asterisk RoPBX (n=nickserv@200.93.34.175) |
| 22:06.20 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 22:07.28 | jblack | I need to find a good, c++ friendly tcp library for talking to the AMI |
| 22:07.30 | jaytee | jblack, hahahaa, that sounds like a good idea! |
| 22:07.43 | jaytee | jblack, what about C#? |
| 22:07.47 | af_ | c++ and frindly? |
| 22:07.47 | RoPBX | Hi All |
| 22:08.02 | jaytee | there's a nice little C# .NET library for that on Sourceforge |
| 22:08.10 | beek | jaytee: A simpler solution is crucifiction of the offending network tech. That'll be an example to the others... |
| 22:08.14 | jblack | jaytee: Oh, you're serious? |
| 22:08.16 | jaytee | I've played with it in Visual C# |
| 22:08.24 | RoPBX | please, someone to ask about TDM problem in Asterisk PBX |
| 22:08.35 | beek | ~ask |
| 22:08.36 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:09.07 | jblack | To put it kindly, I don't work on that platform. |
| 22:12.32 | jaytee | beek, he's an MCSE, CCNA and makes more money than me. He actually threw himself under the bus when I told everyone we had an outage that lasted 5 minutes and the time frame. "Oh, what's the IP of the server? Yeah, that was me!" |
| 22:12.33 | beek | Wow -- that's ballsy. |
| 22:12.33 | beek | Okay, perhaps crucifiction is a bit harsh. |
| 22:12.33 | beek | Don't they teach DHCP in MCSE classes? ;-) |
| 22:12.33 | jaytee | and circumcision of an already circumcised person is cruel and unusual punishment although it's probably not against the law in Indiana. |
| 22:12.33 | [TK]D-Fender | You're right... mercy killing... back of the head, capo-style |
| 22:12.33 | dude7064 | anybody can tell me what's the username is ? |
| 22:12.33 | RoPBX | i have a Rhino tdm card, and the incoming calls get mixed with outgoing calls |
| 22:12.33 | jaytee | hahahaaha |
| 22:12.33 | dude7064 | I just installed AsteriskNow,, and it's asking for the username |
| 22:12.34 | [TK]D-Fender | dude7064: WTF is "IT" |
| 22:12.34 | jblack | ropbx: That's almost certainly caused by a configuration devised by you. |
| 22:12.34 | dude7064 | Centos |
| 22:12.34 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:12.34 | [TK]D-Fender | dude7064: One would think that you'd sign in as ROOT |
| 22:12.34 | dude7064 | I did |
| 22:12.34 | RoPBX | thanks jblack, how can I configure it to fix that problem? |
| 22:12.34 | dude7064 | but I am not getting any GUI |
| 22:12.39 | dude7064 | It is taking me to the command promp |
| 22:12.44 | dude7064 | *prompt |
| 22:12.45 | [TK]D-Fender | dude7064: And how did you you think you would get to a GUI? |
| 22:13.03 | jblack | RoPBX: I'd suggest you look at what contexts you're dropping calls into (both directions), and working up your extensions.conf to separate them according to your deepest, darkest desires. |
| 22:13.27 | dude7064 | in the guide for AsteriskNow , it says simply boot the media with the ISO image and follow the instructions till you get the URL |
| 22:13.33 | drmessano | [TK]D-Fender --> c:\win.com |
| 22:13.54 | [TK]D-Fender | dude7064: its a WEB GUI. You access it via ANOTHER COMPUTER with a WEB BROWSER |
| 22:13.54 | dude7064 | afterwards type this URL and it'll take you to the configuration page with a GUI interactive interface |
| 22:14.30 | jblack | looks at libclaw-net1 |
| 22:14.30 | dude7064 | what is the url then ? |
| 22:14.33 | RoPBX | jblack, incoming and outgoing calls are in separates contexts now |
| 22:14.50 | drmessano | ipaddress or hostname |
| 22:14.51 | jblack | ropbx: take it from there. don't forget to reload files after you change them. |
| 22:14.51 | dude7064 | I'm using VirtualBox to run AsteriskNow |
| 22:14.55 | [TK]D-Fender | dude7064: the IP of the server :80 or :8080 or something to that effect |
| 22:14.59 | dude7064 | and was not presented with any URLs |
| 22:15.07 | jblack | I bet anything with "claw" in the name does a powerfully good job. |
| 22:15.23 | drmessano | jblack: Indeed |
| 22:15.28 | [TK]D-Fender | dude7064: "presented"? There is no "miracle announcement protocol" for this. Its a WEB SERVER, not *MAGIC* |
| 22:15.30 | drmessano | jblack: Just ask Billy Mays |
| 22:15.50 | jblack | lol. "Claw is a C++ Library Absolutely Wonderful" |
| 22:16.03 | jblack | Billy must have a sourceforge account. :P |
| 22:16.29 | drmessano | Any software that is both a description and a review in the same line is DEFINITELY teh awesum |
| 22:16.30 | dude7064 | how can I tell that Asterisk is running after logging in ? |
| 22:16.31 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 22:17.03 | drmessano | The GUI will tell you |
| 22:17.27 | jblack | dude7064: w/ normal asterisk, ps aux | grep asterisk or asterisk -r should suffice. |
| 22:17.41 | jblack | note: running doesn't necessarily mean "working" =) |
| 22:17.50 | RoPBX | jblack, everything works perfect, but there are extrange cases that calls get mixed |
| 22:17.55 | [TK]D-Fender | jblack: thats a grey area :) |
| 22:18.32 | jblack | wtf? why would a stream library have png/jpg/targa support? |
| 22:18.40 | jblack | phear teh libclaw |
| 22:20.25 | drmessano | app_CLAW |
| 22:21.29 | jblack | maybe I should just make something up in perl or python. |
| 22:21.53 | drmessano | python + claw would rock.. Thats as cool as a Bear with a rifle on its back |
| 22:22.09 | [TK]D-Fender | specifically request sharks with friggen lasers on their heads |
| 22:23.18 | RoPBX | jblack, it seems that asterisk is making a bridge with incoming and outgoing calls , that happens when there are a lot of incoming and outgoing calls at the same time, like asterisk didn't realize that the TDM line is busy |
| 22:23.34 | drmessano | .... |
| 22:23.58 | *** part/#asterisk Khratos (n=khratos@190.166.103.111) |
| 22:24.48 | jblack | ropbx: Hmmm. So you think signalling with upstream is getting dropped somehow? |
| 22:24.59 | *** join/#asterisk mib_5ozpug (i=d936d556@gateway/web/ajax/mibbit.com/x-eea7e9caaea58c3a) |
| 22:26.10 | RoPBX | jblack, i don't know what happens there, but you can even hear the other users dialing, and then you can talk |
| 22:26.27 | jblack | You could check your switchtype, signalling and channel with your provider. |
| 22:26.31 | mib_5ozpug | how to istal hud pakage on the server |
| 22:26.31 | mib_5ozpug | how to install hud server |
| 22:27.46 | jblack | Perhaps your zaptel/dahdi configuration is wrong. |
| 22:28.00 | mib_5ozpug | i cant open the package page |
| 22:28.04 | RoPBX | i'm using my local phone line |
| 22:28.07 | jblack | pots? |
| 22:28.11 | mib_5ozpug | to install hud |
| 22:28.33 | jblack | w/ an ata, of course. |
| 22:28.55 | RoPBX | what parameter can i configure in my zaptel configuration to avoid this? |
| 22:29.03 | dude7064 | I have AsteriskNow installed installed in a VirtualBox machine,, |
| 22:29.26 | RoPBX | i set the busydetect=yes, callinprogress=yes, with no change |
| 22:30.28 | dude7064 | how can i access the AsteriskNow from my host machine ? |
| 22:30.35 | dude7064 | it seems very complicated, |
| 22:30.53 | mib_5ozpug | any help me to install hud |
| 22:31.04 | RoPBX | its a very extrange case, this happens when the calls are incoming/outgoing at the very same time |
| 22:31.05 | dude7064 | isn't possible to simply run the GUI from the VirtualBox machine ? |
| 22:31.16 | Kobaz | dude7064: what about #asterisknow? |
| 22:31.34 | brunner | do I have to register if I'm only pushing outbound calls? |
| 22:31.45 | Kobaz | dude7064: we don't know anything about asterisknow in here, so you're not going to get any help |
| 22:31.49 | [TK]D-Fender | dude7064: what part of WEB SERVER do you not understand? |
| 22:32.04 | RoPBX | jblack: why with an ata??? |
| 22:32.13 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 22:32.21 | mib_5ozpug | any body help me to install hud-lite-2 on server |
| 22:32.28 | jblack | ropbx: Let's start over. What equipment are you using? |
| 22:32.36 | RoPBX | jblack: i'm making test with 2 GS-2020 |
| 22:32.41 | jblack | lol |
| 22:32.44 | mib_5ozpug | ok |
| 22:32.58 | jblack | ![TK]D-Fender grandstream-rant |
| 22:33.08 | Kobaz | dude7064: and, honestly... if you think that accessing the gui interface is complicated... maybe setting up your own phone system is not for you |
| 22:33.31 | seanmh | mib_5ozpug, You could try iSymphony. It's pretty easy to install. www.i9technologies.com/isymphony |
| 22:34.11 | RoPBX | jblack: and one Rhino tdm card, 8 ports, but i have only one local line to test |
| 22:35.03 | jblack | RoPBX: Sorry, no suggestions. I'm not familiar with tdm cards. |
| 22:35.10 | RoPBX | ok, thanks |
| 22:35.38 | [TK]D-Fender | brunner: Generally no. |
| 22:35.40 | RoPBX | any one familiar with TDM cards? or another channel? |
| 22:36.37 | drmessano | mib_5ozpug: Astassistant is free, and pretty decent |
| 22:37.08 | jblack | I feel like #asterisk is currently in the twilight zone |
| 22:37.29 | [TK]D-Fender | Rod.... Rod Serling..... is that you? |
| 22:37.33 | [TK]D-Fender | stares at the sky |
| 22:38.04 | jblack | It's either that, or someone's cloned a whole pile of Bizarro's. |
| 22:38.07 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:38.24 | Dovid | evening TK |
| 22:43.01 | brunner | [TK]D-Fender: thanks |
| 22:43.34 | *** join/#asterisk isamar (n=isamar@server1.dw7.telegate-americas.com) |
| 22:43.41 | eppigy | hello |
| 22:43.52 | isamar | hi folks |
| 22:44.05 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 22:44.07 | isamar | anybody using tor3e+astunicall ? |
| 22:44.18 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 22:44.28 | *** join/#asterisk `paul (n=kutimoy@121.97.99.151) |
| 22:44.41 | RoPBX | Zap/1-1 (None) Up Bridged Call(SIP/6001-081feea8 |
| 22:44.41 | RoPBX | SIP/6001-081feea8 1-dial@macro-trunkdi Up Dial(Zap/g1/04141420663) |
| 22:44.53 | RoPBX | sorry |
| 22:44.55 | isamar | need a hand for that to make zaptel running with astunicall patches + tor3e stuff |
| 22:45.25 | brunner | When I call Asterisk from one SIP provider and Dial() out to another, do my RTP packets ever flow directly from one provider to the other, or do they always go through Asterisk? |
| 22:45.28 | `paul | is it possible for 3 people to do video conferencing.... (even if the other one has no video at all) |
| 22:45.29 | RoPBX | dude7064 did you check file /etc/asterisk/manager.conf ? |
| 22:46.03 | isamar | anybody rathern than coppice knows how to do that ? :-) |
| 22:46.39 | [TK]D-Fender | brunner: Depends if you allw reinvites or not |
| 22:47.48 | brunner | how does the authentication work if I allow reinvites? |
| 22:48.32 | [TK]D-Fender | brunner: After the call is accepted then * tries to reconnect both ends together. "shouldn't" affect auth. |
| 22:48.41 | brunner | okay, thanks |
| 22:48.49 | [TK]D-Fender | brunner: Would if you tired a raw Transfer() however |
| 22:48.51 | [TK]D-Fender | tried* |
| 22:56.13 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 23:07.15 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 23:07.35 | isamar | . |
| 23:11.55 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 23:12.10 | RobH | I know this isnt the best place to ask, that being said. Has anyone used a Polycom Soundstation IP 4000 with asterisk? If so, did you experience any horrible echo? (I have enabled echo cancellation and it does nothing that I can tell.) |
| 23:13.10 | jaytee | drmessano, you around? |
| 23:14.00 | isamar | anybody using mfc/r2 ?? |
| 23:14.24 | moy | isamar: the best way to start is just asking, what's your question? |
| 23:14.34 | jaytee | RobH, I haven't used their IP model but I've used a Polycom Soundstation analog plugged into line off of an ATA without any echo issues unless I'm talking on my cell phone over the channel while in the same room. |
| 23:14.55 | RobH | damn |
| 23:15.02 | RobH | I have horrible echo |
| 23:15.02 | RobH | =[ |
| 23:15.35 | RobH | I am messing around with the thresholds now. I have little to no idea what they do, but they are part of the echo cancellation. Worst case I set them wrong, cannot hear anything, and have to reboot the phone ;] |
| 23:15.49 | RobH | Gotta love polycom and the two minute boot process. |
| 23:16.14 | Kobaz | totally |
| 23:18.04 | isamar | moy: I am using tor3e board and I am trying to use astunicall with that... |
| 23:18.26 | isamar | moy: my doubt is if zaptel-tor3e must be patched also for unicall or not... |
| 23:19.04 | moy | isamar: not that I know of, zaptel does not need any patch at all for R2 signaling |
| 23:20.27 | k-man | RobH: no idea if this helps, but my billion ADSL modem's viop ATA thingo also suffered from bad echo, and the solution there was to reduce the mic and speaker gain settings in the modem/ATA |
| 23:20.44 | k-man | RobH: and that solved the problem on the billion |
| 23:20.52 | isamar | moy: ok.. cool.. dude.. that's what I needed to know |
| 23:20.58 | isamar | moy: thanks a lot |
| 23:21.00 | RobH | k-man: I will check that out, thanks! |
| 23:21.23 | k-man | RobH: hope it helps - i wish I uderstood why these devices get echo? its bizzare |
| 23:23.08 | RobH | the worst part is my soundpoint 501 sounds better on handsfree than my expensive soundstation |
| 23:23.19 | RobH | but it doesnt have enough pickup to use throughout the conference room. |
| 23:24.12 | joako | I know this doesn't help much, but I hear echo some times from IP501 <-> IP501 if one is on speakerphone |
| 23:27.59 | voxter | is there a way to disable sip uri dialing on polycom phones? |
| 23:28.12 | joako | voxter: Yes... look in the config files |
| 23:28.29 | voxter | joako: im looking. they're pretty friggin big, and searching for URI has turned up nothing so far. Hmm. maybe url. |
| 23:28.38 | voxter | found it! |
| 23:28.38 | voxter | :) |
| 23:29.24 | *** join/#asterisk jchonig (n=jch@firewall.honig.net) |
| 23:30.41 | `paul | is there a way to automatically just mute the call (disable sound) i just want to use asterisk as video conference server |
| 23:31.28 | [TK]D-Fender | `paul: Disallow audio codecs |
| 23:31.44 | Dovid | TK: wouldnt that cause a media negotiation error ? |
| 23:32.52 | isamar | moy: I am getting undefined symbol error for get_supervisory_tone_set when starting up asterisk with astunicall.. |
| 23:33.03 | isamar | moy: any thoughts? |
| 23:34.39 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 23:34.43 | jchonig | If I have abbreviations for numbers in dialplan (say 9nnnn expands to NXXNX9nnnn for a remote office ext) is there a way to tell the calling SIP phone to display the expanded number (and even update it's redial list)? |
| 23:36.10 | *** join/#asterisk eldorel (n=eldorel@ip68-108-238-144.br.br.cox.net) |
| 23:37.27 | eldorel | Hello everyone. Quick q: is System() depreciated? I can't seem to get it to do anything (asterisk 1.4.21 on ubuntu 8.10 server) |
| 23:39.26 | Dovid | works for me |
| 23:39.33 | Dovid | what are you trying to do ? |
| 23:40.16 | eldorel | i'm recording an audio file into /tmp and then trying to copy it to the aterisk-sounds default after the user saves it. |
| 23:40.42 | Dovid | ru sure asterisk has permission to write there ? |
| 23:40.46 | [TK]D-Fender | eldorel: Consider the PRIVILEGES the * user has <------- |
| 23:40.58 | Dovid | try looking @ the full log and see if you see any erros there |
| 23:41.25 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 23:42.15 | eldorel | yep, chowned the directory after it didn't work the first time just to be sure. and i've been watching the cli with verbose=15. it seemd to execute fine, but nothing happens.... |
| 23:43.17 | *** join/#asterisk Bonix (n=Bonix@200-195-41-212.isimples.com.br) |
| 23:43.35 | Dovid | do u have permission to write there or is the file read only ? |
| 23:43.44 | Dovid | most likely a persnissions issue |
| 23:44.17 | *** join/#asterisk zenfox (n=steven@c-98-213-240-12.hsd1.il.comcast.net) |
| 23:44.18 | Dovid | i am too tired. cant even spell correclty. night ev1 |
| 23:44.24 | eldorel | the /usr/share/asterisk/sounds file is currently owned by user asterisk |
| 23:44.43 | eldorel | night, thanks anyway |
| 23:48.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:52.42 | *** join/#asterisk ingenius (n=alektro@host2.190-31-177.telecom.net.ar) |
