| 00:00.20 | bisonit | I have been reading all day.. my brain is bleeding. I am calling it a day guys. See you later |
| 00:00.33 | norgs | What I am doing is Dial(SIP/100,...) from an external call and then trying to pickup with **100, but it can't seem to find the call |
| 00:00.49 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 00:05.21 | bmoraca | from what i remember, pickup responds to the INITIAL extension the call comes in on, not the currently ringing extension |
| 00:05.42 | bmoraca | so if your call comes in on a did 5551234, your pickup application needs to pickup extension 5551234 |
| 00:05.51 | bmoraca | but it's a long time since i've looked at that application |
| 00:05.59 | bmoraca | may have changed |
| 00:07.25 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:11.12 | voxter | anyone recall when app_queue was changed so the monitor-type became mixmonitor by default? |
| 00:11.57 | blackest_mamba | in sip.conf, the line permit=192.168.1.0/255.255.255.0 will only allow phones with an in 192.168.1.0/24 to regsiter. Is that correct? |
| 00:12.58 | norgs | bmoraca: thanks, that's what I needed |
| 00:19.02 | *** part/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 00:23.51 | *** join/#asterisk Tapout (n=Tapout@unaffiliated/tapout) |
| 00:24.49 | Tapout | I'm in Calgary,AB, Canada. I've got the magic jack, but it kinda sucks in regards to their client software. I was thinking of getting VOIP setup with asterisk, but Im trying to find a provider that will let me have a 336 area code.. anyone have a recommendation (cheap too) :) |
| 00:27.07 | mazpe | setting a variable in 1.6 remains the same right? exten => s,n, SetVar(AUDIO_DIR=/var/lib/asterisk/sounds/) |
| 00:28.30 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:28.31 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
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| 00:29.33 | *** part/#asterisk gulden (n=gulden@mentesvirtuais-mail.netvisao.pt) |
| 00:30.34 | mazpe | i'm getting: WARNING[31960]: pbx.c:3082 pbx_extension_helper: No application 'SetVar' for extension (ivr-recording, s, 2) |
| 00:32.21 | harry_v | mazpe, put your dialplan on pastebin |
| 00:33.01 | mazpe | ok |
| 00:35.08 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 00:35.14 | norgs | Ok, I have a question about the order that extensions are processed in. In [app-pickup], the first line is "include => app-pickup-custom", then in [app-pickup-custom] I have "exten => **100,"... rules then back in [app-pickup] the "exten => _**.,"... rules continue. |
| 00:35.43 | norgs | the _**. rule seems to be trumping my **100 rules |
| 00:36.04 | mazpe | harry_v: http://rafb.net/p/x4BiaI81.html |
| 00:37.00 | mazpe | harry_v: still work in progress |
| 00:37.17 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 00:37.46 | norgs | so anyhow, my question is: Do later matching rules supercede more specific earlier rules? |
| 00:39.39 | harry_v | I need to head out. |
| 00:40.12 | harry_v | put that in www.asterisk.org forums and thay can see what the issue is. Seems okay with me. Paste your errors there also. |
| 00:40.19 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
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| 00:47.41 | Tapout | anyone know of a provider where you pay liek 10 bucks a month for an end-user, choose the phone number, it gives you 1 number that you can use and it's unlimited calling in North America? |
| 00:54.09 | norgs | Re: my previous question, please disregard. I found: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf+sorting |
| 00:54.34 | *** join/#asterisk xp_prg (n=xp_prg3@99.2.31.217) |
| 00:54.41 | xp_prg | hi all what is that gui for asterisk where you feed the frog? |
| 00:54.43 | bmoraca | $10/mo for unlimited calling? good luck. |
| 00:54.56 | norgs | xp_prg: FreePBX |
| 01:04.45 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:12.54 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
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| 01:15.31 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-78-138.nycmny.east.verizon.net) |
| 01:15.45 | Steve_J-obs | hello everybody!! |
| 01:16.04 | pdmmm | 10 bux a month?! |
| 01:16.05 | norgs | hi |
| 01:16.36 | pdmmm | why not make it 5 |
| 01:16.48 | norgs | no, free! |
| 01:16.54 | pdmmm | wiat wait |
| 01:16.57 | pdmmm | get paid for phone service! |
| 01:17.02 | norgs | yah |
| 01:17.16 | norgs | oh wait, isn't that what premium sms is about? |
| 01:17.48 | pdmmm | crazy |
| 01:19.45 | keith4_ | Tapout: Vonage? |
| 01:19.48 | pwell | they have it |
| 01:19.52 | pwell | Magic Jack |
| 01:19.55 | keith4_ | ha |
| 01:19.58 | keith4_ | yah. magic jack |
| 01:20.03 | pwell | USB to RJ23 |
| 01:20.09 | pwell | soft phone plug and play no CS |
| 01:20.11 | pwell | CD* |
| 01:20.25 | pwell | hack the SIP information and can use it with * |
| 01:20.43 | keith4_ | Tapout: I use voicepulse, which is $11 if you want your own number, and < $0.01/min to all of the US |
| 01:20.46 | pwell | get a local number, voicemail and call forwarding through web interface |
| 01:21.04 | pwell | there is bad sides to majic jack |
| 01:21.05 | pwell | many |
| 01:21.17 | pwell | I prefer voiceplus, but magic jack is proof the game is changing |
| 01:21.29 | pwell | they just market well |
| 01:21.39 | pdmmm | unlimited calling is a sham anyway |
| 01:22.25 | pdmmm | there's a wholesale cost per minute |
| 01:22.42 | pdmmm | just like internet bandwidth |
| 01:22.58 | pwell | better off metered |
| 01:23.37 | x86 | heh, does magic jack work with asterisk? |
| 01:23.53 | pwell | have to hack it up |
| 01:23.58 | x86 | nifty |
| 01:24.03 | x86 | $11/year or so? |
| 01:24.12 | pwell | double + 7 |
| 01:24.13 | x86 | I might do that to get a trunk into Asterisk ;) |
| 01:24.32 | x86 | yeah, $30/year is still decent heh |
| 01:24.35 | x86 | for unlimited calling |
| 01:24.52 | pwell | but you break the TOS |
| 01:24.57 | x86 | considering the local cable co is charging $45/mo for their unlimited calling package :P |
| 01:24.57 | pwell | they could fry you any second |
| 01:25.15 | x86 | meh |
| 01:25.26 | *** join/#asterisk RobertLaptop (n=rmiddle@174-146-136-116.pools.spcsdns.net) |
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| 01:28.52 | pwell | you figure they give you that price and they say ok... this person can ONLY talk for 24 hours a day max... that's X amount of bandwidth needed per month / per year. |
| 01:29.17 | pwell | I think they could catch you easy |
| 01:29.31 | x86 | I'm not talking about creating an ITSP with it hah |
| 01:29.35 | x86 | just home usage |
| 01:29.43 | pwell | I just want a better voice mail system, with a little IVR |
| 01:29.55 | x86 | usage would be low anyway -- too low to justify a land line for sure |
| 01:30.41 | pwell | what I wonder is if I can do cheaper than magic jack and still get a local phone number |
| 01:30.54 | pwell | metered would be perfect for me |
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| 01:32.47 | *** part/#asterisk xp_prg (n=xp_prg3@99.2.31.217) |
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| 01:50.38 | Steve_J-obs | pwell: you want something cheaper than magicjack? |
| 01:51.02 | drmessano | lol |
| 01:51.09 | drmessano | Thats kinda funny |
| 01:51.17 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 01:51.18 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 01:51.18 | *** mode/#asterisk [+o denon] by ChanServ |
| 01:51.37 | drmessano | Magicjack is pretty much as cheap as it gets |
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| 01:53.27 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 01:53.33 | Steve_J-obs | pwell: you would have to have a pop on every market in the country |
| 01:54.13 | docid | well, if ya wanna limit yourself to 5 minutes per call ya could always find a free did and use nonoh.net to freedial both sides, wholey hokey, but thats kinda where ya at when looking for cheaper than magicjack |
| 01:54.26 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:55.06 | Steve_J-obs | BTW, do you guys know that skype is making its codec public domain? |
| 01:55.44 | docid | yeah, saw that |
| 01:56.06 | Steve_J-obs | skype and magicjack have a very similar codec |
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| 01:58.00 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 01:58.39 | exothermc | Anyone suggest a non-home built solution that interfaces SIP to a PRI, for legacy PBX installs? |
| 01:58.44 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 01:59.02 | russellb | "non home built solution" ? |
| 01:59.04 | russellb | what does that mean? |
| 01:59.12 | russellb | is an asterisk box a home built solution? |
| 01:59.27 | russellb | if so, you're kind of in the wrong room :-) |
| 01:59.30 | exothermc | russellb: someone cobbling together a atom processor, with a sangoma card. |
| 01:59.41 | pdmmm | russellb: he wants a class 5 switch |
| 02:00.35 | exothermc | pdmmm: Well a true class 5 switch would be overkill. This really just needs to be like an SPA8000 except with a PRI interface instead of FXS ports. |
| 02:00.52 | exothermc | oh ya and full 24 port capabilities. |
| 02:00.52 | pdmmm | exothermc: http://www.red-fone.com/Products/fonebridge2-EC/ |
| 02:01.11 | russellb | blech.. |
| 02:01.49 | exothermc | pdmmm: Isn't that device for converting the other direction? taking legacy PRI feed from the telco and converting to SIP? |
| 02:02.10 | pdmmm | same thing |
| 02:02.31 | pdmmm | hell |
| 02:02.33 | pdmmm | use asterisk |
| 02:02.50 | jaytee | redfone? run away! run away fast!!! |
| 02:03.20 | pdmmm | 2 years ago I had asterisk + digium t1 card |
| 02:03.26 | pdmmm | it worked insanely well back then |
| 02:03.34 | jaytee | it still does |
| 02:03.39 | pdmmm | exactly |
| 02:03.49 | pdmmm | asterisk isnt windows |
| 02:04.34 | carrar | yeah, duh, it's a browser! |
| 02:04.38 | jaytee | which IMHO is a very good thing |
| 02:04.45 | *** part/#asterisk exothermc (n=miles@74.85.89.146) |
| 02:05.00 | drmessano | Thank you, Ric Romero |
| 02:05.06 | pdmmm | was it something i said? |
| 02:05.11 | drmessano | In other news, Chocolate isn't bread |
| 02:06.03 | jaytee | I don't want my phone system to have a blue screen event when I really need it or get more trojans and viruses than all the STDs that hookers in Manila get. |
| 02:06.33 | jaytee | even when chocolate is bread they tend to call it cake instead |
| 02:08.13 | *** join/#asterisk salzh (n=salzh@61.87.216.208) |
| 02:08.40 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 02:08.41 | salzh | hi, how can i mute a channel during a conversation |
| 02:08.49 | pdmmm | hit the mute button! |
| 02:08.58 | pdmmm | :D |
| 02:09.27 | docid | any way to compare dial number with cid number and do something specific if they match? ugg, im feelin braindead today |
| 02:09.46 | salzh | is there some commands in cli |
| 02:10.04 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:10.13 | docid | like if dnid=cid then goto vm |
| 02:10.23 | docid | i guess i should just go read more |
| 02:10.41 | pdmmm | http://www.voip-info.org/wiki-Asterisk+cmd+GotoIf |
| 02:10.55 | docid | thank ya kindly:) |
| 02:11.03 | pdmmm | yer welcome ;) |
| 02:11.36 | pdmmm | why does redfone suck? |
| 02:11.38 | pdmmm | i never used it |
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| 02:17.22 | eric256 | is there anywhere i can put something in the dial plan so that it will always get called on hangup regardless of the context? or better based on the context it came in on? i.e. a call comes in I run a script, they move around the system, then hang up (and i run another script) |
| 02:17.53 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:18.10 | eric256 | currently i can only get the h dial plan to run for the context at the end of the call... |
| 02:18.44 | *** join/#asterisk BeeBuu (n=beebuu@219.132.188.228) |
| 02:19.45 | carrar | eric, import your h context into all your contexts |
| 02:21.24 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 02:21.48 | eric256 | i was afriad you'd say that ;) |
| 02:22.54 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 02:24.44 | eric256 | guess i make a macro store the originating context, and one to run the script if that has been set, and then add that to the ends .... it could work |
| 02:25.01 | *** join/#asterisk jtodd (n=jtodd@150.sub-75-209-72.myvzw.com) |
| 02:25.01 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 02:26.49 | jaytee | pdmmm, redfone sucks because it uses TDMoE from it's box to Asterisk. |
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| 02:28.13 | Micc | anyone tested the wp04 wifi phone? |
| 02:28.17 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
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| 02:37.55 | *** mode/#asterisk [+o Mog] by ChanServ |
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| 02:56.32 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:57.33 | *** join/#asterisk doolph (n=doolph@190.141.68.31) |
| 02:57.38 | doolph | hi |
| 02:59.21 | BeeBuu | hello |
| 03:00.21 | doolph | anyone here using asterisknow with asterisk 1.6? |
| 03:02.06 | *** join/#asterisk stoked (n=df@S01060016b62857a6.vc.shawcable.net) |
| 03:02.56 | stoked | hi all, I'm trying to spoof this handset I have, and everything's working except that the handset always sends a 0111 before the # |
| 03:03.14 | stoked | is there a way to strip this only from this handset? |
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| 03:16.57 | *** join/#asterisk orkid_ (n=orkid@unaffiliated/orkid) |
| 03:19.28 | keith4_ | !asterisknow |
| 03:19.31 | keith4_ | ~asterisknow |
| 03:19.32 | jbot | asterisknow is probably based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
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| 03:24.58 | BeeBuu | hi,all.anyone know is a mysql searching agi faster than a MYSQL command in asterisk? |
| 03:26.08 | KavanS | I'm trying to set a monitor filename....and I find that it's inserting slashes |
| 03:26.30 | KavanS | is there any way to set a variable/replace/etc. to make it so it does not use said "slashes" ? |
| 03:31.56 | *** part/#asterisk stoked (n=df@S01060016b62857a6.vc.shawcable.net) |
| 03:34.57 | *** join/#asterisk _abc_ (n=no@bas3-toronto06-1177890428.dsl.bell.ca) |
| 03:35.02 | _abc_ | hi all |
| 03:35.22 | _abc_ | quick question: how do i set the priority of asterisk ? (niceness) ? |
| 03:35.45 | _abc_ | command line using nice -5 asterisk ... fails, the niceness is reset to 0 |
| 03:35.52 | _abc_ | (default system niceness) |
| 03:35.58 | _abc_ | renicing works fine |
| 03:36.14 | *** join/#asterisk freetown (n=freetown@ipvpn098241.netvigator.com) |
| 03:36.34 | freetown | cor...what a crowd |
| 03:36.38 | freetown | hello all |
| 03:37.24 | freetown | it's has probably come up many times but i wonder what sip phones you would recommend, both ethernet connected and WIFI? |
| 03:37.43 | freetown | s/it's/it/ |
| 03:37.49 | *** join/#asterisk josh77 (n=static@CPE0016b6116a89-CM00111a233e60.cpe.net.cable.rogers.com) |
| 03:37.58 | freetown | nice bot |
| 03:38.27 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 03:41.54 | _abc_ | bot ? it's a parrot wired to irc |
| 03:42.11 | _abc_ | so what about niceness |
| 03:42.12 | _abc_ | ? |
| 03:42.29 | freetown | well...other parrots don't grok s/// |
| 03:42.44 | freetown | or this is the first one i saw |
| 03:42.53 | KavanS | damn |
| 03:43.09 | _abc_ | neural network or bayesian parrots grok whatever 'looks' right |
| 03:43.17 | KavanS | yeah I need that for my asterisk ;) |
| 03:43.25 | KavanS | I set a variable for filename, but it's setting it with a "/" |
| 03:43.31 | KavanS | and making it so monitor doesn't work :( |
| 03:43.54 | _abc_ | so HOW do i set niceness of a process ?! |
| 03:44.02 | KavanS | _abc_: you google that yet? |
| 03:44.08 | _abc_ | yes, no joy |
| 03:44.11 | KavanS | I google'd my question....can't find result |
| 03:44.18 | KavanS | pretty sure nice will set it |
| 03:44.28 | _abc_ | and don't say nice -5 asterisk ... because it resets its niceness to 0 |
| 03:44.31 | KavanS | modify the init script? |
| 03:45.43 | _abc_ | KavanS: there is such a thing as testing. i run the binary directly as above and it sets its own niceness to 0 |
| 03:46.09 | _abc_ | so, short of reading the *"§$%!! source, WHERE is the niceness setting |
| 03:46.14 | KavanS | ok...not sure how to help ya man |
| 03:46.28 | KavanS | that's how I'd do it....and if it became a crazy problem, I'd upgrade my hardware |
| 03:46.35 | KavanS | what do you need to set niceness for? |
| 03:46.43 | josh77 | hi all. I have a system with a tdm800p and 2x4 FXO modules. For some reason, the audio sounds quite rebotic and somewhat in slow motion. dahdi_test shows 99.999% avg so it's not quite clear to me what's wrong. i put a sample at http://qualm.net/wav to demonstrate (playback of included vm-msginstruct.gsm). |
| 03:46.43 | _abc_ | KavanS: not reboot first :) |
| 03:47.31 | josh77 | all 8 channels seem to exhibit the same behaviour |
| 03:47.45 | josh77 | i'm hoping it's not due to the PC :( |
| 03:48.24 | _abc_ | KavanS: prevents dropouts when other programs compete for cpu |
| 03:48.36 | KavanS | sounds like you need a support contract from digium :) |
| 03:48.56 | KavanS | _abc_: or maybe it's time to upgrade hardware? |
| 03:50.17 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 03:50.53 | _abc_ | KavanS: sounds like i need to increase the NICENESS of the virtual cpu this runs on ... |
| 03:51.00 | KavanS | lol |
| 03:51.01 | _abc_ | *decrease |
| 03:51.27 | josh77 | KavanS, _abc_, have you guys used tdm800p before? :) |
| 03:51.35 | *** part/#asterisk jcoffi (n=jcoffi@75.147.157.49) |
| 03:51.38 | KavanS | naw I have tdm410p |
| 03:51.43 | _abc_ | admires the sense of humor of *nix commands: decreasing the niceness of a process makes it more aggressive - a polite point of view |
| 03:52.05 | josh77 | yeah i have another system with a tdm410p and it works well |
| 03:52.13 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 03:52.17 | josh77 | i think it's probably the server that the 800p is in :( |
| 03:52.27 | KavanS | hrm |
| 03:52.48 | KavanS | only way to find out for sure is to put it in another machine during "off hours" |
| 03:53.07 | josh77 | yeah i think so too. |
| 03:54.42 | josh77 | did you hear the sound? http://qualm.net/wav |
| 03:54.50 | josh77 | i once had a system that stuttered a lot |
| 03:54.59 | josh77 | but zttest showed some really low %'s |
| 03:55.11 | josh77 | avg was like 70%, had to scrap the pc |
| 03:55.23 | *** join/#asterisk mahiti-irc (n=mahiti1@203.145.183.210) |
| 03:55.31 | freetown | has anybody here used phones from Planet? |
| 04:00.06 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 04:01.51 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 04:02.10 | KavanS | damn |
| 04:02.17 | freetown | begs for advice |
| 04:02.35 | freetown | the school where i work is contemplating replacing their telephone system |
| 04:02.59 | freetown | i obviously want to put in Asterisk and get appropriate phones for them |
| 04:03.02 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 04:03.08 | freetown | any suggestions? |
| 04:03.24 | carrar | install Asterisk! |
| 04:03.28 | freetown | some nice wireless phones that can traverse access points would be great too! |
| 04:03.45 | freetown | carrar, yes, but you also needs phones man |
| 04:03.54 | carrar | Hatachi wifi phone? |
| 04:04.01 | freetown | hitachi? |
| 04:04.54 | freetown | http://www.mgraves.org/voip/2008/08/hitachi-getting-out-of-wifi-sip-handset-business/ |
| 04:05.21 | freetown | nice... |
| 04:05.29 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:06.00 | freetown | aw come on |
| 04:06.05 | freetown | gets on his knees |
| 04:06.09 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 04:06.26 | freetown | i have all the info i need except good phones to go with asterisk |
| 04:07.10 | freetown | i have found a sip provider here in HK, i have asterisk running with softphones...i just need good hardware phones to complete the proposal |
| 04:07.19 | freetown | please, any suggestions? |
| 04:07.27 | freetown | no aastra available in HK :P |
| 04:07.29 | freetown | :( |
| 04:07.51 | _abc_ | pirelli make one of the wifi/dect phones (wireless) |
| 04:07.53 | _abc_ | there are others |
| 04:08.59 | freetown | hmm...has anybody blanketed their premises with DECT? |
| 04:09.12 | freetown | do phones move from dect station to station? |
| 04:10.19 | *** join/#asterisk DarkRift (n=dark@65.92.250.156) |
| 04:12.02 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 04:15.46 | _abc_ | freetown: no they pair but professional installations have repeaters |
| 04:16.08 | _abc_ | afaik dect cannot cell hop |
| 04:16.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 04:16.53 | freetown | _abc_, ah so wifi phones would be the way to go then i take it |
| 04:18.25 | freetown | the school is getting a wifi blanket so for those who must be available no matter where they are...i guess i need to suggest wifi for them. Classrooms can have their own dect wireless. |
| 04:22.19 | jameswf | wifi phones == total suckage |
| 04:22.54 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 04:23.05 | eric256 | anyone heard of a virus that uses softphones? i have one softphone that made 1100 calls in one day...all bogus |
| 04:23.17 | eric256 | changed the security code on that device and they all went away |
| 04:23.55 | freetown | jameswf, oh...so...no 'roaming' solution on the sip side of things? The only way to go would be DECT + repeaters? |
| 04:24.32 | MaliutaLap | eric256: that would be dependant on the OS and the actual piece of software |
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| 04:27.25 | _abc_ | freetown: sip can roam in software |
| 04:27.36 | _abc_ | it's called handover or something like that |
| 04:28.28 | MaliutaLap | sip is a layer above the things that need to roam |
| 04:29.03 | _abc_ | it roams by logging into another proxy which does the 'roaming'. it is not seamless (but it could be) |
| 04:29.11 | MaliutaLap | the sip would be on an IP network, the "roaming" is done several layers below |
| 04:30.10 | _abc_ | no the roaming in this case would have to be done by sip (different networks, not one) |
| 04:30.36 | _abc_ | think about using a wifi phone in an internet cafe and walking across the street into another with a different net |
| 04:30.44 | _abc_ | that's more or less the picture |
| 04:30.50 | _abc_ | for non-integrated networks |
| 04:32.14 | freetown | _abc_, well...i am talking about a single site...the school |
| 04:33.06 | freetown | i suppose roaming would work like...person moves from ap zone a to ap zone b. Looses connection during move |
| 04:33.13 | freetown | phone reconnects |
| 04:33.27 | freetown | so ongoing conversations will probably be cut |
| 04:33.50 | freetown | unless the wifi traversal works properly |
| 04:38.18 | _abc_ | freetown: it does not |
| 04:38.34 | _abc_ | but depending on size you could get away with a single network and repeaters |
| 04:38.59 | freetown | _abc_, is that DECT you are talking about with 'repeaters'? |
| 04:39.02 | _abc_ | each node will cover some 900 m^2 |
| 04:39.06 | _abc_ | freetown: wifi |
| 04:39.22 | _abc_ | dect repeaters should exist but i have never seen one |
| 04:39.29 | freetown | wifi repeaters? that's new... |
| 04:39.39 | _abc_ | freetown: really ? |
| 04:39.52 | _abc_ | see 'range extender' and the like |
| 04:41.32 | freetown | we are getting a new wireless-n system put in place and it is supposed to be able to allow you to walk with a laptop all over the school without loosing your connection |
| 04:41.42 | _abc_ | sure |
| 04:41.47 | _abc_ | see above |
| 04:41.49 | freetown | supposedly thanks to MIMO support. |
| 04:42.44 | _abc_ | wifi roaming works but there are several methods |
| 04:43.00 | _abc_ | bye |
| 04:43.02 | *** part/#asterisk _abc_ (n=no@bas3-toronto06-1177890428.dsl.bell.ca) |
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| 05:01.25 | mahiti-irc | Hi |
| 05:01.41 | mahiti-irc | anyone knows the GUI setup for asterisk? |
| 05:03.50 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 05:04.08 | MaliutaLap | mahiti-irc: READ THE TOPIC |
| 05:04.17 | MaliutaLap | no gui support here |
| 05:06.14 | mahiti-irc | then where can i found the GUI suport? |
| 05:06.31 | MaliutaLap | READ THE TOPIC |
| 05:06.49 | MaliutaLap | ~topic |
| 05:08.51 | kb3ien | trying to edit a .ulaw file in Audacity (Intrepid Ibex) but the file format is unrecognised. any one else notice this? |
| 05:09.56 | MaliutaLap | kb3ien: use sox to turn it to a wav |
| 05:10.05 | MaliutaLap | edit it then re convert it |
| 05:10.59 | kb3ien | should have thought of that... |
| 05:14.51 | *** part/#asterisk eric256 (n=Administ@c-75-71-188-88.hsd1.co.comcast.net) |
| 05:23.06 | *** join/#asterisk crazyx__ (n=crazyx@41.249.253.247) |
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| 05:29.47 | freetown | now how do you get jbot to do that? |
| 05:30.05 | MaliutaLap | jbot can play with me anytime he likes ... and I won't report him to the cops ;) |
| 05:30.35 | MaliutaLap | freetown: you have to fiddle with him |
| 05:30.52 | freetown | fiddles jbot |
| 05:31.05 | freetown | i think i broke my fiddle |
| 05:31.17 | freetown | MaliutaLap, you owe me one fiddle. |
| 05:31.25 | MaliutaLap | try moving one key to the left |
| 05:31.30 | MaliutaLap | then you get a diddle |
| 05:31.45 | mahiti-irc | newly installed asterisknow |
| 05:31.58 | kb3ien | yuiiro ? |
| 05:32.00 | MaliutaLap | wonders if the term "fiddle" has the same meaning outside .au |
| 05:32.04 | freetown | great, now i can fiddle on my diddle |
| 05:32.11 | kb3ien | sorry i'm a dvorak user. |
| 05:32.18 | MaliutaLap | mahiti-irc: great, no support for that hear either |
| 05:32.30 | freetown | MaliutaLap, well...it's an instrument somewhere |
| 05:32.42 | MaliutaLap | freetown: no, thats called a violin |
| 05:32.59 | MaliutaLap | ever heard the term "kiddy-fiddler"? |
| 05:33.16 | freetown | MaliutaLap, so long as it means instrument in them parts. |
| 05:33.28 | freetown | MaliutaLap, now that one you just mentioned is new to me... |
| 05:33.29 | MaliutaLap | those parts |
| 05:33.43 | freetown | fiddle with your car or what not... |
| 05:33.48 | MaliutaLap | and no, fiddle is not an istrument |
| 05:33.49 | freetown | oh... |
| 05:34.12 | MaliutaLap | so you fiddle yourself? |
| 05:34.15 | freetown | first time i have heard of 'fiddle' being used in that context...messing with yourself... |
| 05:34.59 | MaliutaLap | 'oath |
| 05:36.02 | freetown | you aussie are getting far removed from the Queen's english |
| 05:36.06 | freetown | aussies |
| 05:36.27 | freetown | That first thing i think of is this: http://en.wikipedia.org/wiki/Fiddle ...like you said, voilin |
| 05:36.37 | MaliutaLap | freetown: a damned sight closer than the yanks |
| 05:36.50 | freetown | MaliutaLap, no contest on that one! |
| 05:36.58 | MaliutaLap | freetown: and I don't speak gay persons english :P |
| 05:37.47 | freetown | hahaha, them UK unis must turn every man into a fag then eh? |
| 05:38.33 | kb3ien | hrm: sox soxio: Can't open input file `please-hold-silence.ulaw': unknown file type `ulaw' |
| 05:39.26 | MaliutaLap | freetown: they teach them php and force them to join groups?? you've heard of PHp Abusers Groups??? aka PHAGs? |
| 05:39.39 | MaliutaLap | hmm |
| 05:40.33 | freetown | so MaliutaLap ... what do you think of aastra phones? |
| 05:42.41 | MaliutaLap | kb3ien: what was the full command you used? |
| 05:42.53 | MaliutaLap | freetown: haven't had a chance to play with any |
| 05:43.05 | MaliutaLap | freetown: I only have a cisco 7941 |
| 05:45.23 | freetown | MaliutaLap, oh, so you mainly use softphones with asterisk? |
| 05:46.13 | MaliutaLap | freetown: no, I have 2 different cordless handsets attached via tdm400p and then I have my cisco |
| 05:46.31 | MaliutaLap | freetown: I only use a softphone on the laptop when not at home |
| 05:46.42 | freetown | MaliutaLap, ah. i see. a home pbx :) |
| 05:47.39 | kb3ien | sox -V please-hold-silence.ulaw -r 8000 -c 1 please-hold-silence.wav |
| 05:49.29 | MaliutaLap | freetown: since I'm not working ATM ... yes |
| 05:50.32 | MaliutaLap | kb3ien: sounds like your sox was built without support for ulaw |
| 05:50.43 | MaliutaLap | kb3ien: you could build your own |
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| 05:51.54 | kb3ien | whats odd is that i MADE that file. i dont seem to have the original.... |
| 05:52.08 | kb3ien | it made the file .ulaw... |
| 05:53.12 | freetown | bye all |
| 05:54.26 | MaliutaLap | kb3ien: just re-record the thing in audacity, make it how you want and then try conversion? |
| 05:54.48 | MaliutaLap | kb3ien: then you'd have the wav to turn into other formats if you need it |
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| 06:12.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 06:16.12 | jbjapan | hi |
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| 06:18.06 | crazyx__ | hello everybody. one question plz : where we change the mode of a TC400B from g723 to mixed? |
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| 06:41.21 | k-man | is there some way I can get asterisk to email voicemails in mp3 format? |
| 06:52.31 | stabler | k-man: yea.. record as mp3 |
| 06:52.46 | k-man | stabler: how? |
| 06:54.58 | stabler | k-man: no positive :) |
| 06:56.20 | stabler | k-man: after reviewign my voicemail.conf |
| 06:56.32 | stabler | k-man: i see tehre is something like this: format = wav49|gsm|wav |
| 06:56.46 | k-man | stabler: afaict, you cannot just stick mp3 in there |
| 06:56.57 | *** join/#asterisk Gopaul (n=chatzill@61.17.185.118) |
| 06:57.19 | Gopaul | I am getting liek this error in my asterisk console Got retransmission request sequence numbers greater than 5. Retransmitting 1 message(s). what could be the issue? |
| 06:57.38 | Gopaul | its not a error its a information what it identifies? |
| 06:58.55 | stabler | Gopaul: whaaa? |
| 07:13.29 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
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| 07:18.53 | k-man | how can i call a script after voicemail has been recorded to futher process the voicemail? |
| 07:22.49 | *** join/#asterisk MaliutaLap (n=biteme@203.171.195.75) |
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| 07:42.41 | Gopaul | stabler: my connection was disconnecte |
| 07:43.00 | Gopaul | stabler: actually I am getting this message in my asterisk console |
| 07:43.14 | fcois93 | I have an interresting question about MeetMe in http://forums.digium.com/viewtopic.php?t=67545 |
| 07:43.22 | fcois93 | please have a look for ! |
| 07:44.56 | Gopaul | fcois93: creating a conference is not dependent of the card |
| 07:45.15 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:45.31 | Gopaul | I am getting this message n asterisk console Got retransmission request sequence numbers greater than 5. Retransmitting 1 message(s) what could be the reason? |
| 07:46.14 | fcois93 | Gopaul: MeetMe use a channel. if we have a card, it use the card to create the channel, no? |
| 07:46.53 | Gopaul | fcois93: you mean zap/1? |
| 07:47.14 | fcois93 | Gopaul: "To be able to run MeetMe, your Asterisk server needs a Zaptel timer. This is provided from Digium cards or through various drivers" |
| 07:47.15 | Gopaul | fcois93: zap/1, zap/2, etc..? |
| 07:47.24 | fcois93 | Gopaul: "http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe" |
| 07:47.27 | Gopaul | fcois93: yes |
| 07:47.34 | Gopaul | fcois93: we need timer |
| 07:47.57 | Gopaul | fcois93: for timer if you have the card you can load the zaptel module if you dont have card you can load ztdummy |
| 07:48.17 | fcois93 | Gopaul: if we need timer, with a card is it better? |
| 07:48.45 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 07:48.58 | Gopaul | fcois93: if you have card its better |
| 07:49.34 | fcois93 | Gopaul: is it better to have a card for that ? |
| 07:50.53 | *** join/#asterisk af_ (n=getsmart@88-149-230-241.dynamic.ngi.it) |
| 07:51.02 | Gopaul | fcois93: its not a thing like it will better by using a card |
| 07:51.16 | Gopaul | fcois93: without the card also it will be better with dummy timer |
| 07:51.42 | Gopaul | fcois93: if you feel any difficultied a USB timer is available you can use that |
| 07:52.12 | fcois93 | Gopaul: but, is there a difference between zaptel and dummy for the eprformance ? |
| 07:52.32 | Gopaul | fcois93: no |
| 07:53.20 | fcois93 | Gopaul: I thought that with a card, the CPU and RAM are less use... |
| 07:54.02 | Gopaul | fcois93: most probably yes bcoz the card will handle the timer |
| 07:54.17 | Gopaul | fcois93: if you want to use in prodcution environment you can use card |
| 07:54.30 | Gopaul | fcois93: otherwise you can use dummy timer itself |
| 07:55.11 | fcois93 | Gopaul: I need to have a lots of conferences in one server, I need to have the best performance |
| 07:55.26 | fcois93 | Gopaul: in production environment |
| 07:55.39 | Gopaul | fcois93: then better you use card itself that will give you good performance |
| 07:55.58 | Gopaul | fcois93: use sangoma card its the best one |
| 07:56.15 | fcois93 | Gopaul: is it from digium ? |
| 07:56.31 | Gopaul | fcois93: its different from digium |
| 07:56.38 | Gopaul | fcois93: www.sangoma.com |
| 07:57.00 | fcois93 | Gopaul: and it works with asterisk! |
| 07:57.08 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 07:57.17 | fcois93 | Gopaul: why is it better? |
| 07:57.24 | Gopaul | fcois93: yes its very much compatible with asterisk, yate, freeswitch, 3cx |
| 07:57.52 | Gopaul | fcois93: bcoz sangoma will have difference of 30% less of CPU utlization when comparing to digium |
| 07:58.11 | Gopaul | have any one faced this kind of message in asterisk Got retransmission request sequence numbers greater than 5. Retransmitting 1 message(s)? |
| 07:59.11 | fcois93 | Gopaul: is there a doc for that compare ? |
| 07:59.38 | Gopaul | fcois93: I had one let me check whether its there with me |
| 07:59.38 | fcois93 | Gopaul: is there some doc to install? or is it same as install a digium card ? |
| 07:59.57 | Gopaul | fcois93: for installation you can find here wiki.sangoma.com |
| 08:01.34 | fcois93 | Gopaul: what is the price difference ? |
| 08:01.49 | *** part/#asterisk vlt (n=dm@suez.activ-job.com) |
| 08:02.10 | Gopaul | fcois93: price I am not sure actually where are you from? |
| 08:02.59 | fcois93 | Gopaul: from france |
| 08:03.10 | fcois93 | Gopaul: that can be ok? http://www.sangoma.com/products_and_solutions/hardware/digital_voice_and_data_networking/a101.html |
| 08:03.30 | Gopaul | fcois93: so are you going to use T1 line? |
| 08:03.58 | Gopaul | fcois93: If yes then A101 is fine.. |
| 08:04.50 | fcois93 | Gopaul: I just want a card to do MeetMe from SIP to SIP I dont use T1 or others |
| 08:05.15 | Gopaul | fcois93: then you can buy USB timer device, let me provide the link for you |
| 08:07.16 | fcois93 | Gopaul: I cant see how USB timer can be better than use ztdummy? it is a usb! |
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| 08:16.17 | fcois93 | Gopaul: I cant find USB controler ! |
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| 08:36.15 | awk | hi, hmm, what would be the best way going forwrad to 'encrypt' voice recordings.. copy codec_gsmc to codec_encgsm.c add some encryption function directly on the codec_encgsm.c and set monitor format to encgsm (fake codec). and then i can have codec_encgsm.c read my software license as encryption key? |
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| 08:52.16 | dr_gogeta86 | hi to all |
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| 09:03.04 | Cornev3 | Hi all |
| 09:03.24 | Cornev3 | what codec to load on asterisknow 2.0 beta 6 |
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| 09:39.57 | NewCastleScott | hello all, Im trying my hand at setting up * for a one phone system for now. Im having issues with starting it. I have pasted the debuging info along with the "use flags" (Im on a gentoo box) here ---> http://gentoo.pastebin.ca/1354282 I thank you in advance for any direction one can point me. |
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| 09:58.08 | jermey_g | NewCastleScott:sure we can give you a hand |
| 09:58.58 | NewCastleScott | thanks jermey_g , Is it required to use mysql if the setup is going to be only for outgoing calls with no hold? |
| 10:01.35 | jermey_g | NewCastleScott:no its not. |
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| 10:15.56 | kaldemar | NewCastleScott: you either need to make a configuration file for chan_phone or disable the module. most likely you won't need the module, so you can disable it by adding "noload => chan_phone.so" in /etc/asterisk/modules.conf under [modules]. |
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| 10:22.56 | NewCastleScott | thanks kaldemar |
| 10:24.44 | NewCastleScott | what am I missing for sip dev's not to be found? http://gentoo.pastebin.ca/1354302 Im using a digium IAXY box |
| 10:29.33 | kaldemar | what do you mean by sip dev's? |
| 10:31.51 | NewCastleScott | isnt the digium iaxy box a sip device? |
| 10:33.12 | NewCastleScott | like I said this is all new to me. I know its a powerful platform but can be somewhat simple.... so I have been told |
| 10:35.22 | NewCastleScott | from what I understood I only need a few files; sip.conf and iax.conf |
| 10:35.38 | NewCastleScott | Im not using voicemail or incoming calls for now |
| 10:36.35 | kaldemar | NewCastleScott: no, the iaxy is an IAX device. |
| 10:37.16 | NewCastleScott | ahh so thats why "sip show peers" is not found |
| 10:37.30 | kaldemar | you need more files than sip.conf or iax.conf. your dialplan for example is in extensions.conf, without it, you can't make any calls. |
| 10:38.06 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 10:38.16 | scruz | good day |
| 10:38.19 | kaldemar | in case you haven't already taken a look at this, you'll find this very useful for the basics and also for advanced stuff: |
| 10:38.23 | kaldemar | ~book |
| 10:38.44 | 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 |
| 10:38.44 | NewCastleScott | ahh right on, thanks |
| 10:39.21 | *** join/#asterisk ringing (n=who@212.146.103.154) |
| 10:39.43 | ringing | hello |
| 10:39.51 | scruz | in * 1.4+, chan_ss7 is supposed to be replaced by chan_dahdi, right? |
| 10:40.36 | ringing | running * 1.6.0.6 and having problems matching incoming calls to the right peer |
| 10:41.04 | ringing | i have multiple accounts from the same sip provider, same host but different usernames |
| 10:41.41 | ringing | when a calls comes in, * matches the last peer defined in sip.conf |
| 10:42.12 | ringing | it goes to the right context/extension defined with register, but the channel get set to the wrong peer |
| 10:42.34 | ringing | i've been reading the docs and lists for a couple of days now and can't get it sorted |
| 10:43.25 | ringing | sometime in the past there was chan_sip2, a temporary channel driver that fixed this but it's no longer developed and i don't think it works with 1.6 |
| 10:43.35 | ringing | do you guys have any ideas? |
| 10:44.19 | ringing | how to match incoming calls to peer by username, not by ip/port |
| 10:47.15 | scruz | ringing: may i see a sample from your extensions.conf? i've never had that problem before |
| 10:47.24 | angryuser | ringing: hello, the problem is that some providers dont like more than one trunk registered to 1 Ip , normally if you want to have multiple DID you need to have one trunk |
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| 10:47.58 | angryuser | ringing: and when the negociation comes in * identify a wrong peer |
| 10:48.22 | ringing | angryuser: my provider seems to like to have one trunk for each phone number.. :( |
| 10:48.24 | angryuser | angryuser: look at sip debug carufully and i think you will the the problem |
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| 10:48.36 | angryuser | see the* |
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| 10:49.04 | SparFux | If I don't want to call out with prefix "0" but instead use a prefix for all internal stuff, what would be a good prefix then? |
| 10:49.24 | angryuser | ringing: if you are sure of that have you setup the fromuser for each sip peer ? |
| 10:49.59 | MrChimpy | guys any thoughts on how to monitor the audio coming from a specific channel? ie. just what a caller is saying - not what they're hearing. |
| 10:50.01 | MrChimpy | ? |
| 10:50.10 | lstep | Hello, When in a conversation, I'd like to be able to send a stream to the called party (an mp3 file for example). Is it possible with Asterisk? |
| 10:50.17 | ringing | angryuser: yes, here's an extract from sip debug |
| 10:50.18 | MrChimpy | chanspy does both, which is no good to me :( |
| 10:50.21 | ringing | Using INVITE request as basis request - 00181-UE-0105ee72-2db81ba06@cirpack-e164 |
| 10:50.21 | ringing | No user '07xxxxxxxxx' in SIP users list |
| 10:50.21 | ringing | Found peer '021xxxxx40' for '07xxxxxxx' from 212.xxxxxx:5060 |
| 10:50.40 | ringing | the placed was called however via 021xxxxxx32 |
| 10:51.01 | ringing | yes, the entries in sip.conf for each trunk contain fromuser, fromdomain et al |
| 10:51.06 | scruz | lstep: consider the Playback() application. it won't work if * doesn't have mp3 support built-in, or it can't transcode the media |
| 10:51.52 | angryuser | ringing: paste the sip debug of the invite which comes in with the notoce of the peer who should be ringed and the userd who is actually ringed |
| 10:51.58 | lstep | scruz: ok, but if the conversation is already "running", that means I already have 2 bridged channels, how do I add a third one (that has the mp3) ? |
| 10:52.00 | angryuser | notice* |
| 10:52.03 | scruz | lstep: if the conversation is already ongoing between two people, you might need to activate dtmf |
| 10:52.04 | SparFux | Perhaps whan I use 0* as a prefix for internal stuff, that would be the best. But I wonder wether there would be clashes with some other use of *. |
| 10:52.37 | scruz | and provide for it in the dialplan |
| 10:52.49 | lstep | scruz: doesn't the Dial() function block ? |
| 10:53.29 | scruz | Dial blocks |
| 10:53.36 | lstep | scruz: I thought that if, in my dialplan I've already Dial()'ed, I couldn't get anything more from the Dialplan until someone hangs up |
| 10:53.57 | ringing | angryuser: the dialplan works ok, the call goes where it should.. the problem is that the call gets the channel (and peer-id) of a wrongly matched peer |
| 10:54.03 | lstep | scruz: so how can asterisk get the dtmlf, then ? |
| 10:54.43 | angryuser | ringing: No user '07xxxxxxxxx' in SIP users list who is user '07xxxxxxxxx' ? try to create a peer [07xxxxxxxxx] |
| 10:55.06 | ringing | 07xxxxxx is a mobile phone ringing |
| 10:55.31 | ringing | as far as i see, it's a shortcoming in asterisk |
| 10:56.22 | lstep | scruz: Is there something special in asterisk to intercept dtmf ? |
| 10:56.45 | ringing | all users that bumped into this got a reply that asterisk can't differentiate calls inbound via multiple trunks with same host and port, but different usernames |
| 10:56.53 | lstep | scruz: like there's 'h' for hangup, 't' for timeout in dialplan |
| 11:01.46 | scruz | lstep: maybe the 'G' flag for Dial is what you need |
| 11:02.15 | lstep | scruz: OK, thanks, I'll have a look |
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| 11:03.51 | raasdnil | anyone know how to make use of the making progress sip messages? I want to run "stoptones" when I receive making progress message from a sip peer |
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| 11:37.09 | scruz | how do i set up ss7 for asterisk? install zaptel? dahdi? |
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| 11:42.41 | bobsaccamano | hi..im using a CPE containing two rj-11 lines to talk to asterisk for a VoIP-SIP call...now i get a 401 unauthorized message when i enable both lines..i found out that asterisk has an option to disable authentication on INVITE messages...anyone knows how to do it? |
| 11:44.14 | bobsaccamano | apparently insecure=invite is supposed to do the trick..but it isnt ;( |
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| 11:44.23 | jamespass | Guys, need some assistance with the queue app. Basically it's not functioning right (i'm not a newbie BTW). Calls don't get presented to the agents until >30 seconds pass |
| 11:44.38 | jamespass | we have multiple queues with agents cross populating the queues |
| 11:44.51 | jamespass | I think it's a bug, but i could use a guru's eye on it!! |
| 11:47.30 | bobsaccamano | will the real gurus please stand up? |
| 11:47.31 | bobsaccamano | ;) |
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| 12:36.27 | synthetiq | in asterisk 1.6, how can i get jingle and res_jabber to compile? it doesnt compile by default, and is there more documentation on these modules? all the docus i can find seem to be for v 1.4 |
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| 12:42.02 | bobsaccamano | guys..i have two phones behing a NAT device and when i try to register the phones using SIP i get a 401 unauthorized message. What might be wrong here? |
| 12:42.03 | SparFux | Is there a chance to do SMS over a ulaw compressed line? |
| 12:44.40 | pwell | Spar, is SMS already optimized since you can SMS any # with an email basically? |
| 12:44.46 | pwell | I am new to this |
| 12:45.26 | SparFux | pwell: Ok, I have an analogue phone, a DECT phone, and it would be nice if I could send SMS to it from asterisk, as to be able to notify about stuff. |
| 12:46.28 | SparFux | For now I already do the MWI thingy with an asterisk mailbox. That rocks already :-) |
| 12:46.46 | *** join/#asterisk cianmaher (n=Cian@86.46.249.116) |
| 12:48.27 | cianmaher | Hi all I want to originate a call through manager without it needing 2 channels. For example if i originate now i need to supply an EXTEN to dial then when that answers it will dial the CHANNEL. Is there anyway of dialing the CHANNEL without needing to supply the EXENT? Or any way around this? |
| 12:52.37 | *** join/#asterisk prxtien (n=pro@teamaustralia.net.au) |
| 12:54.23 | scruz | cianmaher: the exten/context/priority simply points to a dialplan context/extension/priority where *processing should continue* after the dialled channel answers the call. |
| 12:54.35 | scruz | ack. he left |
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| 13:00.50 | C4colo | hmm |
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| 13:15.41 | double_cheesburg | I'm looking through some kind of old documentation on Asterisk cmd Curl : http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Curl |
| 13:15.43 | double_cheesburg | Seems to indicate that 1 variable is returned, "Sets Curl channel variable (CURL) upon completion of retrieval" |
| 13:15.45 | double_cheesburg | I'm wondering if anyone knows if the Curl channel variable can be broken down from an array within the dialplan... |
| 13:16.16 | *** part/#asterisk gulden (n=gulden@mentesvirtuais-mail.netvisao.pt) |
| 13:18.14 | [TK]D-Fender | double_cheesburg: "from" an array? it is just a variable, like any other. |
| 13:19.07 | double_cheesburg | Ok, so let's say the variable contains an array. I guess I'm asking if you can extract the array within your dialplan script. Does that make sense? |
| 13:20.10 | double_cheesburg | The Voip Info documentation doesn't seem to indicate that this is the case. |
| 13:20.42 | double_cheesburg | Essentially, you'd be using an array to store multiple variables that you receive from the one cURL call... |
| 13:21.03 | double_cheesburg | and then you'd just distribute those variables throughout the dialplan as needed |
| 13:21.20 | double_cheesburg | I'm just wondering if asterisk supports that |
| 13:21.25 | [TK]D-Fender | double_cheesburg: go read CHANNELVARIABLES.TXT(/ .TEX) |
| 13:21.45 | double_cheesburg | Thx. Where's it at? |
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| 13:22.16 | [TK]D-Fender | double_cheesburg: Int he DOC folder of your source tarball |
| 13:22.28 | double_cheesburg | Cool. Thx |
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| 13:24.46 | scruz | i never knew wanpipe had so many options. i've no idea what to build. all i want to do is be able to use a sangoma e1 card with asterisk and ss7 on an x64 server |
| 13:26.55 | [TK]D-Fender | scruz: Go read their wiki. |
| 13:27.46 | scruz | [TK]D-Fender: thanks. better hop over there |
| 13:29.36 | bobsaccamano | how to restart asterisk so that i can apply config changes? |
| 13:30.16 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 13:31.22 | [TK]D-Fender | bobsaccamano: What kind of changes? |
| 13:31.34 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 13:31.43 | bobsaccamano | [TK]D-Fender, sip.conf |
| 13:31.54 | [TK]D-Fender | bobsaccamano: "sip reload" |
| 13:31.56 | bobsaccamano | i did restart now |
| 13:31.57 | scruz | sip reload |
| 13:32.42 | scruz | bobsaccamano: do you use windows most of the time? |
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| 13:33.00 | bobsaccamano | yeah |
| 13:33.38 | scruz | i was also surprised that asterisk could reload parts of itself without needing to restart the whole service |
| 13:34.06 | scruz | ex: 'extensions reload', 'iax2 reload', 'sip reload' |
| 13:34.29 | bobsaccamano | scruz, any idea how to disable authentication for SIP REGISTER messages on asterisk? |
| 13:35.11 | bobsaccamano | [TK]D-Fender, scruz: thanks for the previous tip |
| 13:35.16 | [TK]D-Fender | bobsaccamano: AFAIK its not possible, nor is that sane |
| 13:35.48 | scruz | you can skip using passwords though |
| 13:35.54 | bobsaccamano | [TK]D-Fender, hmm..i need a way to enable two phones behind a NAT to register with asterisk |
| 13:36.12 | [TK]D-Fender | bobsaccamano: Not the way. |
| 13:36.14 | [TK]D-Fender | ~sipnat |
| 13:36.37 | 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 |
| 13:36.37 | scruz | SIP & NAT don't play nice together |
| 13:36.38 | scruz | kicks jbot |
| 13:36.38 | [TK]D-Fender | ^^^^scruthey do just FINE |
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| 13:36.55 | bobsaccamano | scruz, well its more like a single device - a CPE that has two RJ-11 ports with normal analog phones connected to each |
| 13:37.10 | bobsaccamano | the CPE has a public address |
| 13:37.15 | bobsaccamano | and its own sip stack |
| 13:37.24 | [TK]D-Fender | bobsaccamano: So it isn't behind NAT then? |
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| 13:37.47 | bobsaccamano | [TK]D-Fender, not in the true sense no |
| 13:38.19 | [TK]D-Fender | bobsaccamano: that'd be "false" then if boolean logic serves be correctly. It only does so 50% of the time.... |
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| 13:38.40 | *** mode/#asterisk [+o russellb] by ChanServ |
| 13:39.06 | brunner | does realtime *replace* extensions.conf, or do the tables supplement it? |
| 13:40.51 | [TK]D-Fender | brunner: Supplement. You need to "switch +>" in your dialplan to tell * to use Realtime on a context by context basis |
| 13:41.01 | brunner | thanks |
| 13:41.09 | [TK]D-Fender | bruthere is a new patch to change this. Not sure when its hits mainline.... 1.6.1 perhaps |
| 13:42.24 | bobsaccamano | [TK]D-Fender, okay..its like this: I have a SIP device that can connect to 2 analog phones. A call originated from either phone is tranlated to a sip uri with the respective user name (aka phone number) and the ip address of the sip device (and a different port no). |
| 13:42.50 | bobsaccamano | now i am unable to register both the phones with asterisk |
| 13:42.59 | bobsaccamano | i get a 401 unauthorized message |
| 13:43.10 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:43.21 | bobsaccamano | however when i connect only one phone to the device it works sweet |
| 13:44.03 | [TK]D-Fender | bobsaccamano: And i don't see the SIP debug or your configs. PAStebIN is your friend |
| 13:44.04 | [TK]D-Fender | ~pb |
| 13:44.06 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 13:44.08 | scruz | is there any way to stop the firmware download when building dahdi-linux from the makefile? it borks my setup and dahdi-tools doesn't recognize that dahdi-linx is installed |
| 13:44.47 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 13:46.07 | scruz | basically, when doing 'make install', the make fails on the firmware download (no internet access), and maybe something required isn't installed |
| 13:49.08 | bobsaccamano | [TK]D-Fender, how do i capture them in asterisk? |
| 13:49.14 | bobsaccamano | i have the wireshark tracefile |
| 13:49.17 | bobsaccamano | but its too huge |
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| 13:58.19 | scruz | asterisk has dahdi support built (yay!), but dahdi-tools won't build |
| 13:59.58 | Dovid | scruz: check your dns settings |
| 14:00.03 | Dovid | maybe thats the issue ? |
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| 14:09.38 | SparFux | I am afraid I won't be able to send SMS to my analogue phone connected to ATA because SMS() is bound to PSTN / ISDN. |
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| 14:12.24 | scruz | Dovid: the server *doesn't* have internet access |
| 14:13.35 | scruz | okay... dahdi-tools is now properly configured |
| 14:14.32 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 14:17.36 | jaytee | why should I get this on the CLI if I have call-limit=1 for the peer entry and limitonpeers=yes in sip.conf? |
| 14:17.42 | jaytee | <PROTECTED> |
| 14:18.31 | russellb | is the peer already in use? |
| 14:18.37 | jaytee | doesn't impact my dialplan, the call rolls to VM as intended but is this showing up because I have my verbose setting at 6? |
| 14:18.40 | jaytee | russellb, yep |
| 14:18.41 | russellb | because you did set the limit to 1 :-) |
| 14:19.05 | russellb | yeah, so, only 1 call allowed ... if it's in use, further calls will be rejected |
| 14:19.06 | jaytee | so asterisk considers it an Error? |
| 14:19.20 | russellb | well. it shouldn't be an ERROR |
| 14:19.31 | russellb | NOTICE, maybe. |
| 14:19.46 | jaytee | it's intentionally defined in sip.conf, it should report on the console as NOTICE. This just muddies the waters. |
| 14:19.52 | russellb | agreed |
| 14:19.59 | russellb | i'd call the log level a bug |
| 14:20.06 | russellb | you could submit a patch :-) |
| 14:20.07 | jaytee | russellb, I'm sure you'll get right on that :-) |
| 14:20.27 | russellb | or i could just do it, but that's no fun |
| 14:20.38 | jaytee | me, a patch? messing with c code? do you go around your area handing out dynamite to blind people? |
| 14:21.03 | russellb | heh, just search for "rejected due to usage limit" and change LOG_ERROR to LOG_NOTICE |
| 14:21.30 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 14:22.00 | jaytee | russellb, seriously, I'll dig around and look for the lines in the source code this weekend and look into creating my very first diff file for c code, then I'll submit it as a suggestion :-) |
| 14:22.20 | russellb | yay :-) |
| 14:22.37 | Khratos | Good morning |
| 14:22.51 | jaytee | which file would that be in? sip.c I'd think? |
| 14:22.56 | russellb | the diff is easy to create once you make a change. Just use an svn checkout to make the changes, and run "svn diff" .... svn diff > my_patch.txt |
| 14:23.00 | russellb | chan_sip.c, yup |
| 14:23.32 | jaytee | I'd have to setup svn at home, don't have the wherewithall here at them moment. |
| 14:23.44 | Khratos | Have anyone faced a problem that, when using an analog lines card, the TelCo apparently does not gives the dialtone inmediately and Asterisk found the channel as busy? |
| 14:23.46 | jaytee | but I should do it just so I learn how |
| 14:25.30 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:25.41 | russellb | yeah, and then you can just run trunk on your production boxes |
| 14:25.47 | russellb | updated every hour |
| 14:26.24 | russellb | beyond bleeding edge! |
| 14:26.50 | *** join/#asterisk jermey_g (n=me@static-213-115-44-229.sme.bredbandsbolaget.se) |
| 14:27.14 | jermey_g | what the f is this microsoft unified communication server |
| 14:27.26 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
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| 14:33.56 | FireSlash | What event in the manager is fired when someone picks up a call ringing in from a queue? |
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| 14:43.07 | scruz | yay! dahdi-tools built and now i know the type of card |
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| 14:43.13 | agx | someone has one-way audio between phones with the following message "app.c: No audio available on SIP/RANDOM" with v 1.4 ? |
| 14:43.37 | fbcit | when configuring * w/dahdi --with-dahdi should point to what? (dahdi is already built and installed) |
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| 14:45.25 | russellb | fbcit: usually nothing |
| 14:46.35 | scruz | this is really offtopic, but i want to know how i can get the amount of phys mem on my machine in linux on the cmdline so i can configure wanpipe properly. any help? |
| 14:48.13 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:48.18 | muiro | scruz: cat /proc/meminfo |
| 14:48.37 | scruz | muiro: thankee *windows user* |
| 14:49.04 | muiro | scruz: no worries |
| 14:51.17 | scruz | ok...determined the server has 2GB RAM, but is 64-bit |
| 14:52.00 | jaytee | jermey_g, Microsoft Unified Messaging is a feature in Exchange 2007 that allow voicemail and email integration. We use it here at my work instead of "Comedian Mail". |
| 14:52.26 | fbcit | russellb: I'm actually upgrading from zaptel to dahdi; do I need to rebuild * ? |
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| 15:00.55 | muiro | lol, comedian mail |
| 15:04.43 | *** join/#asterisk XtremXpert (n=XtremXpe@bas5-quebec14-1177724043.dsl.bell.ca) |
| 15:05.05 | XtremXpert | Anybody using AsteriskNow 1.5 Beta??? |
| 15:06.07 | XtremXpert | I have an issue with it and need to know how to report it |
| 15:06.10 | UQlev | XtremXpert: why don't use original one? |
| 15:06.45 | XtremXpert | I am, but I try 1.5 on an other server |
| 15:07.25 | XtremXpert | I tought Beta are to be try, nope :-) |
| 15:07.29 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:07.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:07.46 | UQlev | I would not do it |
| 15:07.59 | XtremXpert | Try a Beta?? |
| 15:08.13 | UQlev | best to configure asterisk manually w/o web-interface |
| 15:08.52 | UQlev | my colleague use it asterisk now with plenty of problems I don't want to study |
| 15:09.24 | jaytee | damn, I forgot about the change to daylight savings time this weekend. |
| 15:09.27 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 15:10.48 | XtremXpert | Well, I don't plan to setup and administrate enought asterisk to learn all the different config file, it's why I rely on prefab distro like asterisknow, trixbox ans elastix |
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| 15:11.22 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-c1048577e3a7b04f) |
| 15:11.30 | jaytee | I really wish they'd change the name of Comedian Mail to something more professional sounding. |
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| 15:11.58 | UQlev | XtremXpert: if you like to bother with CentOS i.o decent OS go on.. |
| 15:12.33 | XtremXpert | I am more a Debian guy |
| 15:13.08 | XtremXpert | but, can live with centos |
| 15:13.20 | UQlev | can't live with goats, camels and CentOS ;) |
| 15:14.02 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:14.03 | XtremXpert | lol, got 2 out of 3, Goats and Centos |
| 15:14.27 | XtremXpert | So you are a debian user |
| 15:14.38 | UQlev | nope, Gentoo and FreeBSD |
| 15:15.35 | XtremXpert | I was using gentoo about 3 or 4 years ago, but never try any bsd |
| 15:15.36 | UQlev | .. and Ubuntu I promote as 1st step to migrate from Windows |
| 15:15.44 | XtremXpert | nice choice |
| 15:15.59 | XtremXpert | I am behind ubuntu for new comer in linux world |
| 15:16.33 | XtremXpert | did you try voicebuntu |
| 15:16.33 | UQlev | FreeBSD is the best option for Internet server, Gentoo - for LAN |
| 15:16.47 | UQlev | I didn't try voiceubuntu |
| 15:17.26 | XtremXpert | May be I gonna give a try at bsd, when I'll have time. It may replace my ClarkConnect in time |
| 15:17.33 | UQlev | and Gentoo is my favorit option for desktop |
| 15:17.45 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 15:17.58 | scruz | yay! e1 card set up |
| 15:18.00 | muiro | can't you just change the sound file that the vm app is referencing? |
| 15:19.25 | XtremXpert | did you give a try at funtoo |
| 15:19.36 | UQlev | no |
| 15:20.23 | UQlev | XtremXpert: I prefere traditional distro rather than specialized |
| 15:20.52 | XtremXpert | in what way funtoo is a specialized distro |
| 15:21.24 | XtremXpert | I tought it was just a fork |
| 15:23.56 | UQlev | is reading funtoo |
| 15:25.12 | *** part/#asterisk khronos (n=khronos@c-76-110-120-247.hsd1.fl.comcast.net) |
| 15:25.15 | XtremXpert | is reading Asterisk on BSD |
| 15:26.49 | [TK]D-Fender | jaytee: I've commented about that for years |
| 15:27.49 | jaytee | [TK]D-Fender, about Comedian Mail? Well, I just think it might in some cases put some potential customers off from using Asterisk. It sounds like a toy or childish. |
| 15:28.12 | muiro | [TK]D-Fender: can you change the sound files that VoiceMail uses? |
| 15:28.53 | [TK]D-Fender | jaytee: you're right, and "DAHDI" and that hardware interface layer named after a film Aussie.... BRILLIANT marketing. |
| 15:29.02 | [TK]D-Fender | muiro: Of course. |
| 15:33.17 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 15:34.13 | jaytee | [TK]D-Fender, I'm having a brain fart when it comes to thinking of the "film Aussie" |
| 15:34.24 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.33.119) |
| 15:34.33 | BlargMaN00 | My company won't switch to * until it can manage it's voicemail through it's e-mail box like "Call Mangler" does |
| 15:34.35 | kannan | hello, I have a occasional link fluctutation on an E1 9the alarm goes RED or YELLOW for a few). What is the steps I can suggest for the telco. Also, How can I monitor the alarm status, eg get an alert by email as soon as the alram is not OK? |
| 15:34.37 | jaytee | but it's been a long, long, week of long, long days. |
| 15:34.53 | kannan | Using Digium single span TE120 card |
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| 15:38.27 | *** join/#asterisk gulden (n=gulden@mentesvirtuais-mail.netvisao.pt) |
| 15:38.32 | jaytee | kannan, only thing I can think of would be to run a script once every minute or two that greps the /var/log/asterisk/messages log for the alarm string and then sends an email/page/alert/something. |
| 15:38.52 | *** part/#asterisk gulden (n=gulden@mentesvirtuais-mail.netvisao.pt) |
| 15:39.41 | kannan | jaytee , thanks i was thinking the same thing, only wondered if there is a ready made thingy available |
| 15:40.25 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 15:40.26 | kannan | any idea what YELLOW alarm indicates |
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| 15:40.52 | jaytee | kannan, not that I'm aware of. Nagios can monitor servers for certain conditions but I don't think the folks at Digium have thought it worthwhile to put "hooks" into their system or write a Nagios plugin for their cards. |
| 15:41.26 | Chesther | Nagios plugins for digium cards/* would be pretty nice. |
| 15:41.31 | jaytee | kannan, yellow alarm depends on what kind of circuit. if you add the yel to your pri defines it will yellow when no channels are in use. |
| 15:41.41 | kannan | oh ok, is this a rare issue then, i am struggling with it for the last week |
| 15:42.19 | kannan | jaytee, actually , an issue is that when the alarm turns yellow, current call channels gets disconn |
| 15:42.42 | kannan | it clears of its own in a couple minutes |
| 15:42.44 | jaytee | who knows? you've just told us a little info. Is this circuit been in service for a long period and just now malfunctioning? |
| 15:42.57 | kannan | just now malfunctioning after 2 months |
| 15:43.28 | jaytee | loose cable? if you loopback and test and it clears and it stays stable then it's the other end. |
| 15:43.38 | jaytee | first check cables, then test in loopback |
| 15:43.59 | kannan | jaytee , ok thanks |
| 15:49.46 | brunner | can anyone recommend a voip provider that offers unlimited outbound trunking? |
| 15:50.45 | doolph | anyone can help me with AsteriskNow?? the logging is not working |
| 15:56.02 | [TK]D-Fender | jaytee: DUNDI <- |
| 15:56.24 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:56.46 | jameswf | ~asterisknow |
| 15:56.46 | jbot | i guess asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 15:59.33 | Kobaz | brunner: search for virtual pri |
| 15:59.59 | brunner | Kobaz: thanks |
| 16:00.36 | Kobaz | brunner: most places will charge about like 8 bucks a month per channel |
| 16:00.46 | Kobaz | brunner: actually, but that's free incomming |
| 16:01.08 | brunner | well that sounds like a pretty good deal |
| 16:01.23 | Kobaz | outgoing will usually be in the range of 1.5 to 2 cents a minute |
| 16:01.41 | brunner | ah |
| 16:01.51 | brunner | are there any that are cheaper than 1.5? |
| 16:02.16 | Kobaz | if you look for them... but mostly you need a high commitment to get below that |
| 16:02.40 | brunner | gotcha |
| 16:02.50 | Kobaz | but if you have a high commitment to a vitual pri, might as well get a real one |
| 16:03.03 | brunner | yeah |
| 16:03.32 | Kobaz | if you're gonna push like 10k minutes a month or more, you'll be able to get around 1 cent or less a minute from a regular telco |
| 16:03.41 | brunner | oh wow |
| 16:04.19 | phix | hi |
| 16:04.28 | phix | <3 |
| 16:04.37 | phix | lets asterisk! |
| 16:04.42 | Kobaz | licks phix |
| 16:04.47 | phix | Kobaz: <3 |
| 16:04.51 | phix | hai |
| 16:04.54 | phix | sup? |
| 16:05.08 | Kobaz | you loves teh axeterisk |
| 16:05.16 | phix | <3 |
| 16:05.28 | phix | 1,1,loveASterisk() |
| 16:05.35 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:05.42 | phix | 1,n,asteriskIsAwesome() |
| 16:05.58 | phix | 1,n,Hangup() |
| 16:06.12 | brunner | Kobaz: are there any voip providers that will do that? I'd really like to do this on a dedicated server instead of installing a PRI and buying a PCI card. |
| 16:06.22 | Kobaz | brunner: do what? |
| 16:06.40 | brunner | I mean, offer rates closer to 1 cent per minute in exchange for a commitment |
| 16:06.54 | Kobaz | oh, you'll have to call |
| 16:06.58 | Kobaz | depends on the provider |
| 16:07.19 | phix | brunner: I has 8 cents untimed to 20 countries including my own cept mobile phones |
| 16:07.48 | brunner | 8 cents untimed? |
| 16:07.56 | phix | yes |
| 16:07.57 | Kobaz | i dont deal with any voip companies in bulk |
| 16:08.04 | Kobaz | phix: 8 cents per call you mean? |
| 16:08.09 | *** join/#asterisk LuisTorres (n=chatzill@a213-22-94-93.cpe.netcabo.pt) |
| 16:08.12 | phix | Kobaz: yes |
| 16:08.14 | brunner | ooh |
| 16:08.16 | brunner | I see |
| 16:08.18 | Kobaz | interesting |
| 16:08.25 | brunner | do you get charged if the call doesn't supervise? |
| 16:08.32 | phix | Kobaz: per call, howeevr that call may be |
| 16:08.39 | eppigy | hello |
| 16:08.41 | eppigy | i am dave |
| 16:08.43 | phix | brunner: sup |
| 16:08.47 | phix | eppigy: hi dave |
| 16:08.51 | brunner | phix: suup |
| 16:08.58 | phix | <3 |
| 16:10.47 | brunner | what's the cheapest SIP trunking service you guys know of that doesn't require a commitment? |
| 16:10.55 | brunner | link2voip is 1.5 cents |
| 16:10.57 | Kobaz | voicepulse is pretty cheap |
| 16:10.59 | phix | brunner: your mum |
| 16:11.04 | Kobaz | ~itsp-usa |
| 16:11.30 | Kobaz | ~itsp |
| 16:11.31 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 16:11.31 | Kobaz | slaps jbot |
| 16:11.31 | Kobaz | ~itsp-list |
| 16:12.40 | *** join/#asterisk enocix (n=jsloan@nat/digium/x-1281915582510e99) |
| 16:13.04 | *** part/#asterisk enocix (n=jsloan@nat/digium/x-1281915582510e99) |
| 16:21.50 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 16:22.24 | Tapout | ~itsplist-ca |
| 16:22.44 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca |
| 16:25.30 | phix | <3 |
| 16:25.50 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:26.48 | brunner | check it out: http://www.flowroute.com/ |
| 16:26.57 | brunner | $0.0098 |
| 16:27.37 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:27.50 | Tapout | brunner, i'm looking for a flat rate.. les.net seems to have one |
| 16:27.52 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:28.36 | brunner | Tapout: sorry, I was referring to something from earlier. I was looking for a flat rate, too, before I spoke to someone about the Fair Use clauses |
| 16:28.41 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:28.46 | phix | you know why they call it tap water? |
| 16:28.52 | phix | because you just tap it on your nuts |
| 16:29.34 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 16:31.17 | pwell | good point |
| 16:31.28 | pwell | its so bad for you |
| 16:31.43 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 16:31.45 | Kobaz | soooo |
| 16:31.52 | Kobaz | i know there's Ringing() to send ringback |
| 16:32.09 | Kobaz | how would i tell the endpoint to just have silence? |
| 16:33.32 | [TK]D-Fender | Kobaz: When? |
| 16:33.33 | *** join/#asterisk trippsss (n=firc_cli@m050e36d0.tmodns.net) |
| 16:34.56 | Kobaz | when Dial()ing |
| 16:34.56 | Kobaz | well actually |
| 16:34.56 | trippsss | calls into my queue using AgentCallbackLogin() are ending up on my cellphone voicemail. Isn't there a setting to require answering agents to press 1 to prevent this? |
| 16:35.17 | mosty | kobaz: well you could use m(silence) - if you create a moh class called silence |
| 16:35.23 | Kobaz | k |
| 16:35.28 | mosty | if there's no simpler way |
| 16:35.43 | [TK]D-Fender | yup, thats the way to do it |
| 16:36.01 | *** part/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 16:36.10 | Kobaz | nifty |
| 16:37.16 | brunner | How much horsepower do I need to handle 20 concurrent calls doing Playback() and no transcoding? |
| 16:37.28 | [TK]D-Fender | brunner: a Timex analog watch |
| 16:37.37 | brunner | okay, so a P4 should handle that fine, right? |
| 16:37.44 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 16:37.49 | [TK]D-Fender | brunner: More than |
| 16:37.55 | brunner | how about 60 calls? |
| 16:38.57 | brunner | are there any benchmarks I could look at? |
| 16:38.57 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
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| 16:39.20 | guax | brunner, you do record the calls? |
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| 16:39.39 | *** part/#asterisk dlewis (i=457e665b@about/security/staff/dlewis) |
| 16:39.43 | brunner | no. |
| 16:40.54 | guax | a core 2 duo 2.5ghz should handle about 120 calls (simultaneous) (without quality decrease) maybe more |
| 16:41.14 | guax | this on our experience in call centers |
| 16:41.25 | pdmmm | How much horsepower do I need to handle 30 concurrent calls with people driving on cellphones and their not at red lights? |
| 16:42.10 | brunner | woah |
| 16:42.12 | *** join/#asterisk kc2tnk (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 16:42.13 | brunner | thanks. |
| 16:42.19 | brunner | guax: what about while recording? |
| 16:43.09 | guax | brunner, we do limit to 60 calls, but the machine can handle more. but for safety and good of all nation... |
| 16:43.30 | brunner | guax: do you record? |
| 16:43.31 | *** join/#asterisk t0rrieri (n=Torrieri@nelug/crew/torrieri) |
| 16:45.43 | trippsss | is there a setting to require queue agents to acknowledge a call, so queue calls going to cell phones don't end up in their voice mail? |
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| 16:50.23 | jasonwoot | will agentCallBackLogin route calls to extensions which are paused? |
| 16:50.55 | Qwell | extensions don't get paused. queuemembers do |
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| 16:52.26 | trippsss | Qwell, do you if anything like I'm asking exists? |
| 16:52.27 | doolph | Qwell |
| 16:52.36 | doolph | are you back? |
| 16:55.04 | jasonwoot | Qwell: I have two queues, one with extensions defined, one with agents defined... |
| 16:55.15 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 16:55.34 | jasonwoot | Qwell: if I have an Agent login using agentcallbacklogin, but they log in from an extension that is a member of the other queue, |
| 16:55.55 | jasonwoot | Qwell: will the agent receive a queue call if the extension is paused? |
| 16:57.45 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 17:01.10 | Qwell | jasonwoot: They are all just queue members. |
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| 17:08.42 | trippsss | ugh got disconnected. does ackcall=yes require agents to ack the call by pressing #? |
| 17:13.17 | *** join/#asterisk djMax (n=chatzill@66.92.91.133) |
| 17:13.17 | djMax | should "manager debug on" show me all the commands submitted via the manager interface? |
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| 17:25.32 | mazpe | isnt this a valid statement to declare a variable in 1.6.0.5: exten => s,n, SetVar(AUDIO_DIR=/var/lib/asterisk/sounds/mypbx) |
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| 17:28.11 | russellb | s/SetVar/Set/ |
| 17:28.26 | russellb | IIRC, SetVar was not valid in 1.4, either |
| 17:28.47 | mazpe | vale |
| 17:28.55 | mazpe | testing.. |
| 17:30.00 | mazpe | yup |
| 17:30.08 | mazpe | thanks russellb |
| 17:30.54 | russellb | np. |
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| 17:41.59 | djMax | is there a way to log all manager commands other than wireshark? |
| 17:45.26 | iMacGyver | how many interrupt misses are 'normal' for a Wildcard TDM800P |
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| 17:55.34 | brunner | I'm sure this is stupid question, but there's no way to blind transfer someone to a phone number on the PSTN under any circumstances, right? |
| 17:55.55 | brunner | my friend says he can do it with his Vonage phone, so I'm wondering how that works and whether it's possible with asterisk |
| 17:59.31 | Kobaz | sure you can |
| 18:00.19 | Kobaz | pick up your phone, hit transfer, enter in the 10 digit phone number |
| 18:00.46 | *** join/#asterisk alecdavis (n=sivad@202-78-149-14.cable.telstraclear.net) |
| 18:00.50 | brunner | yes, but how can I do that with Asterisk? |
| 18:00.56 | Kobaz | Dial() |
| 18:02.28 | brunner | well I meant without using my trunks, but I guess Vonage is listening for the DTMF tones and using up their trunks while the call is occurring |
| 18:05.02 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 18:08.06 | mosty | brunner: your question does not quite make sense |
| 18:09.03 | mosty | brunner, pretty much every sip phone supports call transfer, it does not matter that kind of number the destination is |
| 18:09.19 | mosty | s/that kind/what kind/ |
| 18:09.39 | Khratos | [TK]D-Fender, is there a parameter that I can adjust in zaptel in order to make Asterisk to wait a little before try to send a call to a channel ? |
| 18:11.44 | mosty | Khratos, i believe you can add w to the number to add a half second pause |
| 18:12.27 | Khratos | oohh, thanks mosty, I will check that now... |
| 18:12.54 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 18:13.02 | mosty | i have not had to use it personally, see the dial docs for more info |
| 18:13.35 | Khratos | Actually, the 'w' parameter is used for other purposes |
| 18:14.05 | mort_gib | Khratos: You can use Wait(XX) in extensions.conf |
| 18:14.12 | *** join/#asterisk L2SHO (n=adam@fw01ext.voicepulse.com) |
| 18:14.20 | djMax | so I see a manager logon on the asterisk CLI, which implies to me that the fputs() of the user information must be working. They are immediately followed by more fputs to originate a call, which isn't working. |
| 18:14.31 | djMax | but "manager debug on" doesn't show any of this |
| 18:15.12 | L2SHO | Is there a way to make asterisk only offer video to the B-Leg of a call if the invite from the A-Leg offered video? |
| 18:15.19 | Khratos | mort_gib, the problem with my marvelous(!?) TelCo is that apparently they are not giving the dial tone inmediately, and when asterisk try to send out a call it finds the channel busy |
| 18:15.50 | mosty | Khratos, it's not an option- i think it's part of the channel string, like DAHDI/1/w123 |
| 18:16.04 | mort_gib | Khratos: immediate |
| 18:16.42 | djMax | yeesh, manager doesn't debug to console, only to full log. |
| 18:17.00 | Khratos | Thanks mosty and mort_gib , I will see (I'm using zaptel instead of DAHDI) |
| 18:17.22 | mosty | Wait() won't do what you want |
| 18:17.35 | mort_gib | Khratos: immediate=no should wait until you have finished dialing the number |
| 18:17.54 | mort_gib | mosty: I didn't know in what direction he wanted it :-) |
| 18:18.19 | mosty | my guess is that zap/dahdi needs to pick up the line, wait a second or so and then dial the numbers |
| 18:18.34 | Khratos | Exactly mosty |
| 18:19.00 | Khratos | TelCo's here are a little... prehistoric, I would say, and some strange behaviors hapens |
| 18:19.04 | mosty | try ZAP/1/w1234 |
| 18:19.10 | mosty | or ZAP/1/ww1234 |
| 18:19.39 | mort_gib | mosty: I understood that, but it's really strange that Zaptel starts dialing before it gets a dialtone! |
| 18:19.55 | mort_gib | Is there a divert on the line?? |
| 18:20.20 | Khratos | Wait a minute, let me see what 'divert' is (non native English speaker) |
| 18:20.58 | Khratos | Ah!, no, theres no divert |
| 18:20.59 | mort_gib | locally I can ask my Telco to divert, so when the line is buse the call is passed to, say my mobile |
| 18:21.06 | *** join/#asterisk KU0N (n=kuon@alragore.goyman.com) |
| 18:21.21 | mort_gib | But that means that my dialtone is different for the first 10 sec or so |
| 18:21.22 | SparFux | Can't I do FSK for SMS service over a G711 line? |
| 18:21.59 | Khratos | Here I could to, but Asterisk is detecting it busy because its not receiving de dialtone inmediatelly... I will test with the 'w' |
| 18:23.10 | *** part/#asterisk KU0N (n=kuon@alragore.goyman.com) |
| 18:23.32 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:24.27 | [TK]D-Fender | Khratos: Dial(Zap/g1/wwww1234567890) |
| 18:25.56 | djMax | is there a "better" way to use the manager interface from PHP than fputs/fgets and sockets? |
| 18:26.20 | Khratos | Thanks. Just to know, where was that documented? (Im not an expert, but I think that this kind of things are basic) |
| 18:26.34 | Khratos | I dont know how did I missed that :/ |
| 18:26.45 | [TK]D-Fender | Khratos: Not sure where its best documented... ti is somewhat rare |
| 18:27.06 | [TK]D-Fender | djMax: AMI = sockets. Do you know of another way to do socket work in PHP? |
| 18:27.32 | Khratos | djMax, not at all |
| 18:27.45 | Khratos | there isnt |
| 18:28.03 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 18:28.31 | djMax | ok, it's just seemingly hard to debug this way, so not sure if it's my fault or what |
| 18:29.11 | Khratos | What do you want to debug? |
| 18:29.20 | djMax | the logs say they are getting an Originate command, but no details about it and it doesnt' work |
| 18:29.45 | djMax | I'm on * 1.6, so not sure if the source for this code is out of date in some way, though looking at the docs it doesn't seem so |
| 18:29.45 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 18:30.28 | djMax | The code issues login and originate, then does a fgets and quits |
| 18:30.53 | Khratos | Well, What I do is that things that I write to the socket (I use general purpose socket, not streams oriented socket) I also write it down to a file |
| 18:30.54 | djMax | I just added Async: 1, though that didn't seem to help |
| 18:30.54 | Khratos | And the response is read by a php function, saved in an array |
| 18:30.57 | djMax | (this is all inside sugarcrm) |
| 18:32.20 | Khratos | You could store the complete response inside an array (something like while( $out = @socket_read($Socket, 1024, PHP_NORMAL_READ) ) { do asigning stuff here } ) |
| 18:32.43 | [TK]D-Fender | Khratos: I would recommend using one of the pre-made libraries for this |
| 18:32.45 | Khratos | Then going through the array and getting your response with regexp |
| 18:32.59 | [TK]D-Fender | Khratos: like PHP-AGI ' s AMI stuff |
| 18:33.20 | djMax | yeah, just logged the responses. My first suspicious sighting is that the response I'm reading for login looks a lot like a welcome header. |
| 18:33.44 | *** join/#asterisk airjump (n=airjump@p508BF022.dip.t-dialin.net) |
| 18:34.47 | djMax | this sugarcrm auto dialer plugin is just not especially well written. It ignores the responses from the manager. |
| 18:34.49 | Khratos | djMax, when using php and sockets, you must have a good understanding on how they behave in different conditions... for example, socket_read will hang or do strange reads if you dont use PHP_NORMAL_READ flag parsing AMI responses |
| 18:35.18 | Khratos | reading** AMI responses |
| 18:35.26 | djMax | this is all fgets/fputs, so I assume PHP takes care of common options? |
| 18:36.00 | djMax | Is there a "good" php outbound dial script from some system I might look at? |
| 18:36.06 | Khratos | fgets and fputs is for stream oriented sockets, and are easier to mange that general type sockets |
| 18:36.18 | djMax | I assume the manager is stream oriented |
| 18:36.47 | djMax | I'm aware that the multiple uses of "assumption" should clearly communicate my problem. :) |
| 18:36.51 | Khratos | What you could do, is to write the command that you are sending to the AMI also in a /var/something.txt |
| 18:37.04 | Khratos | so you can copy that to the AMI directly through telnet |
| 18:37.10 | Khratos | telnet yourserver 5038 |
| 18:38.01 | djMax | I think what's happening is the script is disconnecting before reading a response for the Originate command, which must terminate the command or something |
| 18:38.11 | Khratos | oooooooooooohhh |
| 18:38.14 | Khratos | I had that problem |
| 18:38.32 | Khratos | Something changed in Asterisk ast_write function or something like that, and will give you a broken Pipe |
| 18:38.35 | Khratos | error |
| 18:38.37 | *** join/#asterisk dan__t (n=dant@ns1.hitb.net) |
| 18:38.44 | dan__t | Hello, |
| 18:38.58 | djMax | yeah, that's exactly what's happening |
| 18:39.13 | dan__t | So, I got that Programming Asterisk AGI book yesterday. |
| 18:39.20 | djMax | the strange bit is that I would assume I get one response line for each \r\n\r\n |
| 18:39.21 | dan__t | I like it a lot. Just wanted to throw my $0.02 out there. |
| 18:39.27 | Khratos | Your script is sending the command, and is not waiting for asterisk to write the response to the socket... as it has closed the connectiong (before, asterisk did not test if it was opened yet) asterisk warns you about broken pipes |
| 18:39.40 | dan__t | Seems like the author ran out of steam towards the end and used a lot of fluff, but it was all good for demonstration purposes. |
| 18:40.27 | *** part/#asterisk airjump (n=airjump@p508BF022.dip.t-dialin.net) |
| 18:41.15 | Khratos | This is not really Asterisk fault, the problem is that, as PHP execution ends before asterisk has completed the 'dialog' with it, then there come problems |
| 18:42.04 | Khratos | I did a little exec("wait 01.") just before closing the socket (in php) and it worked for me (its dirty, but worked) |
| 18:42.18 | Khratos | exec("wait 0.1"): |
| 18:42.46 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 18:42.55 | Khratos | sorry I meant exec("sleep 0.1") |
| 18:43.06 | djMax | ooh, I see, it's not one line per request, it's \r\n\r\n terminated. |
| 18:44.00 | Khratos | \r\r\r\r does not really ends the connection, It will end if you close it (your php script) or if you send the logoff command |
| 18:44.12 | dan__t | I'm fascinated by FastAGI. |
| 18:44.26 | dan__t | FastAGI + lighty = hardcore |
| 18:47.14 | Khratos | dan__t, what language do you use for AGI ? |
| 18:47.27 | dan__t | PHP. |
| 18:47.44 | dan__t | It was CLI, but now I'm going to see about moving exclusively to FastAGI. |
| 18:48.51 | Khratos | I see, there's a PHP compiler project (PHc) that translates your code to C, so you can compile it (speed of C is... awesome) |
| 18:49.20 | Khratos | But I think it does not support classes yet |
| 18:49.28 | Khratos | (the PHc, I mean) |
| 18:49.46 | dan__t | Eh I could do the same with Zend, ionCube, etc etc |
| 18:50.14 | Khratos | Without encryptiong |
| 18:50.25 | Khratos | no bytecode things, and you have the resulting C |
| 18:51.03 | dan__t | Oh, directly in to C source? |
| 18:51.05 | dan__t | That could be kind of rad. |
| 18:51.41 | Kobaz | a polycom sidecar adds 10 lines? |
| 18:54.32 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 18:55.25 | t_corr | Can an AGI script return a numeric code? So I can do a GotoIf(AGI(blah.agi)?blee:bloo) |
| 18:57.15 | Kobaz | no |
| 18:57.25 | t_corr | k |
| 18:57.26 | Kobaz | use goto within your agi application |
| 18:57.34 | putnopvut | Shaun222: Are you "Shaun R." on the asterisk-users mailing list? |
| 18:59.52 | t_corr | PHP? Yuck. |
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| 19:02.11 | djMax | progress... Getting permission denied now. |
| 19:03.41 | *** join/#asterisk bob_slacker (n=vncsnvs_@189.27.17.25.dynamic.adsl.gvt.net.br) |
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| 19:05.02 | djMax | question is how do I debug permission denied... |
| 19:06.54 | Khratos | mmm... who gave you the 'permission denied' ? Remember that you have to log in into the AMI before sending any command to ti |
| 19:06.56 | Khratos | it*** |
| 19:07.19 | djMax | yeah, login came back ok |
| 19:07.31 | djMax | there's some mail list discussion that originate is now a new privilege? |
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| 19:08.24 | *** part/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 19:08.26 | Khratos | I don't really know, I have not used 1.6 yet... I'm planning to do it soon in order to help in the feedback process |
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| 19:13.42 | *** join/#asterisk bijit (n=chatzill@201.192.89.182) |
| 19:14.32 | bijit | If I want to listen to a call coming from my telco can I use ExtenSpy()? |
| 19:15.25 | [TK]D-Fender | bijit: Depending on what stage of processing your call is in. iw ould sue ChanSpy most of the time |
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| 19:26.57 | djMax | all working now, just needed originate privilege. |
| 19:27.09 | djMax | Is there a way to get Originate NOT to connect the call if it goes to voicemail? |
| 19:27.16 | djMax | (on the * side, not the outbound) |
| 19:27.52 | *** join/#asterisk stevetotaro (n=Steve@c-71-206-29-240.hsd1.md.comcast.net) |
| 19:29.15 | Khratos | erm... I think that you could manage that from DialPlan |
| 19:30.26 | Khratos | [TK]D-Fender, about g1/ww${EXTEN} question, after having done a little grep -irE '(.*g[0-9]+/w.*)' . on asterisk, zaptel, libpri, etc. source, only two lines appear |
| 19:30.36 | Khratos | ./asterisk/asterisk-1.4.18.1/configs/vpb.conf.sample:; exten => _9XXX,1,Dial(vpb/g1/ww${EXTEN:${TRUNKMSD}}) |
| 19:30.40 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 19:30.47 | Khratos | ./asterisk-1.4.23.1/configs/vpb.conf.sample:; exten => _9XXX,1,Dial(vpb/g1/ww${EXTEN:${TRUNKMSD}}) |
| 19:31.59 | ricko73 | anyone familiar with the parking changes in 1.4.23.1? |
| 19:32.01 | [TK]D-Fender | djMax: You are the one sending it to voicemail in the first place. Stop |
| 19:32.30 | djMax | I'm just using a regular extension, but I see your point (I think) - make a separate extension that doesn't go to VM but rings the same device? |
| 19:32.38 | djMax | didn't know if there was some standard dial option for that |
| 19:32.47 | ricko73 | I'm trying to find out why when a call times out, it's no longer going back to the device that parked the call |
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| 19:34.10 | *** mode/#asterisk [+o Mog] by ChanServ |
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| 19:37.00 | harry_v | anyone know how i can take a object and make it lumpy like a chunk of carbon or coal? |
| 19:40.51 | Qwell | harry_v: put it under pressure and wait a million years? |
| 19:49.08 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 19:49.08 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 19:49.19 | harry_v | Qwell, I asked in the wrong channel by accident. Anyway, there are now man made ways to make diamonds with extreem pressure and heat using hydralics and steel rams. It was a interesting documentry on Discovery. |
| 19:51.16 | [TK]D-Fender | harry_v: "now"? More like "how many decades...." |
| 19:51.26 | jplank | harry_v: are you talking about those diamonds that are almost impossible to tell they aren't real? |
| 19:51.37 | jplank | that even a trained jeweler cant tell by eye |
| 19:51.43 | [TK]D-Fender | harry_v: Diamons should have very little intrinsic value and it is almost solely propagated by DeBeers |
| 19:53.16 | coppice | they are real. they just aren't the work of nature |
| 19:53.33 | NewCastleScott | diamonds are not the hardest substance on earth anymore |
| 19:53.46 | jplank | dont they say theres enough diamonds in the world that every person could have a handful? |
| 19:54.20 | coppice | a politician's heart is harder than diamond |
| 19:54.55 | NewCastleScott | http://www.physorg.com/news153658987.html |
| 19:55.52 | *** join/#asterisk froud (n=sean@dsl-243-246-191.telkomadsl.co.za) |
| 19:56.41 | Kobaz | ~book |
| 19:56.41 | jbot | extra, extra, read all about 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 |
| 19:57.41 | froud | Starnge question perhaps, but does anyone know of a addon that will enable DTMF to CGI |
| 19:58.35 | *** join/#asterisk javawebdev (n=me@pool-70-21-211-91.nwrk.east.verizon.net) |
| 19:59.54 | [TK]D-Fender | froud: ... HUH?! |
| 20:00.52 | froud | As I said weired request perhaps |
| 20:00.56 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:01.54 | froud | basic use case is user calls in and enters codes in DTMF which are captures and sent to a server where a bunch of CGI scripts exist that talk SOAP:Lite to a Web Service |
| 20:02.30 | [TK]D-Fender | froud: What yuo have 8 do with DTMF input in your dialplan is up to you. |
| 20:02.34 | [TK]D-Fender | * |
| 20:03.16 | [TK]D-Fender | froud: Have your dialplan call whatever scripts you want. |
| 20:03.57 | jplank | couldn't you use read to capture the DTMF string then just push it out to a script? |
| 20:04.15 | froud | OK so it says dial 1 to add credit, next prompt is enter your meter #, next enter your amount, enter your 4 Digit pin |
| 20:04.34 | froud | XXXXXXXXXX*XXX*XXXX |
| 20:05.01 | [TK]D-Fender | froud: All fine |
| 20:05.36 | froud | jplank yes could do that thx, just wondering if there is any plugin that already provides most of this |
| 20:05.43 | froud | before I start to write the wheel |
| 20:06.02 | [TK]D-Fender | froud: No such thing as a "plug in". What you do is CUSTOM. |
| 20:06.28 | [TK]D-Fender | froud: What kind of plugin do you think could exist that would know what to do with your CGI? |
| 20:07.05 | Micc | anyone have problems with g722 to ulaw transcoding? |
| 20:07.39 | Qwell | Micc: "problems"? |
| 20:07.53 | froud | Basically anything that read the DTMF to a TEXT file will do it. Once in Text I can process the text file and pass it to the CGI |
| 20:08.04 | [TK]D-Fender | froud: All up to you.... |
| 20:08.12 | froud | ok thx |
| 20:08.56 | Micc | Qwell, yeah I get robot audio, or stuttered audio. I'll try some more tests but polycom on g722 to an aastra 480i on ulaw, it stutters. |
| 20:09.05 | Qwell | Micc: What version of Asterisk? |
| 20:09.13 | Micc | Qwell, 1.6.0.6 |
| 20:09.37 | Qwell | should work fine.. |
| 20:09.39 | Qwell | russellb: ^^? |
| 20:09.51 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:09.59 | [TK]D-Fender | Micc: What do you get on a Polycom G722 to Polycom G.711? |
| 20:10.35 | javawebdev | Hi everyone. I was wondering, is it possible to have asterisk send extra info to be displayed on the phone's LCD? For example, Show callers' Caller ID and also show the number of the DID called as well as an extra line of text? |
| 20:11.19 | Micc | TKD-Fender, I'll try that when I get another polycom in. |
| 20:11.49 | *** join/#asterisk tid-wave (n=ovidiu@79.119.154.153) |
| 20:12.14 | Micc | I got a Lookit wp04 wifi phone that does g722. Its got some echo problems for me and its not very english friendly, but its kinda cool. |
| 20:12.27 | tid-wave | hello. I've installed asterisk on a wireless router and it only works with -f option (no forking). any ideas why? thanks |
| 20:13.19 | Qwell | tid-wave: because your router can't fork |
| 20:13.37 | javawebdev | Instead of getting a phone with multiple sidecars for a receptionist, it would be nice if I could send text to the phone to let the receptionist know what DID was dialed and how to greet the caller. Is that possible? Is there a specific feature in a phone I need to look for? |
| 20:14.38 | SparFux | What would you guys suggest for a prefix if I don't want to prefix every outgoing number with "0", but instead want a prefix for numbers going to my asterisk box. "0*" perhaps? |
| 20:14.39 | tid-wave | Qwell: but I see three asterisk processes after starting asterisk |
| 20:14.48 | Qwell | tid-wave: Those are threads. |
| 20:15.39 | tid-wave | Qwell: so I guess I should run asterisk with -f ? |
| 20:15.42 | [TK]D-Fender | SparFux: Whatever you feel like. |
| 20:15.57 | SparFux | Fender: I am afraid of prefix clashes, you know. |
| 20:16.20 | Qwell | tid-wave: if it doesn't start otherwise...yeah, that's probably a good idea. |
| 20:16.30 | tid-wave | ok, thanks for the help :) |
| 20:16.56 | [TK]D-Fender | SparFux: Whats to fear? this is YOUR system. You can't tell already if there is a pattern that will clash? |
| 20:17.00 | NewCastleScott | hey hey all, does anyone know what prerequisites is needed by chan_iax2.so? I cant load it, I get this error "loader.c:326 __load_resource: /usr/lib/asterisk/modules/chan_iax2.so: undefined symbol: ast_parking_ext" |
| 20:17.28 | Qwell | NewCastleScott: res_features. this is documented |
| 20:17.43 | NewCastleScott | I thank you Qwell |
| 20:17.51 | SparFux | Fender: Hm... yes, I can't. I think there are some things predefined. For example I wanted to use *, but there is my ATA which uses **** for a menu and *<numbers> for some codes. |
| 20:19.37 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 20:19.37 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.6 (2009/02/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev |
| 20:20.28 | javawebdev | bmoraca: Thanks. I thought of that but that is not as elegent. I know there is a SendText() app. But not sure where that goes. I've only tried testing it with X-Link softphone so far and it didn't show anything |
| 20:20.30 | SparFux | Fender: the "0" prefix for outgoing numbers is stupid, because it can wreck my phonebook, in case some day I am on a box without this "0". So that's why I am thinking about using some codes for accessing the stuff on my asterisk box instead. |
| 20:21.08 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 20:21.16 | [TK]D-Fender | SparFux: I haven't done prefixes in years and people can re-dial the CID they got from missed calls with impunity |
| 20:21.40 | SparFux | besides, is there an asterisk function which can make me use extension numbers specified as their T9 counterpart in letters. Like 278 for AST and the like. |
| 20:21.51 | *** join/#asterisk MmixX (n=mmixx@61.14.191.142) |
| 20:22.02 | brunner | Using expensive codecs won't take any more CPU power than ulaw as long as the sound clips aren't being transcoded, right? |
| 20:22.06 | bmoraca | javawebdev: not elegant? how so? all you need is 3 letters and a dash and that's more than enough to work it. for instance, if a call is for Service, prepend SVC-. How is that any different than sending text in a method your phone may or may not understand? |
| 20:22.19 | SparFux | Fender: Ah, that's what you mean. Yes, the redial stuff is what I want too. I mean, when using the "0" I cannot simply redial the CID sent in. |
| 20:23.09 | SparFux | Fender: but I have to distinguish phone numbers from all the stuff on my asterisk box. So, I am thinking of using 0* as a prefix for my asterisk instead. |
| 20:23.34 | javawebdev | bmoraca: This would be for a virtual office type situation. So I would like to show the receptionist something that indicates the name of the company or person being called without having to look it up somewhere else. |
| 20:24.20 | [TK]D-Fender | SparFux: "stuff"? Could you be a little more vague please? |
| 20:25.11 | SparFux | Fender: For example, if I want to turn off telephone rings, like at night, there is a number I can dial. For accessing things like this, I would like to use an asterisk specific prefix, like e.g. 0* |
| 20:25.11 | bmoraca | javawebdev: I don't know about you, but it's not hard to translate 3 letter codes into their actual meaning from memory. If it really becomes a problem, you could always overwrite the entire callerid name data and then swap it back when the call is transferred. |
| 20:26.26 | bmoraca | SparFux: if you just use * codes for that, you'll be fine. for example, nightmode could be *65. you just need to plan for it. and if you have more than 100 star codes, you have bigger problems. |
| 20:27.11 | bmoraca | or you could do 3 digit star codes if you need more |
| 20:27.22 | bmoraca | your ATA and asterisk will match the most precise pattern |
| 20:27.33 | SparFux | That's true. Perhaps it is pointless to have all star codes of my ATA be available plus the 50 codes I have for my asterisk box. I cannot keep them in mind anyways. |
| 20:27.40 | bmoraca | so if your ATA uses *X for its features and your asterisk codes are *XX, they will not overlap |
| 20:27.42 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 20:28.28 | SparFux | bmoraca: Yes, and the XX could be some sane T9 code. Then I know the code immediately when looking at the dial pad. |
| 20:29.15 | brunner | Does anyone here have experience with Amazon Elastic Cloud services? |
| 20:29.29 | javawebdev | bmoraca: There could be dozens of 3 digit codes to remember. There is a SendText() app. What is the point of it? Is there a specific feature in a phone that would allow the SEndText() app to work? |
| 20:29.37 | bmoraca | SparFux: well, if that's what you want to do, then you'll need longer numbers. it'll be a tradeoff. |
| 20:30.09 | bmoraca | javawebdev: not all phones support SIP text. have you looked at the wiki page for it yet? |
| 20:30.58 | SparFux | bmoraca: I can keep 3 digit predictive text in mind quite well. And I think I will use the 0* for the master prefix of asterisk, then I have no clashes, dialing is fast, and 5 digits is ok for all the asteirsk functions to access. |
| 20:31.11 | javawebdev | bmoraca: no I have not. Would you happen to have the URL? I didn't even know there was an asterisk wiki |
| 20:31.18 | SparFux | Is there a way to specify extension numbers as letters in predictive text T9 ? |
| 20:31.52 | bmoraca | javawebdev: google the term "asterisk sendtext". |
| 20:32.31 | bmoraca | javawebdev: in fact, any asterisk application or feature can be found in that manner. "asterisk (function)" in google. the first link will almost always be for voip-info.org which is the wiki |
| 20:32.45 | *** join/#asterisk specialist1 (n=research@119.160.105.172) |
| 20:32.50 | specialist1 | hello everyone |
| 20:32.58 | javawebdev | bmoraca: are you talking about voip-info.org? |
| 20:33.10 | bmoraca | yes |
| 20:33.31 | specialist1 | Any folks using a reseller module for opensource billing of astersisk |
| 20:34.19 | javawebdev | bmoraca: Oh. I thought that was some spam site, I avoided clicking on it in google. Anything with a dash and the word info in the domain name triggers my subconcious spam filter :) |
| 20:34.47 | stabler | javawebdev: lol voip-info is baller |
| 20:34.51 | bmoraca | javawebdev: that site should be your friend. |
| 20:35.05 | stabler | i <3 voip-info |
| 20:35.49 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 20:36.15 | javawebdev | just messing arround. I visit the site regularly, but that was my first impression. I just didn't know that was "the wiki" |
| 20:37.30 | stabler | i think TW is throttling my connections :( |
| 20:41.10 | bmoraca | my upstream provider had a service outage 2 days ago...uhg...that was bad... |
| 20:41.12 | stabler | i guess they dont like it when you start moving 5GB+/day in a residential area |
| 20:44.22 | SparFux | It would really be nice to have a function like T9(asterisk) which would become number 27837475 |
| 20:44.33 | SparFux | In the dialplan. |
| 20:44.52 | muiro | SparFux: write it yourself in pure dialplan. Put it in the wiki. |
| 20:45.24 | SparFux | muiro: you think it can be written in the dialplan only? without any other stuff? |
| 20:45.30 | muiro | SparFux: absolutely |
| 20:45.34 | SparFux | hm..... |
| 20:46.08 | muiro | just out of curiosity, what would you use that kind of output for? |
| 20:47.16 | muiro | it could be done in dialplan, but I bet most people would write a parser in python or something and shell out |
| 20:47.41 | muiro | it's probably already been done before |
| 20:48.22 | muiro | lol, you could shell out and replace everything with a long call to sed |
| 20:50.02 | SparFux | muiro: I would use it for defining all the codes I use with asterisk. Like my VoiceMail is VM<mailbox> and VM is 86. So to access mailbox 1234 I dial 861234. But it can become a pain to define longer codes. I can for example define a code to reboot by fiddling with numbers, or simply with exten => T9(reboot),1,System(reboot) |
| 20:51.14 | muiro | hmm, defining an extension that way would be a little weird. You could set a variable equal to ${EXTEN}, run your parser on it, then goto in another context |
| 20:51.31 | muiro | more than a little weird. It isn't possible, far as I know |
| 20:51.51 | muiro | if you want to get into writing a module to parse those, have a blast. |
| 20:52.33 | SparFux | it might not be possible, but as I said, it would be really great. |
| 20:53.29 | muiro | SparFux: to do your vm, you could name the extension something like _86XXXX, then pass ${EXTEN:2:4} to VoiceMail |
| 20:54.08 | *** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca) |
| 20:54.49 | SparFux | muiro: yes, yes, that's what I do. But the point is, the 86XXXX is unreadable. And with longer codes it becomes more and more painful. Nicer would be _T9(VM)XXXX, ... |
| 20:55.44 | muiro | SparFux: I'm sure a smarter mind than my own could come up with a better solution |
| 20:56.07 | [TK]D-Fender | SparFux: thats only for your benifit in reading the exten. You COULD use a little IQ and jsut put a COMMENT on the end of the darn line like any sane person would do. |
| 20:56.30 | muiro | I was about to suggest that |
| 20:57.52 | javawebdev | anybody know if there are any phones that you can program soft keys on the fly from the asterisk dial plan? |
| 20:57.56 | SparFux | Fender: it is even really hard to generate the code oneself. Like "sipgateout" would become, hm... let me see, oh, I use my mobile, type it in and READ the number. That's lame. It should be done by a preprocessor of the dialplan or what. Hey, that's it. M4!! |
| 20:58.08 | SparFux | I could simply m4 the whole extensions.conf |
| 20:59.39 | SparFux | But even then the logging output would be annoyingly complex. Better have it parsed in asterisk. |
| 21:00.19 | [TK]D-Fender | SparFux: You are about the only person who cares about this. Congratulations. |
| 21:00.25 | [TK]D-Fender | :p |
| 21:00.29 | SparFux | Fender :-D |
| 21:00.38 | SparFux | It might be because I am a little weird. |
| 21:00.58 | [TK]D-Fender | SparFux: Don't be modest, you are a complete FREAK! |
| 21:01.13 | SparFux | People say I am totally nuts. |
| 21:01.46 | SparFux | Can I define fixed variables in a context? Like one which would not change? |
| 21:02.25 | [TK]D-Fender | SparFux: What is a variable in a context/ |
| 21:02.40 | [TK]D-Fender | SparFux: * variables have no functional "state" in the first place. |
| 21:03.40 | SparFux | hm... some kind of a fixed expression. Like when I choose a prefix for my fwd-out number and I say FWDOUTPREF=888 and then it would be changed every time to 888. Like a preprocessor thingy. |
| 21:05.12 | SparFux | I notice many of my fixed stuff is defined in [globals] This is overhead. Variables which would not be changed, are defined there. |
| 21:05.14 | SparFux | hm... |
| 21:05.20 | [TK]D-Fender | SparFux: We call these things "tuff you should have under [globals] |
| 21:05.43 | SparFux | as a global variable? hm... |
| 21:05.48 | *** part/#asterisk L2SHO (n=adam@fw01ext.voicepulse.com) |
| 21:06.00 | SparFux | the prefix of fwd-out? I think it will never get changed. |
| 21:06.15 | muiro | "fixed variable". I lol'd |
| 21:06.32 | [TK]D-Fender | SparFux: the word you're looking for is "constant" |
| 21:06.52 | SparFux | ah! yes. Are there constants in asterisk? |
| 21:07.17 | SparFux | no. |
| 21:07.37 | SparFux | Come on, I simply define every prefix for my outgoing voip providers in globals and that's it. |
| 21:10.49 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 21:16.34 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 21:26.46 | k-man | how would i make a voicemail message that would say "to leave a message for X, press 1, for Y press 2 and for Z press 3 |
| 21:27.41 | [TK]D-Fender | k-man: You don't. this is not a "voicemail message". this is YOU making an IVR in the dialplan and dumping them in the appropriate box based on their input. |
| 21:28.16 | k-man | [TK]D-Fender: ok - so do i read up on IVR? |
| 21:28.26 | [TK]D-Fender | ~book |
| 21:28.27 | jbot | well, 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 |
| 21:28.28 | [TK]D-Fender | ~wikis |
| 21:28.29 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:29.11 | [TK]D-Fender | checkout time, heading home, BBIAB |
| 21:31.08 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 21:34.51 | jaytee | quittin time for me too, be back later |
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| 21:43.38 | ayeso | so i have the following: GotoIf($["${CALLERID(ani)}" = "5555555555"] what if i want to match the ani to a list of numbers, can that be done? or do i need multiple GotoIfs? |
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| 21:45.30 | ayeso | anyone here? |
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| 21:59.39 | NewCastleScott | Im trying to configure an IAXy with *. Im getting eather code 50 or 29. can someone point me to a chapter or something to google to solve this issue? thanks in advance |
| 22:09.24 | javawebdev | any of you guys using asterisk on OpenSolaris? |
| 22:11.22 | keith4 | javawebdev: put me down for a "no" on your poll |
| 22:12.27 | Qwell | define "using" |
| 22:18.03 | muiro | undefined symbole "using" in context OpenSolaris |
| 22:18.22 | javawebdev | Qwell: using meaning in production or even testing |
| 22:18.39 | Qwell | define "testing" |
| 22:18.58 | Qwell | or just ask your question and stop making us guess... |
| 22:19.03 | javawebdev | With the performance results reported on thraillingpenguin.com I'm surprised that there isn't more interest with Asterisk on Solaris |
| 22:20.22 | *** join/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 22:20.25 | *** part/#asterisk takashi_85 (n=ahmed@41.196.80.104) |
| 22:20.28 | javawebdev | Qwell: In addition to Solaris 10, I've been trying to get asterisk compiled on OpenSolaris and I ran into this bug http://bugs.digium.com/view.php?id=13704 I was just curious how important getting Asterisk to work on Solaris/OpenSolaris was to the asterisk developers |
| 22:21.17 | *** join/#asterisk Tusker (n=tusker@c-24-98-144-94.hsd1.ga.comcast.net) |
| 22:22.04 | Tusker | heya guys, trying to diagnose a 401 auth denied error for incoming calls from freedigits.net, any idea how to resolve this? I can only see the INVITE in a packet sniff, but not in the CLI. |
| 22:23.44 | *** join/#asterisk nima0102 (n=nima@91.98.215.22) |
| 22:24.33 | [TK]D-Fender | Tusker: enable SIP DEBUG at * CLI |
| 22:24.44 | [TK]D-Fender | Tusker: it doesn't show up unless you enable SIP debugging |
| 22:25.16 | javawebdev | Tusker, isn't freedigits going offline? |
| 22:26.12 | Tusker | javawebdev: not as far as I know, is that true ? |
| 22:26.59 | javawebdev | Tusker, they haven't been accepting new applications for a while. I thought they were having issues. |
| 22:27.30 | Tusker | i do have a number, so maybe they are just unstable... |
| 22:27.31 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 22:29.07 | *** join/#asterisk ingenius (n=alektro@host151.200-117-158.telecom.net.ar) |
| 22:29.53 | javawebdev | Tusker: I just checked the website. If you click on the support.freedigits.com link or on the faxdigits.com link, I get Address Not Found errors. Makes it seem like the site was abandoned |
| 22:30.00 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 22:30.24 | Tusker | ok, too bad then :( |
| 22:33.38 | javawebdev | Tusker: not sure of anything. All I know is I'm having trouble acccessing their websites, including sipnumber.com which I think is the parent site. Not sure if it's just my connection. I don't have an account with them. |
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| 22:41.40 | javawebdev | Quell: have you been able to get any version of Asterisk to compile on OpenSolaris? If so would you mind telling me which versions? |
| 22:42.38 | mazpe | i'm trying to use Realtime queues... any particular reason why i get this errror: app_queue.c: Unable to join queue 'myqueue' |
| 22:45.08 | Qwell | javawebdev: many |
| 22:45.36 | bmoraca | i shouldn't have a problem using a 56k dialup modem plugged in to an FXS port going through asterisk and out a PRI, should i? |
| 22:46.09 | Qwell | bmoraca: Get an analog line for the modem... |
| 22:46.44 | bmoraca | we have one currently, but we're trying to cut it out of the costs |
| 22:46.53 | Qwell | don't. |
| 22:47.02 | bmoraca | will it cause problems? |
| 22:47.13 | Qwell | You're adding 3 extra layers. |
| 22:48.10 | Qwell | I'm not going to say "yes it will work" or "no it won't work". I'm telling you what I would do. |
| 22:48.29 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 22:48.38 | bmoraca | well, i'd keep the analog line, too, but it's not my decision. |
| 22:48.45 | bmoraca | it's used once or twice a month |
| 22:49.03 | bmoraca | and bossman doesn't see the need to pay $15/mo for something we hardly use |
| 22:49.25 | ayeso | I have this gotoif: GotoIf($["${CALLERID(ani)}" = "555555555"] If I want to match the ani to several numbers do I have to make mutiple gotoifs or can i make a list varialble somehow? |
| 22:49.26 | Qwell | until it fails miserably one day when you absolutely need it. |
| 22:49.27 | *** join/#asterisk maddog01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 22:49.32 | Qwell | People do the same with 911. |
| 22:49.40 | Qwell | or PRIs vs SIP |
| 22:50.01 | Qwell | bmoraca: Ask him how much it would cost if it stopped working one day for reasons out of your control. |
| 22:50.11 | Qwell | does that justify the $15/month? Probably, yes. |
| 22:50.25 | bmoraca | it wouldn't cost anything...we use it for testing dialup internet on customers' computers. |
| 22:50.35 | Qwell | *facepalm* |
| 22:50.38 | bmoraca | it's not for anything that's remotely business critical |
| 22:50.52 | Qwell | If it may or may not work, why bother doing the testing? |
| 22:51.07 | Qwell | If you're going to test something, you need an absolutely known good always works setup. |
| 22:51.22 | Qwell | How much is it going to cost you if you mis-troubleshoot an issue? |
| 22:52.18 | bmoraca | nothing. our walkin customers are the dregs of society. we lose money on our inhouse service department already. |
| 22:53.31 | bmoraca | i understand your concern, and I wouldn't do this in a production system, but it's about the lowest priority function in our company...making sure the toilets are flushed ranks higher... |
| 22:53.44 | voxter | bmoraca: so stop offering it. |
| 22:53.54 | Qwell | what he said ^^ |
| 22:53.56 | bmoraca | i'm not. i just want to know if it works. |
| 22:54.05 | voxter | bmoraca: the quick and simple answer is no. |
| 22:54.16 | bmoraca | reason? |
| 22:54.19 | voxter | bmoraca: i bet you could get it to work, but like qwell said, not reliably |
| 22:54.27 | voxter | you're doing like 3 da/ad conversions |
| 22:54.31 | voxter | youre bound to run into problems. |
| 22:54.31 | Qwell | I never even said not reliably. |
| 22:54.49 | voxter | takes back what qwell didnt say |
| 22:54.51 | voxter | :) |
| 22:55.18 | voxter | bmoraca: its the cost of 3 coffee's a month. you're looking at the problem the wrong way, man. |
| 22:55.51 | javawebdev | bmoraca: why not just try it and see if it works? |
| 22:56.04 | voxter | bmoraca: if youre looking to eliminate $15/month cost to you, for the headache of some hacked up thing that ill tell you right now will not be a set it and forget it job, youre much better off just not offering the service to your customers |
| 22:58.10 | ayeso | I have this gotoif: GotoIf($["${CALLERID(ani)}" = "555555555"] If I want to match the ani to several numbers do I have to make mutiple gotoifs or can i make a list varialble somehow? |
| 22:58.17 | bmoraca | well, i'm going to try it and see what happens. i'm not a doom and gloom kind of person, and i can't see any reason why it wouldn't work. if it's not reliable, i'll tell my boss we need to keep the analog circuit. |
| 22:59.59 | javawebdev | Quell: I've been trying all week to get Asterisk latest svn checkout to compile on OpenSolaris but I'm running into bugid 0013704 any thoughts on how to get it to compile? I tried adding -D__EXTENSIONS__=1 in makeopts GC_FLAGS and also trying GMAKE CFLAGS="-D__EXTENSIONS__=1" but that didn't help |
| 23:00.10 | voxter | bmoraca: you blow my mind. |
| 23:00.57 | voxter | bmoraca: I run a large asterisk based ITSP and Qwell is one of the key developers @ Digium. I'm offering you a suggestion based on actual testing that will save you a retarded amount of stress and headache. |
| 23:01.03 | voxter | bmoraca: go ahead though, try it out! |
| 23:01.09 | voxter | I'll wait. :) |
| 23:02.13 | bmoraca | what stress and headache? my expectations are extremely low. |
| 23:02.28 | voxter | Like i said. Go forth and prosper! |
| 23:04.57 | javawebdev | Qwell: if I run 'gmake CFLAGS="-D__EXTENSIONS__=1" I get a different set of errors related to tgetent. 'conflicting types' and 'previous declaration' in editline.c |
| 23:05.17 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-78-138.nycmny.east.verizon.net) |
| 23:05.29 | Steve_J-obs | hello everybody!!! |
| 23:05.55 | javawebdev | hello Steve |
| 23:08.44 | mazpe | in queues how do i tell it to say how many callers are in queue? |
| 23:08.59 | denon | ~itsp |
| 23:09.21 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:09.21 | denon | ~itsp-us |
| 23:09.22 | mazpe | in periodic-anouncement i have queue-thereare |
| 23:09.22 | *** part/#asterisk Khratos (n=khratos@190.166.103.111) |
| 23:09.22 | Qwell | ~itsplist-us |
| 23:09.23 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 23:09.23 | Qwell | or not |
| 23:09.28 | Qwell | smacks jbot |
| 23:09.29 | denon | oh yeah, list |
| 23:09.59 | mazpe | but, it doesnt sayt the actual numbers of callers in the queue.. is there another parameter for that? |
| 23:12.37 | [TK]D-Fender | mazpe: Yes, go read the sample config |
| 23:24.02 | dalbaech | hmmmmm. after recompiling "originate" isn't in my CLI anymore... it is on one box, but not on this one.... |
| 23:24.04 | javawebdev | Qwell: I just got it to install on OpenSolaris. In makeopts I set GC_CFLAGS="-D__EXTENSIONS__=1" and that seemed to do the trick. I didn't have the quotes before |
| 23:24.05 | dalbaech | weirdness |
| 23:25.35 | javawebdev | oops... I mean compile... installing now |
| 23:25.38 | javawebdev | smacks head |
| 23:28.08 | dalbaech | ok; it's listed in applications, so why isn't it in the CLI anymore? |
| 23:28.09 | dalbaech | wtf?Q |
| 23:28.26 | dalbaech | ahh |
| 23:28.27 | dalbaech | it moved. |
| 23:28.28 | dalbaech | nevermind. |
| 23:28.28 | dalbaech | hehe |
| 23:28.30 | dalbaech | :D |
| 23:29.42 | NewCastleScott | Im trying to configure an IAXy with * using voipstunt that uses sip protocol. Im having issues according to debugging about codecs "equested/capability 0x4/0x24 incompatible with our capability 0xf800." I have set it to allow=ulaw and disallow=all. can anyone shed some light on what I might be doing wrong? |
| 23:34.03 | Micc | Is there any kind of sip echo canceling setting in asterisk? |
| 23:34.17 | Micc | I guess this would be a codec specific thing. |
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| 23:35.34 | NewCastleScott | is there any info I can provide to make getting help easier? I dont know about echo canceling |
| 23:35.49 | javawebdev | Does the test with digium by dialing 500 in the sample IVR no longer work? I tried it but keep getting circuit-busy messages |
| 23:36.29 | *** part/#asterisk gego (n=gego@dyndsl-091-096-101-204.ewe-ip-backbone.de) |
| 23:40.04 | NewCastleScott | every once in a while I get the codec error but Im getting a "register_varify" " code 29" most of the time |
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