| 00:00.50 | *** join/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 00:01.25 | [T]ank | anyone here using an ata through firewall? I am setting one up and really don't have any way to test it. Wanted to compare some notes. |
| 00:01.48 | [T]ank | i have asterisk 1.4 on the inside and a grandstream ata on the "outside" |
| 00:02.01 | [T]ank | all i should have to do is open 5060 for sip right? |
| 00:02.16 | mosty | probably wise to port forward the rtp ports too |
| 00:02.30 | [T]ank | 5004? |
| 00:02.34 | [T]ank | tcp or udp? |
| 00:02.39 | [T]ank | all udp, right? |
| 00:05.19 | mosty | see rtp.conf |
| 00:05.37 | [T]ank | thank you |
| 00:12.12 | [T]ank | so how does asterisk handle the destination port. If i send port range 5000:5099 to my asterisk ip address and destination port of 5000, does asterisk handle where to land the traffic? Or do I need a destination range of 5000:5099? |
| 00:13.19 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 00:16.55 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4e336a27f4d6a288) |
| 00:21.24 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 00:24.44 | *** join/#asterisk devhen_ (n=devhen@160.7.235.107) |
| 00:26.06 | jaytee | what port * listens on is in your sip.conf file. It's usually 5060 with few exceptions. That's for SIP to setup the call, the audio uses RTP which with * is normally ports 10000-20000 which is a broad range but not as broad as the RFC for RTP. X-lite softphone likes to try and use 8000 as an RTP port. |
| 00:27.20 | *** join/#asterisk ZeroLux (n=none@96-25-30-144.ral.clearwire-dns.net) |
| 00:29.16 | *** join/#asterisk moy (n=moy@189.169.71.102) |
| 00:42.52 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 00:46.09 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:46.09 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:48.08 | drfreeze | Anyone know how to bring a pri span up? |
| 00:48.32 | hi365_m | try a beer... |
| 00:48.40 | drfreeze | PRI span 1/0: Provisioned, Down, Active |
| 00:50.59 | *** join/#asterisk ZeroLux (n=none@96-25-30-144.ral.clearwire-dns.net) |
| 00:52.58 | *** join/#asterisk [intra]lanman (n=lanman@75-105-17-160.cust.wildblue.net) |
| 00:55.47 | drfreeze | Anyone familiar with PRI cards? |
| 00:56.00 | hi365_m | go on |
| 00:58.23 | *** join/#asterisk ZeroLux (n=none@96-25-30-144.ral.clearwire-dns.net) |
| 00:59.52 | [hC] | drfreeze: chances are that has to happen at your telco's end. |
| 01:03.00 | drfreeze | [hC]: hmmm |
| 01:05.38 | hardwire | hmmm indeed |
| 01:07.04 | drfreeze | [hC]: * box was simply restarted |
| 01:07.14 | drfreeze | unless it is a coincidence |
| 01:08.58 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 01:16.37 | *** join/#asterisk Segnale007 (n=Segnale0@host92-1-dynamic.9-79-r.retail.telecomitalia.it) |
| 01:20.52 | *** part/#asterisk korihor (n=korihor@190.199.171.145) |
| 01:25.05 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 01:28.20 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:31.01 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 01:31.45 | kash | hmmm.. with SIP, is there a way to use one auth line with multiple phones? |
| 01:32.04 | *** join/#asterisk vhatz (n=vhatz@athedsl-409143.home.otenet.gr) |
| 01:32.05 | JT | why would you want to do that? |
| 01:32.45 | kash | my config file is getting very large and i have one user with seven SIP devices |
| 01:32.50 | hardwire | kash: how would you route inbound calls? |
| 01:32.57 | vhatz | hello all |
| 01:33.04 | kash | by just ringing the one peer so they all ring |
| 01:33.09 | hardwire | kash: tried telling that person to sit still? |
| 01:33.50 | hardwire | kash: no dice.. |
| 01:34.06 | hardwire | sip show peers doesn't have a spot for multiple IP's |
| 01:34.08 | hardwire | :) |
| 01:34.17 | kash | same IP |
| 01:34.18 | JT | kash: you could always provision sip accounts from an sql database if it's getting out of hand |
| 01:34.26 | hardwire | kash: they can't all have the same IP |
| 01:34.56 | kash | why not? |
| 01:35.04 | kash | they have different listen ports |
| 01:35.04 | hardwire | because I said so. |
| 01:35.26 | JT | anyway the answer is no |
| 01:35.26 | hardwire | you said multiple phones, right? |
| 01:35.34 | JT | you can't share sip accounts for incoming calls |
| 01:36.02 | hardwire | kash: if you're sick of changing out Dial() arguments to reflect all seven of their devices, start setting channel variables |
| 01:36.10 | hardwire | err |
| 01:36.12 | hardwire | dialplan variables |
| 01:36.20 | hardwire | or start using find/replace |
| 01:36.26 | hardwire | or the database method :) |
| 01:37.05 | kash | hm |
| 01:37.05 | vhatz | Can anyone please let me know whqat I have to do to have early audio on SIP calls? I've tried every combination I could think of in sip.conf (progressinband) and extensions.conf (Ringing() & Progress()) but I can't have early audio... Any suggestions? Asterisk version is 1.4.21.2 |
| 01:37.28 | hardwire | use Answer() first |
| 01:37.33 | hardwire | then do other stuff. |
| 01:37.35 | kash | can the user run a local IAX proxy or something and trunk it out to the server? |
| 01:38.36 | vhatz | hardwire: unfortunately I mustn't give a connect signal to the calling end before the call is actually answered from teh far end... so Answer cannot be used... :( |
| 01:38.51 | hardwire | no connect, no audio. |
| 01:39.36 | vhatz | hardwire: well there is early audio coming from the far end, we just need to pass it on to the calling party before teh call is answered |
| 01:40.27 | hardwire | FSK? |
| 01:40.54 | vhatz | hardwire: it's SIP to SIP calls, no zap cards or anything TDM Related is involved |
| 01:41.11 | hardwire | vhatz: what is the audio? |
| 01:41.15 | JT | hardwire: not correct, you can do early media before answering |
| 01:41.30 | hardwire | JT: what kind of media? |
| 01:41.35 | JT | audio... |
| 01:41.37 | vhatz | hardwire: the audio is ringback and announcements |
| 01:41.39 | JT | that's what media is |
| 01:41.56 | hardwire | so "Please hold" without a "connect"? |
| 01:42.08 | JT | or a ringback tone |
| 01:42.17 | vhatz | hardwire: something like that |
| 01:42.32 | hardwire | sounds neat |
| 01:42.53 | vhatz | I know that in older version that I tried early audio was sent/received by default, but now that I re-installed asterisk it is giving me a hard time |
| 01:42.54 | hardwire | so, play hold music w/o the connect. |
| 01:42.55 | JT | PRIs can do early media too |
| 01:43.25 | hardwire | I just plain had no idea that existed, simply because that seems like a great way to evade USAC on PSTN |
| 01:43.57 | hardwire | esp for information services like time and temp. |
| 01:44.24 | Strom_M | of course, if you do play audio without supervising, the telco will tear your call down after 1-3 minutes if you don't supervise |
| 01:44.29 | kash | hmm |
| 01:44.56 | hardwire | vhatz: whats the dialplan for something like that look like? |
| 01:44.59 | kash | hardwire: what SIP proxy software will allow me to run it behind a NAT so this user can use all his phones |
| 01:45.19 | vhatz | hardwire: is hould be a simple dial as far as I can remember... |
| 01:45.20 | *** part/#asterisk [T]ank (n=ckwall@c-71-199-25-239.hsd1.co.comcast.net) |
| 01:45.27 | hardwire | kash: siproxd :) |
| 01:45.31 | JT | kash: are you actually trying to solve a problem other than your busy dialplan? |
| 01:45.40 | kash | JT: now i am |
| 01:45.41 | kash | :p |
| 01:45.43 | hardwire | vhatz: so you just run Playback() before Dial() ? |
| 01:46.02 | hardwire | I assumed Playback() would initiate a connection. |
| 01:46.10 | hardwire | asme with Ringing() |
| 01:46.19 | vhatz | hardwire: Playback did not work ... there was no audio no matter what I did before the far end answered |
| 01:46.31 | hardwire | You had to Answer() first? |
| 01:46.47 | vhatz | hardwire: no, Answer() must be avoided |
| 01:46.52 | JT | answer answers the call |
| 01:46.56 | JT | can't use it :) |
| 01:47.05 | hardwire | vhatz: did Answer() allow Playback() to work? |
| 01:47.22 | vhatz | if you put Playback after the Answer sure it works |
| 01:47.25 | hardwire | sounds like you said Playback() wasn't sending audio no matter what. |
| 01:47.32 | hardwire | vhatz: ah.. I misunderstood you then. |
| 01:47.45 | vhatz | hardwire: no matter what means before answer... |
| 01:47.49 | wwalker | I can go into voicemailmain and can listen to messages, I can change folders, but if I hit 3 for advanced options, it immediately plays 'vm-starmain' ("Press star to return to the main menu"). any ideas? |
| 01:48.31 | vhatz | early media used to work right out of the box... I don't know why it doesn't work now :( |
| 01:48.48 | hardwire | tests it |
| 01:48.57 | hardwire | I have a 1.2.x box |
| 01:49.00 | *** join/#asterisk Yanik (n=yanik@modemcable218.3-57-74.mc.videotron.ca) |
| 01:50.19 | hardwire | playback auto connects me |
| 01:50.28 | hardwire | vhatz: did you change anything to enable early audio so far? |
| 01:51.06 | hardwire | Ringing() worked fine.. |
| 01:51.22 | hardwire | I assume since I was in ring state, it returned with no action |
| 01:51.41 | *** join/#asterisk devhen_ (n=devhen@160.7.235.107) |
| 01:52.02 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 01:52.16 | vhatz | hardwire: does Ringing give you a ringback generated on your asterisk or does it pass you the remote ringback? |
| 01:52.56 | hardwire | vhatz: I believe it does nothing for SIP->Asterisk |
| 01:53.07 | hardwire | I tried using Progress() then Playback() |
| 01:53.30 | hardwire | vhatz: it doesn't appear to pass any RTP to my phone |
| 01:54.47 | vhatz | as I recall a simple Dial command should be enough to pass early audio beween 2 SIP calls |
| 01:54.56 | vhatz | it seems that this is not the case though |
| 01:55.38 | hardwire | hey |
| 01:55.40 | hardwire | I got it working |
| 01:56.00 | hardwire | <PROTECTED> |
| 01:56.00 | hardwire | <PROTECTED> |
| 01:56.00 | hardwire | <PROTECTED> |
| 01:56.03 | hardwire | short paste |
| 01:56.05 | hardwire | don't buzz me for that. |
| 01:56.27 | hardwire | tests via cell->pri |
| 01:57.34 | hardwire | no worky |
| 01:58.20 | kash | if i'm using users.conf to define a peer with a 'baseexten', say phone 1 registers through it and receives 6000. if another phone registers, does it receive 6001 ? |
| 01:58.21 | hardwire | same with going in/out the same PRI |
| 01:58.57 | hardwire | kash: it doesn't assign registrations for phones |
| 01:59.23 | hardwire | it sets up sip users, iax users, voicemail users, and dialplan contexts |
| 01:59.23 | vhatz | hardwire: this caues te file to played but the dial is executed after the file stops... still this is not real realy audio... :( |
| 01:59.28 | kash | ; Starting point of allocation of extensions |
| 01:59.28 | kash | ; |
| 01:59.28 | kash | userbase = 6000 |
| 01:59.34 | kash | what's that then :/ |
| 01:59.51 | kash | this file has shitty documentation |
| 01:59.53 | hardwire | vhatz: Asterisk 1.2.24 is what I have here |
| 02:00.04 | hardwire | kash: misleading? |
| 02:00.10 | kash | yes |
| 02:00.22 | hardwire | kash: just crash stuff till it doesn't work anymore |
| 02:00.25 | hardwire | make lots of backups |
| 02:00.38 | hardwire | you'll get an instant diploma from asterisk u |
| 02:00.50 | hardwire | can I suggest rsnapshot? :) |
| 02:01.43 | hardwire | JT it's a shame my PRI isn't dealing well with it |
| 02:02.16 | hardwire | it would be nice to use poo flinging monkeys as a telemarketer zapper |
| 02:02.51 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 02:04.58 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 02:20.34 | *** join/#asterisk pkunkra (n=chris@cpe-74-73-8-115.nyc.res.rr.com) |
| 02:21.01 | *** join/#asterisk LemensTS (i=LemensTS@adsl-75-42-148-10.dsl.stlsmo.sbcglobal.net) |
| 02:24.01 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 02:49.22 | *** join/#asterisk rfernandez (n=rfernand@189.136.64.128) |
| 02:49.26 | rfernandez | hiya!! |
| 02:49.35 | rfernandez | can someone please tell me about a good and cheap 4 fxo port gateway? |
| 02:51.35 | carrar | Find a old ADIT600 mounted in a office that someone forgot about |
| 02:51.48 | rfernandez | ? |
| 02:53.02 | carrar | Though most of those are FXS |
| 02:53.29 | carrar | Just suck it up and buy one from Digium |
| 02:53.48 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:53.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:54.00 | carrar | Right Russell? |
| 02:54.17 | carrar | (say yes) |
| 02:54.20 | rfernandez | i mean a analog fxo port like a grandstream linksys or audiocodes.. :S |
| 02:54.26 | rfernandez | for analog lines |
| 02:54.36 | russellb | carrar: maybe! |
| 02:54.41 | carrar | hhahah |
| 02:54.46 | rfernandez | if the customer have e1's i guess he minimum have money for the pbx that support e1's xD |
| 02:55.07 | russellb | i mean, yes |
| 02:55.18 | carrar | See, there you have it! |
| 02:55.28 | rfernandez | lol! |
| 02:55.31 | rfernandez | jejejej xD |
| 02:55.42 | russellb | i can haz money? |
| 02:55.59 | carrar | slides russell a 20 |
| 02:56.03 | russellb | sweeeeet |
| 02:56.05 | LemensTS | !money |
| 02:56.06 | carrar | wait, you should be paying me!! |
| 02:56.20 | russellb | lies |
| 02:58.09 | carrar | goes back to listenign to his confusingly intriguing Japanese punk music |
| 03:02.21 | Wayhigh | sup carrar |
| 03:02.38 | carrar | hey wayhigh |
| 03:02.53 | russellb | he just said listening to confusingly intriguing music |
| 03:02.53 | carrar | When did they let the riftraft in here? |
| 03:03.16 | carrar | Well it is, cause I don't know a lot of Japanese |
| 03:03.44 | Wayhigh | carrar: long time ago.. I hang out here asking dumb questions and being told "we don't support trixbox" |
| 03:03.52 | carrar | hahhah |
| 03:04.07 | carrar | You should install from source like everyone!!! |
| 03:04.18 | carrar | Thats the RIGHT way to do it |
| 03:04.30 | Wayhigh | I've got 1 trixbox, 1 source install, 2 PIAF, etc.. |
| 03:04.37 | carrar | PIAF? |
| 03:04.42 | Wayhigh | pbxinaflash |
| 03:04.44 | *** join/#asterisk ZX81 (n=matt@120.89.80.110) |
| 03:04.51 | carrar | no SwitchVox boxes? |
| 03:04.54 | ZX81 | ~seen Tzafrir_Home |
| 03:05.13 | jbot | tzafrir_home <n=tzafrir@bzq-179-75-202.static.bezeqint.net> was last seen on IRC in channel #asterisk-dev, 29d 5h 18m 42s ago, saying: 'Anyway, time for me to go to sleep anyway'. |
| 03:05.13 | Wayhigh | no.. none of the switchvox boxes |
| 03:05.18 | Wayhigh | oh.. one asterisknow and trixbox pro also |
| 03:05.33 | carrar | Switchvox is nice for customers that want to logs/stats or make their own changes |
| 03:05.37 | ZX81 | Surely I've seen him since then |
| 03:06.09 | JT | eww trixbox pro |
| 03:06.10 | JT | the worst |
| 03:06.13 | Wayhigh | what's the most common thing for people to run as asterisk though? |
| 03:06.22 | jameswf-home | no there is worse :) |
| 03:06.25 | carrar | russellb, You should check out "Linda Linda" from the blue hearts, old Japanese punk band |
| 03:06.30 | russellb | people run asterisk as asterisk i think |
| 03:06.35 | Wayhigh | JT: yeah.. I hear that.. don't even get me started |
| 03:06.45 | russellb | trixbox is teh lame |
| 03:06.52 | carrar | w0rd |
| 03:06.57 | Wayhigh | trixbox is teh haxored |
| 03:07.10 | jameswf-home | trixswitch is wayyyy worse than trixbox |
| 03:07.49 | Wayhigh | ~seen stotaro |
| 03:07.50 | jbot | i haven't seen 'stotaro', Wayhigh |
| 03:08.00 | russellb | jameswf-home: heh, who runs that? |
| 03:08.05 | jameswf-home | ~seen jbot |
| 03:08.06 | jbot | i haven't seen 'jbot', jameswf-home |
| 03:08.29 | Wayhigh | ~wayhigh |
| 03:08.30 | jbot | Asterisk mouse WAZ in his 1U, eatinz his thermo ribbons.. HE R MOUSEKILLA |
| 03:08.49 | Wayhigh | ~carrar |
| 03:08.55 | carrar | What! |
| 03:09.50 | Wayhigh | so who here that installs asterisk from source uses any of the web gui or manager functions? |
| 03:10.08 | Wayhigh | like that new ajam stuff.. |
| 03:11.17 | carrar | CLI or BUST!! |
| 03:11.58 | Wayhigh | the cli leaves a lot to be desired... I use astman more than the cli |
| 03:15.07 | russellb | I personally don't have anything against people that use GUIs ... |
| 03:15.32 | russellb | I have some beef with the trixbox distribution, specifically, but not FreePBX or the people that use/develop FreePBX itself |
| 03:15.55 | russellb | If it were up to me, I'd welcome all the GUI discussion people want in here |
| 03:16.08 | russellb | but the masses that are here more than I am seem to disagree :) |
| 03:16.44 | russellb | FWIW, i guess ... |
| 03:17.11 | Wayhigh | it's all good.. I like the gui ones.. |
| 03:18.02 | jameswf-home | yeah well politics are not exclusive to fonality |
| 03:18.21 | *** join/#asterisk anonymiss (n=user@c-71-234-197-65.hsd1.ct.comcast.net) |
| 03:18.25 | jameswf-home | read a spencer quote today that reaks of fonality attitude |
| 03:18.31 | anonymiss | is asterisk typically unstable on virtual private servers? |
| 03:18.52 | Wayhigh | ya know what drives me nuts is the lack of IAX2 perl module |
| 03:19.03 | jameswf-home | anonymiss: depends asterisk it self is probably fine on a VPS if set up right |
| 03:19.06 | JT | why on earth would you need that, Wayhigh ? |
| 03:19.33 | anonymiss | jameswf-home: have you ever run it on a vps? |
| 03:19.36 | Wayhigh | JT: :) I just do.. |
| 03:19.49 | JT | Wayhigh: pretty good explanation there |
| 03:20.00 | jameswf-home | asterisk is simply the engine once you modify and configure it your way the engine may seize |
| 03:20.20 | Wayhigh | almost as good as fender's explanation of how trixbox works |
| 03:20.54 | jameswf-home | I do most of my development testing on virtual servers, it allows me to test in a broad range of enviroment without clutter |
| 03:20.56 | JT | there's no-one here called fender btw :) |
| 03:21.28 | Wayhigh | ~seen [tk]d-fender |
| 03:21.29 | jbot | [tk]d-fender is currently on #asterisk (6h 32m 14s). Has said a total of 2 messages. Is idling for 5h 50m 15s, last said: '~wikis'. |
| 03:21.41 | JT | yep, not fender |
| 03:21.46 | Wayhigh | hahaha |
| 03:21.48 | jameswf-home | ~seen my ass |
| 03:21.49 | jbot | i haven't seen 'my ass', jameswf-home |
| 03:21.59 | russellb | you can also msg the bot directly ... |
| 03:22.49 | JT | i know if i was him i'd get the shits at people calling me "fender" |
| 03:23.42 | jameswf-home | yes gibson makes much better guitars |
| 03:24.44 | Wayhigh | I was thinking perhaps 'wheelwell' instead? |
| 03:25.35 | JT | iax2 and perl sounds like such an ungodly combination |
| 03:26.02 | Wayhigh | JT: the perl module would probably be a wrapper around iaxclient or something like that |
| 03:26.40 | Wayhigh | you could use chan_iax2 but I'm not sure you'd want or need to |
| 03:26.43 | JT | do you actually need IAX2? |
| 03:27.49 | Wayhigh | sort of.. I have an ATA that uses it.. |
| 03:28.15 | Wayhigh | it's good for testing |
| 03:28.29 | russellb | <3 IAX2 |
| 03:28.46 | jameswf-home | waits for IAX3 |
| 03:28.52 | *** join/#asterisk alancio (n=Alancio@190.75.3.207) |
| 03:29.07 | russellb | i don't see a need for IAX3. There is plenty of room to extend IAX2 as needed. |
| 03:29.18 | russellb | of course, you were probably just making a random comment ... |
| 03:29.20 | JT | wow, an ATA that uses, what is it, an IAXy or something? :P |
| 03:29.42 | jameswf-home | yet another dream killed by russellb |
| 03:30.04 | carrar | I NEED IAX4 |
| 03:30.07 | russellb | I do what I can. |
| 03:30.14 | carrar | cause it uses LASERS |
| 03:30.18 | russellb | heh. |
| 03:31.03 | russellb | yes, forget video, presence, and that crap ... |
| 03:31.08 | russellb | we need lasers ... and sharks ... |
| 03:31.17 | carrar | yeah! |
| 03:31.28 | carrar | Is that so hard to get around here?!?!? |
| 03:41.14 | rfernandez | spa400 4 fxo port its good to use as a trunk interface? |
| 03:49.33 | jameswf-home | I hear if you point lasers at sharks they get pissed |
| 03:50.15 | *** part/#asterisk anonymiss (n=user@c-71-234-197-65.hsd1.ct.comcast.net) |
| 03:50.25 | alancio | you can train dolphins and then equip them with lasers |
| 03:50.41 | alancio | they are smarter than sharks |
| 03:52.55 | jameswf-home | HOLY CRAP!! an X files movie.. |
| 03:55.18 | LemensTS | dpkg |
| 03:59.55 | *** join/#asterisk DaPrivateer (n=matt7229@gateway.66fruit.com) |
| 04:07.38 | *** join/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 04:23.32 | *** join/#asterisk int21 (i=IceChat7@189.104.28.9) |
| 04:24.54 | int21 | hi all |
| 04:27.33 | *** join/#asterisk bsaxon (n=bsaxon@251.sub-75-200-223.myvzw.com) |
| 04:27.36 | int21 | someone have some idea to start a study about integrate asterisk with loquendo to respond questions in a database SQL queries |
| 04:29.08 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 04:29.44 | jaytee | why not use Festival since it's already integrated with Asterisk? |
| 04:30.34 | int21 | thanks bro, but festivalo no have my language (portuguese brazilian) |
| 04:31.03 | jaytee | oh, well don't know what to tell ya then |
| 04:33.45 | int21 | so, I have a client, they have a small database with discount cards in my city ok, he need make a channel to out clients to call and disk your card number and the asterisk will give the money amount (excuse my english) |
| 04:36.18 | Corydon76-dig | You could try the "Americas Spanish" voices in Cepstral and see if they pronounce Brazilian Portugese correctly |
| 04:37.28 | Corydon76-dig | I know it's not an exact match, but it's somewhat close, and anyway, you're providing the text |
| 04:37.58 | int21 | I tried cepstral too, the unique good one is loquendo. I have loquendo here and in windows run perfectly |
| 04:38.33 | int21 | good to my language ok! (sorry) |
| 04:41.31 | Corydon76-dig | I take it from their site that Loquendo isn't cheap |
| 04:44.10 | Corydon76-dig | It's one of the things I like about Cepstral. I have single licenses for most of the English voices |
| 04:44.15 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582704.dsl.bell.ca) |
| 04:44.57 | Corydon76-dig | and I've gone into a customer site before and sold them on Cepstral, merely by having the voice already installed on my laptop and demoing the usage |
| 04:45.09 | int21 | Cepstral for english and spanish language is very good |
| 04:45.37 | Corydon76-dig | Silly people previously thought they were going to record every single prompt |
| 04:47.13 | *** part/#asterisk EnginA (n=engin@88.242.118.114) |
| 04:50.34 | jaytee | I've been working from the other end of things with speech recognition using LumenVox. Kinda tricky at first getting used to creating grammars but I've got it working fairly solid with an Asterisk IVR. |
| 04:52.13 | int21 | speech recognition is not so easy to implement |
| 05:01.45 | Corydon76-dig | Correction. Arbitrary speech recognitiion is not easy to implement. Speech recognition with an artificially limited grammar is much easier. |
| 05:02.55 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582324.dsl.bell.ca) |
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| 05:33.31 | *** join/#asterisk sqwishy (n=sqwishy@d207-81-232-213.bchsia.telus.net) |
| 05:36.43 | sqwishy | how i asterisk? |
| 05:38.10 | JT | an intriguing question |
| 05:38.14 | sqwishy | so i plugged in the telephone cord into where it fits, and installed asterisk. what do i do next? |
| 05:38.29 | JT | ~thebook |
| 05:38.30 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 05:39.02 | sqwishy | the pdfs havce over nine thousand words and a whole bunch of stuff that doesn't apply to me |
| 05:42.26 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:44.36 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177582282.dsl.bell.ca) |
| 05:48.16 | JT | sqwishy: then fail? |
| 05:48.50 | pkunkra | *yawn* |
| 05:49.20 | bbryant | ~whatnow |
| 05:49.21 | jbot | it has been said that now is a good time to tell you that I have 6 gigabytes of data |
| 05:49.45 | pkunkra | 6 GB? that's it? |
| 05:50.09 | pkunkra | looks at jbot |
| 05:50.14 | pkunkra | you're pretty small dude.... |
| 05:51.01 | bbryant | ~nowwhat |
| 05:51.02 | jbot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E |
| 05:52.22 | pkunkra | wow |
| 05:52.32 | pkunkra | someone paid a lot of money to broadcast that |
| 05:53.01 | bbryant | it was on tv here for a while |
| 05:56.13 | pkunkra | that's like a minute long commercial |
| 05:56.21 | pkunkra | airtime must be cheap there. |
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| 07:49.12 | pputman | Anyone ever seen invalid information element 123 errors before? I'm using the latest libpri, and I've looked in the source code and q931 specifications and can't find any information elements with that number. This machine is in south africa, not sure what switch is on the other end, but it's a switchtype euroisdn. |
| 07:59.52 | *** join/#asterisk Magnus_H (n=magnus@90-228-242-54-no71.tbcn.telia.com) |
| 07:59.53 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:59.59 | Chris-NB | hi |
| 08:00.03 | Chris-NB | anyone using aastra sip phones? |
| 08:00.15 | Chris-NB | I got the problem that the phones stop working after a time |
| 08:00.38 | Chris-NB | on the display is a 'No Service' warning and the red light flashes |
| 08:00.46 | Chris-NB | after a reboot the phones work again |
| 08:00.49 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 08:00.51 | Chris-NB | ... for a while |
| 08:00.58 | Chris-NB | someone discovered that? |
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| 08:06.59 | Magnus_H | Anyone knows where to find winmodems that works as X100 fxo's? |
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| 08:20.17 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
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| 08:31.04 | Magnus_H | Found (and bought) |
| 08:31.04 | Magnus_H | http://www.x100p.com/products/FXO.php?mc_gross=119.80&address_status=unconfirmed&item_number1=PP4X100P&payer_id=YZ3ASBFQ42GGW&tax=0.00&address_street=Saerestad+Praestbolet+1&payment_date=01%3A28%3A34+Jul+25%2C+2008+PDT&payment_status=Completed&charset=windows-1252&address_zip=46791&mc_shipping=0.00&mc_handling=0.00&first_name=Magnus&mc_fee=3.77&address_country_code=SE&address_name=Magnus+Hederstad¬ify_version=2.4&custom=&payer_stat |
| 08:31.04 | Magnus_H | us=unverified&business=sales%40x100p.com&address_country=Sweden&num_cart_items=1&mc_handling1=0.00&address_city=Graestorp&payer_email=magnus%40hederstad.se&verify_sign=At89MVuYeUfTuI0P6knfdgvO-ozQAnlkHpd6RXPdYDM-GDN8x1jw3qHH&mc_shipping1=0.00&tax1=0.00&txn_id=35K70838DR872231E&payment_type=instant&last_name=Hederstad&receiver_email=sales%40x100p.com&item_name1=Power+Pack%3A+FOUR+%284%29+x+Authentic+X100P+SE+FXO+PCI+Interface+for+Digiu |
| 08:31.07 | Magnus_H | m+Asterisk+VoIP+PBX+%28Standard+%2F+Low+Profile%29&address_state=&payment_fee=3.77&quantity1=1&receiver_id=X3DFXNH8D3NK2&txn_type=cart&mc_currency=USD&mc_gross_1=119.80&residence_country=SE&payment_gross=119.80&merchant_return_link=Return+to+X100P.com. Thanks anyway! |
| 08:32.12 | Magnus_H | Oh! Sorry for flooding! http://www.x100p.com/ is the right one... |
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| 09:13.24 | dominic1 | hallo? |
| 09:13.34 | dominic1 | oh sorry |
| 09:13.38 | dominic1 | wrong window.... |
| 09:14.41 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 09:24.02 | Vec | Is the password sent to Asterisk in an IAX2 Registration request ? |
| 09:25.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 10:29.14 | penguinFunk | does anyone know if you can do video calling with zap channels ? |
| 10:29.23 | penguinFunk | we have an E1 isdn30 |
| 10:29.26 | penguinFunk | and a pri card |
| 10:30.01 | penguinFunk | according to voip-info.org you can only do video calling with sip or iax2 |
| 10:30.16 | penguinFunk | :/ |
| 10:30.40 | penguinFunk | we want to utilise our isdn channels and save internet bandwidth |
| 10:41.08 | *** part/#asterisk Keypad (n=Keypad@125-238-132-150.broadband-telecom.global-gateway.net.nz) |
| 10:58.21 | Vec | If the ISDN channels are down should zap show channels still list them ? |
| 10:58.21 | penguinFunk | yes |
| 10:58.44 | Vec | penguinFunk : thanks |
| 11:09.18 | BBHoss | video over zap? lol |
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| 11:14.36 | penguinFunk | yes |
| 11:14.46 | penguinFunk | as in video over isdn 30 channels |
| 11:14.59 | penguinFunk | each b-chan is 64kbps |
| 11:15.05 | penguinFunk | so why not ? |
| 11:16.22 | penguinFunk | it is technically possible |
| 11:16.25 | penguinFunk | just not with asterisk |
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| 11:38.21 | mgdm | penguinFunk: In my experience the kit that does that tends to be very proprietary (Cisco stuff, much of the time) |
| 11:41.29 | penguinFunk | i see |
| 11:41.37 | penguinFunk | that's a shame |
| 11:41.56 | tzafrir | yey to XML config files! |
| 11:42.04 | penguinFunk | hoorah |
| 11:42.28 | tzafrir | gajim insists on writing one incorrect character in a certain name (an alias). |
| 11:42.50 | tzafrir | And that completely fails the reading of the config file it generates later |
| 11:43.39 | *** join/#asterisk xenonex (n=xenonex@92.47.1.81) |
| 11:43.45 | tzafrir | With the very helpful error message "can't read config file blist.xml" |
| 11:46.25 | *** join/#asterisk int21 (n=IceChat7@189.104.12.150) |
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| 11:57.37 | int21 | hi all, someone cam show me a way to integrate sql withasterisk, like, the client digit a number and asterisk will check in a database and give a answer? |
| 11:59.32 | pputman | int21 you might want to read up on extensions.conf and realtime, see http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions |
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| 12:08.44 | int21 | Pputman, thanks a lot |
| 12:15.01 | *** join/#asterisk HonestWorker (n=Wothanaz@201.87.225.101) |
| 12:15.10 | HonestWorker | good morning gentlemen |
| 12:16.22 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 12:16.53 | HonestWorker | I the day before yesterday I came by for asking for help on setting the CALLERID(the number displayed for a receiver on a PSTN) to a particular PSTN number for a specific user(SIP channel/extension). |
| 12:17.33 | HonestWorker | I was told that I could set the CALLERID on the dialplan . I did a conditional branch that would trigger the Set(CALLERID(num)=xxxxxxxx) priority. |
| 12:18.05 | HonestWorker | Last night I was reading the chapter 7 of asterisk the future of telephony and I realized that it didnt work because |
| 12:18.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.28 | HonestWorker | that CALLERID is sent by the T1 control protocol . |
| 12:18.39 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 12:19.08 | HonestWorker | So, my question is: is it possible to set the CALLERID to a number that will be displayed on a receiver on PSTN through the dialplan or not? |
| 12:19.53 | HonestWorker | morning [TK]D-Fender . Do you remember my issue on setting CALLERID ?(probably noy). You and another fellow told me that one of the options was to set it on the fly by using the dialplan logic |
| 12:20.13 | [TK]D-Fender | HonestWorker: Not "on the fly. Before you call out. |
| 12:20.15 | HonestWorker | I did so but it didnt work. I think the only choice is to set that at zapata.conf |
| 12:20.30 | [TK]D-Fender | HonestWorker: NO. |
| 12:20.32 | HonestWorker | yes, before you call. I called the Set application |
| 12:20.53 | [TK]D-Fender | HonestWorker: What are you calling out on exactly? |
| 12:21.04 | HonestWorker | but as I was telling...I was reading the future of telephony book last night and on chapter 7 it was talking about signaling protocols for T1 channels |
| 12:21.31 | HonestWorker | I have a digital E1 attached to a digium board for digital links |
| 12:21.45 | [TK]D-Fender | HonestWorker: What signaling exactly? |
| 12:22.30 | HonestWorker | The signaling protocol used over the link by the carrier's ('switch', or endpoint equipment,probably not the right term) |
| 12:22.42 | [TK]D-Fender | HonestWorker: WHAT T1 SIGNALLING? |
| 12:22.43 | HonestWorker | CAS, ISDN, etc |
| 12:22.47 | [TK]D-Fender | E1 rather. |
| 12:23.16 | HonestWorker | ss7, cas, ISDN |
| 12:23.21 | *** join/#asterisk Dr-Linux|home (n=Nothing@117.20.21.66) |
| 12:23.26 | [TK]D-Fender | HonestWorker: You can't be all 3 |
| 12:23.32 | HonestWorker | I know |
| 12:23.38 | Dr-Linux|home | [TK]D-Fender: Hi there :) |
| 12:23.40 | HonestWorker | you've asked what signalling I was talking about |
| 12:23.49 | Dr-Linux|home | SIP debug and SIP trace is the same thing? |
| 12:23.51 | HonestWorker | I was answering to your question by naming a few |
| 12:24.04 | [TK]D-Fender | HonestWorker: No I friggen assked twice which one you were USING <- |
| 12:24.41 | HonestWorker | Calmdown. Its friday. The weekend is coming. Lets be gentle. |
| 12:24.42 | [TK]D-Fender | Dr-Linux|home: for intent, probably |
| 12:25.11 | HonestWorker | I am using pri_cpe. I am guessing that means ISDN |
| 12:25.29 | HonestWorker | ISDN pri and cpe=costumer premisses equipment. That is my guess |
| 12:25.38 | HonestWorker | customer |
| 12:25.45 | Dr-Linux|home | [TK]D-Fender: i see, actually my SIP provider is asking for SIP trace, so that means i should him SIP debug? |
| 12:25.53 | [TK]D-Fender | HonestWorker:pastebin your zaptel & zapata, and a call with PRI debug enabled |
| 12:26.03 | [TK]D-Fender | Dr-Linux|home: Yes |
| 12:26.07 | Dr-Linux|home | great |
| 12:26.16 | HonestWorker | the signalling on zapata.conf says pri_cpe and the switchtype says dms100 |
| 12:26.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:26.32 | Dr-Linux|home | [TK]D-Fender: what is the way i can save console log into a file? |
| 12:26.53 | [TK]D-Fender | HonestWorker: for E1 that should usually be "switchtype=euroisdn" |
| 12:27.02 | [TK]D-Fender | Dr-Linux|home: cut& paste <- |
| 12:27.04 | HonestWorker | the zaptel.conf tells that I am spanning the t1 into 32 channels . |
| 12:27.17 | [TK]D-Fender | HonestWorker: Stop talking and provide the pastebin. |
| 12:27.18 | HonestWorker | [TK]D-Fender, its working fine. euroisdn is commented |
| 12:27.21 | tzafrir | HonestWorker, zaptel.conf is what you write |
| 12:27.31 | tzafrir | T1 has no 32 channels :-) |
| 12:27.38 | HonestWorker | its e1 |
| 12:27.42 | HonestWorker | t1 has 24 |
| 12:27.52 | tzafrir | and it has 30 B channels if you use euroisdn |
| 12:28.03 | [TK]D-Fender | ~e1 |
| 12:28.05 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 12:28.05 | HonestWorker | yes, I do have a e1 |
| 12:28.06 | [TK]D-Fender | ^^^^ |
| 12:28.07 | HonestWorker | I am brazilian |
| 12:28.08 | Dr-Linux|home | [TK]D-Fender: that what all use. So there is not way to save CLI> console log into a file? |
| 12:28.11 | [TK]D-Fender | 31 <- |
| 12:28.16 | HonestWorker | there is no such thing as a t1 digital hierarchy over here |
| 12:28.18 | [TK]D-Fender | 20B + D |
| 12:28.21 | [TK]D-Fender | 30* |
| 12:28.32 | [TK]D-Fender | HonestWorker: Ok, just stop already. |
| 12:28.52 | HonestWorker | Ok, I am standing by. What are your instructions? |
| 12:29.09 | [TK]D-Fender | HonestWorker: I've already asked twice. Go scroll up and read. |
| 12:30.07 | HonestWorker | I am not going to paste my company's information as pastebin when that is not relevant. I have supplied the significant directives. I am not gonna chance the signalling of a link that is working 100% fine. If that sinalling was not supported by the other endpoint the link wouldnt be operational. |
| 12:30.28 | HonestWorker | Before you curse me, I will preffer to standy by for other gentleman to assist me. |
| 12:30.45 | HonestWorker | chance=change |
| 12:30.55 | [TK]D-Fender | HonestWorker: Show us what you're doing or we can't help you. You are the one wasteing time. |
| 12:31.08 | HonestWorker | Here is what I am doing: |
| 12:33.33 | *** join/#asterisk int21 (n=IceChat7@189.104.12.150) |
| 12:34.30 | HonestWorker | http://pastebin.com/d34519dd2 |
| 12:35.49 | HonestWorker | If the call originates from the SIP channel defined as 235 it evaluates to true and send the flow to priority 6 which sets the proper CALLERID |
| 12:35.51 | [TK]D-Fender | HonestWorker: Feel free to ask again when you're ready to provide what's requested of you. You don't seem to be ready to solve your problem currently. |
| 12:36.35 | HonestWorker | [TK]D-Fender, ok, I appreciate your help. I will stand by for other gentlemen to assist me |
| 12:36.53 | [TK]D-Fender | HonestWorker: Nobody can help you with what you've give so far |
| 12:37.07 | HonestWorker | (just to make it clear, I have provided the relevant information related to my link signalling as demanded) |
| 12:37.32 | [TK]D-Fender | HonestWorker: I asked you to show the call with PRI debug. What is the problem in doing so? |
| 12:37.44 | HonestWorker | I don't know how to do that |
| 12:37.59 | HonestWorker | Could you provide me instructions, please? |
| 12:38.05 | [TK]D-Fender | HonestWorker: When your mechanic asks to look under the hood to fix your engine starting probelm, stop trying to give him a guided tour of your trunk. |
| 12:38.23 | [TK]D-Fender | HonestWorker: Go to CLI, enable PRI debug and PASTEBIN THE CALL ATTEMPT |
| 12:38.35 | HonestWorker | ok |
| 12:40.06 | *** join/#asterisk kamui (n=kamui@wsip-66-210-239-81.no.no.cox.net) |
| 12:40.51 | Dr-Linux|home | Asterisk support DS3 cards? |
| 12:42.00 | HonestWorker | I had to change the command to pri debug span 1 |
| 12:42.19 | [TK]D-Fender | Dr-Linux|home: As the direct interface for TDM, no. |
| 12:42.43 | Dr-Linux|home | [TK]D-Fender: in what case yes |
| 12:43.04 | [TK]D-Fender | Dr-Linux|home: Never directly |
| 12:43.50 | Dr-Linux|home | [TK]D-Fender: DS3 line is from our provider |
| 12:44.19 | [TK]D-Fender | Dr-Linux|home: Just stop already. If you have a DS3>SIP gateway, then * can use that, but there is no card that * can manage |
| 12:44.41 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:44.41 | [TK]D-Fender | Dr-Linux|home: And where else would the DS3 be coming from other than your "provider"? |
| 12:45.07 | dominic1 | how can I see which channels was dialed on a hangup |
| 12:45.20 | dominic1 | with ${CHANNEL} I alwas see the incoming channel |
| 12:46.06 | [TK]D-Fender | dominic1: Go set a variable during your call |
| 12:46.35 | dominic1 | with a maro in the dialcommand? |
| 12:47.00 | [TK]D-Fender | dominic1: SOMEWHERE |
| 12:47.02 | dominic1 | how can I read out the destinationchannel? |
| 12:47.07 | dominic1 | BRIDGEPEER? |
| 12:48.45 | *** join/#asterisk Dr-Linux|work (n=Nothing@117.20.21.66) |
| 12:50.30 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 12:55.19 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 12:55.26 | ghenry | Hi |
| 12:55.26 | HonestWorker | [TK]D-Fender, may I private message? |
| 12:55.31 | ghenry | What does "Really destroying SIP dialog" mean? |
| 12:55.49 | [TK]D-Fender | HonestWorker: Why? |
| 12:56.21 | [TK]D-Fender | HonestWorker: Nobody cares what your # is and who you are calling. |
| 12:56.22 | HonestWorker | Because I would rather send the pri debug information directly to you instead of giving out the url to everybody. |
| 12:56.54 | HonestWorker | You can't really answer for everybody. |
| 12:57.45 | [TK]D-Fender | HonestWorker: Do you want to be more paranoid and unproductive? |
| 12:58.02 | [TK]D-Fender | HonestWorker: I'm sure its possible. |
| 12:58.26 | HonestWorker | Ok, you are right. I am gonna edit the zapata.conf and try to sort that out . Thank you for your time. |
| 12:58.57 | [TK]D-Fender | HonestWorker: FFS. Just PM the damn link |
| 12:59.18 | HonestWorker | I realize I must to supply information needed but at t he same time I am a information technology professional who is expected to take good care of my company's information security. |
| 13:00.21 | HonestWorker | [TK]D-Fender, its alright, I appreciate your attention and time. I dont think the debug would be informative as it didnt have any telephone numbers on it. I believe I didnt gather the information you wanted. |
| 13:00.23 | [TK]D-Fender | HonestWorker: Just send the link |
| 13:01.06 | dominic1 | CDR(dstchannel) worked for me! |
| 13:01.20 | [TK]D-Fender | HonestWorker: Ok, you have just wasted half an hour on 1 stupid pastebin. |
| 13:01.45 | [TK]D-Fender | HonestWorker: pastebin the ENTIRE CALL. Not some tiny clipping. from beginning to end. |
| 13:02.06 | HonestWorker | Yeah, so I have realized |
| 13:02.15 | [TK]D-Fender | HonestWorker: And the zapata.conf & zaptel.conf I asked for over half and hour ago. |
| 13:02.26 | HonestWorker | I did set verbose to 1 and ran the debug but there were still lots of outputs |
| 13:02.48 | HonestWorker | how to I save the debug output to a file? |
| 13:02.50 | [TK]D-Fender | HonestWorker: verbose 10 <- |
| 13:02.56 | [TK]D-Fender | HonestWorker: Cut & paste. |
| 13:02.56 | kamui | might be a stupid question here, but I just configured asterisk yesterday for the first time, just basic sip. Im trying to connect my N95 to the sip server (dmz on my network) and Im getting a registration failed on the client. However, on the asterisk cli (-rvvvv) I get this: -- Registered SIP '1000' at xxx.xxx.239.81 port 31958 expires 3600 |
| 13:03.02 | kamui | what does this mean? |
| 13:03.12 | kamui | did it connect successfully or not? |
| 13:04.08 | kamui | thats the account I set up for the N95, and I have no other clients configured |
| 13:04.17 | [TK]D-Fender | kamui: pastebin the complete SIP debug of the regitration attempt. |
| 13:04.36 | kamui | [TK]D-Fender: mind telling me how to get that |
| 13:04.49 | kamui | Im on chapter 4 of the oreileys book :) |
| 13:04.56 | [TK]D-Fender | kamui: go to * CLI, and "sip debug". |
| 13:05.01 | HonestWorker | [TK]D-Fender, my terminal history will exhaust . I think it would be better if I could redirect the output to a file |
| 13:05.08 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.121) |
| 13:05.30 | [TK]D-Fender | HonestWorker: Get a bigger buffer. |
| 13:05.55 | *** join/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 13:06.13 | *** part/#asterisk dawebber (n=dawebber@209.136.161.135) |
| 13:07.23 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 13:08.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 13:09.50 | cesar_CR | hello guys how can I see if I am reciving the callerid from a zap channel in the cli ? |
| 13:10.15 | penguinFunk | core set verbose 40 |
| 13:10.25 | [TK]D-Fender | penguinFunk: Healthy :) |
| 13:10.46 | [TK]D-Fender | cesar_CR: "core show applicaiton NoOp" |
| 13:10.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:12.28 | *** join/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 13:12.42 | HonestWorker | [TK]D-Fender, I think the conditional branching isnt evaluating as it should(I must to fix that before anything). Are you sure that setting CALLERID variable is enough to send a specific number |
| 13:12.50 | HonestWorker | to a receiver on a PSTN network ? |
| 13:13.02 | [TK]D-Fender | HonestWorker: Please provide what was requested. |
| 13:13.55 | harryjr | doesn't RECORD() return anything? something like DIAL_STATUS? |
| 13:13.58 | HonestWorker | Its ok, Fender, I must to fix the dialplan logic before. I'd just like to know the Theory. Is the CALLERID set going to be sent to the receiver on a PSTN ? |
| 13:14.04 | [TK]D-Fender | HonestWorker: and no, ":" is not "=" |
| 13:14.09 | cesar_CR | [TK]D-Fender, thanks |
| 13:14.24 | HonestWorker | Isnt DIALSTATUS for the DIAL() application ? |
| 13:14.25 | *** join/#asterisk steliosk (n=Stelios@athedsl-394773.home.otenet.gr) |
| 13:14.30 | harryjr | yep. |
| 13:14.40 | [TK]D-Fender | HonestWorker: Stop wasting time, go fix your other little error, and come back with what was requested. |
| 13:14.57 | [TK]D-Fender | HonestWorker>Isnt DIALSTATUS for the DIAL() application ? <- No, it isn't |
| 13:15.07 | [TK]D-Fender | harryjr: No. |
| 13:15.10 | HonestWorker | what is the ':' operator for? substring matching ? |
| 13:15.30 | *** part/#asterisk harryjr (n=harry@67-207-147-205.slicehost.net) |
| 13:15.32 | [TK]D-Fender | HonestWorker: Go read the chapter on "Asterisk Evaluations" again |
| 13:16.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:16.15 | HonestWorker | I am reading the applications reference |
| 13:16.23 | HonestWorker | page 398 GotoIf() |
| 13:17.05 | [TK]D-Fender | HonestWorker: Don't. |
| 13:17.25 | [TK]D-Fender | HonestWorker: Go lookup "asterisk expression" on the WIKI and READ |
| 13:17.28 | [TK]D-Fender | ~wikis |
| 13:17.29 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:17.56 | [TK]D-Fender | HonestWorker: http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 13:18.41 | *** join/#asterisk [intra]lanman (n=lanman@va-71-0-90-168.dyn.embarqhsd.net) |
| 13:19.02 | HonestWorker | [TK]D-Fender, :P I am trying to justify the money I spent importing that book |
| 13:19.24 | *** join/#asterisk LemensTS (n=matthew@adsl-75-42-148-10.dsl.stlsmo.sbcglobal.net) |
| 13:19.29 | [TK]D-Fender | HonestWorker: No, you are trying to waste a nuce round hour of my time minimum. |
| 13:19.34 | [TK]D-Fender | nice* |
| 13:19.59 | HonestWorker | Oh, c'on. Dont be that nervous. Its not healthy |
| 13:20.22 | LemensTS | Everytime I install freePBX on top of asterisk, it will not let me view webpage folders until I chmod them to 777. I must be doing something wrong in the installation, ive tried it 3 times now |
| 13:20.38 | [TK]D-Fender | HonestWorker: And you haven't been paying attention. I already gave you the answer. |
| 13:20.50 | HonestWorker | I did |
| 13:20.53 | LemensTS | *when i mean webpage folders, i mean like html/folder/page.html |
| 13:20.57 | HonestWorker | I should use the '=' operation instead |
| 13:23.03 | kamui | [TK]D-Fender: http://pastebin.com/d795d9f4c |
| 13:23.07 | kamui | theres the output |
| 13:23.13 | [TK]D-Fender | ~freepbx |
| 13:23.14 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 13:24.19 | LemensTS | TKD-Fender: i searched for #freepbx before and it didnt come up, heh. But it is now wierd. Thanks |
| 13:24.33 | [TK]D-Fender | kamui: Ther are all "SIP/2.0 401 Unauthorized" and "Contact: <sip:1000@192.168.1.2>" shows me you have not set up * properly to work behind NAT. |
| 13:24.39 | [TK]D-Fender | kamui: Go read the guide : |
| 13:24.41 | [TK]D-Fender | ~sipnat |
| 13:24.42 | 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 |
| 13:24.59 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:24.59 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:25.03 | [TK]D-Fender | LemensTS: This is not 2nd level support for FreePBX. |
| 13:25.04 | kamui | [TK]D-Fender: thanks I was worried that it was nat related, thats like a whole extra chapter and I still dont have a grasp on this yet |
| 13:25.41 | [TK]D-Fender | kamui: Your auth is wrong so its failing, but * is also failing to return the failure notive properly as well. |
| 13:26.20 | [TK]D-Fender | kamui: So fix your NAT issue first and then once communication is at least functioning properly you can work on the fact you don't like what its saying :) |
| 13:30.52 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:31.25 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 13:32.31 | *** join/#asterisk moy (n=moy@nat/ibm/x-0c805d446be410e2) |
| 13:32.53 | dominic1 | is it possible to see from which context the call jumped to exten hangup? |
| 13:33.07 | dominic1 | from_context or something like that? |
| 13:33.12 | [TK]D-Fender | dominic1: Its int he current context. |
| 13:33.36 | dominic1 | sorry not context extension |
| 13:34.40 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 13:35.30 | [TK]D-Fender | dominic1: go read CHANNELVARIABLES.TXT |
| 13:35.46 | *** join/#asterisk jm|home (n=jm|home@dilbert.jamiem.com) |
| 13:37.17 | kamui | [TK]D-Fender: thanks, Im gonna read your guide now |
| 13:41.21 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 13:41.25 | UnixDog | morning |
| 13:41.36 | dominic1 | there is no variable from_extension |
| 13:41.41 | UnixDog | well I think I about have the new bsd port for 1.6 done |
| 13:41.58 | UnixDog | doing a test build now |
| 13:42.26 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:42.26 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:43.20 | [TK]D-Fender | dominic1: Then odds are you can't get your answer. |
| 13:44.41 | *** join/#asterisk AlexTO (n=alex@75.149.245.109) |
| 13:45.11 | kamui | [TK]D-Fender: just dawned on me. Is it the client I need to worry about? or the server? because * is running on a NAS that IS the dmz... |
| 13:46.06 | [TK]D-Fender | kamui: DMZ is overkill in the forwarding dept, and is not enough . * needs to tell the other side where to send responses to. thats the Contact: hearder which is sending private addresses because it doesn't know any better |
| 13:46.16 | [TK]D-Fender | kamui: so YES, you need to fix this. |
| 13:46.20 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 13:47.24 | *** join/#asterisk ToTo (n=ToTo@207.176.6.186) |
| 13:47.58 | kamui | [TK]D-Fender: thanks, I understand now. The addresses being sent back to my device are internal because * doesn't know my external address, or my dynamic hostname & domain |
| 13:48.42 | [TK]D-Fender | kamui: So go read the guide and fix your sip.conf |
| 13:51.10 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 13:53.57 | kamui | [TK]D-Fender: since my ip changes moderately frequently, can I set the externip= to my dyndns hostname? |
| 13:54.22 | [TK]D-Fender | kamui: No, but there is another parameter you can use. Go look a bit for it. |
| 13:54.44 | kamui | on it |
| 13:55.45 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:58.18 | dominic1 | ${CDR(dst)} ist the solution to get the Information of the context which was dialed before hangup |
| 13:58.30 | *** join/#asterisk ManxPower (n=manxpowe@108.sub-75-202-107.myvzw.com) |
| 13:58.49 | dominic1 | do you know a variable for the uniqueid of the dialed channel? |
| 13:59.31 | *** join/#asterisk bkw__ (n=brian@70.234.182.169) |
| 13:59.53 | [TK]D-Fender | dominic1: "core show application dial" |
| 14:02.33 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 14:02.36 | hsv-al | hello fellow internet addicts |
| 14:02.48 | hsv-al | are we all looking forward to another long & glorious weekend of irc? :) |
| 14:02.57 | dominic1 | can you give me a more detailed hint |
| 14:03.34 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 14:03.42 | dominic1 | please |
| 14:04.34 | hsv-al | heh, apple is ripping off * |
| 14:04.42 | hsv-al | their releasing AGI - apple gateway interface, same concept |
| 14:04.48 | hsv-al | normal languages development for iphone |
| 14:04.59 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:05.28 | *** part/#asterisk korihor (n=korihor@190.39.163.45) |
| 14:05.56 | [TK]D-Fender | dominic1: go read channelvariables.txt again. |
| 14:07.30 | x86 | gah, I've got a hunt group of 4 CO lines coming into an FXO channel bank, and randomly they seem to stop accepting inbound calls |
| 14:07.45 | hsv-al | since when did x86 make the transition from |
| 14:07.50 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:07.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:07.52 | hsv-al | "general networking", to asterisk focused learning :) |
| 14:08.11 | x86 | hsv-al: I've been doing Asterisk stuffs for ~4 years |
| 14:08.23 | hsv-al | well the past 5 years ive just seen you mainly chat in |
| 14:08.24 | hsv-al | #networking |
| 14:08.27 | hsv-al | #cisco etc |
| 14:08.39 | [TK]D-Fender | x86: And still can't let go of his silly channel banks :) |
| 14:09.00 | *** join/#asterisk DragonBall-Z (n=Adnan@202.133.78.60) |
| 14:09.24 | coppice | do channel banks suffer from the sub-prime crisis? |
| 14:09.26 | x86 | most of the time when inbound calls aren't working, all the lines are in a dialplan loop. I took out the loop (waitexten, t,1,Goto(foo,s,1)), and now none of the channels are in use at all on Asterisk, but still inbound calls just ring and ring but never show ring state on Asterisk |
| 14:09.56 | x86 | [TK]D-Fender: normally I would use a 4-port POTS card, but I had a spare T1 port and channel bank, so I figured I'd just use that instead of spending more money |
| 14:10.04 | coppice | I bet they are seeing falling interest |
| 14:10.14 | x86 | coppice: hah |
| 14:11.15 | [TK]D-Fender | coppice: Channel banks are all robbed-bit signalling... |
| 14:11.40 | dominic1 | is it possible that I can only read out the uniqueid of the inboundchannel? I only find information about uniqueid and CDR(uniqueid) both are for the inbound channel, not the connection to the internal channel |
| 14:12.47 | coppice | most, but not all |
| 14:12.47 | x86 | [TK]D-Fender: so any idea why the channels would not be in use, but when you call them they don't ring asterisk? |
| 14:13.55 | [TK]D-Fender | x86: anything on debug? |
| 14:14.22 | [TK]D-Fender | dominic1: Read the file closer, and stop centering on those 2 stupid values. |
| 14:15.02 | x86 | [TK]D-Fender: hmm, lemme check |
| 14:22.45 | tzanger | is there a way to turn off the "check timer" feature of chan_zap on startup? |
| 14:22.57 | tzanger | i.e. just don't load chan_zap.so if the timer doesn't work, instead of killing off asterisk entirely? |
| 14:23.45 | kamui | yes! |
| 14:24.01 | kamui | [TK]D-Fender: ok, that nat problem looks to be fixed, now im having this auth mismatch problem |
| 14:24.15 | tzafrir | tzanger, a sinple source patch? |
| 14:24.25 | tzanger | tzafrir: ys that will certainly work |
| 14:24.29 | [TK]D-Fender | kamui: go look at your peer, and waht you put in your phone. |
| 14:24.32 | tzanger | was hoping there was already a config option to turn it off |
| 14:24.44 | tzanger | dont_crash_on_bad_timer => yes or something |
| 14:24.49 | kamui | [TK]D-Fender: the [ ] header for the phone doesn't have to match the username= field does it? |
| 14:24.51 | tzafrir | tzanger, but what is the timing source you actually use? |
| 14:25.01 | kamui | I thought that could be anything I wanted as a discriptor |
| 14:25.11 | [TK]D-Fender | kamui: pastebin the new sip debug and your peer. |
| 14:25.17 | [TK]D-Fender | (masking only the password) |
| 14:25.29 | kamui | ok |
| 14:25.49 | DragonBall-Z | hello all we are running asterisk 1.2.26 today we receive this error continuously and the CPU goes to 100% for quick fix i remove the notice entry in logger.conf from messages entry error is "Jul 25 09:52:55 NOTICE[32273]: callerid.c:373 callerid_feed: Unknown IE 0" |
| 14:25.53 | *** join/#asterisk Alpha_AI (n=Ben@d122-109-17-74.rdl14.qld.optusnet.com.au) |
| 14:27.18 | *** join/#asterisk cplx (n=cplx@ettamo.lnk.telstra.net) |
| 14:27.35 | cplx | hi guys.. i'm running CME (call manager express) and trying to auth with a Asterisk box |
| 14:27.42 | cplx | getting the following debug msgs: |
| 14:28.00 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:28.35 | cplx | http://pastebin.org/57006 |
| 14:28.46 | cplx | SIP/2.0 603 Declined |
| 14:29.25 | [T]ank | i am trying to get a sip phone to work through my firewall. I am using smoothwall express. Asterisk server is behind the firewall and the sip phone is outside. I have opened ports 5060 tcp and udp as well as 10000-10010 udp for rtp. The phone registers, and I can dial its telephone number and make it ring, however I am getting no sound. Where can I go from here? |
| 14:29.58 | cplx | anyone? |
| 14:30.21 | [TK]D-Fender | cplx: Do another pastebin with * SIP debug |
| 14:30.31 | dominic1 | [TK]D-Fender: I can not find the right function or variable. Can you please help me again? |
| 14:30.35 | [TK]D-Fender | [T]ank: Go read the guide : |
| 14:30.37 | [TK]D-Fender | ~sipnat |
| 14:30.37 | 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 |
| 14:30.39 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:31.15 | [T]ank | headed there... thank you |
| 14:31.38 | *** join/#asterisk mikeshank (n=sam@c-68-37-250-134.hsd1.pa.comcast.net) |
| 14:31.41 | [TK]D-Fender | dominic1: try every damn variable in that doc if you have to. You don't seem to have been inspired by the names or descriptions in there so I guess you'll just have to forego that and jsut test them ALL |
| 14:31.53 | cplx | [TK]D-Fender; that's all I can get at the moment from the CME box - getting those debugs when i try an outbound call |
| 14:32.19 | [TK]D-Fender | cplx: I just get the debug from ASTERISK CLI, not CME. |
| 14:32.59 | cplx | [TK]D-Fender - i'm trying to connect to a ITSP (they are the ones running the Asterisk box) |
| 14:33.10 | cplx | [TK]D-Fender; thats whats giving me the SIP/2.0 603 Declined |
| 14:33.32 | cplx | [TK]D-Fender - they are out of support hours at the moment, so i can't contact them.. that's why im here :P |
| 14:33.35 | [TK]D-Fender | cplx: And like I said twice get the comprehensive SIP debug from THAT BOX |
| 14:33.43 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 14:33.48 | [TK]D-Fender | cplx: Then you're running blind. |
| 14:34.09 | cplx | [TK]D-Fender - any quick ideas that might be the issue? or no? |
| 14:34.33 | kamui | [TK]D-Fender: http://pastebin.com/d1cb3b73e |
| 14:34.36 | cplx | [TK]D-Fender - the 603 Declined error could be many reasons causing that? |
| 14:34.41 | *** join/#asterisk spokra (n=spokra@host093-179-153.sea0.speakeasy.net) |
| 14:34.44 | kamui | you're going to love this, Im geting a ton of those errors every few seconds |
| 14:34.47 | [TK]D-Fender | cplx: Yes |
| 14:35.01 | dominic1 | I still think I cannot get the uniqueid of another channel and there is nothin in the document explaining something different. I asked cause I thought anybody of you know a different solution. I read this document thousand times and tested so many variables you can't imagine |
| 14:35.12 | cplx | [TK]D-Fender - anything that it most likely is? according to that debug output? |
| 14:35.54 | [TK]D-Fender | cplx: Sorry, we cannot help you with so little. |
| 14:36.22 | kamui | [TK]D-Fender: http://pastebin.com/d377be73a (here's everything that I could paste in the buffer, came out in less than 3 minutes) |
| 14:36.30 | [TK]D-Fender | dominic1: The Dial() application uses the following variables: |
| 14:37.41 | [TK]D-Fender | kamui: Is 68.225.79.9 indeed the proper IP to your * box, and what have you forwarded to it exactly? |
| 14:38.36 | tzanger | tzafrir: zaptel tdmoe, but if the link isn't up, it shouldn't kill asterisk, it should just prevent the loading of chan_zap |
| 14:38.45 | tzanger | the same as when I don't have a T1 card in but tell asteirsk I do |
| 14:38.49 | *** join/#asterisk pbrown985 (n=na@wh-gtw-0001.woolfharris.com) |
| 14:38.52 | tzanger | it complains, refuses to load chan_zap, and continues |
| 14:39.24 | tzafrir | tzanger, the issues in question were general timing issues (problems with playback). Not issues with chan_zap |
| 14:39.28 | kamui | [TK]D-Fender: it is indeed my router address |
| 14:40.29 | tzanger | tzafrir: if I ahve a zapless system, why does it still work? |
| 14:40.40 | tzafrir | What would it take, then, to re-start timing if it is still in a bad state? |
| 14:41.01 | tzafrir | If there's no /dev/zap/pseudo, you won't get that message |
| 14:41.16 | tzanger | tzafrir: that's not the point; if I normally have a t1 card in there and I rip it out, asterisk will still load and work, although anything requiring timing wont', obviously. but hte PBX works. |
| 14:41.56 | tzafrir | tzanger, the point is that the symptoms were not limited to chan_zap. Hence the fix you suggest won't help. |
| 14:42.06 | DragonBall-Z | can anyone shed some light on it 'Jul 25 09:52:55 NOTICE[32273]: callerid.c:373 callerid_feed: Unknown IE 0' |
| 14:42.08 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 14:42.24 | tzafrir | That is not to say I like crashing Asterisk. But please sugest a better fix :-( |
| 14:42.41 | tzafrir | IE >= 5.5 is required? |
| 14:43.15 | dominic1 | fender: do you read another channelvariables than I. Where is there a DIALEDPEERUNIQUEID or something like that? |
| 14:43.47 | kamui | [TK]D-Fender: its forwarded to to 192.168.1.2 |
| 14:44.16 | tzafrir | DragonBall-Z, Not really sure. But what version of Asterisk? Talking to what on the other side? |
| 14:44.34 | mikeshank | hi all. not sure this is the right place to ask, I'm trying to start festival with festival_server and get command not found, festival is installed and i can start it with festival --server. festival_server doesnt seem to exist in my install, how would I get it? |
| 14:45.58 | DragonBall-Z | asterisk 1.2.26 OS debian etch TDM2400 FXO connected with PSTN lines |
| 14:49.09 | Dr-Linux|work | does asterisk support PAI or the RPID ? |
| 14:50.11 | *** join/#asterisk Hastalavi (n=kumar@mail.netvita.com) |
| 14:50.26 | Hastalavi | Hi |
| 14:50.38 | [TK]D-Fender | kamui: Details please. |
| 14:50.49 | Hastalavi | can anyone suggest a server machine to use asterisk and TE407P card ? |
| 14:51.04 | UnixDog | any pc should do |
| 14:52.37 | [TK]D-Fender | dominic1: Where do you get this impression that doing a dial creates another UNIQUEID? |
| 14:52.49 | tzafrir | DragonBall-Z, "IE" can come from e.g. IAX or PRI. Do you have any IAX connections? |
| 14:53.42 | [TK]D-Fender | dominic1: if I dial Zap/1/123456, the other side doe not have a uniqueid. That is something tied to the concept of CDR |
| 14:53.44 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:54.05 | [TK]D-Fender | dominic1: the other side isn't placing a call so it has nothing to record. |
| 14:54.10 | kamui | [TK]D-Fender: ok, my external address is 68.225.79.9, and my internal address is 192.168.1.2, which is also the DMZ. Im not sure why it appears to be connecting but my sip client continues to say registration failed. The client is also behind a nat on a different wireless network. Do I need to set up a proxy? |
| 14:54.29 | kamui | and then there is the mysterious retransmitting issue |
| 14:54.30 | DragonBall-Z | nope no IAX or PRI connections call came from analog lines(POTS) |
| 14:55.04 | [TK]D-Fender | kamui: Not sure at this point. |
| 14:55.36 | Dr-Linux|work | <PROTECTED> |
| 14:56.31 | kamui | [TK]D-Fender: ok, well Ill keep hammering away at it. |
| 14:56.40 | kamui | im much closer than I was yesterday |
| 14:56.47 | [TK]D-Fender | Dr-Linux|work: http://www.google.ca/search?hl=en&q=asterisk+rpid+support&btnG=Google+Search&meta= |
| 14:56.54 | kamui | I can get local clients to connect, just not remote clients it seems |
| 14:57.07 | kamui | and by local clients, only xlite has worked so far |
| 14:57.54 | *** join/#asterisk GhOnDiE (n=ghondie@dsl78-143-210-38.in-addr.fast.co.uk) |
| 14:57.59 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 14:58.49 | [TK]D-Fender | kamui: You're trying to get a Nokia to work remotely, right? |
| 14:59.02 | kamui | yes |
| 14:59.22 | kamui | at this point, Im going to try a regular client to make sure its not just a problem with the phone |
| 14:59.48 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:00.24 | *** join/#asterisk courchea (n=someone@office.prival.ca) |
| 15:00.35 | courchea | Hi, anyone can enlight me to the status or SRTP and SIP over TLS ? |
| 15:00.45 | [TK]D-Fender | kamui: While I don't know the specifics, this should be Google-able. I think Iv've heard something to the effect that they don't work too well behind NAT. Go take a look |
| 15:00.59 | dominic1 | the management tells me a uniquedid for my inbound ISDN Channel connection and a uniqueid for the connection to the sip device. I get a inbound connection via isdn, this connection has a newchannel event with a uniqueid. This call goes to the context isdn_incoming. There is a dial command to my internal device. It throws a newchannel and with another uniqueid than the inbound isdn uniqueid. If I echo uniquedid I get the uniquedid of the inbound isdn channe |
| 15:01.52 | [TK]D-Fender | courchea: http://bugs.digium.com/view.php?id=5413 |
| 15:02.35 | courchea | Hi TK, yup found that one, so it is still dev work? Will be included in 1.6 I guess? |
| 15:03.06 | [TK]D-Fender | courchea: Maybe. Dunno, I think I heard that 1.6.0 is feature-locked. I may be mistaken however. |
| 15:03.49 | courchea | Ok. I'll e-mail the asterisk-dev list. the asterisk-dev channel is pretty much unresponsive to my question... |
| 15:04.03 | [TK]D-Fender | courchea: Mind you the 1.6 dev cycle is changing and will include more regular feature-add updates |
| 15:04.07 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:04.21 | dominic1 | ctooley from -dev helped me: http://bugs.digium.com/view.php?id=11816 |
| 15:05.02 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
| 15:05.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 15:06.35 | [TK]D-Fender | dominic1: OK. |
| 15:08.32 | kamui | [TK]D-Fender: should sip show registry show me connected clients? |
| 15:08.50 | kamui | because wengophone says its online, but I can't confirm it |
| 15:08.53 | [TK]D-Fender | kamui: No, that shows what * is registered to. |
| 15:09.01 | [TK]D-Fender | kamui: "sip show peers" |
| 15:09.09 | kamui | damn, I tried that too |
| 15:09.10 | kamui | shows nothing |
| 15:09.12 | [TK]D-Fender | kamui: "sip show peer[peerwithoutbraces]" |
| 15:10.11 | *** join/#asterisk korihor (n=korihor@190.199.171.145) |
| 15:10.14 | kamui | [TK]D-Fender: I can see all my accounts. they all say unmonitored, except the nokia which says unknown |
| 15:10.37 | [TK]D-Fender | kamui: "qualify=yes <- all peers should have |
| 15:10.51 | kamui | [TK]D-Fender: however gizmo and the wengo phone show ip's |
| 15:11.08 | kamui | [TK]D-Fender: ok let me fix that, btw, when I do a dialplan reload I get an error, no such command |
| 15:11.41 | [TK]D-Fender | kamui: what version of *? |
| 15:12.29 | kamui | 1.2.13 |
| 15:15.49 | [TK]D-Fender | kamui: thats why |
| 15:15.59 | [TK]D-Fender | kamui: that format wasn't introduced until 1.4 IIRC |
| 15:16.07 | [TK]D-Fender | kamui: just "reload" |
| 15:16.39 | [TK]D-Fender | kamui: Aside from the fact that's 1.2, its not even anywhere near the latest version within that family. |
| 15:17.15 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 15:17.20 | kamui | I see, I should probably apt remove this version and download the latest source then |
| 15:17.52 | kamui | just takes ages to build stuff on this nas with its slow arm cpu |
| 15:22.10 | Hastalavi | Can asterisk recognise 32 bit or 64 bit Linux and can work accordingly ?? |
| 15:22.51 | courchea | FYI From Russell: |
| 15:22.51 | courchea | SIP over TLS is supported in Asterisk trunk / 1.6, but is currently |
| 15:22.51 | courchea | marked as "experimental" |
| 15:23.01 | courchea | SRTP has not yet been included in Asterisk trunk. |
| 15:24.41 | damjan | Hastalavi: what? |
| 15:25.07 | damjan | Hastalavi: if you compile from source it will be compiled acording to the platform |
| 15:29.05 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:29.06 | *** join/#asterisk jameswf-mac (n=jameswf-@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:30.01 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:30.17 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:31.26 | *** join/#asterisk thansen|laptop (n=thansen@160.7.243.19) |
| 15:36.01 | *** join/#asterisk jpcansa (n=jpbenavi@201.237.76.66) |
| 15:40.18 | *** join/#asterisk l2cache (n=l2cache@179.190.204.68.cfl.res.rr.com) |
| 15:41.29 | *** join/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 15:41.30 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:42.02 | l2cache | Is there any wildcards you can use when doing IF statements in the dialplan? For example, if my customVAR has data in the middle that I want to match. Kind of like mysql 'where field like '%data$' |
| 15:42.11 | *** join/#asterisk dverzolla (n=dverzoll@proxynet.fcl.com.br) |
| 15:42.27 | *** join/#asterisk Assimilate (n=Assimila@216.83.78.108) |
| 15:42.48 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 15:44.10 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 15:44.45 | [TK]D-Fender | l2cache: "core show function REGEX" |
| 15:44.46 | dverzolla | The asterisk manager interface is limited? |
| 15:47.28 | kamui | [TK]D-Fender: this is a good sign. I've been able to successfully connect my sip server to sipphone.com |
| 15:47.44 | kamui | [TK]D-Fender: I think my problem is still in the NAT configuration for my clients, so Im going to do more reading |
| 15:47.50 | [TK]D-Fender | dverzolla: No, you just need to do "load res_omnipotence.so" and you can conquer entire nations with it. |
| 15:48.16 | dverzolla | [TK]D-Fender: :D |
| 15:51.00 | jeev | pump up the jam |
| 15:51.06 | jeev | fender, my 330's are being delivered today! |
| 15:52.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:53.58 | kamui | [TK]D-Fender: finally, it looks like I gotta use a stun server, which appears to be just ssh tunnelling |
| 15:54.51 | *** join/#asterisk Penggu (n=penggu@220-245-200-87.static.tpgi.com.au) |
| 15:55.25 | Penggu | hi all. if an extension is matched, and i only want to set a variable... and let further extensions continue to be matched.... how can it be done? |
| 15:56.08 | Penggu | eg. context [publicphone], exten => _X.,1,Set(callingFromPublicPhone=1), exten => _X.,2,<keep going to other extensions further on> |
| 15:56.28 | Penggu | and then i might have an include => or something to other contexts |
| 15:56.43 | *** join/#asterisk jjshoe_ (n=jjshoe@72.37.252.50) |
| 15:56.49 | Penggu | where checking for callingFromPublicPhone variable might occur |
| 15:57.39 | *** join/#asterisk remibemol (n=remibemo@81-66-205-202.rev.numericable.fr) |
| 15:58.27 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 16:00.27 | *** join/#asterisk Magnus_H (n=magnus@90-228-242-54-no71.tbcn.telia.com) |
| 16:01.09 | *** part/#asterisk Magnus_H (n=magnus@90-228-242-54-no71.tbcn.telia.com) |
| 16:03.54 | *** join/#asterisk geek_cl (n=geek@190.54.42.62) |
| 16:08.51 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.95.130) |
| 16:09.23 | geek_cl | hi all, somebody run asterisk on a alix.2 ?? |
| 16:10.16 | [TK]D-Fender | geek_cl: Yes, I've heard of several |
| 16:11.00 | geek_cl | anyone already? |
| 16:11.23 | l2cache | Does anyone know a good way to get how many seconds that extensions are on a call. I did a script that gets the information from the 'Status' command. But wondering if there is a cleaner way to get the info. |
| 16:11.35 | l2cache | vi AMI |
| 16:12.15 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:18.44 | *** join/#asterisk alexhopper (n=a27386@mctnnbsa24w-142167059025.pppoe-dynamic.nb.aliant.net) |
| 16:22.51 | *** join/#asterisk horsesgofaster (n=dcantera@pool-71-168-142-212.cmdnnj.east.verizon.net) |
| 16:25.03 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 16:25.03 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:27.08 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 16:27.28 | Kobaz | is there a configurable delay before starting to proces the dialplan when a call comes in |
| 16:27.48 | Kobaz | i have something simple like: |
| 16:27.49 | Kobaz | [services] |
| 16:27.49 | Kobaz | exten => 2400,1,VoiceMailMain() |
| 16:28.07 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:28.12 | [TK]D-Fender | Kobaz: "core show application wait" |
| 16:28.14 | Kobaz | i dial 2400, and i hear "elcome to..."; |
| 16:28.19 | UnixDog | wait() and fille the() with how long in millisec you want to wait |
| 16:28.20 | Kobaz | the beginning is always cut off |
| 16:28.42 | Qwell | "fille the()"? |
| 16:28.43 | Kobaz | yeah but, i shouldn't need to put a wait in front of every sequence |
| 16:29.04 | Kobaz | if i do a wait(1), and then play, that does work, but is there a better way? |
| 16:29.16 | l2cache | Does anyone know a good way to get how many seconds that extensions are on a call. I did a script that gets the information from the 'Status' command via AMI. But wondering if there is a cleaner way to get the info. |
| 16:30.27 | l2cache | Kobaz: You usually want to do an Answer(), then wait(1) before any initial playbacks. Then wait(1) for subsequent playbacks. |
| 16:30.47 | Kobaz | l2cache: this is just local stuff, just an extension in a random context |
| 16:30.48 | [TK]D-Fender | Kobaz: Thats it. |
| 16:30.52 | Kobaz | heh |
| 16:31.17 | [TK]D-Fender | l2cache: What isn't "clean" about your current approach? |
| 16:31.25 | l2cache | No call duration people? |
| 16:31.51 | l2cache | [TK]D-Fender: So you're saying that is the best approach? I am just looking for any other options |
| 16:32.32 | l2cache | I want the least overhead possible. So filtering output from the Status command in AMI is it? |
| 16:32.38 | [TK]D-Fender | l2cache: No, I'm saying I don't know the details of your current method. What aspect of it are you unhappy with? |
| 16:32.46 | [TK]D-Fender | l2cache: Whats the down-side? |
| 16:33.21 | l2cache | Well, I would like to just gather the extensions and duration from AMI. If I run this on a callcenter with 200 phones, the output will be crazy huge. |
| 16:33.54 | l2cache | Because Status returns a huge amount of data. |
| 16:33.55 | [TK]D-Fender | l2cache: "core show channels concise" |
| 16:34.34 | l2cache | you gotta be kidding me |
| 16:34.49 | l2cache | Thank you [TK]D-Fender |
| 16:35.13 | [TK]D-Fender | pushes the "Easy" button |
| 16:35.25 | l2cache | "That was easy" |
| 16:36.28 | jjshoe_ | cube next to me has an easy button hacked with a record your own sound chip |
| 16:36.31 | jjshoe_ | makes for some fun |
| 16:36.40 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:36.43 | *** join/#asterisk mags2 (n=mags2@ampulex.whoi.edu) |
| 16:36.46 | jjshoe_ | 'That was easy chicken fscker!' |
| 16:44.52 | Kobaz | [TK]D-Fender: speaking of timeing... what about when i dial out of a PRI... the behavoir when dialing many numbers is that the first second or half second of audio is always missed, especially when going to soneone's voicemail at verizon or etc |
| 16:48.43 | errr | how can I play a sln audio file? People are saying my moh is way to loud and Im trying to listen to it to see which file it might be but I dont know how to play the sln files.. |
| 16:51.21 | [TK]D-Fender | errr: Where did they come from? |
| 16:51.39 | errr | [TK]D-Fender: I have no idea |
| 16:51.54 | [TK]D-Fender | errr: You have no idea where your MoH files come from? |
| 16:52.13 | errr | [TK]D-Fender: nope, I didnt put them there, someone else did who is no longer with us |
| 16:52.17 | [TK]D-Fender | errr: What are the complainers callin in on? Is the situation constant across all who encounter MoH? |
| 16:52.52 | errr | [TK]D-Fender: no just certian music, it was descirbed to me as sounding like "porn music" |
| 16:53.05 | hardwire | who was it yesterday that wanted notification of a blind tranfer that was failed? |
| 16:53.08 | *** join/#asterisk neurosys (n=neurosys@adsl-067-032-132-060.sip.bct.bellsouth.net) |
| 16:53.48 | errr | [TK]D-Fender: so I just tarred up all the files and put them on my desktop to listen to them find the "porn music" and Ill remove it from the server |
| 16:55.41 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 16:57.55 | rwaite | lol |
| 16:57.59 | rwaite | bow chicka bow wow |
| 16:58.04 | errr | lol |
| 17:02.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:11.05 | WhiteWolf | i have a moh class just for the sleazy porn music, and a feature to instnatly transfer a call to it <.< perfect for unwanted telemarketers ^.^ |
| 17:12.18 | WhiteWolf | s/nat/ant/ |
| 17:12.47 | *** join/#asterisk qdk (n=qdk@195.242.194.41) |
| 17:13.41 | errr | WhiteWolf: lol classic |
| 17:14.15 | UnixDog | clear |
| 17:14.16 | UnixDog | ls |
| 17:14.31 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:15.16 | mags2 | "I can't define porn music but I know it when I hear it" |
| 17:15.17 | *** join/#asterisk Tako-san (n=jmkiffia@24.108.192.144) |
| 17:15.41 | *** join/#asterisk gr0mit (n=tim@lawlm2.plus.com) |
| 17:16.22 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:16.24 | gr0mit | hi - anyone recommend a swiss voip provider? |
| 17:16.36 | gr0mit | need some Basel DID numbers |
| 17:16.44 | *** part/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 17:17.50 | *** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 17:17.58 | WhiteWolf | that was unfortunate |
| 17:18.07 | WhiteWolf | not quite sure how i managed that one |
| 17:22.45 | gr0mit | hmmm - seems no-one can advise on Swiss numbers! |
| 17:23.28 | Katty | hai. |
| 17:26.14 | _ShrikE | Hi Katty! |
| 17:26.29 | Katty | hugs _ShrikE |
| 17:29.15 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:29.29 | jpcansa | hi, any idea why my channels got stucked like this: http://imagebin.ca/view/tj1x8r.html |
| 17:29.45 | jpcansa | this is my extensions.conf: http://pastebin.com/m6230bd85 |
| 17:30.54 | *** join/#asterisk curious101 (i=curious1@119.95.80.169) |
| 17:31.52 | curious101 | hi, i can't seem to find a resource about backing up ang restoring asterisk settings from one system to another. can anyone point me to a specific link? |
| 17:32.04 | [TK]D-Fender | curious101: "man cp |
| 17:32.23 | WhiteWolf | curious101: pretty much only need /etc/asterisk/ or wherever your configs are |
| 17:32.37 | [TK]D-Fender | curious101: And your voicemail folder |
| 17:32.40 | WhiteWolf | perhaps any custom sounds, music, scripts, AGI in a share folder |
| 17:32.59 | kamui | [TK]D-Fender: well, it took me all this time to figure out where the problem was |
| 17:33.13 | curious101 | so all are in the asterisk folder? I see. thanks |
| 17:33.24 | [TK]D-Fender | kamui: What'd you find? |
| 17:34.01 | kamui | [TK]D-Fender: I think I need a proxy server. Im able to connect and register fine via HSDPA |
| 17:34.08 | kamui | but its the internal NAT at work that doesn't work |
| 17:34.43 | kamui | when * tries to send back anything (I guess handshaking) its redirected to my external address at work on a port that wont reforward back |
| 17:34.57 | kamui | I thought thats what the stun server was for, but I must not understand how to properly use it |
| 17:35.23 | kamui | I tried specifying stun.fwdnet.net on the default stun port in my config |
| 17:35.31 | kamui | didn't help |
| 17:35.57 | WhiteWolf | kamui: a fairly common solution is to use static translating on the NAT/PAT firewall/router |
| 17:36.08 | kamui | I can't configure the firewall at work |
| 17:36.22 | WhiteWolf | kamui: sip is pretty hostle if the network isn't configured for it |
| 17:36.28 | WhiteWolf | sip & nat anyway |
| 17:36.33 | curious101 | oh sorry, I didn't mention that the other configuration is from a FreeBSD system. I'm going to restore the configs to a CentOS 5.2 one. Is this okay? |
| 17:37.00 | WhiteWolf | curious101: the config is the same on all oses for a given asterisk version |
| 17:37.09 | WhiteWolf | minor differences, but it's mostly platform independent |
| 17:37.12 | *** part/#asterisk LemensTS (n=matthew@adsl-75-42-148-10.dsl.stlsmo.sbcglobal.net) |
| 17:37.14 | curious101 | I see. One more thing please... |
| 17:37.27 | [TK]D-Fender | curious101: asterisk.conf <- tells you where the other folders are. |
| 17:38.31 | curious101 | ok. thanks for your help, WhiteWolf and [TK]D-Fender. be back later. |
| 17:39.43 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:40.12 | *** join/#asterisk bildo (n=bildoto@bildo.tk) |
| 17:40.44 | WhiteWolf | kamui: if possible, you could try to use IAX |
| 17:41.49 | WhiteWolf | which is inheriently more nat friendly |
| 17:44.18 | WhiteWolf | would be so much better of a quit message if it said clue > 0, since storing a truth value as a varchar is inefficient... |
| 17:45.10 | kamui | WhiteWolf: I wanted to, but my phone doesn't support iax |
| 17:45.53 | WhiteWolf | kamui: =( |
| 17:46.32 | *** join/#asterisk bildo (n=tobbe@bildo.tk) |
| 17:48.12 | kamui | yea, there isn't apparently even a dedicated S60v3 client for IAX, which I would also accept (though the sip integration in S60 is terrific) |
| 17:58.11 | *** join/#asterisk mecheng (n=owenfred@69.12.128.128) |
| 17:59.27 | mecheng | I am experiencing some lag when answering and placing calls. Basically about 3 seconds but enough to make it a pain when saying HELLO. Any suggestions? |
| 17:59.50 | [TK]D-Fender | mecheng: Describe each leg of the call in detail. |
| 18:00.54 | mecheng | Call my DID from my cell, hear normal rings, answer the voip phone and say hello but the cell doesn't play my hello. |
| 18:01.24 | [TK]D-Fender | mecheng: Details please. |
| 18:01.52 | mecheng | I am new to this what more do you need? |
| 18:02.17 | [TK]D-Fender | mecheng: I just asked for you to provide complete details concerning each leg of the call. |
| 18:03.50 | *** join/#asterisk moy (n=moy@nat/ibm/x-b3d2fb6e7b0d9342) |
| 18:03.58 | mecheng | What do you mean by legs of the call |
| 18:04.18 | [TK]D-Fender | mecheng: How does the call arrive into * in the first place? |
| 18:05.02 | mecheng | I call from my cell phone to my DID from my VOIP provider |
| 18:05.21 | [TK]D-Fender | mecheng: What protocol, what provider? |
| 18:05.43 | mecheng | voicenetwork.ca |
| 18:05.55 | mecheng | and I believe it is ulaw??? |
| 18:06.36 | jeev | it's sys admin appreciation day today!! |
| 18:07.02 | [TK]D-Fender | mecheng: go to * CLI and pastebin a failed call at verbose 10, sip debug enabled |
| 18:07.09 | [TK]D-Fender | ~pb |
| 18:07.10 | 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 |
| 18:07.11 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 18:07.15 | jeev | FENDER! |
| 18:08.09 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:08.20 | mecheng | calls don't fail but just have a lag of about 3-4 seconds before voice works |
| 18:08.41 | mecheng | but after that initial lag everything is fast and works |
| 18:08.42 | [TK]D-Fender | mecheng: And once it starts, is it ok for the remainder? |
| 18:08.54 | mecheng | yes |
| 18:09.42 | [TK]D-Fender | mecheng: could simply be a problem with the provider then. |
| 18:09.56 | *** join/#asterisk [intra]lanman (n=lanman@va-71-0-90-168.dyn.embarqhsd.net) |
| 18:09.57 | [TK]D-Fender | jeev: Just annihilated the Earth again :) |
| 18:10.18 | [TK]D-Fender | pets his new genocidal bacterium |
| 18:10.20 | mecheng | When * registers SIP with the provider I get a 113 ms ping |
| 18:10.38 | [TK]D-Fender | mecheng: 3s is not the issue what that. |
| 18:12.18 | mecheng | huh? |
| 18:13.00 | [TK]D-Fender | mecheng: Your ping can't account for 3 seconds of delay for setup |
| 18:13.40 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 18:13.46 | hardwire | anybody know the syntax to have hylafax shrink a document to fit in a faxed page? |
| 18:14.25 | mecheng | So ping time doesn't matter for call setup? |
| 18:14.41 | [TK]D-Fender | mecheng: Not for the size of your problem. |
| 18:14.49 | [TK]D-Fender | mecheng: Test against something else |
| 18:15.27 | mecheng | test against what? |
| 18:15.35 | mecheng | ps |
| 18:15.56 | [TK]D-Fender | mecheng: Another SIP service |
| 18:16.03 | jeev | e |
| 18:16.16 | jeev | you haven't seen genocidal bacterium until you see what comes out of my ass when i fart |
| 18:16.53 | [TK]D-Fender | jeev: Thats because it instantly dissolves your optic nerves first :) You never see it coming! |
| 18:17.01 | jeev | it blinds you. |
| 18:17.04 | jaytee | now that's enough to put me off my feed |
| 18:17.10 | jeev | waits patiently for fedex to deliver his phones from telephonydepot |
| 18:17.34 | mecheng | any good test providers? |
| 18:17.57 | [intra]lanman | callcentric? tollfreegateway? |
| 18:18.05 | jeev | Fender, i have a viatalk account, i'd like to connect it to my asterisk for only outgoing calls, use it as a calling card, is that possible? if i call a special DID, put a code and be able to dial out using that line ? |
| 18:18.08 | [TK]D-Fender | mecheng: PICK ONE. Doesnt' matter. Just go and do it. Sign up with FWD. Tryt he echo test, etc |
| 18:18.31 | [intra]lanman | oh yeah, fwd too... forgot about them |
| 18:18.41 | [TK]D-Fender | jeev: Sure |
| 18:18.43 | jeev | how! |
| 18:18.52 | [TK]D-Fender | ~osmosis |
| 18:18.53 | jbot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:18.56 | [TK]D-Fender | jeev: ^^^^^^^^^^^^^^^^ |
| 18:19.08 | jeev | doesn't enjoy reading anything other than IRC |
| 18:20.04 | mecheng | I read it may be my extensions.conf ?? |
| 18:20.51 | [TK]D-Fender | mecheng: No. |
| 18:21.37 | [TK]D-Fender | mecheng: If you're ringing and you answer, then you are in Dial uto and preceeing. Lack of audio is stream setup at your provider or other poor conditions. Not * config |
| 18:22.15 | mecheng | I will try a different provider then |
| 18:22.33 | mecheng | I tried to tell my provider this but they deny it. |
| 18:23.12 | [TK]D-Fender | mecheng: Stop running in circles and go test it. |
| 18:24.26 | jeev | Fender, so you suggested i look in the book for it ? |
| 18:24.40 | mecheng | I wish there was a way to prove it is the provider from my side? |
| 18:25.47 | *** join/#asterisk Hertzy3 (n=ahertz@ahertz.atlantic.net) |
| 18:27.09 | [TK]D-Fender | jeev: If you want to do a CC setup you need to learn about CDRs and a whole buch of other stuff. Go try stuff |
| 18:27.22 | [TK]D-Fender | mecheng: You do that by proving that someone else works fine. |
| 18:27.29 | jeev | i dont really care for a calling card set up, i just wanna set it u pfor my friend |
| 18:27.33 | jeev | he'll be the only one calling |
| 18:27.41 | jeev | what's the best way to do so ? |
| 18:27.44 | mecheng | okay thanks fender |
| 18:28.09 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:28.26 | *** join/#asterisk qwequ777 (n=stefan@ip-80-73-240-23.cyberservice.net) |
| 18:28.40 | qwequ777 | hi there |
| 18:29.03 | [TK]D-Fender | jeev: A call is a call is a call. Setup something so he can call in, and then let him dial out. |
| 18:29.13 | jeev | hmmmmmmmmm |
| 18:29.36 | qwequ777 | i'm planning to install asterisk... just one question - has SRTP already been implemented? |
| 18:30.12 | [TK]D-Fender | qwequ777: No |
| 18:32.40 | Hertzy3 | My asterisk console turns my terminal grey. Does anyone know how to clear that back without closing my terminal? |
| 18:32.46 | hardwire | any voip phones that have a lifter input port, so that I don't need a lifter? |
| 18:33.06 | hardwire | with ring feedback? |
| 18:33.44 | [TK]D-Fender | hardwire: What kind of phone comes with a lifter built in? |
| 18:33.53 | hardwire | not a real lifter |
| 18:34.04 | hardwire | just a port for some common lifter systems. |
| 18:34.19 | hardwire | that way it tells the phone it's off-hook or not |
| 18:34.28 | hardwire | and the phone can send a ring notification through the port. |
| 18:34.40 | hardwire | without having to deal with the detectors and actual mechanical lifters |
| 18:34.44 | qwequ777 | [TK]D-Fender, but couldn't asterisk also act as a pure sip-server? then the users could use any protocol they like for audio transportation |
| 18:35.02 | [TK]D-Fender | hardwire: Polycom 550/560/650 support the Jabra standard. Not sure if thats compatible with your needs though. |
| 18:35.11 | *** join/#asterisk xenonex (n=xenonex@89.218.233.159) |
| 18:35.34 | [TK]D-Fender | qwequ777: No, * is not a proxy, it is a B2BUA |
| 18:35.35 | hardwire | [TK]D-Fender: it's an RJ port on the back? |
| 18:35.47 | [TK]D-Fender | hardwire: Yup |
| 18:35.51 | hardwire | [TK]D-Fender: thanks! |
| 18:35.57 | [TK]D-Fender | hardwire: Just go look it all up |
| 18:36.31 | hardwire | I am, now that I know where to look |
| 18:36.52 | qwequ777 | [TK]D-Fender, yes i read that in the book that you can download; just thought that something in this direction might have happened in the meantime. well, no problem, i can live without it^^ |
| 18:37.02 | jjshoe | I just have all calls I initiate auto-answer my aastra on speaker, save the need for a lifter or anyting like it :) |
| 18:37.29 | [TK]D-Fender | jjshoe: Now try that in a crowded call center :) |
| 18:37.42 | jjshoe | [TK]D-Fender you could do the same scenario to a headset |
| 18:38.12 | [TK]D-Fender | jjshoe: Never tested to see if you can choose the answering device. |
| 18:38.26 | [TK]D-Fender | (on a Polycom anyways) |
| 18:38.38 | *** join/#asterisk hadronzoo (n=user@user-0c8h205.cable.mindspring.com) |
| 18:38.39 | jjshoe | [TK]D-Fender honestly not sure if you can set that on a polycom. |
| 18:38.45 | [TK]D-Fender | jjshoe: Have you succeeded in doing so on an Aastra? |
| 18:38.52 | jjshoe | [TK]D-Fender yup. |
| 18:39.07 | [TK]D-Fender | jjshoe: Wellfor the 3 people needing that... more power to'em! |
| 18:39.11 | *** join/#asterisk zeppelin_ (n=zeppelin@201.21.237.179) |
| 18:39.21 | jjshoe | [TK]D-Fender everyone I know around here uses it :) |
| 18:39.29 | jjshoe | [TK]D-Fender 100+ employees |
| 18:39.30 | *** part/#asterisk hadronzoo (n=user@user-0c8h205.cable.mindspring.com) |
| 18:39.36 | jjshoe | it's really handy |
| 18:39.52 | [TK]D-Fender | jjshoe: Poor, poor agents! |
| 18:40.31 | jjshoe | [TK]D-Fender ? |
| 18:40.41 | jjshoe | [TK]D-Fender do you make all your agents sit at a table with no sound barriers? |
| 18:40.57 | jjshoe | that'd be pretty harsh |
| 18:41.23 | jjshoe | I don't care if it's handset, headset, or speaker, unless you're in at minimum cubes that block sound, blech. |
| 18:41.44 | n3hxs | thinks that they must be the ones calling me... lots of background noise :) |
| 18:42.26 | jjshoe | n3hxs yeah, so unprofessional |
| 18:42.38 | jjshoe | I don't want to hear what your co-workers did on their weekend |
| 18:42.43 | *** join/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 18:42.58 | hardwire | [TK]D-Fender: thats exactly what I need |
| 18:43.05 | hardwire | orders some 320's |
| 18:43.10 | n3hxs | With my hearing, I have trouble understanding without the background noise. |
| 18:43.36 | [TK]D-Fender | hardwire: No, the 320/330 don't have RJ headset jacks |
| 18:44.11 | [TK]D-Fender | hardwire: Stop jumping and seriously read first and make sure your "lifers" are certified compatible |
| 18:44.38 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-169.dsl.tul2ok.sbcglobal.net) |
| 18:45.32 | hardwire | [TK]D-Fender: check this out: http://www.google.com/url?sa=t&ct=res&cd=4&url=http%3A%2F%2Fwww.jabra.com%2Fsites%2FJabra%2FGNImages%2FCampaigns%2FPolycom%2FPolycom_Brochure_3753.pdf&ei=tx2KSJOzDYzmpgTb0sHADg&usg=AFQjCNEPMnAWwaTer3i0tonGFz35MtZgJA&sig2=eKy4WfkX46ht9GVesDWjdQ |
| 18:45.43 | hardwire | sorry.. can't seem to get direct links out of google search results |
| 18:45.57 | mecheng | fender I posted my SIP debug on www.pastebin.com if you want to check it out. |
| 18:46.23 | mecheng | under the user owenfredericks |
| 18:46.37 | hardwire | [TK]D-Fender: and other docs say the EHS works well with the 320's |
| 18:46.56 | [TK]D-Fender | hardOk, if their 2.5mm jack supports it and they say so, sure. |
| 18:47.22 | [TK]D-Fender | mecheng: the LINK to your post would be nice. |
| 18:47.23 | brodiem | or they just go in line/replace the handset |
| 18:47.43 | hardwire | hard0k? |
| 18:47.52 | hardwire | heh |
| 18:48.14 | [TK]D-Fender | mecheng: Contact: <sip:7076573860@192.168.1.101> <-- your * is not setup properly to handle NAT. Go read the guide : |
| 18:48.16 | brodiem | then automatically requiring external amplifier |
| 18:48.17 | [TK]D-Fender | ~sipnat |
| 18:48.18 | 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 |
| 18:48.19 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 18:48.22 | mecheng | sorry http://pastebin.com/d2a59ec4a |
| 18:48.35 | mecheng | first time using it |
| 18:50.02 | mecheng | I will be going to lunch in about 10 mins but I will return about 45 min after that. |
| 18:50.32 | [TK]D-Fender | mecheng: save the link to the guides, go read, and follow it |
| 18:52.57 | mecheng | okay I emailed them to myself for later |
| 18:54.22 | mecheng | could this cause the delay I am seeing? |
| 18:56.08 | MooingLemur | do any wifi (or otherwise wireless) cordless (battery/rechargeable operated) conference speakerphones exist? |
| 18:57.38 | [TK]D-Fender | mecheng: Shouldn't |
| 18:58.01 | [TK]D-Fender | MooingLemur: Polycom SoundStation2W + ATA |
| 18:58.02 | *** join/#asterisk wonderworld (n=ww@ip-62-143-163-199.hsi.ish.de) |
| 18:58.58 | MooingLemur | aha.. sounds good.. thanks [TK]D-Fender |
| 19:01.21 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 19:02.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:03.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:04.00 | hardwire | the GN 9350 headset has a usb port |
| 19:04.06 | hardwire | I just used ekiga and it to make a call |
| 19:04.07 | hardwire | very smooth |
| 19:04.16 | hardwire | need to get indications and stuff working with it tho |
| 19:04.30 | hardwire | or use the polycom software, what a drag. |
| 19:06.00 | HonestWorker | I have got to go. Good bye, guys. |
| 19:06.06 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 19:06.24 | rwaite | is it 5pm yet? |
| 19:06.36 | MooingLemur | in some time zone :) |
| 19:24.46 | *** join/#asterisk Ast001 (n=uros@81.18.55.102) |
| 19:26.52 | *** join/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 19:29.19 | *** join/#asterisk dlynes (n=chatzill@S01060016b68219f1.vs.shawcable.net) |
| 19:29.40 | dlynes | Does anyone know what dtmf payload type I should use for asterisk? |
| 19:29.50 | dlynes | Should it be 101? 96 doesn't seem to be working well |
| 19:32.07 | *** join/#asterisk ta^3 (n=tacvbo@189.146.188.205) |
| 19:34.02 | [TK]D-Fender | dlynes: Whichever your endpoint supports |
| 19:34.31 | ta^3 | Just wonder, what does 'overlap dialling' stands for? (allowoverlap=yes/no) |
| 19:36.55 | *** part/#asterisk Ast001 (n=uros@81.18.55.102) |
| 19:39.22 | kaii | is it a common problem that extension monitoring is hard to get working in Asterisk 1.4 ? |
| 19:39.46 | [TK]D-Fender | kaii: Care to explain what you mean by that exactly? |
| 19:40.18 | *** join/#asterisk nny_1 (n=Scott@64.203.237.47) |
| 19:40.41 | kaii | { |
| 19:42.09 | nny_1 | i have a recently installed t1 card (digium) with hardware echo cancel. I am not clear as to whether or not there is any additional steps to enable the echo cancel, and whether or not it needs to be tuned (like fxotune) once it is up. Any advice or links appreciated |
| 19:42.31 | [TK]D-Fender | nny_1: "echocancel=yes" in zapata.conf. Thats it |
| 19:43.09 | nny_1 | [TK]D-Fender: ok thanks. Apart from that, if they complain about echo on the line, is there anything else that should eb done? |
| 19:43.46 | [TK]D-Fender | nny_1: Shouldn't. Perhaps try playing with the gains a bit |
| 19:44.29 | *** join/#asterisk CanWood (n=chatzill@24.108.64.80) |
| 19:44.36 | nny_1 | [TK]D-Fender: will do. I have the milliwatt test app set up as an extension, so that should be easy |
| 19:44.37 | nny_1 | thanks |
| 19:50.56 | *** join/#asterisk damjan (n=damjan@217.16.95.15) |
| 19:53.06 | tzafrir | nny_1, fxotune will do nothing with that card |
| 19:54.33 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 19:56.11 | *** join/#asterisk rootlogin (n=root@saturn2.franken.de) |
| 19:58.44 | kaii | [TK]D-Fender: sry, was busy on the phone. regarding extension monitoring: in sip.conf i have limitonpeers=yes and llowsubscribe=yes, also notifyringing=yes and notifyhold=yes |
| 19:59.06 | kaii | i have hints for every SIP peer in the start context of my sip phones |
| 19:59.13 | kaii | (like in asterisk 1.2) |
| 19:59.47 | kaii | but asterisk 1.4 does not change states (i checked with "show hints", all idle) nor allows subscriptions |
| 19:59.57 | *** join/#asterisk moy (n=moy@nat/ibm/x-5d4570909f6ba19d) |
| 20:00.04 | kaii | my phone recieves 404 as answer to the SUBSCRIBE |
| 20:01.32 | kash | what kind of stuff do i have to look for when upgrading to 1.6 from 1.4? |
| 20:03.03 | *** join/#asterisk marc7 (n=marc@S0106001c1024382d.gv.shawcable.net) |
| 20:03.42 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:03.43 | kaii | kash: see UPGRADE.txt in the source archive |
| 20:04.19 | kash | found it :p |
| 20:06.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:06.23 | [TK]D-Fender | kaii: remove all of those. Set "type=peer", "call-limit=100". Apply and test. If failure pastebin the SIP debug and hint dump |
| 20:11.27 | *** join/#asterisk edibrac (n=edibrac3@75.149.50.41) |
| 20:11.52 | edibrac | the last time our T1-PRI provider had problems I saw a "Red Alarm" in my /var/log/asterisk/messages .. but "Red Alarm" could also indicate a bad voip card? |
| 20:12.12 | edibrac | because in addition to that, I see other Red Alarms in the past, maybe a few times each month, lasting up to 5 seconds...happening at different times. |
| 20:13.03 | edibrac | could XO be testing or calibrating things? |
| 20:14.22 | tzafrir | edibrac, what card do you have? |
| 20:15.19 | edibrac | tzafrir: Digium TE121P |
| 20:16.12 | tzafrir | "red" means basically "not connected". Or otherwise your side and the remote side are not talking at a very basic level |
| 20:16.37 | tzafrir | So, do you have anything plugged into that card? |
| 20:16.42 | edibrac | also: rpath Linux 1.07, asterisk 1.2.12, kernel 2.6.24.7 |
| 20:17.10 | tzafrir | asterisk 1.2.12? wow |
| 20:17.11 | edibrac | tzafrir: our PRI line? |
| 20:17.42 | *** join/#asterisk BBHoss (n=hoss@c-68-62-170-33.hsd1.al.comcast.net) |
| 20:17.47 | tzafrir | (and it's not a voip card. No IP is involved) |
| 20:18.01 | marc7 | besides the asterisk TFOT, are there any good resources that describe hardware requirements for high-volume asterisk servers? |
| 20:18.43 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:18.59 | tzafrir | edibrac, this is something of the sort of: bad wiring, incorrect span= line in zaptel.conf, or bad definitions at the telco side |
| 20:19.21 | tzafrir | In what country are you? What telco? |
| 20:19.29 | tzafrir | hopes for a saner E1 |
| 20:20.14 | edibrac | tzafrir: span line should be fine, this is on a working setup that's been on for a few months. we have had 2 "major" official outages from our telco (XO) but were late in the day |
| 20:20.38 | edibrac | the thing I was wondering about were the other red alarms (up to 5 seconds) i see in my logs. |
| 20:21.44 | tzafrir | well, I'll leave this to someone who actually uses T1... |
| 20:22.09 | edibrac | wiring from our asterisk box to the MPOE is new also ..but who knows maybe that particular one is bad |
| 20:22.13 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 20:26.37 | *** join/#asterisk tloges (n=tiagolog@mail.abyzti.com.br) |
| 20:28.28 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 20:29.24 | angler | edibrac, upgrade Zaptel to the latest(not sure when the code changes we introduced but it was sometime after zaptel 1.2.12) and theres a good chance the problem will go away. |
| 20:35.14 | x86 | is there a way I can bring just a single zaptel span down? |
| 20:35.30 | *** join/#asterisk irieKen (n=chatzill@rrcs-74-87-28-55.west.biz.rr.com) |
| 20:35.39 | x86 | I've got a down voice T1, and I was wanting to take it down and reload the interface (it's a sangoma card) |
| 20:35.54 | x86 | i.e., make the entire span unavailable to asterisk |
| 20:36.05 | x86 | without bothering the other spans, which are working fine |
| 20:36.56 | bbryant | does she have to sell herself to get some? |
| 20:37.07 | bbryant | I know she's a whiny teen star, but damn |
| 20:37.14 | *** part/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 20:37.21 | irieKen | Hello, I was wondering if anyone could help me resolve an asterisk FXS outbound calling problem. |
| 20:38.09 | irieKen | I am using an AA50, and I can receive calls just fine over the FXS port, but I can't seem to call out; I get 503 errors on my phone. |
| 20:40.10 | *** join/#asterisk bbryant (n=brett@c-71-228-178-34.hsd1.al.comcast.net) |
| 20:43.11 | Idle | looks around |
| 20:43.57 | irieKen | Hey Idle:) |
| 20:46.17 | bbryant | wrong window |
| 20:47.54 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 20:49.14 | *** join/#asterisk tobeya (n=chatzill@unaffiliated/tobeya) |
| 20:51.21 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 20:52.31 | angryuser | have you read the mailing list lately?, that 'who is better' thread |
| 20:55.57 | angryuser | fender you here ? |
| 20:58.40 | angryuser | I have configured fring client for testing purpose, so registered = Nokia phone=>Nat=>asterisk i am using standart ports 5060 and 10000-20000 , routed to * box , how come fring managers to register to port 50563 ? |
| 20:59.53 | angryuser | Is the udp port number passed on sip handshake ? |
| 20:59.54 | edwin_quijada | I installed a T1 card into my box so now I am not using this line but aSterisk doesnt want upload without this line. |
| 21:00.09 | tzafrir | angryuser, unlike most flames on that list, there was a pretty decent s/n rate on that thread. Though I'm not sure ohw long it will stay that way |
| 21:00.10 | edwin_quijada | I need to disconeect this card?: |
| 21:00.40 | tzafrir | "doesn't want to upload" ==? |
| 21:00.46 | tzafrir | Does not want to start? |
| 21:01.27 | edwin_quijada | tzafrir: Yes, start I get an error about this zaptel card but when I connect the line everithing is fine |
| 21:01.29 | angryuser | tzafrir , what s/n means ? |
| 21:01.39 | tzafrir | If you want to use that card for timing only, put in its span line in zaptel.conf: span=1,0,<rest-of-ther-line> |
| 21:01.55 | tzafrir | angryuser, sound-to-noise ratio |
| 21:02.21 | tzafrir | Useful content vs. random noise that should be ignored |
| 21:02.26 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:02.32 | tzafrir | (or actually: filtered out) |
| 21:02.41 | *** join/#asterisk D3b|4n (i=X@unaffiliated/lynxnica) |
| 21:02.49 | edwin_quijada | Thks, tzafrir |
| 21:03.15 | D3b|4n | i have a problem |
| 21:03.23 | D3b|4n | Jul 25 14:43:42 NOTICE[3908]: chan_sip.c:11151 handle_request_subscribe: Got SUBSCRIBE for extension 7540@default from 190.53.33.25, but there is no hint for that extension |
| 21:03.23 | D3b|4n | <PROTECTED> |
| 21:03.47 | angryuser | tzafrir yes, sometimes arguing goes too far |
| 21:04.42 | D3b|4n | ? |
| 21:04.50 | angryuser | D3b|4n you need to sort your hints ? |
| 21:05.46 | D3b|4n | i need active the hints for erase the error? |
| 21:06.00 | angryuser | yes but it is not critical |
| 21:06.23 | tzafrir | You need to have a 'hint' priority for that extention to refer to the actual device |
| 21:07.02 | tzafrir | e.g.: exten => 7540,hint,MGCP/1234 |
| 21:07.10 | D3b|4n | ok |
| 21:07.18 | angryuser | your phone subscribing to context and cant find hint, do you use them ? do you want that phone use hints ? |
| 21:07.49 | D3b|4n | yeah |
| 21:08.10 | tzafrir | The phone does not use the hints. Hints are internal for Asterisk. |
| 21:08.12 | D3b|4n | my extensions are configured like this |
| 21:08.41 | angryuser | verify what context it subscribing , and hint ther for that device, as tzafrir wrote |
| 21:08.45 | tzafrir | The phone subscribes through SIP to be notified on the status of an "extension" on the server |
| 21:09.07 | D3b|4n | exten => _NXXNXXXXXX,1,Monitor(wav|${EXTEN}-${DNID}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}.wav|a) |
| 21:09.07 | D3b|4n | exten => _NXXNXXXXXX,2,Dial(SIP/panama/${EXTEN}) |
| 21:09.14 | D3b|4n | so the new way should be like this |
| 21:09.26 | tzafrir | You should be able to see that 'hint' in 'dialplan show <context-name>' in the CLI |
| 21:09.42 | tzafrir | e.g: dialplan show default |
| 21:09.56 | tzafrir | And also: core show hints |
| 21:10.28 | angryuser | also there is subscribecontext= in general, it's changes the context where hints placed |
| 21:10.53 | *** part/#asterisk brodiem (n=brodiem@rrcs-24-227-171-186.sw.biz.rr.com) |
| 21:10.55 | D3b|4n | sipserver*CLI> show hints |
| 21:10.55 | D3b|4n | There are no registered dialplan hints |
| 21:11.04 | angryuser | no wrong , it changes where SUBSCRIBE is sended |
| 21:12.54 | *** join/#asterisk RipeR-81 (n=riper81@190.53.33.25) |
| 21:13.31 | RipeR-81 | angryuser im working with D3b|4n with this asterisk |
| 21:13.51 | RipeR-81 | issue ... when you subscribe from x-lite to a contact... |
| 21:13.57 | RipeR-81 | we keep getting that error |
| 21:14.03 | RipeR-81 | so as your explanation on hint... |
| 21:14.07 | angryuser | create a context like 'myhints' change value of subscribecontex=myhints , create a file hints.conf add hints extensions, then do #include=hints , that what i do usially |
| 21:14.18 | RipeR-81 | ok |
| 21:14.21 | RipeR-81 | will try that way... |
| 21:14.55 | RipeR-81 | angryuser i tought that by just adding a first step.. and putting the hint, then 2nd step monitoring (so a recording is made) and 3rd to actually dial. |
| 21:15.34 | *** join/#asterisk nickjqw (n=webbn@c-71-231-94-228.hsd1.or.comcast.net) |
| 21:16.02 | RipeR-81 | angryuser the #include=hints should be on extensions.conf right ? |
| 21:16.14 | RipeR-81 | ~centos52bug |
| 21:16.15 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 21:17.23 | nickjqw | I'm going crazy trying to figure out why I can't register with my service provider over iax2... I just upgraded on a separate box and if the old box is registered, then the new one will to. If the old box is off, I can't get the new one to register, just times out. Any ideas? |
| 21:17.52 | angryuser | RipeR-81 hints are managed by system , it's a state, it will send automaticly the state of device tu subscribed phones, you can manage the state of device or whatever in dialplan, what do you want exactly ? |
| 21:18.19 | x86 | what's an RAI alarm? |
| 21:18.34 | x86 | I know what RED and OOF alarms are, but RAI? |
| 21:18.51 | angryuser | RipeR-81 place that include inside your subscribecontext |
| 21:19.41 | RipeR-81 | angryuser to be quite honest dont use that feature, how ever since we installed x-lite and added people to the contact list i started noticing that error message on the asterisk CLI |
| 21:20.05 | RipeR-81 | RipeR-81 so i requested D3b|4n to get in here and do a little research since i was fighting with cisco call manger |
| 21:20.08 | RipeR-81 | :) |
| 21:21.04 | angryuser | RipeR-81 discard it then, it's not critical |
| 21:22.11 | RipeR-81 | angryuser thanks for the explanation though... |
| 21:22.19 | RipeR-81 | X) |
| 21:22.20 | *** join/#asterisk Qapf (n=Qapf@mail.oldworlddoor.com) |
| 21:22.22 | irieKen | Hey, anyone know why an Asterisk appliance would be able to receive calls, but wouldn't make outgoing calls (gives 503 error)? |
| 21:22.59 | angryuser | irieKen pastebin all output |
| 21:23.20 | Qapf | hey, i was wondering if anyone had an asterisk box up on a vps or other shared server arrangment and how it coexisted in terms of call quality and what. im paying for a dedicated server right now but its a bit of an overkill |
| 21:23.21 | angryuser | irieKen and sip trace would be nice |
| 21:24.40 | angryuser | Qapf wherever your server is , when you have a bad outbound provider, it's bad ;) |
| 21:25.49 | Qapf | angryuser: that is what is really bugging me about this, i have a server up at 1and1 im using for asterisk and its nice and reliable, but its also $100 a month for a call volume of maybe 5-6 active calls at once |
| 21:25.56 | irieKen | Hmm, I'm not quite sure how to do the SIP trace. |
| 21:26.00 | Qapf | and i also know that 1and1's support is just about useless |
| 21:26.10 | Qapf | and im simply lucky that i have never needed them |
| 21:26.24 | angryuser | irieKen just pastebin cli output first |
| 21:26.43 | angryuser | you use 1and1? |
| 21:26.45 | irieKen | How will I get CLI output on the appliance? |
| 21:27.10 | jeev | hey, is there a difference between PoE or ethernet cable? is it the same cable ? |
| 21:27.11 | Qapf | angryuser: yea, i know |
| 21:27.15 | Qapf | its a problem waiting to happen |
| 21:27.20 | angryuser | irieKen log to cli 'asterisk -vvvvvvvr' make that call copy&paste to pastebin.ca |
| 21:27.25 | Qapf | but the connectivity is solid |
| 21:27.29 | Qapf | and the ping times are consistant |
| 21:27.37 | irieKen | Angryuse: Ok, BRB. Thanks. |
| 21:27.55 | angryuser | Qapf it's a french provider ? |
| 21:28.02 | Qapf | 1and1.com, us provider |
| 21:28.07 | Qapf | datacenter in nyc |
| 21:30.30 | tzafrir | ok. Let the record show I did my best to stop that flame :-) |
| 21:31.12 | tzafrir | irieKen, can you ssh into it? |
| 21:31.32 | irieKen | Yeah, I have SSH access. Posting results of failed call now: http://pastebin.com/m35978ecc |
| 21:32.34 | angryuser | Qapf i dont see any question here ;) |
| 21:32.41 | kash | root 27429 0.0 0.0 2800 720 pts/1 S 14:29 0:00 astcanary /var/run/alt.asterisk.canary.tweet.tweet.tweet |
| 21:32.43 | kash | what the hell. |
| 21:32.59 | irieKen | angryuser: http://pastebin.com/m35978ecc |
| 21:33.17 | Qapf | angryuser: im just wondering if anyone has their asterisk going on some kind of shared server situation and if they have anything good to say about it. |
| 21:34.07 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 21:34.17 | jameswf | would not use 1and1 |
| 21:34.23 | angryuser | irieKen you have some syntax problems pastebin your extensions.conf |
| 21:34.54 | angryuser | Qapf i have used ovh dedicated, never failed |
| 21:34.59 | tzafrir | irieKen, dialplan show numbering-plan-custom-1 |
| 21:35.24 | tzafrir | There seems to be a variable there whose value is accidentally empty |
| 21:35.45 | angryuser | like Sip |
| 21:35.51 | jameswf | some say there are no accidents |
| 21:36.04 | kash | ok, uhm, i upgraded to 1.6 from 1.4 and now get this |
| 21:36.05 | kash | Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 21:36.21 | Qapf | jameswf: i dont want to keep using 1and1, but i gotta find somewhere else to go :P |
| 21:36.22 | irieKen | angryuser: Dialplan http://pastebin.com/d338710c6 |
| 21:36.24 | jameswf | well does it ? |
| 21:36.38 | kash | it does. |
| 21:37.33 | tzafrir | exten=_91700XXXXXXX!,1,Macro(trunkdial,${}/${EXTEN:1}) |
| 21:37.47 | tzafrir | Nice :-) |
| 21:37.59 | jameswf | kash: what does "pgrep asterisk" show |
| 21:38.18 | kash | onyx:~/asterisk# pgrep asterisk |
| 21:38.19 | kash | 28402 |
| 21:38.36 | irieKen | tzafrir: It's an AA50, so it comes pre-packed with goodness;) |
| 21:38.53 | angryuser | tzafrir it's some king of gui generated i suppose, i he modyfy by hand, gui can just go south |
| 21:38.55 | tzafrir | So ask whoever packed it with goodness :-( |
| 21:39.10 | jameswf | kash how are you starting asterisk |
| 21:39.14 | kash | init.d |
| 21:39.30 | tzafrir | angryuser, actually that GUI is designed to work with manually-edited config files |
| 21:39.36 | jeev | hey, is there a difference between PoE or ethernet cable? is it the same cable ? |
| 21:39.44 | jameswf | does asterisk.conf have /var/run or /var/run/asterisk/ |
| 21:39.46 | Qapf | jeev: don't lick the poe cable |
| 21:39.50 | Qapf | it will hurt |
| 21:39.58 | irieKen | tzafrir: Digium isn't much help <-- They pre-loaded the AA50 device. |
| 21:39.59 | tzafrir | Either a bug in it, or something not set in the GUI, or both |
| 21:40.01 | jeev | Qapf, coujld any cat5e use PoE ? |
| 21:40.11 | angryuser | tzafrir that's nice, but in most times it's not ;) |
| 21:40.23 | Qapf | jeev: there is no difference, if you read the standard poe is power over ethernet, or simply enough power over cat5e |
| 21:40.35 | jeev | ;) |
| 21:40.37 | Qapf | and my eariler tip remains in effect |
| 21:40.40 | Qapf | dont lick the poe cable |
| 21:40.42 | tzafrir | irieKen, try maybe #asterisk-gui |
| 21:40.52 | jeev | Qapf, i'll ignore your recommendation towards licking the cable and do it anyway |
| 21:41.01 | tzafrir | Unless someone here can help debug that |
| 21:41.24 | *** join/#asterisk nn (n=nn@unaffiliated/nn) |
| 21:41.31 | irieKen | tzafrir: Ok. Though, I don't think that it is a GUI problem. |
| 21:41.40 | angryuser | i give up , we know the problem, we cant solve it, welcome ti gui ;) |
| 21:41.45 | angryuser | tp* |
| 21:41.50 | tzafrir | it's an error in the dialplan generated by the GUI |
| 21:41.56 | angryuser | oh.... to* |
| 21:42.19 | tzafrir | numbering-plan-1 is generated directly by the javascript code of the GUI |
| 21:42.32 | angryuser | i have an idean you generated custon trunk dont you ? |
| 21:42.45 | irieKen | tzafrir: Oh... Well, I can manually edit the dialplan. |
| 21:42.49 | edibrac | how does asterisk decide which zap channel to use? basically if zap/1 is used, it will go to zap/2.. and so on? |
| 21:42.49 | tzafrir | maybe try to figure out the missing name of the variable |
| 21:42.55 | irieKen | angryuser: Yeah, I can probably create a custom trunk. |
| 21:42.58 | angryuser | irieKen and you a using sip outbound ? |
| 21:43.02 | tzafrir | Or replace it with the actual value that should be there |
| 21:43.13 | irieKen | angryuser: No, analog over the FXO port. |
| 21:43.19 | tzafrir | irieKen, which looks like the trunk itself |
| 21:43.26 | tzafrir | e.g. SIP/peername |
| 21:43.30 | tzafrir | but I'm not sure |
| 21:43.48 | angryuser | so ypu need to put Zap somewhere |
| 21:44.02 | tzafrir | if it's analog: try: Zap/<num-of-channel> |
| 21:46.01 | irieKen | tzafrir: ok, so how would I change this? exten=_9XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid}) |
| 21:46.22 | *** join/#asterisk NirS (n=NirS@77.127.137.52) |
| 21:46.56 | tzafrir | irieKen, core show globals |
| 21:47.07 | tzafrir | what value does trunk_1 have? |
| 21:47.21 | NirS | Tzafrir, I uploade the hebrew patch for app_voicemail today to the tracker |
| 21:47.35 | NirS | let's see how much time it will take me this time to get the patch in :-) |
| 21:47.54 | irieKen | tzafrir: trunk_1_cid=asreceived <-- That's all that's there. |
| 21:48.36 | *** join/#asterisk nickjqw (n=webbn@c-71-231-94-228.hsd1.or.comcast.net) |
| 21:48.42 | tzafrir | irieKen, so it's not set |
| 21:49.04 | irieKen | tzafrir: What is not set? |
| 21:49.06 | tzafrir | you should probably also set in in the [globals] section |
| 21:49.15 | tzafrir | trunk_1 |
| 21:49.37 | NirS | ~centos52bug |
| 21:49.37 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
| 21:50.31 | irieKen | tzafrir: so, what should I set there? |
| 21:50.39 | wasabi | Hey so how should one trap voip traffic in iptables? |
| 21:50.42 | wasabi | Static port or what? |
| 21:51.01 | wasabi | (sip) |
| 21:52.39 | wasabi | Also, I'm still having an annoying Polycom digitmap problem I cannot seem to solve. Don't suppose anybody is familiar with this? It's when doing a blind transfer. When I type *5, it exits out of hte blind transfer window. |
| 21:53.31 | angryuser | wasabi trap sip traffic ? for dumping ? |
| 21:53.38 | wasabi | for Qos |
| 21:53.44 | wasabi | not sure how best to identify it |
| 21:54.07 | irieKen | tzafrir: should I add trunk_1 = Zap/g1 ? |
| 21:54.17 | tzafrir | I think |
| 21:54.31 | angryuser | wasabi so you want to find Qos solutins for your network? outgoing queues ? |
| 21:54.47 | irieKen | tzafrir: BTW, what does the G indicate in "trunk_1 = Zap/g1" ?:) |
| 21:54.56 | tzafrir | that is: if you have those channels with group=1 in zapata.conf (or is it: in users.conf?) |
| 21:54.58 | wasabi | angryuser: i want to ensure the tos fields are set properly using iptables, is all. |
| 21:55.04 | x86 | irieKen: group number |
| 21:55.09 | x86 | irieKen: actually no |
| 21:56.03 | x86 | irieKen: it's how it picks the channel from that group... g is sucession starting on the first channel (lowest numbered), and going up from there... G starts with the last channel (highest number), and goes down from there, r and R pick a channel randomly, iirc |
| 21:56.30 | irieKen | THANK YOU GUYS SOOOOOO MUCH! |
| 21:56.57 | *** join/#asterisk genioreal (n=real@200.27.193.98) |
| 21:57.12 | irieKen | I've been working on this thing for 2 days, crawling google and all... And all it took was one command line addition!:) |
| 21:57.52 | angryuser | that's why gui sometimes is not nice for start |
| 21:58.03 | genioreal | hi, im installing a new server with asterisk on linux i got to hard drives... i was wondering on hw the partitioning must go with partitions will keep more data ? so i can make it bigger the server is just for asterisk any ideas? |
| 22:00.18 | angryuser | genioreal most place taked by audio messages, aterisk is not very hdd intensive |
| 22:00.36 | angryuser | genioreal id you dont have 500 users of course |
| 22:00.51 | genioreal | i will have a lot of users |
| 22:01.53 | wasabi | Anybody ever tried asterisk in KVM or XEN? :0 |
| 22:01.56 | angryuser | create one partition for audio, i dont see anything more, tzafrir ? |
| 22:02.10 | *** join/#asterisk ta^3 (n=tacvbo@189.146.188.205) |
| 22:02.19 | *** part/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 22:03.33 | tzafrir | angryuser, if you use the same sound file often, it is likely to already be in memory |
| 22:04.11 | tzafrir | recording and writing many logs is more I/O intensive |
| 22:04.24 | angryuser | tzafrir i mean user recorded conversations |
| 22:04.34 | angryuser | just a storage, no more |
| 22:04.39 | *** part/#asterisk damjan (n=damjan@217.16.95.15) |
| 22:05.08 | angryuser | tzafrir that's depends on verbose |
| 22:05.31 | angryuser | tzafrir look's like your comment on flame was responded |
| 22:05.37 | tzafrir | BTW: one thing I like about the asterisk-gui is that it creates a simple and quite sane dialplan |
| 22:05.43 | tzafrir | That can be hand-edited later |
| 22:06.40 | angryuser | includes everywhere, i get lost with them |
| 22:10.36 | tzafrir | I actually referred mostly to other parts of that flame |
| 22:10.44 | tzafrir | s/flame/thread/ |
| 22:14.13 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:17.23 | tzafrir | NirS, that was fast :-) |
| 22:17.30 | angryuser | Well it doen not matter anymore |
| 22:18.44 | angryuser | misstel ^^ |
| 22:25.55 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:42.37 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:43.23 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:52.50 | *** join/#asterisk ManxPower (n=manxpowe@132.sub-70-223-56.myvzw.com) |
| 23:00.32 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 23:02.16 | *** part/#asterisk Deeewayne (n=Deeewayn@216.207.245.1) |
| 23:02.37 | ManxPower | waves from Canon GA |
| 23:05.20 | TJNII | waves back from Ames IA |
| 23:05.44 | angryuser | i see waves , blue waves |
| 23:06.00 | TJNII | I see corn. Lots and lots of corn. |
| 23:06.19 | [TK]D-Fender | "I see stupid people" |
| 23:06.27 | jeev | FENDER! |
| 23:06.28 | angryuser | TJNII wake up |
| 23:06.34 | jeev | i got all 20 phones up in 15 minutes :) |
| 23:06.48 | jeev | loves dhcpd.conf and addphone.sh that i built |
| 23:06.55 | *** join/#asterisk ][ologramMan (n=giuseppe@host35-206-dynamic.2-87-r.retail.telecomitalia.it) |
| 23:07.00 | ][ologramMan | Hi all |
| 23:07.05 | angryuser | Fender i see fender, what is tk ? |
| 23:07.13 | jeev | tittykaka |
| 23:07.45 | angryuser | stop flaming it's a serious question |
| 23:07.58 | angryuser | ^^ |
| 23:08.23 | jeev | i'm not flaming lol |
| 23:09.00 | ][ologramMan | anyone willing to deal with newbie question? |
| 23:09.15 | ManxPower | you might be surpized at what I see. |
| 23:09.29 | jeev | just ask the questions |
| 23:09.34 | angryuser | dont ask, to ask, ask |
| 23:09.36 | jeev | if we could deal with ManxPower, we could probably deal with you |
| 23:10.11 | ManxPower | jeev: The moment Pidgin has a Perm Ignore feature, you'll be the first on on it. |
| 23:10.27 | ManxPower | Don't you feel special? 8-| |
| 23:10.41 | jeev | i love you too |
| 23:10.42 | ][ologramMan | I wanted to use an external sip provider from my wireless lan, and configure a phone to access it without asterisk |
| 23:10.52 | jeev | hey, does the PoE standard use more than 2 pairs on the ethernet cable ? |
| 23:11.29 | angryuser | jeev look at ppoe specs |
| 23:12.00 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:12.03 | ][ologramMan | the question is, if i configure the phone to point to asterisk box (in my lan), is it possible to have a trick that will allow me to automagically route the packets to the external sip provider when asterisk box is powered off |
| 23:12.04 | jeev | i want fender back. |
| 23:13.07 | jeev | http://www.altair.org/labnotes_POE.html |
| 23:13.19 | jeev | hmm |
| 23:13.42 | angryuser | ][ologramMan depends on phone, some have more than one sip account possible |
| 23:13.49 | jeev | if possible, would it be safe to have 2 phones running off 1 cable through the walls? |
| 23:13.59 | ][ologramMan | just discovered that N78 is not among them... |
| 23:14.20 | angryuser | ][ologramMan you are using nokia sip client ? |
| 23:14.37 | TJNII | jeev: As long as ther is no PoE. I've seen it done. |
| 23:14.42 | ][ologramMan | r u kidding me? no sip client on n78, I am poking with fring |
| 23:14.53 | jeev | i'm asking bout PoE |
| 23:15.10 | jeev | right now, there is only a single ethernet going into that room and i need two phones.. |
| 23:15.14 | *** join/#asterisk macros73 (n=cs@c-67-163-224-69.hsd1.pa.comcast.net) |
| 23:15.15 | jeev | if i split it, will it be dangerous ? |
| 23:15.15 | angryuser | ][ologramMani never said that, there is one |
| 23:15.24 | TJNII | jeev: Don't know. Doesn't true PoE current monitor? |
| 23:15.42 | jeev | no idea, i guess i'll try to run another one. |
| 23:15.47 | ][ologramMan | Not in Italy, how do you launch it? there are settings for sip, BUT no client |
| 23:15.52 | TJNII | Don't think it would be dangerous, don't know if it will work. |
| 23:15.57 | jeev | ok |
| 23:16.22 | TJNII | If running another cable is a pain, why not inject PoE in the room or not use PoE? |
| 23:16.44 | ][ologramMan | Angry user: If you could point me on how to run/enable it I would be grateful |
| 23:16.59 | angryuser | ][ologramMan i dont see what you are telking about, i have used both, niki sip client and fring, still , what was the question ? |
| 23:17.09 | angryuser | nokia* |
| 23:17.26 | jeev | well, i have 1 adapter |
| 23:17.31 | angryuser | fring support only one sip cient configuration |
| 23:17.37 | jeev | injector is so expensive, considering i bought a 24 poe switch for 400 bux |
| 23:17.40 | jeev | i'l probably run another wire. |
| 23:17.44 | ][ologramMan | Angryuser: Sorry for the dumb question, but could you tell me how to run the nokia one? |
| 23:18.10 | ][ologramMan | fring does work but as you said, just one profile |
| 23:19.20 | angryuser | ][ologramMan i have a different phone n95, it has multiple sip profiles posible, the only issue, it does not send any packet's to keep nat part open |
| 23:19.29 | angryuser | port* |
| 23:19.46 | angryuser | qualify= does not help either |
| 23:19.48 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 23:20.28 | angryuser | ][ologramMan n73 got symbian 3rd edition ? |
| 23:21.26 | ][ologramMan | Yes 9.3fp2 Ok, so back to square one: is it feasible to have my n78 configured in such a way that will use a single profile and point to a asterisk box (on my lan) when the box is up and to the sip provider when the asterisk is down? |
| 23:22.00 | ][ologramMan | I was thinking to fiddle with the ip and/or firewall rules plus some scripts |
| 23:22.35 | angryuser | find the answer for n 73 'You can add a SIP profile into the N73 but you cannot use SIP' so frin only |
| 23:23.13 | angryuser | ][ologramMan probably not |
| 23:24.46 | ][ologramMan | but if the firewall could detect the asterisk is down it could be done activating a rule, or maybe if I reference the ip of the asterisk with a dyndns name... |
| 23:24.54 | ManxPower | ][ologramMan: Many phones support SIP DNS SRV records, check the docs for your phone. Also most phones support primary and backup server configs, maybe with even the ability to have different auth details |
| 23:25.29 | ManxPower | ][ologramMan: Ah, a nokia. Hell if I know. |
| 23:25.36 | ][ologramMan | Thanks ManxPower, sadly N78 is not prone to this, and fring seems not having these features |
| 23:25.55 | ][ologramMan | maybe I am wrong, just started the journey into this and asterisk |
| 23:26.03 | angryuser | you need to use a proxy without a downtime |
| 23:26.18 | angryuser | sip proxy* |
| 23:27.05 | ][ologramMan | Angry and ManxPower: the weirdest idea that comes to my mind is to set as primary dns the asterisk box, that would correctly resolve its own name as the sip provider |
| 23:27.37 | ][ologramMan | if the box is down, the secondary dns would be the firewall that would point the phone to the provider...weird enough? |
| 23:27.47 | angryuser | ][ologramMan you need to use a sip proxy |
| 23:28.25 | ][ologramMan | I only have one sip proxy that is up 24/7 and it is the provider |
| 23:28.45 | ][ologramMan | I would not use asterisk this way |
| 23:29.32 | ][ologramMan | at all I mean, there would be a complete bypass |
| 23:29.40 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 23:29.40 | *** join/#asterisk Victor_Yure (n=aaa@unaffiliated/victoryure/x-837844) |
| 23:36.37 | angryuser | why this channel accepts only registered users ? |
| 23:36.55 | [TK]D-Fender | angryuser: Troll-reduction |
| 23:37.32 | angryuser | i think some new to irc users are unable to get here maybe |
| 23:37.59 | [TK]D-Fender | jeev: Don't |
| 23:38.21 | [TK]D-Fender | jeev: Each will try to put a load on the circuit. Use one PoE, and passthrough to the other and usea brick |
| 23:38.43 | [TK]D-Fender | angryuser: And ChanServ tells them they have to register, and they do all the time |
| 23:41.15 | angryuser | i am sure of that he does, but 'very new users to irc', and how big is that Troll problem was ? |
| 23:41.42 | jeev | brick ? |
| 23:41.47 | jeev | but will PoE go passthru ? |
| 23:42.01 | jeev | if i have PoE go into the phone, when it's switched, does it send power to the switched port? |
| 23:42.03 | Qwell | If you can't figure out how to register after ChanServ tells you...you probably wouldn't be able to figure out how to fix the problem you came here about. |
| 23:42.12 | *** join/#asterisk jeffspeff (n=jeff@c-98-240-112-228.hsd1.ky.comcast.net) |
| 23:43.04 | [TK]D-Fender | jeev: No, it won't passthrough, thats why I said use a brick on the other |
| 23:43.22 | [TK]D-Fender | Qwell: Like I said... troll prevention :) |
| 23:44.05 | *** join/#asterisk dinominant (n=dinomina@S0106000d882cf7f3.cg.shawcable.net) |
| 23:45.41 | dinominant | Hi, I've been following the gentoo wiki article on getting asterisk installed on my system and I'm stuck. The wiki suggested I come here for help |
| 23:48.50 | dinominant | typing sudo -u asterisk asterisk -vvvvvc gives me several errors |
| 23:49.50 | dinominant | I'll concentrate on fixing this one first: |
| 23:49.51 | dinominant | pbx_dundi.c:4580 set_config: Unable to load config dundi.conf |
| 23:50.32 | [TK]D-Fender | dinominant: first guess is however you installed it you did not get sample configs to help start you off. |
| 23:51.00 | dinominant | I did "sudo emerge asterisk" (I'm running gentoo) |
| 23:51.05 | [TK]D-Fender | dinominant: Go see what other packages you may be missing, or trashe what you've done, and compile from source direct off asterisk.org |
| 23:54.30 | jeev | Fenderino |
| 23:55.51 | jeev | Fender, what i was meaning to say is |
| 23:56.05 | jeev | take the A portion of the ethernet into phone 1 and B portion into phone 2 |
| 23:56.19 | jeev | and A portion into swtich port x and B portion into switch port x+1 |
| 23:57.19 | jeev | so 2 pair running into each phone |
| 23:57.28 | [TK]D-Fender | jeev: Don't even dream of shoving it on a Y cable |
| 23:57.49 | [TK]D-Fender | jeev: wall (PoE) to the first, chain the 2nd into the first, and that one is going to need a brick |
| 23:57.50 | jeev | not a Y! |
| 23:58.26 | Qwell | dinominant: That isn't an error. Next? |
| 23:58.28 | angryuser | can someone tell me where MFC/R2 signalling is used, and what for ? |
| 23:58.45 | jeev | Fender, i'm saying if i have 1 line into the wall, i could split the cable with 4 pins in each jack |
| 23:58.52 | jeev | would that be OK? |
| 23:59.02 | jeev | so the single cable with 4 pair would be split, each phone would have 2 pair. |
| 23:59.50 | [TK]D-Fender | <PROTECTED> |
| 23:59.52 | Qwell | splitting 2 pairs? what? |