| 00:00.30 | jks | adorah, choose a different provider |
| 00:01.11 | adorah | <[TK]D-Fender>I don't have an access to SIP debug since the provider blocks by firewall any attempt to get data I get only what data they send me |
| 00:01.13 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 00:01.39 | [TK]D-Fender | adorah: wher is the * SIP debug data? |
| 00:01.41 | adorah | <jks>Curently they r the only one that provide locally SIP Trunk |
| 00:02.17 | jks | adorah, you provider cannot block by firewall the data _you_ send so that you cannot record it |
| 00:02.20 | *** join/#asterisk Bananaskin (n=Banana@94-193-31-47.zone7.bethere.co.uk) |
| 00:02.36 | jks | adorah, that sounds pretty weird then... make your own provider then |
| 00:03.33 | adorah | jks: regulatory problems..this is neither US nor Sweeden.. |
| 00:03.56 | jks | adorah, where is it then? |
| 00:04.04 | jks | adorah, ireland? |
| 00:04.08 | adorah | jks: Israel |
| 00:04.16 | adorah | even worse..hehe |
| 00:04.18 | jks | Israel... hmm.. you can't buy phone lines there? |
| 00:04.45 | adorah | <jks>we can but sip trunk has some merits-when it works.. |
| 00:05.03 | jks | but it doesn't :-) |
| 00:05.42 | adorah | <jks>well when the analog PCI card is full this is the most economical option |
| 00:06.42 | adorah | Also with analog line one doesn't get some options like DIDs etc. |
| 00:06.44 | hardwire | orkid: hai? |
| 00:07.22 | jks | adorah, buy another pci card.. DID you should be able to buy |
| 00:08.01 | adorah | <jks>I can buy the customer doesn't want to spend money on it.. |
| 00:08.05 | jks | adorah, if your think your provider is clueless, you cannot solve the problem yourself, and you're unable to give the channel the requested information necessary to examine the case closely, then your options are limited |
| 00:08.20 | jks | adorah, then tell your customer that you cannot deliver |
| 00:08.50 | adorah | <jks>well for one I bought g729 codec licenses and installed it may be there is a BW problem.. |
| 00:09.14 | orkid | hardwire: so they wont do RCF , to a local number, and to a long distance number it would cost money per minute + some fee. no point in paying twice. i wonder if the cableco listed in my remote will do rcf or something for me, like just fwd that stuff to a sip :) |
| 00:09.19 | orkid | i doubt it though |
| 00:09.50 | hardwire | ew |
| 00:10.41 | hardwire | that sucks |
| 00:11.01 | jks | adorah, your traces would show you if it were |
| 00:15.57 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 00:23.04 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:33.08 | *** join/#asterisk stevie_ramjet (n=putnopvu@c-71-228-178-34.hsd1.al.comcast.net) |
| 00:33.08 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 00:41.19 | Micc | someone told me what to do to change the up/down arrow keys in menuconfig to something else because when I hit arrow keys it just exits. |
| 00:43.35 | jaytee | cool! I got the C# Asterisk.NET libs working so I can make one phone call another. Woohoo! |
| 00:44.26 | *** join/#asterisk cods (n=cods@rsbac/developer/cods) |
| 00:48.01 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 00:49.42 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3a633a25b2e03d90) |
| 00:51.06 | *** join/#asterisk workdraft (n=acxide@203.215.94.239) |
| 00:56.27 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 00:57.10 | *** join/#asterisk chuck (n=charlie@wikimedia/cmelbye) |
| 00:57.33 | hesco | I got a second connection to the console, without the -vv's hoping it might be a quiet place to work and study the documentation. Now I have two CLI consoles giving me too much information. I'd like to dial it back to one or no v's and see it, without interrupting what is going on right now. |
| 00:57.40 | chuck | Is there any way to make X-Lite act smarter? It's trying to tell a remote asterisk server that my WAN IP address is my local one |
| 00:58.01 | hesco | Can the verbosity be reset without stopping the server and restarting it? |
| 00:58.09 | jaytee | sure |
| 00:58.10 | [TK]D-Fender | chuck: no need. you tell * to ignore the IP it sends and jsut look where its sending from. |
| 00:58.16 | jaytee | core set verbose 8 |
| 00:58.22 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:58.24 | jaytee | core set verbose 4 |
| 00:58.26 | [TK]D-Fender | chuck: "nat=yes" , "qualify=yes" for your peer entry |
| 00:58.28 | jaytee | yadda yadda |
| 00:58.38 | chuck | [TK]D-Fender, in sip.conf? |
| 00:58.45 | [TK]D-Fender | chuck: Yes |
| 00:59.24 | chuck | awesome, the second I restarted it worked, thanks again [TK]D-Fender |
| 01:06.50 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:09.55 | *** join/#asterisk chendy (n=chatzill@219.134.30.97) |
| 01:31.28 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:32.31 | chuck | Are there any services to get free phone numbers? |
| 01:32.57 | *** join/#asterisk Bad_Robot- (n=Bad_Robo@cpe-76-173-219-25.socal.res.rr.com) |
| 01:33.59 | jaytee | wow, I was just reading an article where Sergie Brin is saying they're going to shutdown Google due to the overwhelming success of the #asterisk IRC channel. |
| 01:35.39 | stevie_ramjet | jaytee, what? |
| 01:35.51 | stevie_ramjet | was the article in the onion or something? |
| 01:36.23 | jaytee | stevie_ramjet, no, it was a fictitious article that existed only in my warped sarcastic mind :-) |
| 01:36.35 | [TK]D-Fender | chuck: http://www.ipkall.com/ |
| 01:36.49 | stevie_ramjet | jaytee, ha! :) |
| 01:37.54 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:38.03 | jaytee | I mean, I'm just waiting for the day (and I know it'll come) when someone comes in here and asks, "Hey, my wife and I are trying to decide what to have for dinner. She wants chicken and I'm leaning towards meatloaf. What do you all think?" |
| 01:38.10 | chuck | [TK]D-Fender, I tried them, but I just get busy signals when I call my number |
| 01:38.27 | [TK]D-Fender | chuck: They work fine, you just didn't set things up right |
| 01:38.43 | chuck | I'm not sure if I configured it correctly though, the extension I want to call goes in SIP Phone Number and the SIP Proxy is my asterisk server, right? |
| 01:38.58 | [TK]D-Fender | chuck: Thats about right., |
| 01:39.43 | [TK]D-Fender | chuck: you also need "allowguest=yes" and "context=somewhere" under [general] in sip.conf to permit their incoming call. |
| 01:39.53 | chuck | ah! |
| 01:40.02 | [TK]D-Fender | chuck: I'd advise dumping them into a VERY limited context. |
| 01:40.20 | [TK]D-Fender | chuck: And of course enable SIP debug to watch the call attempt to see if you screw up along the way |
| 01:41.00 | chuck | okay, ipkall makes more sense to me now :P |
| 01:42.28 | chuck | hmm |
| 01:42.40 | chuck | [TK]D-Fender, Is it a long distance call if I'm in Minnesota? |
| 01:42.48 | chuck | It's working great by the way, perfect sound quality, I love it |
| 01:43.02 | [TK]D-Fender | chuck: its a DC local number. Do the math |
| 01:43.03 | *** join/#asterisk digime (n=digi@adsl-75-3-206-197.dsl.sndg02.sbcglobal.net) |
| 01:43.23 | [TK]D-Fender | chuck: Well at least their first bunch of area codes... not sure about the others |
| 01:43.26 | digime | hi, is there a time card add on for asterisk 1.4 that lets employees "clock in" and out, and generates reports, etc? |
| 01:43.27 | chuck | doesn't know how the rates all works, but I'm assuming it is. :'( |
| 01:44.01 | [TK]D-Fender | digime: "extensions.conf" |
| 01:44.45 | [TK]D-Fender | digime: * doesn't generalte reports. * processes calls. thats it. Your dialplan is your job to configure |
| 01:44.53 | digime | right |
| 01:45.08 | digime | I can configure the dialplan but I want to have some reporting as well. |
| 01:45.20 | chuck | maybe you can use that AstDB thing I was reading about? |
| 01:45.36 | [TK]D-Fender | chuck: Horrible tool for this job |
| 01:45.38 | digime | what is AstDB? |
| 01:45.54 | chuck | digime, It's a berkely DB that you can manipulate inside of your dialpan |
| 01:45.56 | chuck | *plan |
| 01:45.58 | digime | I figured that asterisk, being as powerful as it is, should be able to handle my need |
| 01:46.07 | chuck | but it's a horrible tool for the job apparently :P |
| 01:46.10 | digime | hmm |
| 01:46.12 | digime | apparently... |
| 01:46.22 | digime | but maybe not? i will look into it |
| 01:46.26 | [TK]D-Fender | digime: * is not a word processor, a spreadsheet, and acconting package, or a video game. |
| 01:46.29 | digime | what is horrible about it, it seems like a good idea |
| 01:46.47 | [TK]D-Fender | digime: * give you the ability to ADD this stuff that you make or find YOURSELF |
| 01:46.56 | jaytee | digime, you could use either mysql or postgre sql and create a custom app in your dialplan to do what you want. |
| 01:47.09 | jaytee | it would be alot of work though |
| 01:47.09 | digime | okay, that sounds more like it |
| 01:47.12 | digime | ok |
| 01:47.17 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 01:47.46 | digime | is there something already on the market for this? I haven't seen anything. I guess there hasn't been enough demand. Seems like a good idea to me, though |
| 01:48.16 | [TK]D-Fender | digime: Plenty of time-card programs out there.. NONE of them involving *. |
| 01:48.31 | [TK]D-Fender | digime: Go find one you think you can integrate or code it yourself |
| 01:48.40 | jaytee | friend of mine does that with a product called Voiceguide and some custom software he's written in .NET. He makes serious money selling the systems and support. |
| 01:49.38 | digime | I'll check i tout |
| 01:51.02 | [TK]D-Fender | I'm out for a bit. |
| 01:51.04 | [TK]D-Fender | BBL |
| 01:51.08 | digime | thanks for your help guys |
| 01:51.17 | digime | I have found some threads where people are trying to do this |
| 01:55.24 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 01:56.00 | digime | who do y'all use for your 800 DID providers? |
| 01:56.17 | digime | I have been using 3U but they have a known DTMF issue and we don't always get our calls! |
| 01:59.16 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:11.12 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:29.08 | chuck | Does Asterisk support mp3 hold music by default |
| 02:32.34 | chuck | Hmm, I can't seem to hear any audio coming from people talking into the IP Kall number :'( |
| 02:36.10 | *** join/#asterisk bijit (n=chatzill@200.122.188.156) |
| 02:37.16 | bijit | anyone can give me directions on where to go (read) how to route voip incoming calls? |
| 02:38.50 | *** join/#asterisk BeeBuu (n=beebuu@218.13.97.130) |
| 02:39.04 | BeeBuu | ~book |
| 02:39.04 | jbot | book is probably probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:39.35 | bijit | without did? |
| 02:51.17 | *** join/#asterisk Segnale007 (n=Pietro@host86-135-19-235.range86-135.btcentralplus.com) |
| 02:52.57 | jaytee | ~itsplist-us |
| 02:52.58 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 02:58.23 | bijit | ~ |
| 02:58.29 | bijit | ~? |
| 03:02.05 | bijit | ~voxbone |
| 03:17.36 | BeeBuu | is asterisk make call one by one when i move .call files to /var/spool/asterisk/outgoing? |
| 03:20.22 | *** join/#asterisk bitfrost (n=bitfrost@190.12.5.106) |
| 03:20.35 | bitfrost | Hello :) |
| 03:22.03 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-31-164.dsl.hrlntx.sbcglobal.net) |
| 03:22.52 | WimpMan | BeeBuu: No, parallel. |
| 03:23.29 | TrentCreek | SERIAL |
| 03:24.39 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.21) |
| 03:26.25 | *** join/#asterisk etfonhomey_ (n=chatzill@mobile-166-214-215-020.mycingular.net) |
| 03:28.21 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 03:29.21 | jaytee | CEREAL? |
| 03:30.24 | jaytee | I don't see anything in the book that specifies whether * handles the call files all at once or one at a time |
| 03:33.13 | bitfrost | I have a little problem, I have a FWT GSM base, but actually my X100p does not recognize the ring, maybe too low volts? |
| 03:36.16 | jaytee | bitfrost, you're plugging a phone into an FXO port? |
| 03:36.37 | *** join/#asterisk fudpucker (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net) |
| 03:36.39 | bitfrost | jaytee of course, thanks for answer |
| 03:37.28 | jaytee | bitfrost, I'm not familiar with a FWT GSM base but phones only get plugged into FXS ports, not FXO. |
| 03:37.44 | bitfrost | I I plugged that directly from my PSTN line it works fine, but not with that FWT terminal |
| 03:38.35 | bitfrost | FWT is like an ATA, it gives Dial tone |
| 03:38.55 | bitfrost | but it is for Cellular Chips |
| 03:39.39 | jaytee | hence the GSM |
| 03:39.45 | jaytee | got a link? |
| 03:40.42 | BeeBuu | TrentCreek: SERIAL? |
| 03:41.00 | TrentCreek | yes..instead of parallel |
| 03:41.33 | BeeBuu | is asterisk make call one by one when i move .call files to /var/spool/asterisk/outgoing? |
| 03:41.46 | BeeBuu | TrentCreek: you agree? |
| 03:48.37 | *** join/#asterisk talntid (n=eric@c-67-185-179-75.hsd1.wa.comcast.net) |
| 03:52.56 | *** join/#asterisk synchris (n=synchris@athedsl-04914.home.otenet.gr) |
| 03:57.30 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:58.23 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-5037a92eb11ce949) |
| 04:04.00 | bitfrost | jaytee, of course the zaptools shows GREEN |
| 04:05.21 | jaytee | bitfrost, great! sounds like you're well on your way to reverse engineering the universe. let us all know how that works out. |
| 04:06.03 | bitfrost | ??? I don´t understand :( |
| 04:06.37 | bitfrost | sorry jaytee if that bottered you |
| 04:06.45 | fudpucker | what is the replacement config option for insecure=very? |
| 04:06.47 | jaytee | it didn't |
| 04:06.59 | jaytee | insecure=invite,port |
| 04:07.10 | fudpucker | thx |
| 04:07.18 | jaytee | yw |
| 04:07.41 | bitfrost | Where I can find more info about that issue? |
| 04:07.51 | jaytee | about what issue? |
| 04:08.47 | *** join/#asterisk illizit (n=cengroba@c-71-206-66-218.hsd1.fl.comcast.net) |
| 04:09.04 | [TK]D-Fender | bitfrost: How well does an analog phone on your GSM device work? |
| 04:09.05 | bitfrost | That my X100p don`t detect the FWT "ring" it just ignore it |
| 04:09.56 | [TK]D-Fender | bitfrost: So are verbose 10, core debug 10 you see nothing for an incoming call? |
| 04:10.23 | bitfrost | I works Ok, when I call the GSM device the analog phone Rings |
| 04:10.45 | bitfrost | No, nothing at all |
| 04:11.01 | bitfrost | it is like It does not receive any signal |
| 04:11.12 | [TK]D-Fender | bitfrost: pastebin your zapata.conf or chan_dahdi.conf (whicever you use), and your extensions.conf |
| 04:11.14 | [TK]D-Fender | ~pb |
| 04:11.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:11.29 | bitfrost | :( maybe a voltage problem |
| 04:11.57 | bitfrost | Ok i Will do that, jbot thanks for the explain |
| 04:12.26 | bijit | ~jbot |
| 04:12.27 | jbot | i heard jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 04:12.40 | jaytee | jbot botsnack |
| 04:12.40 | jbot | thanks, jaytee |
| 04:12.41 | bitfrost | sorry, hahaha that was the bot, I am going nuts |
| 04:13.00 | [TK]D-Fender | ~areyouadog ? |
| 04:13.01 | jbot | Bark! Bark! |
| 04:13.02 | TrentCreek | Wrong jbo! pb is PeanutButter |
| 04:14.11 | bijit | ~bug |
| 04:14.12 | jbot | somebody said bug was n: A son of a glitch. An error in design or programming in hardware or software. Effects range from cosmetic errors to system crash and loss of data. See also Feature. |
| 04:15.26 | bijit | ~wiki |
| 04:15.52 | fudpucker | i am having a problem dialing one of my extensions, it coems back and says:app_dial.c:1450 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 04:16.22 | jaytee | you don't have the device defined in your sip.conf |
| 04:16.27 | fudpucker | it is trhere |
| 04:16.46 | [TK]D-Fender | fudpucker: PASTEBIN is your friend. |
| 04:16.49 | jaytee | pastebin your sip.conf and your extensions.conf MASKING any passwords |
| 04:17.03 | fudpucker | i have another extensions registered on that phone as well and it has no problems. |
| 04:17.06 | fudpucker | ok, give me a sec |
| 04:17.16 | bijit | how does asterisk detect a bad analog line? |
| 04:17.48 | jaytee | bijit, with a red alarm usually |
| 04:18.01 | jaytee | if the line is dead |
| 04:18.14 | Maliuta | bijit: oh, I think I heard this one ... nope, how _does_ asterisk detect a bad analog line? |
| 04:18.16 | Maliuta | :) |
| 04:18.48 | jaytee | oh, how not how does it indicate |
| 04:18.58 | drmessano | So whats a good cli app for combining files under *nix? |
| 04:18.58 | bijit | do I have to add a extra setting for it to jump that one? Until it is fixed? |
| 04:19.05 | bijit | Oh sry my bad. |
| 04:19.08 | Maliuta | drmessano: cat |
| 04:19.17 | bijit | sometimes don't know how to use the right words. |
| 04:19.24 | Maliuta | drmessano: patch? |
| 04:19.36 | drmessano | hmmm |
| 04:19.49 | drmessano | Will cat echo or do I need to pipe? |
| 04:20.04 | jaytee | bijit, if that line is part of a group you can comment it out or remove it from your zapata.conf or chan_dahdi.conf |
| 04:20.06 | drmessano | What I want to do is |
| 04:20.09 | fudpucker | pastebin isn' t responding |
| 04:20.26 | *** join/#asterisk RobH (n=RobH@130-227.127-70.tampabay.res.rr.com) |
| 04:20.28 | [TK]D-Fender | ~pb |
| 04:20.29 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:20.40 | [TK]D-Fender | fudpucker: there's *6* of them. |
| 04:20.51 | jaytee | pastebin.ca is working fine here |
| 04:21.14 | bijit | jaytee: thanks I will go read more about the red alarm.. |
| 04:21.40 | drmessano | echo <beginning of XML file> + macaddress.xml + basephoneconfig.xml + </beginning of XML file> to newfile, for every macaddress.xml in /path |
| 04:22.31 | jaytee | drmessano, makin configs for polycoms? |
| 04:22.37 | drmessano | Linksys devices |
| 04:23.03 | WimpMan | Yes, cat sounds right. |
| 04:23.43 | jaytee | ah, I wrote a script to use sed and just pass the unique info as command line args so it creates the files I need for each phone with all the info in it. |
| 04:23.54 | *** join/#asterisk neoalex (n=chatzill@user-387h2mq.cable.mindspring.com) |
| 04:24.22 | neoalex | hello, what's the most reliable provider I can get a single DID from in the US? |
| 04:24.29 | neoalex | looking to replace stanaphone |
| 04:24.30 | jaytee | AT&T |
| 04:25.08 | neoalex | ok... free DID should've mentioned |
| 04:25.09 | jaytee | or if you mean SIP? |
| 04:25.16 | neoalex | yes, SIP |
| 04:25.16 | jaytee | ~itsplist-us |
| 04:25.17 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 04:26.10 | jaytee | note the words, "starting with the most respected ones". that doesn't necessarily mean most reliable but most people don't respect unreliable providers. |
| 04:27.00 | jaytee | hmmmm, looks like www.pastebin.com is down |
| 04:27.11 | jaytee | but all the others are up |
| 04:27.54 | neoalex | pastebin?!, they do voip now? |
| 04:28.02 | bijit | jaytee: is there a automatic way I can jump the "bad" lines? Or do I have to set it by removing it from my config? |
| 04:28.06 | fudpucker | here is my sip.conf: http://paste.lisp.org/display/71297 |
| 04:28.53 | jaytee | bijit, you can modify your dialplan so it checks if the channel is available or you can remove it from the config. your choice. |
| 04:29.59 | bijit | jaytee: app I should be reading? |
| 04:30.19 | jaytee | ~book |
| 04:30.20 | jbot | hmm... book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 04:30.46 | jaytee | and your zapata.conf or chan_dahdi.conf sample files |
| 04:31.11 | fudpucker | here is my extensions.conf: http://paste.lisp.org/display/71298 |
| 04:31.17 | *** part/#asterisk baliktad (i=baliktad@c-24-16-23-12.hsd1.wa.comcast.net) |
| 04:31.17 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 04:32.13 | bijit | jaytee: "dialplan so it checks if the channel is available" extension.conf? |
| 04:32.29 | [TK]D-Fender | fudpucker: And now the CLI output of your failure and the sip peer dump accordingly... |
| 04:33.04 | [TK]D-Fender | bijit: Yes you have to change your config for this. |
| 04:34.10 | jaytee | bijit, check the book and the channelvariables.txt file in your source tarball for Asterisk and you can figure out how to use the DIALSTATUS variable and CHANUNAVAIL to jump to a different priority but with zap or dahdi channel groups it's still not the best way. the best way is just to remove the offending channel from the group defined in zapata.conf or chan_dahdi.conf. |
| 04:34.33 | fudpucker | sip show peers: 2925/person (Unspecified) D 0 UNKNOWN |
| 04:34.54 | jaytee | phone isn't registered |
| 04:35.05 | [TK]D-Fender | fudpucker: The device has not resgistered and * has no idea where to send the call |
| 04:35.12 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 04:36.35 | bijit | jaytee: and [TK]D-Fender thanks for the explanation |
| 04:37.34 | bitfrost | Hi finally |
| 04:37.45 | bitfrost | http://pastebin.solid-ec.org/?page=view&id=1228106067 |
| 04:37.54 | bitfrost | http://pastebin.solid-ec.org/?page=view&id=1228106214 |
| 04:38.02 | bitfrost | I seems that pastebin.com is down |
| 04:38.07 | jaytee | man, I wonder what the producer was thinking when he signed up for making Lost Boys: The Tribe. I thought most sequels sucked but this one goes far beyond any suckage expectations. |
| 04:38.39 | bitfrost | I used one of free pastebin sites |
| 04:39.14 | bitfrost | member:identifier:[tk]d-fender that are my config files |
| 04:39.34 | jaytee | bitfrost, this is not the freePBX channel btw. |
| 04:39.36 | [TK]D-Fender | FreePBX? Ok, i'm off this one... |
| 04:39.47 | fudpucker | i just power cycled the phone and all is good. byt before, the phone was telling me it was registered |
| 04:40.33 | jaytee | [TK]D-Fender, I saw #include zapata_additional.conf in the first pastebin and a shudder ran down my spine. |
| 04:41.15 | bitfrost | :( ok thanks anyway :) |
| 04:42.58 | *** join/#asterisk robba (n=robert@mail.ampwest.com.au) |
| 04:43.36 | robba | Hi All. |
| 04:43.58 | robba | is there a way to disconnect a sip call from the CLI? |
| 04:44.18 | [TK]D-Fender | robba: |
| 04:44.26 | [TK]D-Fender | robba: "soft hangup [channel]" |
| 04:44.45 | robba | [TK]: tried that |
| 04:44.58 | robba | just returned channel could not be found |
| 04:45.18 | [TK]D-Fender | robba: try showing us your looking at the channel, then trying to hang it up |
| 04:45.23 | *** part/#asterisk bitfrost (n=bitfrost@190.12.5.106) |
| 04:45.50 | robba | well i'll give you a better background. |
| 04:46.04 | robba | we have 2 asterisk servers. |
| 04:46.07 | robba | one local |
| 04:46.08 | [TK]D-Fender | robba: PASTEBIN is all the background I should need |
| 04:46.10 | robba | one external |
| 04:46.24 | [TK]D-Fender | robba: Don't need a story, jsut need the clear output of 2 CLI commands. |
| 04:47.24 | robba | ~pb |
| 04:47.25 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 04:47.39 | jaytee | and then when I was 6 we moved to a new house on the other side of town.......... |
| 04:48.24 | jaytee | ok, I'm outta here. nite all |
| 04:48.51 | bijit | nite man |
| 04:48.59 | robba | http://rafb.net/p/FbkYd886.html |
| 04:49.29 | [TK]D-Fender | robba: "core show channels concise" |
| 04:49.36 | [TK]D-Fender | robba: that is not a channel name you used |
| 04:50.40 | robba | http://rafb.net/p/SmGTjM18.html |
| 04:50.55 | robba | do i use the whole string for the disconnect? |
| 04:51.24 | [TK]D-Fender | robba: 1st parm is the channel |
| 04:51.56 | *** join/#asterisk Pryon (n=Pryon@irc.animalcules.com) |
| 04:52.15 | [TK]D-Fender | robba: up untin the 1st "!" |
| 04:53.47 | chuck | what are some good services like ip kall but are toll free numbers (and probably a pay service) |
| 04:54.53 | robba | thanks heaps. |
| 04:56.03 | [TK]D-Fender | chuck: Depends on your needs |
| 04:56.22 | robba | the sip show channels command seems to get quite a few connections with (None) under the User/ANR heading what does this mean? |
| 04:58.31 | robba | example http://rafb.net/p/ILba9s47.html |
| 04:59.39 | [TK]D-Fender | robba: those are not CALLS. |
| 04:59.45 | [TK]D-Fender | robba: nothing to hang up there |
| 05:00.02 | [TK]D-Fender | robaNow stop using "sip show channels' youa re only getting yourself in trouble |
| 05:00.15 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 05:01.21 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 05:01.36 | robba | [TK]: in those instances when the (None) records are present it still holds open the channel to our telco and we get billed for it. |
| 05:02.01 | mchou | Anyone know any good (but relatively inexpensive) soho ip pbx appliances on the market? |
| 05:02.41 | mchou | something like a home router that doesnt suck power like a computer would |
| 05:03.25 | robba | mchou: OpenWRT on Linksys WRT router running asterisk? |
| 05:03.55 | mchou | robba: umm, most of those routers are no longer available on the market |
| 05:04.16 | mchou | robba: something like that would have been ideal, of course |
| 05:04.42 | robba | mchou: The WRT54GL router is still available |
| 05:05.08 | mchou | robba: lol, dont be expecting to run asterisk on that. not enough RAM |
| 05:05.33 | robba | mchou: how many extensions you wanna run? |
| 05:05.52 | mchou | robba: not many, mainly just sip really |
| 05:05.52 | robba | mchou: i have done it with 4 concurrent calls |
| 05:06.24 | mchou | robba: how did you deal with storage? |
| 05:07.36 | *** join/#asterisk JohnnyBeGood (n=JohnnyBe@c-98-232-40-217.hsd1.wa.comcast.net) |
| 05:07.48 | [TK]D-Fender | mchou: Get a PC Engines board. |
| 05:08.10 | [TK]D-Fender | mchou: http://www.pcengines.ch/alix.htm |
| 05:08.31 | robba | mchou: with OPENWRT it has enough space for asterisk and a basic ( REAL Basic ) web interface, however there is a new ddwrt with asterisk pre compiled thats much easier to use |
| 05:09.22 | robba | mcou: and for extra space, you can connect the openwrt to share on a basic nas or pc |
| 05:12.54 | mchou | alix looks pretty interesting |
| 05:13.09 | drmessano | The dd-wrt mega? |
| 05:13.36 | mchou | mega?? |
| 05:14.36 | drmessano | Yes |
| 05:14.37 | drmessano | mega |
| 05:18.43 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-0eb222a32f3f1e81) |
| 05:22.42 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 05:25.44 | *** join/#asterisk GoRK (n=gork@209.40.175.194) |
| 05:27.18 | GoRK | hello everyone; I am testing out moving from 1.4 to 1.6 and I am trying to figure out the change to Dial option f.. In 1.4 it took a parameter that let me specify outgoing callerid. in 1.6 it seems to take only from the dialplan hint. Is there a way to emulate the 1.4 behavior such as way to override the hint for that extension on the single call? |
| 05:27.57 | [TK]D-Fender | GoRK: Hint's are for presence what does this have todo with CID & Dial? |
| 05:28.33 | GoRK | Good qestion... 1.4 help text for cmd_dial: f(x) - Force the outgoing callerid to 'x'. |
| 05:29.03 | GoRK | 1.6 help text: f - Force the callerid of the *calling* channel to be set as the |
| 05:29.03 | GoRK | <PROTECTED> |
| 05:29.57 | GoRK | for the life of me i cant tell why anybody would change it. moreover there seems to be no other dial option that allows one to force outgoing CID in 1.6 with a similar dialplan technique |
| 05:30.28 | [TK]D-Fender | GoRK: Set it before you dial |
| 05:31.05 | GoRK | the hint or a variable? what is the var? |
| 05:33.03 | [TK]D-Fender | GoRK: No, just set the CID |
| 05:35.55 | GoRK | oh gotcha well that seems obvious. maybe the 1.4 "f" option behavior was kind of useless then |
| 05:36.30 | GoRK | ill give it a shot; thanks |
| 05:41.06 | GoRK | [TK]D-Fender: That works great. Im feeling dumb. You have things in your dialplan for a couple of years and you just expect that you are doing things the right way i guess! |
| 05:45.24 | *** join/#asterisk tessier (n=treed@kernel-panic/sex-machines) |
| 05:45.29 | tessier | Hello all! |
| 05:46.34 | tessier | I just migrated from asterisk 1.4.15 or so to 1.4.22 and everything used to work perfectly but now I get an error: [Nov 30 21:40:13] WARNING[13399]: chan_sip.c:2933 create_addr: No such host: teliax |
| 05:46.56 | tessier | This happens when I dial out. My dialplan contains: exten => _91NXXNXXXXXX,2,Dial(SIP/teliax/${EXTEN:1}) |
| 05:47.11 | tessier | It appears to be treating teliax like a host instead of a SIP device. I have a device named teliax in my sip.conf |
| 05:47.30 | tessier | Did something change in asterisk or have I somehow messed up a config and not realized it? |
| 05:54.19 | [TK]D-Fender | tessier: pastebin is your friend... |
| 05:56.20 | tessier | [TK]D-Fender: http://pastebin.ca/1272062 |
| 05:57.34 | [TK]D-Fender | tessier: Check your DNS on your box, and dumpt your peer to verify |
| 05:59.03 | tessier | [TK]D-Fender: dumpt? |
| 05:59.15 | [TK]D-Fender | dump |
| 05:59.21 | [TK]D-Fender | "sip show peer teliax |
| 05:59.40 | tessier | DNS seems to work fine. host=voip.lax.teliax.com resolves |
| 05:59.58 | tessier | Peer teliax not found. |
| 06:00.02 | tessier | hmm...that's a problem... |
| 06:00.37 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0e4d54e82568fc6d) |
| 06:00.50 | tessier | ah-HAH |
| 06:01.19 | tessier | I commented out a bunch of stuff with # instead of ; in sip.conf who knows how long ago and hadn't reloaded the config in weeks |
| 06:02.14 | *** join/#asterisk suahmed (n=Rubel@58.65.224.5) |
| 06:02.48 | tessier | [TK]D-Fender: Thanks! |
| 06:03.04 | *** join/#asterisk jack_sparo (n=jack_spa@dxb-b136578.alshamil.net.ae) |
| 06:03.18 | [TK]D-Fender | np |
| 06:03.46 | tessier | jack_sparo: How are things over in the United Arab Emirates these days? |
| 06:04.24 | jack_sparo | bad man |
| 06:04.30 | tessier | oh? Why is that? |
| 06:04.53 | jack_sparo | one of the biggest comapnanie here called Emaar |
| 06:05.38 | jack_sparo | its for the govern, the price was for each 36 dirhams around 10$, it is now 1.90 dirhams |
| 06:05.46 | jack_sparo | more explanations |
| 06:05.55 | jack_sparo | firing people from work |
| 06:06.10 | jack_sparo | all prices still same, i mean rent food cars and stuff |
| 06:06.16 | jack_sparo | and people loosing their jobs |
| 06:06.39 | jack_sparo | tessier, i have a problem with my zap, are u an expert/ |
| 06:06.40 | jack_sparo | ? |
| 06:08.33 | jack_sparo | im getting all circuits are busy now on outgoing calls using zap, but trunks and routes and i can see the 4 channels in the FOP |
| 06:09.44 | tessier | jack_sparo: Unfortunately, no. I really try to avoid zap stuff. I used to do a lot with it but it was always a real pain to set up. I don't do it often enough to maintain proficiency. |
| 06:10.09 | tessier | jack_sparo: We have some similar problems here. The whole world has economic problems like that it seems. |
| 06:10.39 | tessier | jack_sparo: My wife is from Vietnam. They have huge inflation there right now. And of course jobs are being lost as less money comes in as the US buys less from them. |
| 06:10.57 | jack_sparo | where u from tessier |
| 06:11.00 | [TK]D-Fender | jack_sparo: pastebin the failed call attempt along with a channel dump, and your zapatal.conf. |
| 06:11.16 | tessier | jack_sparo: I am from San Diego, California, USA |
| 06:11.18 | jack_sparo | ok [TK]D-Fender, i will do it |
| 06:11.30 | tessier | [TK]D-Fender is the asterisk rock star in here tonight :) |
| 06:11.37 | jack_sparo | :D |
| 06:11.58 | tessier | I've been using asterisk for 4 or 5 years but still occasionally get stuck on something dumb. The past year or two I haven't done enough with it to maintain proficiency. |
| 06:12.07 | tessier | I've got a bunch of asterisk systems running and...they all just run! |
| 06:12.27 | tessier | Only when I mess with something do I have a problem. When asterisk breaks it's because I broke it. :P |
| 06:13.25 | *** join/#asterisk moy (n=moy@189.169.61.116) |
| 06:14.03 | jack_sparo | pastebin is not loading :| |
| 06:14.36 | [TK]D-Fender | ~pb |
| 06:14.37 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 06:14.39 | *** join/#asterisk t0rrieri (n=torrieri@nelug/crew/torrieri) |
| 06:14.46 | [TK]D-Fender | jack_sparo: 5 others to choose from. |
| 06:14.54 | [TK]D-Fender | Hurry up, I'm about to checkout for the night |
| 06:15.13 | jack_sparo | http://rafb.net/p/80vcar52.html |
| 06:16.57 | *** part/#asterisk BeeBuu (n=beebuu@218.13.97.130) |
| 06:17.42 | jack_sparo | [TK]D-Fender got the link? |
| 06:19.41 | [TK]D-Fender | jack_sparo: If you're going to ask for support for a FreePBX system in here, please have the brains to include all of the INCLUDED files.... |
| 06:19.49 | [TK]D-Fender | #include zapata-channels.conf |
| 06:19.54 | [TK]D-Fender | #include zapata_additional.conf |
| 06:20.33 | jack_sparo | moment plz |
| 06:21.11 | jack_sparo | nothing there dude |
| 06:21.14 | jack_sparo | empty |
| 06:21.54 | [TK]D-Fender | jack_sparo: then you have no zaptel channels at all |
| 06:22.30 | jack_sparo | what to do? |
| 06:23.15 | [TK]D-Fender | jack_sparo: You have no zap channels defined. what you need to do is go CONFIGURE TEHM. |
| 06:23.35 | jack_sparo | this was my ques |
| 06:23.37 | jack_sparo | how? |
| 06:23.44 | jack_sparo | i can see them in the FOP |
| 06:23.50 | jack_sparo | they exist there |
| 06:24.00 | [TK]D-Fender | jack_sparo: meaningless garbage |
| 06:24.10 | [TK]D-Fender | jack_sparo: forget FOP, * see nothing' |
| 06:24.28 | [TK]D-Fender | jack_sparo: you do not have any defined channels. Go set them up |
| 06:24.38 | jack_sparo | pbx*CLI> zap show status |
| 06:24.39 | jack_sparo | Description Alarms IRQ bpviol CRC4 |
| 06:24.39 | jack_sparo | Wildcard TDM410P Board 1 OK 1 0 0 |
| 06:25.18 | [TK]D-Fender | jack_sparo: Board doesn't matter, you have defined no CHANNELS |
| 06:25.28 | jack_sparo | how to define that/ |
| 06:25.32 | jack_sparo | i dnt know |
| 06:25.55 | [TK]D-Fender | jack_sparo: Go to #freepbx and they'll walk you through it. |
| 06:26.09 | jack_sparo | they are all dead |
| 06:26.19 | jack_sparo | why it is so hard to ask someone question man |
| 06:26.32 | [TK]D-Fender | jack_sparo: Your GUI owns your ass. Go learn how it works. As in their channel, check their boards, etc |
| 06:26.42 | [TK]D-Fender | jack_sparo: because no GUI's are supported here. |
| 06:26.54 | jack_sparo | pfffffffff |
| 06:29.38 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 06:29.57 | [TK]D-Fender | jack_sparo: http://www.freepbx.org/support/documentation/administration-guide/interfacing-to-a-pstn |
| 06:30.07 | [TK]D-Fender | thats all for tonight. |
| 06:30.09 | [TK]D-Fender | later all |
| 06:39.33 | *** join/#asterisk loconut (n=jessica@webtrotter.com) |
| 06:40.40 | loconut | hello. i've got a double nat scenario. one nat i have control of (server side) the other i do not. the server has nat=yes, externip=(outside ip), im also forwarding all the the rtpstart-rtpend ports, yet i get no audio either direction. any ideas? |
| 06:43.21 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 06:43.55 | loconut | hello joako |
| 06:44.16 | joako | Good Morning! |
| 06:44.28 | loconut | i suppose it is morning |
| 06:44.32 | loconut | only just here. |
| 06:45.04 | loconut | i hope you have more solutions than problems, otherwise you and i will be waiting a while it seems... |
| 06:45.42 | joako | My IRC client auto-opens #asterisk... I hope I don't have any problems |
| 06:45.53 | loconut | well, i do =) |
| 06:47.06 | joako | Well feel free to ask |
| 06:47.28 | loconut | well, if it will please, i'll re-send my question from buffer. |
| 06:47.32 | loconut | hello. i've got a double nat scenario. one nat i have control of (server side) the other i do not. the server has nat=yes, externip=(outside ip), im also forwarding all the the rtpstart-rtpend ports, yet i get no audio either direction. any ideas? |
| 06:47.58 | loconut | i've followed http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html that page |
| 06:48.50 | joako | Put asterisk boxes on each end and use IAX between them. Problem solved. You can run Asterisk on a Linksys router. |
| 06:49.23 | loconut | i've got a GL thats about out of space, thought about opensips/openser. |
| 06:50.30 | loconut | unfortunately, this is a low budget operation ;) |
| 06:50.54 | loconut | it's my home asterisk server and im trying to get a phone to work on the private network at work so i dont need to pay for cell time. |
| 06:51.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 06:51.32 | loconut | joako: what could i be missing? |
| 06:51.38 | loconut | aside from the dual asterisk setup |
| 06:52.24 | loconut | :) |
| 06:53.00 | joako | You said you do not have access to one of the NAT devices, can you work with its administrator to open the same ports you did on your end? |
| 06:53.14 | joako | I would think doing so might solve your issues.... |
| 06:54.35 | loconut | well, in the short term, i can do that, but had hoped for another phone. thanks for at least talking to me joako. |
| 06:55.23 | loconut | i do appreciate your time |
| 06:55.47 | joako | If you can replace the phone(s) look for those with IAX support |
| 06:56.35 | loconut | no iax firmware for Cisco 7960s right? |
| 06:56.49 | joako | No |
| 06:57.28 | loconut | c'est la via ;) |
| 06:57.48 | loconut | i'll hack around a bit and see if i can do some more debug peer and figure out what the dealie is |
| 06:58.24 | loconut | anyway, time to move along ;) |
| 06:58.27 | loconut | thanks again |
| 07:08.35 | *** join/#asterisk ixx (n=ixx@70.114.153.233) |
| 07:09.16 | ixx | Is there a solution for Dial() timeout when the VOIP provider is answering the call while still dialing? |
| 07:09.24 | ixx | Some providers just give |
| 07:09.54 | ixx | 180 Ringing |
| 07:10.37 | joako | If they send 180 ringing they are not answering the call and correctly passing the progress |
| 07:10.52 | ixx | sorry that was not suppose to be split... |
| 07:11.05 | ixx | yes the ones working correctly send 180 Ringing |
| 07:11.19 | joako | I've seen some numbers not correctly send answer supervision... 800-PICK-UPS used to do this, did not "answer" while you were on the IVR, only when an agent answers |
| 07:11.54 | joako | Ok, then if the provider just "answers" when you send them the call the only thing you can do is ring detection, which from what I have heard is not reliable |
| 07:11.54 | ixx | some others seem to not do that and I get a 200 OK instead |
| 07:11.59 | ixx | while it is still ringing |
| 07:12.11 | ixx | sipphone/gizmo for example does this |
| 07:12.16 | ixx | it does send SIP/2.0 100 Giving a try. before that |
| 07:12.33 | ixx | telasip and teliax both use 180 Ringing first and only 200 OK after someone picks up |
| 07:13.08 | ixx | darn :( |
| 07:13.20 | ixx | I was hoping to avoid ring detection... |
| 07:18.49 | liri | morning |
| 07:18.51 | ixx | Teliax uses 183 Session Progress instead of 180 Ringing... or maybe the timeout (SIP CANCEL) was sent before the Ringing could arrive |
| 07:19.47 | ixx | ok this seems to be a known issue with gizmo at least - http://www.voipuser.org/forum_topic_8560.html |
| 07:20.45 | ixx | as far as I know something similar is happening with voicepulse |
| 07:21.19 | joako | FWIW those providers aren't using an RFC-compliant SIP implementation |
| 07:21.52 | joako | A SIP call will normally proceed 100 Trying --> 180 Ringing --> 200 OK |
| 07:21.59 | ixx | yes... unfortunately the person I was trying to help is already on voicepulse |
| 07:22.35 | joako | I could see if say you dialed an IVR why it would matter if it went straight to 200... |
| 07:22.40 | ixx | oh well. provider issue... passed info onto them. i hate ring detection and vm detection in asterisk |
| 07:22.54 | joako | FWIW I don't like Voicepulse |
| 07:23.18 | joako | Besides the call quality issues, they told me to run "PingPlotter" (Windows program) on a colocated server |
| 07:23.24 | ixx | they were good early on... like well 5 years ago |
| 07:24.22 | joako | when I informed them it was a Windows program and we had colocated linux servers they suggested there was "some emulator" that might let it run... sure in X and last time I checked it was not advised to run asterisk and X on the same machine (I don't know... don't install X on my servers to begin with) |
| 07:24.31 | joako | And their thing about the channels is so silly |
| 07:24.45 | joako | 1 account has 5 channels or whatever the number is |
| 07:25.00 | joako | Buy 500 DID and you still have those 5, they make you pay extra for each channel |
| 07:25.17 | joako | Or put each DID in its own account each DID gets 5 channels (or whatever that number is) |
| 07:26.07 | *** join/#asterisk enyawix (n=enyawix@68-114-138-145.dhcp.jcsn.tn.charter.com) |
| 07:27.21 | drmessano | ROFL |
| 07:27.32 | drmessano | They wanted to run PingPlotter in an emulator |
| 07:27.44 | drmessano | Dumbasses |
| 07:28.36 | enyawix | good Asterisk distro? I am looking at AstLinux and PBX in a Flash |
| 07:28.53 | ixx | Backgrounddetect looks like it may help |
| 07:29.21 | tessier | enyawix: asterisk.tar.gz from digium.com is the best asterisk distro |
| 07:31.38 | enyawix | tessier thanks looking |
| 07:31.54 | tessier | enyawix: That's a joke... |
| 07:31.57 | tessier | But not really |
| 07:32.13 | liri | can I build asterisk with app_meetme support but without zaptel? |
| 07:32.16 | joako | Well its a Win32 app so it runs in WINE, an opensuse implementation of the Win32 API on Linux |
| 07:33.20 | joako | Wine Is Not an Emulator... "some emulator" is VoicePulse's words... ironic part is this was pretty soon after they renamed the service to "VoicePulse Connect FOR ASTERISK" |
| 07:33.25 | enyawix | and i wend looking |
| 07:33.33 | enyawix | went* |
| 07:34.38 | enyawix | joako no point |
| 07:34.45 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:35.03 | joako | liri, no. But you can use zt_dummy without having a zaptel card |
| 07:35.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:35.41 | liri | I have the ztdummy module already compiled |
| 07:37.07 | liri | joako: when I specify ./configure --with-zaptel=DIR which dir do I need to provide? as I already have the zt_dummy driver installed (but I don't have the original sources) |
| 07:37.31 | enyawix | anyone used AstLinux or PBX in a Flash? |
| 07:40.45 | enyawix | anyone running other servers on their asterisk box? |
| 07:41.03 | enyawix | apache mysql postfix etc? |
| 07:41.11 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 07:41.17 | hesco | . |
| 07:42.14 | enyawix | <PROTECTED> |
| 07:42.19 | hesco | On voip-info's Asterisk+cdr+csv page it states: "By default, Asterisk generates CDR records in comma-separated text files " How is this default changed? |
| 07:47.37 | joako | liri: either use the asterisk by the same person that packaged your zaptel or build zaptel from source after you remove the package |
| 07:47.48 | joako | or if you just deleted zaptel source, recompile |
| 07:48.47 | joako | hesco vi /etc/asterisk/cdr.conf |
| 07:50.30 | *** join/#asterisk moy (n=moy@187.133.21.100) |
| 07:58.17 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 08:00.10 | IPkaf | hi |
| 08:07.09 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 08:11.11 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
| 08:14.58 | IPkaf | i just find an intersting software with is managing phone calls on cybercafe |
| 08:16.15 | IPkaf | and the name is : ipcash |
| 08:16.44 | IPkaf | the only probleme is : it compatible ony on windows environement |
| 08:17.04 | IPkaf | or i need that soft my linux os |
| 08:17.46 | IPkaf | is there any existing software wich is compatible with linux environement |
| 08:17.48 | IPkaf | ? |
| 08:25.35 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 08:30.47 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
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| 08:49.24 | hesco | joako: I've got * on three different boxes, I've checked multiple times and multiple ways but still can not figure out why I'm only getting cdr for incoming calls, but not for outgoing calls. Any ideas? |
| 08:49.42 | hesco | This problem is only for one box. |
| 08:50.37 | *** join/#asterisk chendy (n=chatzill@219.134.30.97) |
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| 09:33.06 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
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| 09:56.40 | key2 | is there any softphone for standardist where its possible to do a drag and drop for transfering calls... ? |
| 10:00.46 | liri | pbx_exec launches the AGI in the foreground so that until it finishes the dialplan progress can't continue and it's locking the channel... is there a background (possibly forking) function to do that? |
| 10:01.41 | jql | you could have the AGI itself fork |
| 10:01.57 | jql | but that would only work based on what interaction you want it to have with * |
| 10:06.09 | *** join/#asterisk ZachFlem (n=zach@d58-105-178-246.dsl.vic.optusnet.com.au) |
| 10:06.54 | ZachFlem | hey folks, im looking to build a cluster to host my voip server, can anyone help me find me feet? |
| 10:08.48 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:11.19 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 10:20.35 | angryuser | ZachFlem: hello, what are you looking to achieve ? |
| 10:21.10 | liri | jql: uhmm, I modified app_meetme to run pbx_exec() for an AGI script on an event, the AGI script runs a bash script which waits for user input and then completes. this makes the participant who issued the event wait until the input is typed in. |
| 10:21.16 | ZachFlem | high availabilty hosted PBAX |
| 10:21.41 | liri | jql: I tried running the bash script in the background via & but that didn't solve it. any recommendations? |
| 10:22.01 | ZachFlem | I work for a communications company in a medium sized city, we want to experiment with a cluster for our hosted systems |
| 10:22.02 | jql | the bash script would need to run another script using & |
| 10:22.26 | ZachFlem | any suggestions? |
| 10:27.32 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 10:28.30 | JonOnt | Hey guys, im following a guide over at powerpbx.org and I guess something was ommited from the guide, how do I start the asterisk manager interface? |
| 10:31.47 | liri | jql: that doesn't work |
| 10:38.13 | *** part/#asterisk ZachFlem (n=zach@d58-105-178-246.dsl.vic.optusnet.com.au) |
| 10:48.21 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
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| 11:06.21 | *** join/#asterisk benneton (n=DELL@adsl-16-47.teol.net) |
| 11:06.33 | benneton | Hi guys! |
| 11:06.49 | benneton | I have question about licencing. |
| 11:07.16 | benneton | Can I sell asteriks based products (PBX) |
| 11:07.18 | benneton | ? |
| 11:07.23 | benneton | asterisk |
| 11:07.24 | benneton | :D |
| 11:08.23 | benneton | or I need to buy licenced software? |
| 11:09.12 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 11:14.10 | *** part/#asterisk benneton (n=DELL@adsl-16-47.teol.net) |
| 11:17.50 | *** join/#asterisk benneton (n=DELL@adsl-16-47.teol.net) |
| 11:18.44 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 11:29.25 | linuxstb | benneton: Yes - Asterisk is licensed under the GPL, so you can sell it as long as you comply with that license - http://www.asterisk.org/about (I am not a lawyer - you should read and understand Asterisk's license yourself) |
| 11:44.40 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:47.26 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 11:47.27 | xrmx__ | should i worry about chan_sip.c:3989 copy_via_headers: No header field 'Via' present to copy? |
| 11:47.29 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
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| 11:53.43 | *** join/#asterisk donnib (n=aaaa@0x555281d0.adsl.cybercity.dk) |
| 11:53.51 | donnib | what commands do i need to run to start zaptel when i don't have hardware |
| 11:53.59 | donnib | i need the conferencing options to work |
| 11:54.06 | donnib | <PROTECTED> |
| 11:55.05 | pokui | hi all, we have an inhouse asterisk + sangoma solution to terminate voip lines. "management" wants now to invest in as turn-key solutions as possible and are pushing that we use quintum boxes (see quintum.com) with the ss7 addons. does anyone have pointers to useful reviews while I google? or pointers to more asterisk based (read open) turnkey solutions? |
| 11:55.25 | *** part/#asterisk liri (n=lirant@bzq-179-128-126.static.bezeqint.net) |
| 11:55.31 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 11:55.37 | angryuser | donnib: hi if zaptel is not loaded on boot try "modprobe zapter" and "ztcfg -v" and start * |
| 11:55.48 | angryuser | modprobe zaptel* |
| 11:56.01 | farkus_ | I want to capture the logging that normally goes to TTY9 to a file, so I added verbose to logger.conf. This doesn't capture the logging from AGI scripts, though. Is there a way to redirect the AGI script output? |
| 11:56.53 | *** join/#asterisk iurz (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 11:59.10 | donnib | angryuser thx. i needed to run modprobe ztdummy as well |
| 11:59.17 | donnib | how do i make it work on boot ? |
| 11:59.51 | *** part/#asterisk benneton (n=DELL@adsl-16-47.teol.net) |
| 12:01.08 | angryuser | donnib: do a simple bash script for example run on boot ;) |
| 12:02.16 | donnib | hmm ok i'll look on the net how to do that. thx |
| 12:02.27 | angryuser | or try to do "make config" in your zaptel source dir, it will install startup scripts |
| 12:03.31 | *** join/#asterisk Karlitoo (n=proscom@213.137.110.67) |
| 12:03.38 | donnib | another question. normal all trafic will go thru the asterisk server. i know you can make clients connect to eachother by calling on the same extension by just using the asterisk as registration and to figure out the ip addess. how are u guys running your asterisk ? |
| 12:04.01 | donnib | is it totally normal and recommended to run all trafic thru the asterisk server ? i mean the speach as well ? |
| 12:04.09 | donnib | angryuser: thx will try that |
| 12:04.22 | iurz | hi to all |
| 12:04.56 | iurz | i got a pentium 3 pc where asterisk runing on it |
| 12:05.02 | angryuser | donnib: all traffic run through * if you have set canreinvite=no for all peers |
| 12:05.16 | iurz | i can do 5 simultaneous using that server |
| 12:05.44 | donnib | yes i have canreinvite=no but is that recommended ? |
| 12:06.09 | donnib | i am just trying to learn if that is the recommended setup or people do this differently |
| 12:06.22 | iurz | now i want to sign lots sip provider on my asterisk |
| 12:07.10 | iurz | if i do it how many sip provider simultaneous can it be capable ? |
| 12:07.27 | angryuser | donnib: "recommended" is not the term, if you set it to yes the rtp traffic will go directply from one peer to another, but you need to be sure that peer1 will know the way reaching peer2 |
| 12:08.34 | joat | donnib: depending on your network config, it may not work (i.e., NAT complicates things) |
| 12:09.07 | angryuser | donnib: also if you have nat involved for example peer1>>nat>>asterisk >>peer2 if you set canreinvite=yes in that case peer2 will not be able to reach peer 2 unleast you have the sip proxy installed |
| 12:09.12 | donnib | let's say that i am running an internal server on my own network where all clients are internal so there is no NAT then what would you recommend ? |
| 12:09.29 | angryuser | oh sorr it's the peer1 not 2 ;) |
| 12:09.39 | donnib | yes i understand the problem but i gues since it's internal there won't be such problems. |
| 12:09.41 | joat | as long as you don't care about record keeping... |
| 12:09.50 | donnib | what about voicemail ? |
| 12:09.57 | joat | what about it? |
| 12:10.03 | donnib | would that work ? |
| 12:10.17 | joat | yes |
| 12:10.34 | angryuser | iurz: without transcoding more, with, less ;) measure |
| 12:10.36 | joat | reinvite means the call is initiated via the asterisk box but then is moved off of it |
| 12:10.38 | donnib | if i run canreinvite=yes would i be able to log the calls that are in progress and get statistics on them ? |
| 12:10.52 | donnib | joat: yes i understand. |
| 12:11.07 | Karlitoo | hi, I know that most people don't use h323 but I need the ability to call from sip to h323 and my problem is that I can make the call and the connection is made but there is no sound... I get an error from asterisk WARNING[5817]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_3 |
| 12:11.16 | joat | if the call doesn't go through, it gets passed to voicemail (if you have it configured) |
| 12:11.25 | Karlitoo | I would really apreciate the help |
| 12:11.27 | donnib | oh ok... |
| 12:12.01 | angryuser | donnib: how many clients ? |
| 12:12.06 | joat | i'm not sure about statistics though... call termination wouldn't involve the end points |
| 12:12.13 | joat | would it? |
| 12:12.20 | donnib | but the progress and stats would that work with canreinvite=yes ? i mean does the phones tell asterisk when they are done with the call and when they are in a call and so on ? |
| 12:12.25 | donnib | angryuser: about 40 |
| 12:13.01 | donnib | angryuser: are u thinking on the load on the server when u have many people and use canreinvite=no ? |
| 12:13.02 | angryuser | so you dont need to optimise the traffic i suppose ? just set to "no" and forget about it |
| 12:13.35 | angryuser | donnib: yes |
| 12:14.05 | iurz | just an precision i just my asterisk just for signing sip provider account |
| 12:14.29 | iurz | just an precision i just use my asterisk just for signing sip provider account |
| 12:14.31 | donnib | ok thx for the help guys |
| 12:14.46 | iurz | how many simultaneous call can it possible ? |
| 12:14.59 | angryuser | iurz: how many you want ? |
| 12:15.36 | iurz | 50 calls |
| 12:15.43 | iurz | is it possible ? |
| 12:15.53 | iurz | using a pentium 3 pc ? |
| 12:15.55 | angryuser | iurz: do you need transcoding ? |
| 12:16.28 | iurz | n o transcoding |
| 12:17.27 | angryuser | iurz: it's a p3 what ? 600 more ? |
| 12:17.50 | iurz | sorry |
| 12:18.07 | iurz | i don't understand ur qUESTION |
| 12:18.33 | angryuser | it's a pentium 3 450mHz more ? |
| 12:19.08 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096590388.dsl.bell.ca) |
| 12:19.55 | angryuser | try to generate 50 call's and see for yourself, 50 call's with p3 could be not enough |
| 12:20.49 | iurz | r u from france ? |
| 12:21.17 | *** join/#asterisk Hick0rd (n=Hick0rd@196.218.200.202) |
| 12:21.29 | angryuser | maybe ;) |
| 12:21.40 | Hick0rd | hello, any good tutorial for howto configure asterisknow. |
| 12:25.12 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 12:25.22 | JonOnt | Anyone awake? |
| 12:26.32 | angryuser | Hick0rd: i hear only about a book http://www.packtpub.com/asterisknow/book/ |
| 12:26.37 | angryuser | heard* |
| 12:29.20 | coppice | does anyone here use asterfax? |
| 12:29.20 | Hick0rd | angryuser, great. let me check it out. |
| 12:29.45 | *** join/#asterisk SibRphrek (i=SibrPhre@cpe-67-243-43-136.nyc.res.rr.com) |
| 12:29.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:30.42 | JonOnt | Hi guys. I'm trying to setup my autoprovisioning script, but its not working, the phone waits about 60 seconds at the point where it updates the config, but never goes into the setup app, any one feel like helping me debug? |
| 12:33.26 | *** join/#asterisk MrNeutr0n (n=DrLexus@c-98-199-63-121.hsd1.tx.comcast.net) |
| 12:34.03 | JonOnt | Ok, i fixed the ftp server |
| 12:34.48 | MrNeutr0n | Does anyone know how I can get a 7960 to show the name of the extension I'm dialing? |
| 12:35.13 | MrNeutr0n | Like, when dialing an extension, it says "Line 1\nTo\nXXXX" |
| 12:35.33 | MrNeutr0n | but there's nothing next to "To" |
| 12:35.49 | Hick0rd | angryuser, It's not free. |
| 12:35.49 | MrNeutr0n | sorry to be a pain about something like that - I don't even know if this is the right place to ask |
| 12:36.54 | angryuser | Hick0rd: try the asterisknow channel, or google |
| 12:38.22 | MrNeutr0n | angryuser, I've tried on google for quite a bit, but I can't seem to figure out a decent query for something like this |
| 12:38.36 | MrNeutr0n | 7960 dialed extension to field |
| 12:38.42 | MrNeutr0n | any ideas? |
| 12:38.48 | iurz | signing an sip accound providing from an sip provider on my asterisk server, is it the same as using an extension which is creating from my asterisk server , |
| 12:39.08 | iurz | my question is the sip account that i sign on is it acting same as one of extension that is generated by asterisk server ? |
| 12:40.25 | angryuser | MrNeutr0n: i am not sure to understand what do you want to do, provide more info |
| 12:41.11 | angryuser | maybe query some 'destination' in db ? |
| 12:41.30 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.195.82) |
| 12:41.48 | MrNeutr0n | angryuser, sure thing! I would like to be able to press "NewCall", then "6901", then "Dial" and as it is ringing ext. 6901, it shows "To: Gary" then the next line "6901" |
| 12:42.12 | MrNeutr0n | I thought about that - querying a database, but I seem to remember getting it setup before without doing so |
| 12:42.23 | MrNeutr0n | I was thinking it might be a function of SIP |
| 12:42.35 | MrNeutr0n | for instance, if 6901 is registered with the name "Gary" somewhere in there |
| 12:43.01 | angryuser | Sounds like a phonebook |
| 12:43.03 | MrNeutr0n | then when _I_ dial it, it might be able to tell me that is the name assoc. with that extension - but I am just speculating and I don't know how a 7960 implements it |
| 12:45.05 | angryuser | MrNeutr0n: it is usefull when you receive call, some phones do association with the internal phonebook |
| 12:46.01 | angryuser | "when you call out" |
| 12:46.05 | MrNeutr0n | angryuser: I actually tried experimenting with that for quite a while, but I was unable to reproduce it |
| 12:46.20 | MrNeutr0n | right - so you're saying when someone calls me first, it stores their name with the extension |
| 12:46.24 | JonOnt | Any one familiar with aastra xml startup script, I got everything setup, but now I need to provision an extention so that I have one to login to |
| 12:46.33 | MrNeutr0n | so that next time i call the extension, it reads back the stored name |
| 12:46.40 | MrNeutr0n | but i couldn't get the 7960 to do that |
| 12:46.55 | angryuser | MrNeutr0n: no this in fo is sent by the * sip |
| 12:47.03 | MrNeutr0n | oh really? |
| 12:47.20 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:47.25 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 12:47.29 | MrNeutr0n | oh yes, I see what you're saying - but I am talking only about internal extension-to-extension dialing |
| 12:47.35 | angryuser | MrNeutr0n: you can do whatever you want, read about channel variables |
| 12:48.11 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:51.12 | angryuser | MrNeutr0n: you can send message to a phone lcd screen with that info, if the phone support it |
| 12:51.55 | MrNeutr0n | angryuser, exactly! Do you know if the 7960 supports it and how I might set that up? |
| 12:52.00 | *** join/#asterisk invalidrecord (n=fares@87.115.79.148) |
| 12:52.12 | JonOnt | arg, forgot to apply settings |
| 12:54.57 | angryuser | MrNeutr0n: Usage: SEND TEXT "<text to send>" for example if we have a db entry in astdb like this users/6901="Gary 6901" you need to do SEND TEXT "${DB(users/6901)}" i am not sure if it will work, and about the syntax or how to do it with mysql, test and read |
| 12:55.31 | MrNeutr0n | hmm... Thanks angryuser, you've given me a couple of ideas. |
| 12:58.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:02.19 | *** join/#asterisk sasargen (n=chatzill@173.100.87.21) |
| 13:02.49 | Karlitoo | hi, I know that most people don't use h323 but I need the ability to call from sip to h323 and my problem is that I can make the call and the connection is made but there is no sound... I get an error from asterisk WARNING[5817]: chan_ooh323.c:1054 ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_3 |
| 13:03.00 | Karlitoo | I would really apreciate the help |
| 13:03.19 | invalidrecord | hi having difficulty with getting a realtime extension if i set extension to do say playback pbx-invalid it works but if i do Dial SIP/1001 it wont: http://pastie.org/327708 if anyone can help would be very great full |
| 13:05.15 | *** join/#asterisk macros73_ (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
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| 13:18.06 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
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| 13:20.49 | key2 | if I want to add "00" in front of a callerid, how can I do it ? |
| 13:21.23 | *** join/#asterisk qdk (n=qdk@94.191.252.224.bredband.3.dk) |
| 13:22.37 | JonOnt | I am sooo close to making this work |
| 13:23.03 | angryuser | key2: look how to use Set() and set you callerid |
| 13:24.50 | *** join/#asterisk feeds (n=feeds@85-135-237-245.adsl.slovanet.sk) |
| 13:26.35 | JonOnt | I need help with Aastra 57i |
| 13:27.10 | key2 | angryuser: yeah but what is the var for the current callerid in the dialplan |
| 13:27.12 | JonOnt | I have the tftp setup and the auto provisioning script, when i boot the phone is asks me for a exention and password |
| 13:27.55 | angryuser | key2: http://www.voip-info.org/wiki-Asterisk+variables |
| 13:28.09 | JonOnt | But when i put in the password, the phone says Data Timeout and the CLI shows connect attempt from 127.0.0.1 unable to athenticate |
| 13:28.41 | *** join/#asterisk kerframil (n=kerframi@gentoo/user/kerframil) |
| 13:29.06 | JonOnt | I would apreciate any help at all, it SOO close to working |
| 13:30.32 | invalidrecord | JUon |
| 13:30.48 | invalidrecord | JonOnt: i have them whats the issue may be able to help |
| 13:31.09 | invalidrecord | nm i scrolled |
| 13:31.43 | JonOnt | invalidrecord, well, first thing that seems bad, is the CLI says connect attempt from 127.0.0.1 unable to athenticate |
| 13:31.51 | JonOnt | ahh |
| 13:32.17 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 13:32.35 | invalidrecord | mm why is the localhost trying to connect? |
| 13:32.46 | invalidrecord | shouldnt the phone be a seperate ip? |
| 13:33.03 | JonOnt | invalidrecord, maybe not, its probably the setup script |
| 13:33.12 | invalidrecord | i dont know the way the3 autoprovision works thats for next job |
| 13:33.41 | JonOnt | theres this auto provisioning script, asks you for a exten and voice mail pass, it then is supposed to make a config for you |
| 13:33.57 | JonOnt | and then reboot the phone |
| 13:33.59 | invalidrecord | JonOnt: got link? |
| 13:34.09 | JonOnt | but I think my script isnt working.. |
| 13:34.23 | invalidrecord | i would have thought the phone would still make the request |
| 13:34.29 | invalidrecord | on reboot |
| 13:34.56 | JonOnt | http://www.aastratelecom.com/cps/rde/xbcr/SID-3D8CCB6A-B14651E6/03/XML_API_PA-001008-03-REV00_2.4.0_0811.zip |
| 13:35.17 | invalidrecord | cheers |
| 13:35.36 | JonOnt | invalidrecord, it does make the request, but through a xml script running on httpd, so the manager sees it as local cause the script is local |
| 13:36.05 | invalidrecord | ahh ok make3s sense |
| 13:36.14 | JonOnt | there must me somewhere where i setup the manager pass and stuff, but I thought I did that |
| 13:36.57 | *** part/#asterisk suahmed (n=Rubel@58.65.224.5) |
| 13:37.01 | invalidrecord | http://pastie.org/327708 anyone see why this isnt working ? |
| 13:37.36 | *** join/#asterisk pawpro (n=IRC@213.166.12.34) |
| 13:37.36 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:37.53 | invalidrecord | the extension works if i use playback but not with the sip dial |
| 13:39.15 | pawpro | Hi everybody. Could you tell me how can i remove the delay from running command like asterisk -rx "sip show channels"? It will stall for 1 sec at the end. I assume is "running las minute cleanups". |
| 13:40.12 | Corydon76-dig | pawpro: upgrade to the latest version |
| 13:40.17 | pawpro | 1.6 |
| 13:40.22 | pawpro | ? |
| 13:40.49 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:40.58 | Corydon76-dig | and it doesn't stall for a second. It stalls for 500ms |
| 13:41.06 | pawpro | :) |
| 13:41.22 | Corydon76-dig | It's giving the command time to complete |
| 13:41.28 | pawpro | just round up in case some human beings are present |
| 13:42.14 | Corydon76-dig | Because the command is executing remotely, there is no way for it to know when a command is complete, which is why it uses a timeout |
| 13:42.52 | pawpro | I rune it on the localhost. How else can I do it? |
| 13:43.08 | pawpro | would php manager be faster? |
| 13:43.18 | Corydon76-dig | pawpro: there's a astcli Perl script in 1.6 that you can use to execute the command via Manager |
| 13:43.43 | pawpro | i cant switch to 1.6 i'm afraid right now. It's too many machines |
| 13:43.43 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 13:43.48 | IsUp | hello |
| 13:44.06 | IsUp | i am trying to setup my GSM gateway, with PRI. |
| 13:44.18 | IsUp | i am getting Primary D-Channel on span 1 up |
| 13:44.20 | IsUp | and Primary D-Channel on span 1 down |
| 13:44.29 | IsUp | "No D-channels available! Using Primary channel 16 as D-channel anyway!" |
| 13:44.44 | IsUp | ive tried to change timing source, but it doesnt works |
| 13:44.51 | Corydon76-dig | pawpro: you can use the script with 1.4 |
| 13:45.09 | pawpro | Corydon76-dig: can i write to the socket myself? to get the list of sip and zap channels? |
| 13:47.31 | IsUp | any ideas? |
| 13:48.44 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:49.37 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 13:53.12 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-227-95.telkomadsl.co.za) |
| 13:53.13 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:53.55 | IsUp | hey [TK]D-Fender |
| 13:54.07 | invalidrecord | anyone done realtime extensions?? |
| 13:58.29 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:58.37 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 13:59.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:05.40 | *** join/#asterisk carranca (n=carranca@pampero.itba.edu.ar) |
| 14:07.10 | carranca | Hi, if i have an AGI script that controls the logic of a call and in some point the call gets modified by an AMI program, does the AGI script get the "notification" of the change? |
| 14:08.03 | carranca | for example i want to do "kind of an" ivr in the AGI script but at some moment the call may be redirected. |
| 14:08.13 | carranca | by the AMI program of course |
| 14:09.15 | *** join/#asterisk chazz (n=chazz@nat/digium/x-5dd50fef05a56a60) |
| 14:09.30 | *** join/#asterisk Leddy (n=dpreuss@72.54.198.194) |
| 14:10.21 | Leddy | Is it possible to change the caller id "number" when using the ami Originate? I can change the Caller Id: but it still shows "asterisk" as the # |
| 14:10.40 | [TK]D-Fender | LeddyYes, is you Originate the right kind of CHANNEL |
| 14:10.58 | Katty | morning |
| 14:11.02 | [TK]D-Fender | Katty: Mew |
| 14:11.06 | Katty | [TK]D-Fender: mew. |
| 14:11.10 | Leddy | SIP/Extension |
| 14:11.31 | [TK]D-Fender | Leddy : Clearly that will not do it... go read over the list of channel types. |
| 14:13.14 | *** join/#asterisk Tweety84 (n=spam@sub18.rz-zw.fh-kl.de) |
| 14:13.32 | Leddy | Which channel type should I be looking for? |
| 14:14.02 | [TK]D-Fender | Leddy : quick freebie : Channel: Local/1234@contextwithextensthatdialbutdon'thitvm&setCIDfirst/n) |
| 14:14.16 | [TK]D-Fender | Leddy : a LOCAL channel. And leave off the ")" |
| 14:14.29 | [TK]D-Fender | Leddy : And you'll want the "/n" thats on the end... that's legit |
| 14:14.56 | [TK]D-Fender | Leddy : think of it as dialplan on side A, and dialplan on side B |
| 14:15.17 | [TK]D-Fender | laddjust that the Originate side will indeed call out to your SIP device once having done its prep work. |
| 14:16.22 | [TK]D-Fender | carranca: What "change"? |
| 14:16.29 | *** join/#asterisk qdk_ (n=qdk@94.191.224.98.bredband.3.dk) |
| 14:18.28 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 14:18.50 | Leddy | tk: You need Caller Name <Caller Number> |
| 14:18.58 | Leddy | in the CallerID: field |
| 14:21.26 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:24.13 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.demon.co.uk) |
| 14:24.35 | [TK]D-Fender | Leddy : Suppose you could sue that too :) |
| 14:24.38 | [TK]D-Fender | use* |
| 14:25.01 | Leddy | easier too :) |
| 14:25.06 | [TK]D-Fender | Leddy : Its a great place to do things like Auto-answer as well... |
| 14:25.10 | [TK]D-Fender | Speeds up the outbound dial |
| 14:28.56 | kerframil | RypPn: as it happens, I do now have a few questions about sccp (whenever you have some time) |
| 14:29.13 | RypPn | kerframil go for it :) |
| 14:29.38 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 14:31.25 | mcargile | So I am trying to connect my asterisk system to a Broadworks setup. Everything is working fine accept inbound. My provider is saying this is because on registration I have the wrong contact information. |
| 14:31.29 | kerframil | RypPn: ok ... will one hurdle I ran into was constand "Bad address" errors in sccp_socket.c (chan_sccp-20071213 with asterisk-1.4.22 on amd64). I read some vague post somewhere that hinted that it might be an amd64-specific problem but it helpfully didn't elaborate. for now, that seems to have been resolved by downgrading to asterisk-1.4.21.2 :| |
| 14:31.50 | mcargile | They say my contact line needs to include ";transport=UDP" at the end of it |
| 14:32.08 | mcargile | I cannot find a way in asterisk to do this without changing source code. |
| 14:32.53 | *** join/#asterisk _Krieger_ (n=krieger@91.195.10.82) |
| 14:33.01 | mcargile | and even then I doubt that this is the issue |
| 14:33.33 | kerframil | RypPn: worse, I managed once to get my Cisco 7940 to negotiate with chan_sccp and get itself set up. that's gone out of the window since I upgraded to the latest firmware (P00308001000). I think it may be because it's assuming that my router is a SRST gateway (?). I can see it constantly trying to access 192.168.254.254 (my router) as well as 192.168.254.2 (my sccp server). |
| 14:33.36 | RypPn | kerframil use svn version r308 |
| 14:33.45 | kerframil | RypPn: alright |
| 14:34.02 | RypPn | kerframil what tftp are you using? |
| 14:34.16 | kerframil | RypPn: tfpt-hpa |
| 14:34.19 | kerframil | tftp-hpa |
| 14:35.01 | kerframil | RypPn: any idea as to how I might disable SRST? I was thinking of maybe simply not pushing the routers DHCP option from dhcpd.conf but I can't figure out a way to scope that just to the cisco devices. |
| 14:35.03 | RypPn | kerframil have you reversed the slashes (remembering the phone is expecting a windows box) ? |
| 14:35.46 | kerframil | RypPn: I'm not experiencing any problems with tftp at all. that's worked fine right from the beginning. my XML files are parsed correctly, the new firmware image was pushed fine. |
| 14:36.20 | _Krieger_ | ChanSpy delivers both talkers audio to spyer, or only spied channel audio? |
| 14:36.22 | kerframil | RypPn: it's only since I upgraded from the stock firmware that it no longer 'settles down' - it keeps hitting 192.168.254.254. |
| 14:36.25 | kerframil | RypPn: (my router) |
| 14:36.59 | RypPn | did you setup the alternate tftp in the menu? |
| 14:37.06 | RypPn | phone menu |
| 14:37.09 | kerframil | RypPn: no. why would I need to do that? |
| 14:37.54 | RypPn | kerframil well, is the tftp on the same box as the dhcp server? |
| 14:38.03 | kerframil | RypPn: that address is my router (option routers 192.168.254.254). when I look at the phone's network settings after it's got its lease, it seems to assume that it's a Cisco router (CallManager 1 = 192.168.254.2, CallManager 2 SRST = 192.168.254.254) |
| 14:38.11 | kerframil | RypPn: yes. everything's on 192.168.254.2 |
| 14:38.21 | kerframil | RypPn: I don't want it touching 192.168.254.254 |
| 14:39.20 | kerframil | RypPn: it wasn't picking up that CallManager 2 setting before I updated the firmware. I'd like to just temporarily disable the router dhcp option to see what happens but my colleagues would be nonplussed ;) |
| 14:39.55 | RypPn | kerframil I'm using that same firmware on a 7940 I have, where do you see those srst requests? |
| 14:41.10 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:41.58 | RypPn | kerframil in my menu I only have callmanager1 showing with any settings |
| 14:42.55 | kerframil | RypPn: I am certain that this is the problem. I wonder why it is identifying my router as a SRST enabled gateway merely on account of upgrading the firmware? I assure you that I changed no other settings. |
| 14:43.13 | kerframil | tell you what, I will disable the router option - most everyone is out to lunch now anyway |
| 14:43.19 | kerframil | then I'll reboot the phone and see what happens |
| 14:44.59 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 14:45.42 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:45.57 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:46.11 | _Krieger_ | ChanSpy delivers both talkers audio to spyer, or only spied channel audio? |
| 14:46.22 | kerframil | RypPn: well, CallManager2 is not set now. but it times out "Opening 192.168.254.2" (no errors in asterisk which I'm running un-forked, and no errors in the status window/log on the device itself). unless you have any other ideas, I'll check out r308 as you suggested. |
| 14:48.08 | kerframil | RypPn: meh, "Bad address" again ... time for a checkout :) |
| 14:48.24 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 14:48.56 | donnib | guys in which folder is zaptel ? |
| 14:49.03 | donnib | i need to run make config |
| 14:51.06 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:51.17 | tuxx- | depends on where you downlaoded it ;-) |
| 14:51.36 | tuxx- | its a seperate package, downloadable @ asterisk.org |
| 14:51.52 | donnib | oh i think i found it /usr/src/zaptel |
| 14:51.54 | donnib | thx anyway |
| 14:51.56 | tuxx- | great \oi |
| 14:51.57 | tuxx- | \o/ |
| 14:52.09 | [TK]D-Fender | M |
| 14:52.11 | [TK]D-Fender | C |
| 14:52.12 | [TK]D-Fender | A |
| 14:52.14 | [TK]D-Fender | ! |
| 14:52.19 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 14:52.22 | tuxx- | !google MCA |
| 14:52.22 | [TK]D-Fender | is teh funneh |
| 14:52.23 | tuxx- | ?:D |
| 14:55.03 | *** join/#asterisk jeffik (n=jeffik@69-196-165-181.dsl.teksavvy.com) |
| 14:57.59 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-f76dbb095930594f) |
| 14:57.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:01.13 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 15:04.38 | *** join/#asterisk mog (n=mog@nat/digium/x-877dd3c1f8cd3ec9) |
| 15:04.38 | *** mode/#asterisk [+o mog] by ChanServ |
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| 15:09.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:09.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 15:23.50 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
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| 15:28.00 | *** join/#asterisk nfi|ermes (i=ErMeS@host232-32-dynamic.3-79-r.retail.telecomitalia.it) |
| 15:28.18 | *** join/#asterisk killfill (n=killfill@200.63.96.244) |
| 15:28.18 | nfi|ermes | hi [TK]D-Fender |
| 15:28.20 | killfill | hi.. |
| 15:28.32 | killfill | [Dec 1 12:27:23] WARNING[4780]: chan_zap.c:899 zt_open: Unable to specify channel 1: Device not configured |
| 15:28.38 | killfill | what would that mean?... |
| 15:28.50 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-aa2366943e37398d) |
| 15:28.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:29.06 | killfill | oh nevermind. |
| 15:32.57 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:34.18 | *** part/#asterisk oej (n=olle@ns.webway.se) |
| 15:34.20 | *** part/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 15:37.48 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:37.50 | *** join/#asterisk moy (n=moy@189.169.207.39) |
| 15:37.51 | nfi|ermes | a problem with callerid: it isn t shown in asterisk at incoming calls |
| 15:37.58 | nfi|ermes | my zapata.conf : http://pastebin.com/m16ba5d93 |
| 15:38.08 | nfi|ermes | asterisk 1.4.22 with WCTDM/0 "Wildcard TDM400P REV E/F Board 1" (MASTER) |
| 15:38.21 | nfi|ermes | to debug in my extensions.conf i have: exten => s,5,NoOp(${CALLERID(num)}), and the result is : Executing [s@from-pstn:5] NoOp("Zap/1-1", "") in new stack |
| 15:38.48 | [TK]D-Fender | nfi|ermes: And your zaptel.con... |
| 15:39.46 | [TK]D-Fender | nfi|ermes: immediate=yes <- THIS is your problem |
| 15:39.49 | tzafrir_laptop | nfi|ermes, any good reason to use immediate = yes ? |
| 15:39.51 | [TK]D-Fender | nfi|ermes: NEVER do this |
| 15:42.11 | nfi|ermes | ok, i try |
| 15:44.18 | nfi|ermes | it look like it didn t solved |
| 15:47.21 | nfi|ermes | my zaptel.conf: http://pastebin.com/d469f7a73 |
| 15:47.56 | [TK]D-Fender | nfi|ermes: If you've restarted * and it still isn't working, call up Digium for support. Your zone my not be very well supported for CID |
| 15:48.02 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 15:48.34 | nfi|ermes | i have not a digium card, but an openvox |
| 15:50.31 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:50.56 | [TK]D-Fender | nfi|ermes: Then try calling them up. |
| 15:50.58 | [TK]D-Fender | And.... |
| 15:51.00 | [TK]D-Fender | ~wglwat |
| 15:51.01 | jbot | well, wglwat is well, good luck with all that |
| 15:53.01 | *** join/#asterisk Hymie (i=hymie@l8r.net) |
| 15:53.48 | Hymie | dudes! Hey, has anyone tried the Aastra 57i CT or 9480i CT? |
| 15:54.14 | [TK]D-Fender | Hymie: 57i CT = meh |
| 15:54.23 | Hymie | I'm having a hard time telling if there are generational differences betwee3n the 57i CT and the 9480i CT, or if their differences are just the number of lines, etc |
| 15:54.35 | Hymie | hey! TK dude, you're still here! ;) |
| 15:54.59 | Hymie | [TK]D-Fender: don't like the 57i CT, or all CT from Aastra? |
| 15:55.03 | [TK]D-Fender | Hymie: No wieth, rubbery shit buttons, poor LCD viewing angle, crashes on mass-page, tinny speakerphone, poor LCD usability (char matrix drivers for a pixel screen = stupid), etc |
| 15:55.06 | Hymie | or all Aastra ;) |
| 15:55.24 | [TK]D-Fender | Hymie: My 57i CT made me yearn for my old bedside Polycom IP 301 |
| 15:55.43 | [TK]D-Fender | Hymie: Anything lower than a 57i can only be considered a lower value |
| 15:55.56 | [TK]D-Fender | Hymie: And seriously not worth thinking about IMO |
| 15:56.05 | Hymie | my thing is I really want a new cordless phone, my old is hosed, and I want to go VOIP this time around, and these CT models seem like the only way to get a quality name reasonably priced |
| 15:56.15 | Hymie | hum |
| 15:56.19 | [TK]D-Fender | Hymie: the 1 thing Aastra really has going for it is the Godly attendant console... |
| 15:56.31 | Hymie | I have the odd feeling you don't like Aastra outside of that |
| 15:56.44 | Hymie | [TK]D-Fender: how's the cordless phone side of things, though? |
| 15:57.05 | [TK]D-Fender | Hymie: if you can live with the cordless being tied to that base (cannot really operate independant of it), and that the base can intercept calls the would ring there, etc... then ok/fine/sure |
| 15:57.11 | Hymie | [TK]D-Fender: is there a speaker phone on the cordless part? I didn't check |
| 15:57.22 | [TK]D-Fender | Hymie: Not that I recall.... maybe |
| 15:57.45 | Hymie | [TK]D-Fender: bleh. Any ideas on just a good cordless SIP phone? |
| 15:58.12 | [TK]D-Fender | Hymie: Seimens & Polycom both make DECT phones as well |
| 15:58.23 | [TK]D-Fender | Hymie: Those would be better options. WiFI = suck |
| 15:58.27 | Hymie | I can't seem to get Seimens domestically, at least not easily |
| 15:58.46 | Hymie | I thought the polycoms were $1000 or some such though, with the base unit |
| 15:59.01 | Hymie | rechecks, for he could be very wrong |
| 16:00.32 | Hymie | hmm, yeah.. I mean that Polycom Kirk thing might be alright in a corporate env, but I Guess not for me |
| 16:01.17 | Hymie | [TK]D-Fender: did you try out those Aasura MBU 400s and such? Keep in mind I really, really am very happy with Polycom, but just can't afford their cordless phones for single person use |
| 16:01.51 | [TK]D-Fender | Hymie: ATA + cordless |
| 16:02.39 | Hymie | [TK]D-Fender: I have that now, but bah :( |
| 16:02.54 | Hymie | [TK]D-Fender: people don't like the snom m3 either? |
| 16:03.11 | [TK]D-Fender | Hymie: Iffy range & battery from what I heard |
| 16:03.18 | [TK]D-Fender | Hymie: No personal experience though |
| 16:03.29 | [TK]D-Fender | Hymie: a "maybe" |
| 16:03.50 | Hymie | [TK]D-Fender: ok, outside of how the buttons feel on the 57i, and that I won't be using much of the XML stuff, the 57i doesn't seem horrible to you? |
| 16:04.13 | [TK]D-Fender | Hymie: No point. |
| 16:04.16 | Hymie | is the tinny speakerphone on the listening side, or on how your voice comes through to the remote side? |
| 16:04.30 | [TK]D-Fender | Hymie: Undoing the XML is just another great reason not to get the Aastra |
| 16:05.03 | Hymie | [TK]D-Fender: my reasoning for the Aasura: the cordless phone, plus a speaker phone at my work desk |
| 16:05.42 | [TK]D-Fender | Hymie: Aastra.. get the name right... and it may be OK for you if you want to replace your desk phone as well... |
| 16:05.49 | [TK]D-Fender | Hymie: But you've been warned |
| 16:05.53 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 16:05.56 | *** join/#asterisk propellerhead (n=yogurt2u@host130.190-226-46.telecom.net.ar) |
| 16:06.43 | Hymie | [TK]D-Fender: this really annoys me. I was two years ago that I last looked for a good cordless solution, and apparently I'm still hosed :( |
| 16:07.15 | Hymie | [TK]D-Fender: I get so _+#($@#_+ pissed off that v-tech and others just make all these lame assed, tied to lame name company, voip phones |
| 16:10.27 | Hymie | [TK]D-Fender: ok dude, thanks. You've given me food for thought. |
| 16:10.30 | *** part/#asterisk Hymie (i=hymie@l8r.net) |
| 16:12.26 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:14.57 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 16:15.31 | *** join/#asterisk SQLDarkly (n=dakendri@192.147.57.6) |
| 16:16.13 | *** join/#asterisk nny_1 (n=scott@64.203.244.146) |
| 16:16.48 | SQLDarkly | So when a meetme room is running you are able to see how many users are in the said room plus any marked users. This is using the CLI. Now, in the dialplan is there a way to detect the marked user has entered so all participants that join past the marked user are not prompted for a PIN |
| 16:17.07 | nny_1 | if you had to use an ata to connect a couple of analog phones to a IP based PBX, which would you choose? Looking at the Linksys SPA series, but curious if anyone has a better choice |
| 16:17.19 | nny_1 | something in the realm of 4 ports |
| 16:17.50 | SQLDarkly | I just need a way to grab that information. I would simply use a gotoif providing that is available somewhere to grab |
| 16:20.30 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:22.09 | SQLDarkly | Anyone? |
| 16:22.39 | SQLDarkly | Yes I could write a small php script to use the manager api to grab that status I suppose. I simply would like to know if there is a native way to do this. |
| 16:23.16 | [TK]D-Fender | nny_1: 2 small SPA, or 1 SPA-8000 |
| 16:23.49 | [TK]D-Fender | SQLDarkly: "core show functions" , "core show applications" |
| 16:29.23 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 16:29.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:35.32 | nny_1 | [TK]D-Fender: thanks |
| 16:38.39 | angryuser | SQLDarkly: manager api is pretty much a native way ;) |
| 16:39.10 | *** join/#asterisk lchristensen (n=lchriste@adsl-218-122-86.asm.bellsouth.net) |
| 16:40.06 | jaytee | [TK]D-Fender, what's with the RJ-21 jack on the SPA-8000? |
| 16:40.54 | [TK]D-Fender | jaytee: Whats there to say? Its as obvious as it looks... |
| 16:42.06 | lchristensen | I'm a relative newbie with a question. I am trying to use call files with Asterisk Business edition. Asterisk places the call and goes to the specified extension, but doesn't wait for the called party to pick up before plunging ahead with my IVR scripts. How do I get Asterisk to wait for the called party to pick up? None of the Wait applications (Wait, WaitExten, ...) seem applicable? |
| 16:42.26 | *** join/#asterisk hi365_m (n=hi365@213.151.60.154) |
| 16:42.42 | SQLDarkly | yes it is ;) php however is not |
| 16:42.45 | jaytee | [TK]D-Fender, so it lets you hook up a 25 pair cable to it obviously but it's not going to terminate all 24 lines off the cable as FXS ports. |
| 16:43.15 | [TK]D-Fender | jaytee: Correct, but it allws you to use your existing patch pane for quicker integration. |
| 16:43.34 | jaytee | [TK]D-Fender, I can see where that might come in handy and save time. |
| 16:43.55 | jaytee | per port it's a great price |
| 16:43.56 | [TK]D-Fender | lchristensen: It does wait until the channel has answered. You just haven't seen WHERE the answer is coming from./ |
| 16:44.07 | [TK]D-Fender | jaytee: It is an awsome value. |
| 16:44.59 | lchristensen | D-fender: It is supposed to wait. But by the time my cell phone rings, about 6-8 seconds of prompts have already played. |
| 16:45.08 | jaytee | I really like their SPA-2102 and SPA-3102's but this looks like it would make a better fit for some of our larger out buildings where we want to use many cordless base phones. |
| 16:45.34 | [TK]D-Fender | lchristensen: As I said you have not looked at WHERE the answer takes place. |
| 16:45.50 | [TK]D-Fender | lchristensen: its HOW you get to the PSTN that causes the call to be considered answered |
| 16:46.01 | lchristensen | D-Fender: I was about to ask where I should look for where the answer occurs. |
| 16:46.14 | [TK]D-Fender | jaytee: It is a good choice.. 1 box to power / configure, sleek too, and wiring OPTIONS 9not force-fed) |
| 16:46.32 | [TK]D-Fender | lchristensen: Look in your dialplan,CLI output, and call-file |
| 16:46.57 | [TK]D-Fender | lchristensen: PASTEBIN is your friend... |
| 16:46.59 | [TK]D-Fender | ~pb |
| 16:47.00 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:47.01 | [TK]D-Fender | ^^^^^^ |
| 16:47.12 | jaytee | [TK]D-Fender, yeah I like the idea of not having 4 xformers plugged into a power strip off a UPS just to get power to 4 SPA-2102's for 8 FXS ports. this is much cleaner. |
| 16:47.40 | jaytee | Glad I saw you type that. I hadn't seen that model yet. I usually go to telephonydepot and they don't list it. |
| 16:47.48 | [TK]D-Fender | jaytee: I haven't handled it physically, but I first recommended it to jblack and helped him configure it |
| 16:47.56 | [TK]D-Fender | jaytee: He was very pleased with it |
| 16:48.27 | [TK]D-Fender | jaytee: I know... VoiPSUPPLY does though, and there are other retailers.. |
| 16:48.32 | [TK]D-Fender | jaytee: I'm wondering if there is some agreement that stops TD from carrying it |
| 16:48.36 | lchristensen | D-Fender: What am I looking for? An answer application? |
| 16:49.12 | jaytee | [TK]D-Fender, I was going to call them today anyways so I'll ask about it. |
| 16:49.40 | [TK]D-Fender | lchristensen: You are going to show us your call file, related dialplan and CLI output of the attempt in a pastebin. |
| 16:49.51 | [TK]D-Fender | jaytee: Please do.. perhaps its an oversight |
| 16:51.01 | jaytee | [TK]D-Fender, I finally made some progress at home with the .NET libraries for Asterisk. Pretty soon I'll have a HUD Lite/FOP replacement written in C#. :-) |
| 16:51.18 | [TK]D-Fender | jaytee: Neato... |
| 16:51.33 | coppice | the SPA8000 seems nice, but note that is has a fan |
| 16:51.34 | [TK]D-Fender | jaytee: How are you planning on releasing it? |
| 16:52.09 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:52.15 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:52.17 | lchristensen | D-Fender: Let me propose an alternative. I started with a dialplan someone else put together with a GUI. I've been hand editing. I'm going to save off the GUI dialplan and create a stripped down one of my own and test. If that resolves the issue, I'll dig deeper on the auto-gened. |
| 16:52.29 | jameswf | anyone know if its possible to delete voip-info pages |
| 16:53.02 | lchristensen | D-Fender: If not, I'll post the stripped down variant. |
| 16:54.13 | [TK]D-Fender | lchristensen: We probably only need the CLI output. |
| 16:54.33 | [TK]D-Fender | lchristensen: I would not change anything just yet as you could screw up the evidence |
| 16:54.35 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:56.02 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 16:56.09 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:56.42 | lchristensen | D-Fender: I'll save off the evidence. I'll have to sign off to get to it, though. Access is through VPN, which messes up my internet and IRC links. The CLI indicates that the Answer is occurring where on the dialplan line where I think it should. Back in 10 minutes ... |
| 16:58.38 | *** part/#asterisk lchristensen (n=lchriste@adsl-218-122-86.asm.bellsouth.net) |
| 17:00.58 | SQLDarkly | I do apologize if I am missing the obvious. I have looked through the apps and only thing I see is to do meetmecount to a variable but this will not allow me to see who is marked...... |
| 17:01.48 | SQLDarkly | Perhaps inserting a CDR record when Marked Enters and have a check for that field ..... |
| 17:01.51 | [TK]D-Fender | SQLDarkly: And that in itself is the answer |
| 17:02.12 | [TK]D-Fender | SQLDarkly: AKA : You can't. So go write your external script now. |
| 17:02.31 | SQLDarkly | Doh :) |
| 17:02.40 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 17:03.04 | SQLDarkly | After I get this working with external php. D-Fender if you have that information in your head I would like to know the correct way to accomplish this. |
| 17:04.25 | [TK]D-Fender | SQLDarkly: Have what info? |
| 17:04.59 | SQLDarkly | Oh nevermind then. I got the impression you had an ace up your sleeve for doing this without the need for an external script. |
| 17:05.03 | [TK]D-Fender | SQLDarkly: "correct" is whatever way works as cleanly as possible. |
| 17:05.20 | SQLDarkly | Very true statement. |
| 17:06.24 | [TK]D-Fender | lunch, BBIAB |
| 17:07.44 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 17:08.19 | *** join/#asterisk tobias (n=tobias@cpe-076-182-095-118.nc.res.rr.com) |
| 17:12.15 | *** join/#asterisk lord_nikon (n=lord@host-216-153-131-74.roc.choiceone.net) |
| 17:14.58 | lord_nikon | im having a bit of trouble using Transfer properly, can someone assist perhaps ? my situation is that i am attempting to redirect any calls that enter a given dialplan context to a second asterisk server, using Transfer(SIP/2323@192.168.1.38) however the server at 192.168.1.38 never responds |
| 17:16.27 | *** join/#asterisk lchristensen (n=lchriste@adsl-218-122-86.asm.bellsouth.net) |
| 17:19.31 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:21.25 | lchristensen | D-Fender: The pastebin with my info is at http://www.pastebin.ca/1272477. Thanks for looking at it. |
| 17:22.10 | SQLDarkly | lord_nikon: Do you see any incoming traffic on the second box from the first one in the CLI? |
| 17:22.50 | SQLDarkly | lord_nikon: Second, you should do a traffic capture from BoxA when teh transfer happens and monitor BoxB while the capture is going on. Take both results find the issue. Solve the problem |
| 17:23.25 | lord_nikon | SQLDarkly: i see the invites comming in, but nothing seems to get done about them |
| 17:24.26 | SQLDarkly | What does the CLI on BoxB say? Can you pastebin the output please |
| 17:24.38 | lord_nikon | with sip debugging on? |
| 17:25.13 | SQLDarkly | Yes. I simply want to see what your seeing and maybe it is something obvious. |
| 17:25.30 | lord_nikon | gimme a second |
| 17:26.13 | SQLDarkly | Do they get transfered to an existing context? Does the said context or priority exist? If so has the dialplan been reloaded after said contexted was added? Are teh IPs or Hostnames correct? |
| 17:27.19 | SQLDarkly | Also do you have any conflicting contexts / priorities? All of these things need to be checked and rechecked for accuracy. 9 times out of 10 it is one of these issues |
| 17:31.55 | *** join/#asterisk etfonhomey (n=chatzill@www2.askpri.org) |
| 17:32.09 | lord_nikon | SQLDarkly: i dont specify a context, just an extension |
| 17:32.30 | lord_nikon | SQLDarkly: http://pastebin.com/d75eff08a |
| 17:32.37 | SQLDarkly | Pastebin the sectiono f your dialplan that makes the magic happen also |
| 17:33.46 | lord_nikon | SQLDarkly: http://pastebin.com/d54b3ef4 |
| 17:35.46 | SQLDarkly | ok and is exten 195 available in the correct context on BoxB. Meaning if I place a call to extension 123@default but 123 is really @incomingfromboxA then it will not work |
| 17:36.13 | lord_nikon | yea, its a valid ext |
| 17:36.31 | lord_nikon | it doesnt even apear to get into any context tho |
| 17:36.32 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:36.50 | lord_nikon | i never see any sort of response or anything from the second server |
| 17:36.52 | SQLDarkly | yes its valid but is it in the right place..... In your CLI dump i dont see it attempting anything nor do I see a hangup. This cannot be your complete dump |
| 17:37.01 | lord_nikon | it is :/ |
| 17:37.18 | lord_nikon | the invites just repeat a bunch of times |
| 17:37.23 | SQLDarkly | ok so its not leaving box1 then..... |
| 17:38.08 | lord_nikon | it seems as if its trying to, but box2 just ignores it |
| 17:39.57 | SQLDarkly | if box2 would ismply ignore it. There would be some sort of a response on the CLI on said box |
| 17:40.33 | SQLDarkly | Could you open up logger.conf and please enable debugging for the console. Recap and Repaste |
| 17:40.45 | flujan | hello guys. I have two boxes running asterisk. |
| 17:41.00 | flujan | The first one answers a call and transport it to box2 using sip. |
| 17:41.02 | SQLDarkly | Your box should give some sort of response saying that the call could not transfer because of X |
| 17:41.29 | flujan | I am loosing a lot of calls on this machine. I debug sip and got this http://pastebin.com/m468d8814 |
| 17:41.44 | SQLDarkly | lord_nikon: Have you successfully called an extension from BoxA to BoxB ? |
| 17:41.53 | flujan | machine 1 is answering passing it to machine two... Then machine one SIMPLY send a BYE and destroy the sip dialog... |
| 17:41.55 | flujan | any ideas? |
| 17:42.00 | lord_nikon | SQLDarkly: yes, using plain Dial works fine |
| 17:42.08 | lord_nikon | i also have IAX2 working fine |
| 17:42.17 | SQLDarkly | interesting. |
| 17:42.44 | SQLDarkly | ENable debugging on the CLI and repaste. If you got the other techs working then it shouldnt be a big issue. |
| 17:43.15 | flujan | here goes my sip.conf |
| 17:43.19 | flujan | http://pastebin.com/d72fdfc25 |
| 17:44.35 | *** join/#asterisk lchristensen (n=lchriste@adsl-218-122-86.asm.bellsouth.net) |
| 17:45.35 | SQLDarkly | lord_nikon: Also in your sip.conf make sure you have the correct hosts set. Also set a default context where it will be dropped |
| 17:45.56 | SQLDarkly | So you could specify sip:123@10.10.10.10/1234 |
| 17:46.19 | lord_nikon | Stopping retransmission on '0fc2c069da980d32@192.168.2.81' of Response 20149: Match Not Found |
| 17:46.27 | lord_nikon | that looks to be the problem |
| 17:46.43 | SQLDarkly | Try what I just suggested it should clear it up |
| 17:47.14 | lord_nikon | roger |
| 17:47.16 | SQLDarkly | make sure when you done to turn off debug in your logger.conf so you dont fill your log file |
| 17:47.26 | lord_nikon | yup |
| 17:47.41 | lord_nikon | sip.conf on box2 correct ? |
| 17:48.13 | lchristensen | I am working on Asterisk Business Edition trying to use Call Files. Asterisk is placing the call, but is proceeding with the dialplan before the call is being answered by the called party. Dialplan, call file and CLI output are posted at http://www.pastebin.ca/1272481. Increasing the verbosity level from 3 to 8 indicates that Asterisk thinks the call is being answered immediately, which is not shown in the paste bin. My ques |
| 17:49.22 | file | if it is an analog line then they are treated as answered as soon as dialing is complete |
| 17:51.10 | lchristensen | It is an analog line. |
| 17:51.18 | seanbright | then there you go |
| 17:51.26 | seanbright | that'll be $125 |
| 17:51.29 | quentusrex | How do I disable an asterisk module? I think Dictate is crashing asterisk... |
| 17:51.30 | quentusrex | <PROTECTED> |
| 17:51.30 | quentusrex | app_dictate.so => (Virtual Dictation Machine) |
| 17:51.31 | seanbright | payable to seanbright |
| 17:51.40 | quentusrex | that's the last lines I get when starting asterisk. |
| 17:51.44 | seanbright | quentusrex: in modules.conf put noload => app_dictate.so |
| 17:52.19 | SQLDarkly | Stepping out for 10 minutes brb |
| 17:52.24 | lchristensen | Is there any way to prevent that or do a "hello" detect? Or do I have to use a T1 for my in house test and development system? |
| 17:53.20 | quentusrex | seanbright: still crashes... |
| 17:53.28 | seanbright | then it's not app_dictate.so |
| 17:53.33 | quentusrex | right. |
| 17:53.38 | seanbright | glad i could help. |
| 17:53.39 | seanbright | heh |
| 17:53.55 | seanbright | asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvg |
| 17:53.58 | seanbright | and pastebin the output |
| 17:54.00 | quentusrex | seanbright: do you know where I might be able to get some kind of info? |
| 17:54.20 | seanbright | quentusrex: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvg |
| 17:54.23 | seanbright | and pastebin the output |
| 17:55.30 | quentusrex | http://pastebin.com/d3ba57f26 |
| 17:55.45 | seanbright | i still see app_dictate.so |
| 17:55.53 | quentusrex | right. I enabled it again. |
| 17:56.04 | quentusrex | since it wasn't the cause... |
| 17:56.13 | seanbright | are you getting a core file? |
| 17:56.23 | quentusrex | What ever the next step in the process is what killing it... nope.. no core... |
| 17:57.06 | seanbright | even with the -g? it should be... hrm |
| 17:57.48 | seanbright | do a 'ulimit -c unlimited' |
| 17:57.51 | seanbright | and try again |
| 17:58.32 | *** join/#asterisk ElDios (n=ElDios@213-140-6-112.ip.fastwebnet.it) |
| 17:58.35 | ElDios | yo dogs |
| 17:58.38 | ElDios | =) |
| 17:58.56 | ElDios | any idea on howw to disable the # key during a voicemail registration |
| 17:58.56 | ElDios | ? |
| 17:58.59 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:59.22 | ElDios | I mean, I want that during a registration no key is working.. |
| 18:00.25 | quentusrex | http://pastebin.com/d471ff34e |
| 18:00.29 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 18:01.12 | seanbright | and no core file |
| 18:01.13 | seanbright | ? |
| 18:01.39 | seanbright | gdb /wherever/its/installed/asterisk |
| 18:01.44 | seanbright | run -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvg |
| 18:02.55 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 18:03.25 | Katty | wibbles |
| 18:03.33 | Katty | i don't get this place. |
| 18:04.01 | Katty | first, i thought we were doing asterisk. then we brought on talkswitch cause we thought it would be cheaper. it wasn't. |
| 18:04.07 | Katty | so now they want to sell samsung crap. |
| 18:04.15 | Katty | and then last week i got an invite for toshiba crap. |
| 18:04.24 | JonOnt | Hey guys, quick question on my SIP trunk, host=xxx.xxx.xxx.xxx or does it need to be host=xxx.xxx.xxx.xxx:5060 ? |
| 18:04.27 | [TK]D-Fender | Katty: unload chan_telecomwhores.so |
| 18:04.29 | Katty | now /today/ they're wanting me to go setup a SQL server. |
| 18:04.36 | Katty | is confused. |
| 18:04.39 | [TK]D-Fender | JonOnt: Typically no |
| 18:04.53 | JonOnt | [TK]D-Fender, hey, nice to see you in here again |
| 18:05.05 | Katty | wonders what her job discription is. |
| 18:05.18 | quentusrex | http://pastebin.com/m1262d2a3 |
| 18:06.02 | JonOnt | [TK]D-Fender, now my sip trunk provider say i need to configure a seconday gateway, do I just do a second host= ? |
| 18:06.11 | seanbright | quentusrex: wtf? it's not even crashing |
| 18:06.41 | Katty | [TK]D-Fender: oh it's worse than you think. |
| 18:06.47 | Katty | [TK]D-Fender: we're mainly a kyocera/copystar dealer |
| 18:06.53 | Katty | [TK]D-Fender: what's a copier company doing selling phone systems? |
| 18:06.57 | quentusrex | seanbright: alright... good catch. it's not crashing... Why is it just quiting? |
| 18:07.05 | nny_1 | jesus ... |
| 18:07.07 | seanbright | i have no idea |
| 18:07.27 | Katty | [TK]D-Fender: did you know we sell furniture? and sprint phones too? and operate a duplication service (now that i would excpect at a copier dealer) |
| 18:07.28 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 18:07.37 | nny_1 | ok does anyone know of a ITSP that works? |
| 18:08.04 | nny_1 | vitelity: no DTMF. teliax: (try their number for a laugh) 18884835429 |
| 18:08.43 | nny_1 | that and I haven't gotten outbound working with teliax as of yet |
| 18:09.07 | quentusrex | seanbright: any idea how I can find out? |
| 18:09.29 | seanbright | try adding a couple dddd's to the command line |
| 18:09.35 | seanbright | asterisk -cvvvvvvvvvvvvvvvvvvvvvvvdddddddddddddddddddddddddddg |
| 18:09.47 | seanbright | maybe that will spit out something useful |
| 18:09.58 | quentusrex | same thing... |
| 18:10.11 | JonOnt | Any one have any idea how to setup a secondary gateway ip address for a sip trunk? |
| 18:10.36 | seanbright | quentusrex: other than turning off autoload and adding modules by hand, i don't know what else you can do at this point |
| 18:12.41 | quentusrex | seanbright: that worked. |
| 18:12.48 | quentusrex | one of the modules is killing it... |
| 18:12.57 | quentusrex | or one of the config files is fscked up... |
| 18:13.01 | seanbright | right |
| 18:13.12 | seanbright | i'll leave that detective work to you |
| 18:13.14 | seanbright | :) |
| 18:13.23 | quentusrex | lol |
| 18:14.03 | quentusrex | what does asterisk try to load after modules? |
| 18:14.08 | quentusrex | and how does it get a list of the modules? |
| 18:14.15 | quentusrex | dictate isn't in modules.conf |
| 18:14.23 | [TK]D-Fender | nny_1: les.net is pretty solid |
| 18:14.28 | Yourname` | Hi. I removed AgentLogin() because agents now want their queue calls to RING. So I changed it for AQM. However, AQM is more like no asking for username/password. Is AgentCallbackLogin the replacement for that then? |
| 18:14.33 | seanbright | quentusrex: autoload loads everything, basically. |
| 18:14.53 | quentusrex | everything from where? |
| 18:14.53 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 18:15.13 | seanbright | quentusrex: wherever your modules are installed... /usr/lib/asterisk/modules for me |
| 18:15.39 | quentusrex | ok... |
| 18:15.45 | quentusrex | now... which one is it... :) |
| 18:16.01 | seanbright | if i knew the answer to that question, we wouldn't be having this conversation |
| 18:16.31 | generalhan | hey all ... i need some serious help !! i had realtime setup using a MySQL DB. i tried to turn that off and remove all the references to the DB in my configs ... but something went haywire ! my CLI just keeps posting the same message over and over again: WARNING[8481]: app_voicemail.c:2262 inboxcount: Failed to obtain database object for 'asterisk'! |
| 18:17.01 | quentusrex | Do you know how asterisk determines which modules to load in which order? |
| 18:17.02 | seanbright | quentusrex: run this from your shell: for x in /usr/lib/asterisk/modules/*.so; do echo module load $x; done |
| 18:17.15 | seanbright | copy the output, paste into asterisk |
| 18:17.17 | seanbright | :) |
| 18:17.29 | generalhan | but i removed the references from extconfig, res_mysql, and voicemail, so i dont know what i could have missed. |
| 18:18.26 | jaytee | [TK]D-Fender, sorry I was called away earlier. In answer to your question about how I plan to release it. As cheaply as possible while still managing to make something from my efforts :-) |
| 18:18.41 | generalhan | anyone have any ideas ? whatever has happened has made it impossible for any user to receive a voicemail. |
| 18:19.49 | quentusrex | seanbright: that still works. |
| 18:19.53 | quentusrex | asterisk is still working... |
| 18:20.32 | seanbright | quentusrex: don't know what to tell you. |
| 18:20.53 | quentusrex | seanbright: I think it could be a fscked config file... |
| 18:21.05 | seanbright | then a 'module load' should make it die, as well |
| 18:21.09 | quentusrex | what does asterisk try to do after it loads modules? |
| 18:21.46 | key2 | when a call comes from zap and I want to see it on hudlite, to what extension am I supposed to send it ? |
| 18:22.12 | *** join/#asterisk Leddy (n=Leddy@72.54.198.194) |
| 18:22.37 | seanbright | quentusrex: a ton of stuff |
| 18:22.53 | quentusrex | How do I find out what the 'next step' is? |
| 18:23.11 | nny_1 | [TK]D-Fender: thanks |
| 18:23.15 | seanbright | look at the source? |
| 18:25.58 | jaytee | key2, try asking that in #trixbox |
| 18:26.14 | *** join/#asterisk Segnale007 (n=Pietro@host86-135-19-235.range86-135.btcentralplus.com) |
| 18:26.55 | [TK]D-Fender | Yourname`: Looks like chan_agent is heading out the window and the only app that references it is agentlogin from 1.6 |
| 18:27.04 | quentusrex | [TK]D-Fender: what would it mean if asterisk died with a status code of 1 ? |
| 18:27.11 | [TK]D-Fender | Yourname`: Time to revamp your methodology |
| 18:27.24 | key2 | jaytee: mmh thx, i thought someoone out of the 270 peops in here would be able to answer ;) |
| 18:27.27 | [TK]D-Fender | quentusrex: error itself is meaningless |
| 18:27.38 | [TK]D-Fender | key2: This isn't #hudlite |
| 18:27.51 | quentusrex | [TK]D-Fender: ok... Is there a way I can get more info from asterisk? |
| 18:28.06 | [TK]D-Fender | quentusrex: When does it happen? |
| 18:28.08 | jaytee | key2, Hudlite is not an Asterisk product. it's from Fonality, makers of Trixbox |
| 18:28.41 | quentusrex | [TK]D-Fender: I installed asterisk, and it worked for a moment. I restarted the server and every time I try to start asterisk now it dies with the error code of 1. |
| 18:29.04 | [TK]D-Fender | quentusrex: then start it manually and see at which point it bombs |
| 18:29.09 | quentusrex | [TK]D-Fender: if I tell it not to auto load modules it lives, but when I tell it to manually load the modules it dies. |
| 18:29.27 | [TK]D-Fender | quentusrex: pastebin is your friend. |
| 18:29.27 | quentusrex | no, I mean when I manually tell it to load the modules it lives... |
| 18:29.28 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 18:29.46 | quentusrex | but any auto load of the modules kills it... |
| 18:30.32 | quentusrex | http://pastebin.com/d471ff34e |
| 18:30.44 | quentusrex | and the gdb of asterisk is http://pastebin.com/m1262d2a3 |
| 18:31.17 | Yourname` | [TK]D-Fender: What?! You mean the only option other than AQM and AgentLogin, there's nothing else in 1.6? |
| 18:32.08 | [TK]D-Fender | Yourname`: "core show applications like queue" , "core show applications like agent" |
| 18:33.23 | [TK]D-Fender | quentusrex: Try spcifically noload-ing that module |
| 18:33.30 | Yourname` | [TK]D-Fender: Thanks |
| 18:34.25 | *** part/#asterisk intralanman (n=lanman@va-67-76-163-209.sta.embarqhsd.net) |
| 18:34.55 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 18:35.58 | quentusrex | [TK]D-Fender: I noloaded the dictate module and asterisk still dies... |
| 18:39.12 | [TK]D-Fender | quentusrex: less talk, more show |
| 18:39.29 | *** join/#asterisk monstertruck (n=Tanenbau@70.2.83.55) |
| 18:40.22 | monstertruck | hello children |
| 18:40.25 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:40.50 | monstertruck | i need to send post dial dtmf using php agi |
| 18:40.55 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:40.57 | monstertruck | is this even possible? |
| 18:41.41 | monstertruck | trying with anything similar to $agi->exec('Dial IAX2/16688@voipjet/17869752108','D(wwww011$extension#)'); hasnt worked |
| 18:41.43 | [TK]D-Fender | monstertruck: this is don'e in Dial itself |
| 18:41.48 | jblack | I don't know how to get asterisk to dial after dial at all. |
| 18:42.27 | quentusrex | [TK]D-Fender: I'm going to reinstall... |
| 18:42.31 | [TK]D-Fender | monstertruck: Here's a thought : I'm quite certin the secont parameter to dial isn't OPTIONS. |
| 18:43.26 | [TK]D-Fender | monstertruck: http://phpagi.sourceforge.net/phpagi2/docs/phpAGI/AGI.html#methodexec_dial |
| 18:44.49 | *** join/#asterisk clintc (n=clintc@n128-227-5-110.xlate.ufl.edu) |
| 18:46.19 | *** join/#asterisk fudpucker (n=here@75.151.177.173) |
| 18:50.46 | *** join/#asterisk cvnet (n=dahitler@74.210.103.241) |
| 18:50.49 | cvnet | hello |
| 18:51.10 | cvnet | default port for sip is 5060 can you change that? I'm sure you could, but where? |
| 18:52.27 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 18:52.45 | hi365_m | cvnet: in sip.conf. read the samle config for more info |
| 18:52.52 | Yourname` | [TK]D-Fender: That's unfortunate. |
| 18:53.12 | *** join/#asterisk fingerlickin (n=chatzill@216.65.195.52) |
| 18:53.56 | fingerlickin | anyone know if there are wireless sip phones that are NOT wifi, but the base has an Ethernet connection? |
| 18:54.14 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:54.28 | grandpapadot | fingerlickin: Aastra 480i CT |
| 18:54.28 | Yourname` | [TK]D-Fender: The only reason why I don't want AgentLogin anymore is because of the agents now want on-hook calls, rather than the permanently off-hook feature of AgentLogin. And AQM doesn't ask for username/password. I think it's AgentCallbackLogin for now then, and never think of lookin at 1.6 |
| 18:54.30 | Yourname` | heh |
| 18:55.05 | jaytee | yep, AgentCallbackLogin is gone, gone, gone in 1.6 |
| 18:55.06 | cvnet | hi365_m thanks |
| 18:55.45 | fingerlickin | grandpapadot: cool that will work. thank you. |
| 18:56.32 | Yourname` | I like AQM too, but to make it ask for username/password of the agent is a biznitch. |
| 18:56.56 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:57.05 | [TK]D-Fender | Yourname`: unload chan_whine.so |
| 18:57.14 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 18:57.28 | [TK]D-Fender | Yourname`: Few lines of dialplan... |
| 19:03.18 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:06.26 | *** join/#asterisk ezrafree (i=ezra@gware/developer/ezrafree) |
| 19:06.45 | ezrafree | hello |
| 19:06.50 | Assimilate | In Asterisk 1.4.22 if you do not have a retry in the queue.conf will the queued call go to invlaid or the next step if an agent does not answer? |
| 19:08.58 | hi365_m | can asterisk be started without any configs? |
| 19:09.21 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 19:09.40 | hardwire | do you think I'll need echocan if I'm just bridging and mixmonitoring between ports on the same T1 card? |
| 19:09.56 | hardwire | seems like it could cause more problems than it could solve with that low of latency |
| 19:13.26 | hardwire | and can you buy the current digium echocan card as an addon to the TE405P? |
| 19:15.23 | Yourname` | [TK]D-Fender: chan_whine, lol, good one there buddy. |
| 19:16.08 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 19:16.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:34.51 | *** join/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 19:36.11 | cvnet | is there a softphone out there thant you could change the port of it ? |
| 19:36.59 | meuserj | I'm trying to set up app_voicemail_imap.. and it's not accepting my self signed cert. So, I can either open up LOGIN for unencrypted connections (which I would rather not do) or get asterisk to accept the cert... is there a config option to do the latter? |
| 19:45.23 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 19:45.29 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:45.30 | kannan | hello, i am usng a sip service provider to call usa PSTN, with eyebeam softphones. I have not setcaller id , and want to show asrestricted number. However, on some calls there is a random caller id being displayed. How do I check whether this is sething on the Asterisk side? We are not registering with the sip service, its IP authentication |
| 19:45.33 | *** join/#asterisk errr_ (n=errr@fedora/errr) |
| 19:45.33 | *** join/#asterisk MrNeutr0n (n=DrLexus@209-253-217-62.ip.mcleodusa.net) |
| 19:45.33 | ezrafree | is asterisk likely to be a good solution for performing interviews with people in remote locations? what kind of client could one use to connect to a server running asterisk? |
| 19:45.35 | kannan | it happens even when i use the softphone direct to the sip service , i.e not thru asterisk box |
| 19:45.35 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 19:45.35 | MrNeutr0n | Hi there - does anyone know where I should ask a question about how the 7960 works with asterisk? |
| 19:45.53 | grandpapadot | MrNeutr0n: Only use firmware 8.2 where NAT is involved. G.729 only works on one channel at a time. |
| 19:46.49 | grandpapadot | Other than that, works great. |
| 19:47.09 | MrNeutr0n | grandpapadot, Well, actually I don't think NAT is a problem with this one. I am actually wondering how to get the name of the extension you're calling to show up as "To" on the screen |
| 19:47.15 | [TK]D-Fender | ezrafree: * is a telephony toolkit with which people typically configure as a PBX. |
| 19:47.30 | [TK]D-Fender | MrNeutr0n: ... |
| 19:47.32 | [TK]D-Fender | ~cpid |
| 19:47.33 | jbot | [~cpid] Called-Party ID is possible with * using patches on Mantis. See : http://bugs.digium.com/view.php?id=8824 |
| 19:47.34 | [TK]D-Fender | ^^^^^ |
| 19:47.41 | [TK]D-Fender | MrNeutr0n: YMMV |
| 19:47.44 | MrNeutr0n | So, if I dial a local extension, I want it to show some sort of name - really? |
| 19:47.47 | MrNeutr0n | Okay, thank |
| 19:47.48 | MrNeutr0n | s |
| 19:49.11 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:49.27 | ezrafree | [TK]D-Fender: would it be suitable for use as a voip solution if i wanted to perform real-time audio interviews between two individuals over the internet? |
| 19:49.51 | *** join/#asterisk mog (n=mog@nat/digium/x-b0c4b2614efdbd4a) |
| 19:49.51 | *** mode/#asterisk [+o mog] by ChanServ |
| 19:50.19 | *** join/#asterisk sasargen_ (n=chatzill@173-100-87-21.pools.spcsdns.net) |
| 19:54.54 | [TK]D-Fender | ezrafree: Certainly |
| 19:55.34 | [TK]D-Fender | ezrafree: However that typically requires the client to install software on their end which would probably only be needed for your solution. Kind of a waste |
| 19:55.52 | [TK]D-Fender | ezrafree: This is where it becomes worthwhile using a Skype channel, etc |
| 19:59.29 | [TK]D-Fender | ezrafree: At which point you could jsut use Skype yourself and save the trouble. What specific functionality would * add over normal Skype in your case? |
| 20:02.54 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 20:03.10 | *** join/#asterisk propellerhead (n=yogurt2u@host89.190-136-111.telecom.net.ar) |
| 20:07.20 | *** part/#asterisk fudpucker (n=here@75.151.177.173) |
| 20:07.53 | tzafrir_laptop | spammed back on the list and feels much better |
| 20:12.20 | *** part/#asterisk nny_1 (n=scott@64.203.244.146) |
| 20:18.22 | *** join/#asterisk kc2tnk (n=fskrotzk@host198.textwise.com) |
| 20:19.18 | *** join/#asterisk ziram19 (n=chatzill@41.226.158.169) |
| 20:20.29 | *** join/#asterisk lunaphyte_ (n=lunaphyt@unaffiliated/lunaphyte) |
| 20:27.51 | ziram19 | hi |
| 20:28.21 | ziram19 | how to change uri in the header from |
| 20:29.15 | ezrafree | [TK]D-Fender: okay i see what youmean, thanks for the information. so really it would be a waste of my time to develop the client which is the reason for using a solution like skype, ekiga, or googletalk |
| 20:29.49 | [TK]D-Fender | ezrafree: Its just a question of what value each solution brings you and the complexity of each |
| 20:30.06 | [TK]D-Fender | ezrafree: If your need is "disposible" then it should be easy for the end-user |
| 20:30.24 | Fairman | Can somebody tell me if this is possible w/ MOH... I need MOH to play music to be played like normal when on hold, but ever 15-30 seconds randomize a message ("Did you know, blah") |
| 20:30.45 | Fairman | ^ hope that made sense... |
| 20:30.55 | ezrafree | [TK]D-Fender: yes my need is disposable, i'd probably only be interviewing each person once |
| 20:31.05 | [TK]D-Fender | Fairman: Use an extrenal streaming source that will aloow this, or bluid a SINGLE MoH file that has your message pre-mixed into it |
| 20:31.12 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 20:31.14 | [TK]D-Fender | build* |
| 20:31.19 | Fairman | [TK]D-Fender: thanks |
| 20:31.37 | grandpapadot | Fairman: You could transfer the caller to a Queue and probably achieve close to what your after or just mix in an audio message in your MoH audio files. |
| 20:31.41 | ezrafree | [TK]D-Fender: i guess my goal was to try to automate the recording of these interviews with server-side scripts but it wouldn't much extra trouble to just upload them. |
| 20:31.56 | Fairman | grandpapadot: thanks! |
| 20:32.04 | grandpapadot | ezrafree: If you just need to record calls, try http://udigits.com, neat. |
| 20:33.35 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:35.39 | [TK]D-Fender | grandpapadot: EW |
| 20:35.46 | Alan_Hicks | I'm at my wit's end here guys. |
| 20:35.48 | [TK]D-Fender | grandpapadot: Horrific. |
| 20:36.01 | [TK]D-Fender | Alan_Hicks: It will die a lonely death! |
| 20:36.26 | Alan_Hicks | I've got an asterisk server with Polycom phones that has a constant echo problem. I've set echocancel=yes, echotraining=yes (and no) and I've changed out the cards. |
| 20:36.59 | Alan_Hicks | Currently I'm running a Sangoma analogue card with two FXO modules, but I've used a Digium TDM410 with two FXO modules without any noticeable difference. |
| 20:37.35 | [TK]D-Fender | Alan_Hicks: What card model exactly, what * version, what zaptel/DAHDI version. |
| 20:37.46 | Alan_Hicks | I'm not sure where else to go from here. I've even eliminated all potential grounding problems by plugging everything into the same UPS. |
| 20:37.58 | Alan_Hicks | [TK]D-Fender: Give me a second and I'll get that for you. |
| 20:38.21 | grandpapadot | [TK]D-Fender: LOL, what's horrific? |
| 20:38.38 | [TK]D-Fender | grandpapadot: using a Queue as MoH |
| 20:38.47 | Alan_Hicks | asterisk-1.4.21.2, zaptel-1.4.11 |
| 20:38.54 | [TK]D-Fender | grandpapadot: How will you specifically grab your caller back? |
| 20:39.04 | Alan_Hicks | Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 20:39.09 | [TK]D-Fender | grandpapadot: and think of the CDR mess... |
| 20:39.17 | [TK]D-Fender | Alan_Hicks: What EC routine? |
| 20:39.31 | grandpapadot | [TK]D-Fender: Well, I DID suggest just mixing in the periodic recordings with his MoH which is what I would do. |
| 20:39.44 | Alan_Hicks | [TK]D-Fender: EC routine? |
| 20:39.54 | grandpapadot | [TK]D-Fender: yea, you're right, would be really flakey to use a Queue like this. |
| 20:40.22 | [TK]D-Fender | Alan_Hicks: There are multiple algorithms for EC you know... which is Zaptel using? |
| 20:41.07 | Alan_Hicks | I'm not sure. Would that be defined in /etc/zaptel.conf? |
| 20:41.23 | [TK]D-Fender | Alan_Hicks: "ztcfg -vvvv" |
| 20:41.49 | Alan_Hicks | MG2 |
| 20:41.51 | Alan_Hicks | Thanks. |
| 20:41.58 | [TK]D-Fender | Alan_Hicks: Try another. |
| 20:42.10 | [TK]D-Fender | Alan_Hicks: if you don't like any of the others, try OSLEC next |
| 20:42.12 | [TK]D-Fender | ~oslec |
| 20:42.13 | jbot | [~oslec] OSLEC is the Open Source Line Echo Canceler. It is an superior alternative to the native SWEC routines in Zaptel/DHADHI and debatably a small notch below that of Digium's HPEC and Sangoma's SoftEcho in effectiveness. Web site : http://www.rowetel.com/ucasterisk/oslec.html , Mailing list : https://lists.sourceforge.net/lists/listinfo/freetel-oslec |
| 20:42.52 | Alan_Hicks | [TK]D-Fender: Thanks. I'll get to reading the documentation on different echo cancellers. |
| 20:45.22 | *** join/#asterisk rti (n=chatzill@41.226.158.169) |
| 20:45.58 | *** join/#asterisk af_ (n=getsmart@88-149-241-38.dynamic.ngi.it) |
| 20:52.07 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 20:52.12 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 20:52.34 | VxJasonxV | So I had a crazy idea the other day, I learned that iChat does SIP for it's voice and video conferencing stuff... |
| 20:52.41 | VxJasonxV | ...is there a way to get iChat and Asterisk to play nice together? |
| 20:53.00 | VxJasonxV | I have a debug log of the SIP packets sent between two users on different networks, and it's really intriguing to me |
| 20:54.39 | Alan_Hicks | Quick dumb-ass question if I may. |
| 20:54.44 | VxJasonxV | :( |
| 20:54.54 | VxJasonxV | oh, you weren't talking about my question... XD |
| 20:54.59 | VxJasonxV | comprehention is a good thing |
| 20:55.04 | Alan_Hicks | When there is echo on the line, both parties should hear it? Or is it possible that only one party would hear the echo? |
| 20:56.12 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 20:56.23 | orkid | the latter is possible |
| 20:56.46 | Alan_Hicks | Thank you. The latter is exactly what I'm seeing^Whearing here. |
| 20:56.53 | jasonwoot | which version of kernel-smp-devel is required to compile zap? |
| 20:56.57 | Alan_Hicks | goes back to reading. |
| 20:58.43 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 20:58.53 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 20:59.00 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 20:59.04 | *** part/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:59.41 | dssman | can someone tell me what ports need to be forwarded to my asterisk server... I had it running a year ago, no changes have been made, but now I dont have any voice... only call setup is working |
| 21:00.00 | *** join/#asterisk ziram19_ (n=chatzill@41.226.158.169) |
| 21:00.15 | [TK]D-Fender | ~sipnat |
| 21:00.16 | 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 |
| 21:00.18 | [TK]D-Fender | dssman: ^^^ |
| 21:01.21 | dssman | thx, goin to read :) |
| 21:01.38 | [TK]D-Fender | jasonwoot: The one that matches your kernel |
| 21:01.40 | Alan_Hicks | I must recompile zaptel in order to use a different echo canceller? |
| 21:01.46 | *** part/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 21:02.06 | [TK]D-Fender | Alan_Hicks: Zaptel yes, DAHDI, no. OSLEC Yes for Zaptel, and unsure for DAHDI |
| 21:02.40 | Alan_Hicks | Thank you. I was reasonably sure of that, but wanted to be positive before spending the time recompiling zaptel and wanpipe. |
| 21:03.20 | jasonwoot | Fender, it didn't look like the version numbers matched, but I'll check again |
| 21:10.02 | dssman | DOHHHHHHHHHH it was my external IP |
| 21:10.14 | dssman | I didnt change it in my SIP.conf file :( |
| 21:14.05 | dssman | thx [tk] |
| 21:15.53 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 21:16.57 | *** join/#asterisk AiT (n=airani@66-146-175-59.skyriver.net) |
| 21:18.10 | AiT | how can i define a phone's IP address in SIPmacaddress.cnf ? If not, how would I define an IP for a phone configured through TFTP? |
| 21:19.57 | AiT | anyone alive? |
| 21:20.43 | grandpapadot | AiT: FreePBX? TrixBox? |
| 21:20.53 | AiT | grandpapadot, Freepbx |
| 21:20.58 | grandpapadot | Try #freepbx |
| 21:21.15 | AiT | I figured, thanks |
| 21:29.52 | [TK]D-Fender | BBIAB |
| 21:39.03 | *** join/#asterisk kerx (n=prepro@adsl-68-123-206-108.dsl.irvnca.pacbell.net) |
| 21:40.44 | *** join/#asterisk teknoprep (n=cmr@unaffiliated/teknoprep) |
| 21:41.29 | teknoprep | hi all.. i have an IVR setup... wth option 1 being that if this is pressed it will email out a message to a predetermined list and then drop into Dial Tone so the user can then dial out. |
| 21:41.43 | teknoprep | the problem is i do not know how to email out from asterisk script |
| 21:41.52 | teknoprep | everything else works fine |
| 21:45.45 | telnettech | ok i have another strange problem. I have a customer that has mailbox that even if you select the delete option, the messages dont seem to be getting deleted.....it is only 1 mailbox. Has anyone had this same issue and any possible solution |
| 21:46.12 | telnettech | BTW it is Asterisk 1.2.10 version |
| 21:48.19 | *** part/#asterisk meuserj (n=meuserj@indianalifesciences.com) |
| 21:50.10 | Yourname` | OTQ: Anyone know when you provision a new Polycom IP 330 and it doesn't let you dial a number, rather a SIP URI? What needs to be done? |
| 21:50.46 | lmadsen | Yourname`: need to change the default dialing style to be numbers and not SIP URIs |
| 21:50.51 | lmadsen | the admin manual should have something about that |
| 21:51.23 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 21:54.14 | jeffik | anybody running a live conference call service? I need it for a cusomer |
| 21:55.46 | teknoprep | is it possible to run a shell script from asterisk script ? |
| 21:56.20 | Qwell | teknoprep: asterisk script? |
| 21:56.29 | teknoprep | from an asterisk dial plan |
| 21:59.16 | kerx | teknoprep, you can run System() and have it run your shell script. |
| 21:59.19 | kerx | for example: |
| 21:59.42 | kerx | s,n,System("/path/to/shell_script.sh arg1 arg2") |
| 22:00.59 | *** join/#asterisk martyn-dev (n=martyn@200.71.48.212) |
| 22:01.02 | martyn-dev | Hii |
| 22:01.03 | martyn-dev | :D |
| 22:01.07 | martyn-dev | I need some help .. |
| 22:02.37 | martyn-dev | I'm tryning install patch for app_quee in logger.c.. is a patch publish for Vixtor user on digium page.. somebody know how it ? |
| 22:03.47 | martyn-dev | I have a problem. I've installed ok, but in the moment when i launch a call to Queue dont show nothing in mysql or asterisk log :'( |
| 22:04.01 | martyn-dev | Some help ? |
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| 22:27.18 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
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| 22:29.06 | FruitBasket | help? Line: GotoIf($[$[${fetchedacct}=0] | $[${acctenable}=0] | $[${fetched}=0] | $[${extenable}=0]]?dialDisabled,1) -- all brackets seem to match. Error: syntax error: syntax error, unexpected '=', expecting $end; Input: =0; and for the same line: syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input: |
| 22:29.07 | FruitBasket | 0 | 0 | 1 | <blank space> |
| 22:30.15 | FruitBasket | may have larger issues |
| 22:31.13 | *** join/#asterisk JonOnt (n=Jon@72.34.90.74) |
| 22:33.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:34.04 | FruitBasket | got it.. database fetch failed, so... variable was never set.. or something, and $[] failed, causing the if to fail. |
| 22:34.18 | FruitBasket | I need to check the "fetched" var first to make sure I actually got something from the DB. |
| 22:41.01 | *** join/#asterisk stevie[xxx] (n=stevie@e177151081.adsl.alicedsl.de) |
| 22:47.10 | martyn-dev | Hi : Ihave "error: unknown field âdestroy_funcâ specified in initializer" error in make on addons on 1.4 .. |
| 22:49.20 | stevie[xxx] | is there a known problem with mISDN and asterisk 1.6? i dont get free channels |
| 22:53.13 | *** join/#asterisk telecos (n=sergio@154.166.219.87.dynamic.jazztel.es) |
| 22:55.17 | phix | FruitBasket: nice |
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| 23:08.34 | *** join/#asterisk emiller (n=ed@c-76-124-139-140.hsd1.pa.comcast.net) |
| 23:09.52 | emiller | i have a client saying that only 4 lines of their 800p are being used, they have analog lines connected to 6 of them. remotely, how can i see if the two ports are being utilized |
| 23:10.16 | phix | zap show channels ? |
| 23:10.35 | martyn-dev | yeah. zap show * |
| 23:11.40 | phix | <3 |
| 23:11.56 | phix | Now how do I get a Sipura 3002 working correctly ? :) |
| 23:12.13 | phix | I know it isn't entirely related to asterisk :P |
| 23:12.20 | emiller | it shows all 8 channel extensions, however, in dahdi-channels.conf channels 5-8 do not have "(in-use)" in the comment: http://pastebin.com/dd5e2ee3 |
| 23:12.56 | phix | but still :) THe line is quite as shit, I get echos, and when I ring a number then hang up, it still rings the person. |
| 23:13.12 | *** join/#asterisk ldsjohn (n=jbunn@c-24-22-52-147.hsd1.wa.comcast.net) |
| 23:13.31 | [TK]D-Fender | emiller: You only show 1 part of the config. Show ALL of it, and CLI channel dumps |
| 23:14.02 | ldsjohn | anyone know how to stop the vm-login from playing, im sure im missing something in the voicemailMain docs, im trying to get rid the "Comedian mail, mailbox " voice that plays |
| 23:14.15 | [TK]D-Fender | ldsjohn: Remove the sound file |
| 23:16.09 | emiller | [TK]D-Fender: dahdi-channels: http://pastebin.com/d64b8260d cli dump: http://pastebin.com/d1680fa48 |
| 23:16.15 | *** join/#asterisk loather-work (n=khudson@internal-nat.djnetworks.net) |
| 23:16.41 | ldsjohn | when I remove the sound file |
| 23:16.45 | ldsjohn | it hangs up |
| 23:16.51 | ldsjohn | cause there is no sound file |
| 23:17.28 | [TK]D-Fender | ldsjohn: then copy a silence file over it |
| 23:18.16 | [TK]D-Fender | emiller: No. "core show channels concise" |
| 23:21.18 | emiller | [TK]D-Fender: http://pastebin.com/d13051bec |
| 23:21.34 | martyn-dev | problem solved: ) |
| 23:22.00 | [TK]D-Fender | emiller: there you have it 2 in use |
| 23:22.17 | martyn-dev | When you try use queue storage on mysql with logger.c check sock in logger.conf of mysql: sock=/var/run/mysqld/mysqld.sock for debian etch. byt |
| 23:22.19 | martyn-dev | bye |
| 23:22.23 | phix | wtf is this wank we have to do now with prefix some commands with core? |
| 23:22.36 | phix | can't it assume we are speaking about core if nothing is prefixed? |
| 23:23.25 | [TK]D-Fender | ~assume |
| 23:23.26 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav It makes an (ass) out of (u) and (me) |
| 23:24.16 | emiller | [TK]D-Fender, martyn-dev: i dont think i was clear in my question. or maybe i am not picking understanding your help. they are saying that they can only have 4 simultaneous calls at one time, even though they have 6 analog lines plugged into the card |
| 23:24.31 | [TK]D-Fender | emiller: BS |
| 23:24.55 | [TK]D-Fender | emiller: Show us the FAILURE |
| 23:25.48 | emiller | im going off of straight hearsay at this point. going off my of dahdi-chan.conf, it seems like i am able to use all 8 ports |
| 23:26.13 | [TK]D-Fender | emiller: Go verify the equivalent of zaptel.conf as well |
| 23:26.15 | emiller | ill see if i can replicate the problem. thanks [TK]D-Fender and martyn-dev |
| 23:26.36 | martyn-dev | XD |
| 23:27.25 | emiller | zaptel.conf: fxsks=1,2,3,4,5,6,7,8 |
| 23:28.36 | [TK]D-Fender | emiller: /etc/dahdi/system.com |
| 23:28.39 | [TK]D-Fender | emiller: /etc/dahdi/system.conf |
| 23:29.24 | emiller | sorry, i still get a little cloudy with the whole dahdi/zaptel switch... |
| 23:29.47 | emiller | http://pastebin.com/d4ee6d896 |
| 23:32.02 | *** part/#asterisk martyn-dev (n=martyn@200.71.48.212) |
| 23:32.26 | [TK]D-Fender | emiller: Ok, the files check out... proof is in the pudding. |
| 23:32.57 | emiller | [TK]D-Fender: thanks, just wanted to make sure i wasn't missing something before coming back and telling them to stick it. |
| 23:32.59 | emiller | :D |
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| 23:39.10 | orkid | wth? x-lite does not want to show me line/call information |
| 23:39.21 | orkid | anyone know how to get it? (like codec used, etc) |
| 23:40.11 | orkid | according to their pdf you're supposed to hover mouse over the 'line 1' button |
| 23:40.11 | orkid | while on a call on line 1 |
| 23:42.35 | [TK]D-Fender | orkid: Keep reading their manual |
| 23:43.30 | orkid | ... |
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| 23:50.22 | FinboySlick | Hello gang. I have a bit of a situation here where we apparently keep busting FXO modules, both sangoma and digium... This isn't lightning season and the lines appear to be fine, anybody care to provide an insight? Would some sort of filter/surge protector really help, what should I be looking for? |
| 23:52.10 | [TK]D-Fender | FinboySlick: Just your typical filer. Many power basrs have them |
| 23:52.27 | [TK]D-Fender | bars* |
| 23:52.58 | FinboySlick | [TK]D-Fender: That's already in the plans.. Wouldn't help if the line is 'hot' though, right? |
| 23:53.18 | [TK]D-Fender | orkid: FinboySlick It should keep things withing reasonable spec |
| 23:56.41 | echinos | I'm getting "peer is not supposed to register" when my softphone tries to reg - what do I put in sip.conf to make registration OK or required? |
| 23:56.54 | echinos | I imagine it is better to have softphones register... |
| 23:57.04 | [TK]D-Fender | orkid: Does appear the doc was wrong |
| 23:57.07 | echinos | but I'm an * noob |
| 23:57.33 | [TK]D-Fender | echinos: it means "stop setting 'host=someip' and do "host=dynamic' instead" |
| 23:57.52 | echinos | ah, well, I'm not doing either ;) |
| 23:58.12 | [TK]D-Fender | echinos: pastebin the actuall error and your matching config |
| 23:58.27 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 23:58.29 | echinos | meh, let me go fiddle after that bit of wisdom |
| 23:58.52 | echinos | This is the first time I;ve played with * for... many moons |
| 23:58.55 | [TK]D-Fender | echoOH? You mean now you're going to actually look? |
| 23:59.48 | echinos | I'm not clear on what stuff is required, what all does what yet |