| 00:00.32 | Nugget | I've got two guns, one for each of you. |
| 00:00.42 | Qwell[] | ph|ber: John...what? |
| 00:00.48 | Qwell[] | there are plenty of John's |
| 00:01.20 | ph|ber | bigalow |
| 00:01.26 | ph|ber | DUECE! |
| 00:01.32 | Qwell[] | never heard of him |
| 00:01.41 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:01.41 | ph|ber | john bigalow? |
| 00:01.44 | ph|ber | he works at * |
| 00:01.53 | ph|ber | been on the phone with him all day |
| 00:01.56 | Qwell[] | ph|ber: * isn't a company |
| 00:01.59 | Qwell[] | Digium, however, is |
| 00:02.01 | ph|ber | digium |
| 00:02.11 | Qwell[] | What is he, sales? |
| 00:02.16 | ph|ber | support |
| 00:02.36 | Qwell[] | call back tomorrow? |
| 00:03.07 | ph|ber | i figured hes gone. |
| 00:04.40 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 00:09.30 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
| 00:11.37 | [TK]D-Fender | ph|ber : Only reason I can thing of is permissions |
| 00:11.53 | ph|ber | ya checked that |
| 00:11.54 | ph|ber | 666 |
| 00:14.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:15.18 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:15.19 | [TK]D-Fender | ph|ber : <mac>-directory.xml or 00000000-directory.xml? |
| 00:15.24 | ph|ber | mac |
| 00:15.31 | ph|ber | 0004f2038e06-directory.xml |
| 00:15.50 | ph|ber | and i had the 0000 |
| 00:15.53 | ph|ber | as a default |
| 00:16.01 | ph|ber | for me to edit and save to other. |
| 00:16.06 | ph|ber | could that be the problem? |
| 00:16.44 | [TK]D-Fender | ph|ber : No... |
| 00:16.55 | [TK]D-Fender | Sure your poly still has the right login for that folder? |
| 00:17.20 | ph|ber | all the configs are tftp with the directory.xml |
| 00:17.22 | ph|ber | the configs load. |
| 00:17.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-23-125.red.bezeqint.net) |
| 00:17.33 | ph|ber | just the *-directory.xml doesnt |
| 00:22.01 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 00:25.24 | [TK]D-Fender | ph|ber : Hmmm |
| 00:25.39 | ph|ber | any ideas? |
| 00:25.43 | [TK]D-Fender | ph|ber : Change the file (add an entry) |
| 00:26.07 | [TK]D-Fender | ph|ber : tftp doesn't see file changes the same way FTP does (no date) |
| 00:26.13 | [TK]D-Fender | ph|ber : give that a show |
| 00:26.15 | [TK]D-Fender | shot* |
| 00:26.21 | ph|ber | k.. hold brb |
| 00:26.27 | [TK]D-Fender | BBIAB |
| 00:27.19 | *** join/#asterisk [Thomas] (n=thomask@Barrie-ppp216096.sympatico.ca) |
| 00:27.46 | [Thomas] | Greetings and Salutations all |
| 00:28.50 | *** part/#asterisk nailbags|work (n=neilbags@149.171.94.134) |
| 00:28.51 | [Thomas] | can someone help me with my asterisk config? Its not answering the line... |
| 00:29.28 | *** join/#asterisk nailbags|work (n=neilbags@149.171.94.134) |
| 00:30.27 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:32.08 | *** part/#asterisk [Thomas] (n=thomask@Barrie-ppp216096.sympatico.ca) |
| 00:32.27 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 00:38.38 | *** join/#asterisk andymul (n=iCallAnd@cpe-69-203-217-237.nyc.res.rr.com) |
| 00:39.03 | *** join/#asterisk Echelo (i=Echelo@c-67-181-14-240.hsd1.ca.comcast.net) |
| 00:39.11 | andymul | Anyone in here that knows PHP and Asterisk that's interested in some programming work please PM me |
| 00:42.48 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 00:48.15 | Dovid | what does no valid handler mean ? |
| 00:48.22 | *** join/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 00:53.18 | Dovid | Argh. For some reason asterisk stopped loading anything new that I add to mysql |
| 00:53.29 | Dovid | i rebooted the machine but nothing seems to work |
| 00:59.04 | CrashHD | I have an issue where rtp debug is not showing any rfc2833 packets being sent to to another asterisk even though both servers are clearly set to rfc2833, anyone have any thoughts on how to troubleshoot this? |
| 00:59.19 | CrashHD | I do however see them recieved from the phone |
| 01:00.17 | CrashHD | dtmf is being sent somehow, I believe it is inband |
| 01:01.18 | CrashHD | definitely no sip info messages being sent |
| 01:06.55 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 01:13.35 | *** join/#asterisk [-Thomas-] (n=thomask@Barrie-ppp200596.sympatico.ca) |
| 01:14.15 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 01:15.50 | [Thomas] | Greetings and Salutations! |
| 01:15.53 | tzafrir_laptop | [Thomas], ask a more specific question |
| 01:16.33 | tzafrir_laptop | yes, I figure some people could help you with config fils if you gave a more specific description of the problem |
| 01:17.42 | [Thomas] | tzafrir_laptop: I shall ;) |
| 01:18.52 | [TK]D-Fender | tzafrir_laptop : Likely AMP/A@H related given the previous departure... |
| 01:19.45 | [Thomas] | Last year I had a asterisk system put in my home by a friend that wanted to lengthen his dialing zone, without the long distance charges. I simply want to modify it from a broadband link to his system now (I no longer have broadband) and want it to answer the phone, give the caller a couple options and dump to voicemail if the person being called is not available. Right now the box is not even answering, can someone assist me in gettin |
| 01:20.52 | [Thomas] | the card is a 00:10.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537, and I am running Asterisk CVS-v1-0-01/31/05-18:28:04 |
| 01:21.29 | hads | Mmm CVS Asterisk is old. |
| 01:21.44 | [TK]D-Fender | Jan '05 is OLD |
| 01:21.51 | [Thomas] | Yes, it is.. ;) |
| 01:22.11 | [Thomas] | however I dont really want to update to a new current version being on dialup. |
| 01:22.22 | hads | It's not that big! |
| 01:22.37 | [Thomas] | how big? |
| 01:23.05 | hads | ~10M for Asterisk |
| 01:23.30 | hads | and ~1M for zaptel |
| 01:23.40 | [Thomas] | oh, and here I thought it was over 100 mb :/ heh. What benefits would I get over that old a version to the new version? |
| 01:26.36 | quid2478 | dialup? whats that? |
| 01:26.44 | [Thomas] | hads: is there a best place to download an 'all inclusive' version, or just cherry pick from asterisk.org |
| 01:27.16 | file | all inclusive version? that makes us sound like a resort... |
| 01:27.46 | hads | http://ftp.digium.com/pub/asterisk/ has all the deluxe luxary version |
| 01:28.00 | [Thomas] | heh.. the way some ppl make asterisk sound is all inclusive ;) |
| 01:28.10 | [Thomas] | thanks hads |
| 01:28.30 | hads | It is! |
| 01:29.01 | tzafrir_laptop | that's only asterisk. If you happen to need zaptel, check http://asterisk.org as well |
| 01:29.34 | [Thomas] | so it seems, I've even subscribed to the taug's mailing list (toronto asterisk users group) ;) |
| 01:29.34 | [Thomas] | ok |
| 01:30.00 | [Thomas] | thanks |
| 01:30.07 | hads | Or just traverse directories http://ftp.digium.com/pub/zaptel/ |
| 01:30.38 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 01:37.03 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 01:43.27 | quid2478 | Thomas: You are plan on using Asterisk over dialup? |
| 01:44.13 | quid2478 | Never mind... read the rest of your message.... you want a high-tech answering device :) |
| 01:44.27 | *** join/#asterisk madfactor (i=madfacto@74.128.34.115) |
| 01:45.21 | madfactor | got a general question |
| 01:45.24 | andymul | Anyone in here that knows PHP and Asterisk that's interested in some programming work please PM me |
| 01:47.44 | madfactor | aside of TDMoE, is there a way to setup asterisk using a Digium TDM card to connect two Meridian Option 11C boxes, that are positioned at opposite ends of a point to point T1 using IAX2? |
| 01:48.53 | Nivex | madfactor: You could do some grouping and dialplan magic. Kinda depends on how you mean "connect" |
| 01:49.25 | Un1x | yo can i run ASterisk on a Computer that is 600megs ram, and a Sempron 2200+ comp? |
| 01:49.35 | Qwell | Un1x: sure |
| 01:49.43 | Qwell | run it on a P1 133 |
| 01:49.45 | Qwell | with 32mb ram |
| 01:49.45 | Un1x | as long as i stay away from the processor intensive g.729 codece right? |
| 01:49.51 | Nivex | run it on a wrt54g! |
| 01:50.10 | madfactor | OH K... the two Meridian boxes connect using a traditional T1 multiplexer box, where 12 channels communicate directly on the T1. |
| 01:50.15 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:50.19 | Un1x | «Qwell» whats a good codec that don't use much processor resources, but wont exceed 90KB ...? |
| 01:50.21 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:50.21 | madfactor | and 11 channels are ethernet. |
| 01:50.25 | Qwell | Un1x: ulaw |
| 01:50.39 | Un1x | Qwell Ulaw will stay under 90KB? |
| 01:50.41 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 01:50.43 | docelmo | Anyone know where I can get the app_swift code? |
| 01:50.48 | Qwell | 83 or so, I believe it is |
| 01:50.51 | franck | Hi all |
| 01:50.54 | Un1x | kb or 83Kb? |
| 01:50.56 | Qwell | docelmo: What's that? |
| 01:50.58 | Qwell | Un1x: kbit |
| 01:50.59 | JunK-Y | Un1x: dont trust him, hes lying you, use g711u :P |
| 01:50.59 | Un1x | err KB or kb |
| 01:51.02 | Un1x | so kb |
| 01:51.04 | Qwell | JunK-Y: :P:P |
| 01:51.17 | Qwell | or alaw |
| 01:51.24 | Un1x | alay? |
| 01:51.26 | Un1x | *alwa? |
| 01:51.28 | Un1x | err |
| 01:51.28 | JunK-Y | swift, u mean app_cepstral? |
| 01:51.30 | madfactor | Can the zaptel TDM cards, be connected directly to the DTI/PRI cards, and "passthrough" the 12 digital channels to an asterisk boxes on the other side using IAX2? |
| 01:51.30 | Qwell | give up |
| 01:51.33 | Un1x | crap alwa is new one? |
| 01:51.38 | Qwell | ... |
| 01:51.50 | Qwell | Un1x: just say "that codec with an a" |
| 01:51.58 | Qwell | You seem to be having difficulty spelling it :p |
| 01:52.01 | docelmo | sure.. Where can I get it? :) |
| 01:52.07 | madfactor | using only a simple "tie" dialplan?? |
| 01:52.10 | Qwell | docelmo: ugh...don't do it |
| 01:52.28 | Un1x | Qwell Ulaw and Alaw are both uncompressed right? |
| 01:52.30 | madfactor | point to point configuration, just like the physical T1 channels? |
| 01:52.30 | docelmo | What's the easiest way to intergrate cepstral into it? |
| 01:52.34 | docelmo | Un1x yes |
| 01:52.53 | Un1x | hmm i hope it dont produce some, kinda voice lag |
| 01:52.54 | franck | is it me or I have noticed that all SIP phones between the moment you finishing dialling the number and the moment the SIP protocol has established the connection and audio channels the phone stay silent? If you do on asterisk some enum lookups and other things like this, the user on the phone, think his phone is dead |
| 01:53.02 | Un1x | like when i speak and the guy hears me after like 10 seconds |
| 01:53.38 | madfactor | I don't mean to use asterisk as a voip gateway for other phones yet... just as a point-2-point for connecting the boxes? |
| 01:54.15 | franck | can anybody point me to a phone that has a comfort ring, like ekiga has... |
| 02:01.19 | madfactor | ??? |
| 02:06.57 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 02:08.18 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 02:09.20 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
| 02:10.26 | clyrrad | correct me if I am wrong - but incomming calls - do you not set their context=incoming in iax.conf which will point to the respective DID in extensions.conf under the [incoming] context? |
| 02:12.19 | *** part/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 02:13.13 | *** join/#asterisk Skarmeth (n=Skarmeth@201009002118.user.veloxzone.com.br) |
| 02:15.59 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 02:16.46 | shmaltz | wb tzafrir |
| 02:17.18 | shmaltz | anybody from voipsupply here? |
| 02:18.19 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 02:25.10 | Scrye | what could be a reason why my grandstream 101 is no longer sending the digits i press after a call has started |
| 02:25.34 | Scrye | im trying the dial the console and it dials it but then the buttons dont work from just that phone |
| 02:26.13 | Scrye | one on the next ip up, a ut starcomm f1000g works fine |
| 02:26.14 | Scrye | no nat involved |
| 02:27.44 | tzafrir_laptop | shmaltz, hi |
| 02:28.18 | shmaltz | tzafrir, hi |
| 02:28.28 | shmaltz | did you get the answers for me on the FXO? |
| 02:28.34 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 02:29.31 | *** join/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 02:31.15 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 02:31.52 | *** join/#asterisk trbldwine (i=troubled@c-71-194-161-170.hsd1.il.comcast.net) |
| 02:39.23 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 02:42.52 | *** join/#asterisk magikh0e (n=magik@node-8467.tor.pppoe.execulink.com) |
| 02:43.29 | magikh0e | Anyoen know of an appliance you can by pre built, or maybe a document explaining on hardware needed etc to build an asterix network to support 2 sites with about 15 users? |
| 02:43.50 | *** join/#asterisk foxmjay (i=foxmjay@adsl196-72-34-217-196.adsl196-10.iam.net.ma) |
| 02:43.51 | *** join/#asterisk jgoo (n=foo@ppp134-168.adsl.forthnet.gr) |
| 02:43.59 | shmaltz | magikh0e, check the wiki |
| 02:44.18 | shmaltz | magikh0e, also xorcom sells somthing like this: |
| 02:44.20 | shmaltz | www.xorcom.com |
| 02:44.27 | jgoo | Hey guys. I am looking to write my first dialplan script - I would just like to print a list of variables to console as a first tr |
| 02:44.40 | shmaltz | http://www.xorcom.com/ts-1/features.html |
| 02:44.57 | shmaltz | jgoo, this is not ssh |
| 02:45.08 | jgoo | *try, wondering if there is a good script / dialplan 'best practicesä |
| 02:45.11 | [TK]D-Fender | shmaltz : Whats your question? |
| 02:45.22 | jgoo | shmaltz: the dialplan calls scripts though right? |
| 02:45.27 | shmaltz | [TK]D-Fender, none for the moment |
| 02:45.46 | shmaltz | jgoo, if thats what you tell it to do |
| 02:45.48 | jgoo | I am not talking about the macros, but isnät there a simple bash call from extensions? |
| 02:45.59 | shmaltz | jgoo, system will do |
| 02:46.17 | shmaltz | jgoo, if you need feedback then agi will do |
| 02:46.28 | jgoo | shmaltz: sounds good then. Is this the best way to do something liek this (of course, printing a list of variables is a bit inane right now) |
| 02:46.47 | shmaltz | jgoo, no it's not the best way |
| 02:46.56 | shmaltz | jgoo, what exactly do you want to do? |
| 02:46.57 | jgoo | what I would like, I guess, is to make a db call, in the script, and return an extension number into a call forward (if busy or DND) |
| 02:47.20 | [TK]D-Fender | jgoo : What kind of variables? Printed where/how? what is your REAL goal? |
| 02:48.02 | jgoo | [TK]D-Fender: I would like to print the number associated with the line, to be sure that this inforation is passed correctly into the system |
| 02:48.09 | shmaltz | jgoo, something like this will do: |
| 02:48.10 | shmaltz | exten => _X.,1,Dial(${DB(CF/${EXTEN})}) |
| 02:48.47 | shmaltz | if CF/${EXTEN} is set to something like SIP/1234 |
| 02:49.26 | jgoo | shmaltz: but that will be dynamic? what do you mean by set? I guess I need to learn the syntax of extensions really quick :) |
| 02:49.36 | jgoo | (reading asteriskguru right now= |
| 02:50.02 | shmaltz | jgoo, and dont forget the wiki |
| 02:50.11 | shmaltz | jgoo, and read and read and read |
| 02:50.13 | shmaltz | ~docs |
| 02:50.17 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:50.17 | shmaltz | ~wiki |
| 02:50.34 | shmaltz | ~wiki asterisk |
| 02:50.53 | CrashHD | anyone have a problem where a featuremap key sequence conflicts with trying to use an IVR? |
| 02:51.08 | shmaltz | CrashHD, it shouldn't |
| 02:51.08 | jgoo | shmaltz, [TK]D-Fender, no doubt, I am not allergic to reading, I would like to get a feel as well. As a 'bare minimum' script: |
| 02:51.23 | CrashHD | shmaltz: it does, tested and confirmed |
| 02:51.49 | shmaltz | CrashHD, I'm not conviced |
| 02:52.07 | jgoo | where should I put the script + how does asterisk pass (and what does it pass) variables to the script (and how would I pass the number associated with the dialled line). sorry if that all sounds too newbish |
| 02:52.09 | shmaltz | CrashHD, it's imposible |
| 02:52.20 | shmaltz | because featuremaps only work in the dial application |
| 02:52.21 | jgoo | I know I can develop more ideas from there if how this interaction works |
| 02:52.26 | CrashHD | I have blindtransfer set to ## |
| 02:52.29 | shmaltz | IVRs are not in a dial app |
| 02:52.32 | CrashHD | I use Tt in my dial() |
| 02:52.36 | CrashHD | and when I call out |
| 02:52.51 | CrashHD | if I press # once it is not sent |
| 02:53.00 | CrashHD | if I press another key it will get sent |
| 02:53.11 | CrashHD | but # will not be sent until another key is pressed |
| 02:53.25 | shmaltz | oh, you meant that way |
| 02:53.30 | shmaltz | yeah I've seen it |
| 02:53.44 | CrashHD | what should I use for featuremap keys then? |
| 02:53.49 | shmaltz | thats why * is more popular |
| 02:53.50 | CrashHD | since all keys are used in ivrs |
| 02:54.01 | shmaltz | nah, * is not so popular |
| 02:54.05 | CrashHD | hmm |
| 02:54.12 | shmaltz | in IVRs that is |
| 02:54.21 | CrashHD | to exit out of menu's and such? |
| 02:54.41 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:54.41 | Scrye | so i call from one extension to another and when i hit a key on either phone it blanks out |
| 02:54.44 | Scrye | and you cant hear the key hit |
| 02:54.44 | shmaltz | CrashHD, still not so popular |
| 02:54.54 | CrashHD | is this not a huge bug? |
| 02:54.58 | CrashHD | *sigh* |
| 02:54.58 | shmaltz | Scrye, what blanks out? |
| 02:55.05 | shmaltz | CrashHD, nah |
| 02:55.14 | CrashHD | everyone seems to shrug at it |
| 02:55.40 | CrashHD | and featuredigittimeout seems to have no affect |
| 02:55.43 | shmaltz | CrashHD, you understand what Scrye wants? |
| 02:55.51 | shmaltz | CrashHD, of course not |
| 02:56.08 | jgoo | So, on what you said:: exten => _X.,1,Dial(${DB(CF/${EXTEN})}) ;; this means match any numeric pattern, first priority is Dial (Macro) $ signifies the value from DB(), where would DB(... be defined? |
| 02:56.17 | CrashHD | shmaltz: I was not reading what Scrye was saying |
| 02:56.30 | shmaltz | CrashHD, no scrollback? |
| 02:56.40 | shmaltz | CrashHD, he has the same problem as you |
| 02:56.51 | CrashHD | no he doesn't |
| 02:56.56 | CrashHD | one extension to another |
| 02:56.58 | CrashHD | it blanks out? |
| 02:57.04 | CrashHD | my dtmf is not being sent |
| 02:57.13 | CrashHD | dunno how that is blanking out? |
| 02:57.21 | Scrye | when you hit a key |
| 02:57.27 | shmaltz | CrashHD, because that what it is |
| 02:57.41 | shmaltz | CrashHD, it blanks out for the duration of the key hit |
| 02:57.45 | shmaltz | Scrye, right? |
| 02:57.47 | Scrye | yes |
| 02:57.52 | Scrye | plus prolly another .2 seconds |
| 02:57.54 | shmaltz | Scrye, what key? |
| 02:57.57 | Scrye | any |
| 02:58.01 | CrashHD | nothing to do with what I'm talking about |
| 02:58.08 | shmaltz | Scrye, then you have a different problem |
| 02:58.14 | shmaltz | CrashHD, I realized |
| 02:58.21 | Scrye | digits or special keys |
| 02:58.22 | shmaltz | Scrye, what phone you using? |
| 02:58.32 | Scrye | 1 GS-100 and 1 F1000g |
| 02:58.37 | Scrye | (grandstream and ut starcomm) |
| 02:58.49 | shmaltz | and on both you have this problem? |
| 02:59.08 | Scrye | yes |
| 02:59.16 | jgoo | shmaltz: so speaking of best practice, I am guessing http://www.voip-info.org/wiki-Asterisk+AGI would actually be the best way to do what I am looking to do |
| 02:59.17 | shmaltz | Scrye, pb your sip.conf and comment wich one is the GS and which one is the F1000g |
| 02:59.21 | shmaltz | ~pb |
| 02:59.22 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 02:59.32 | shmaltz | jgoo, not at all, the dp would be |
| 02:59.50 | Scrye | so it has to be somthing i hosed |
| 03:00.09 | shmaltz | Scrye, pb sip.conf and extension.conf and I'll take a look at it |
| 03:01.28 | shmaltz | ~dp |
| 03:01.30 | jbot | [debianplanet] http://www.debianplanet.org: Free Beer for the users of the Free Linux of a Free Planet. Debian community site. Founders: robster + Netsnipe Editors: agl, solomon, flugh Uberguru: ZaNaGa, or DP present no content, in a tactless manner that is un-paralleled by other site that lack talent |
| 03:01.35 | Scrye | yeah well it was 50$ new |
| 03:02.19 | Scrye | http://pastebin.ca/101716 |
| 03:02.29 | shmaltz | Scrye, so was my brand new bike, but my bike can't make phone calls, and so can't the gs |
| 03:02.30 | Scrye | both .confs were from the examples plus the changes i just pasted |
| 03:03.19 | Scrye | well i was told to get a gs because it was cheap and worked with asterisk |
| 03:03.25 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:03.32 | *** join/#asterisk BryanFE2 (n=chatzill@ip72-194-209-224.sb.sd.cox.net) |
| 03:03.39 | shmaltz | Scrye, take out the t in the dial line so that it reads: |
| 03:03.41 | shmaltz | exten => 10001,1,Dial(SIP/ryan2,20,r) |
| 03:03.42 | shmaltz | reload and try again, let me know |
| 03:04.59 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 03:05.19 | Scrye | ok |
| 03:05.19 | Scrye | walking into the other room to test it |
| 03:05.25 | Scrye | the phone rings but now there is no voice |
| 03:05.50 | *** join/#asterisk tempest1 (n=asf@adsl-12-51-176.chs.bellsouth.net) |
| 03:06.11 | shmaltz | Scrye, thats because of the nat, will fix that by adding canreinvite=no to sip.conf for each phone |
| 03:06.43 | Scrye | ok |
| 03:06.50 | Scrye | 1 there is no nat |
| 03:06.52 | shmaltz | add to sip.conf: |
| 03:06.54 | shmaltz | canreinvite=no |
| 03:06.54 | Scrye | anymore i ripped it out |
| 03:06.55 | shmaltz | add it after the secret line for each device, then do a sip reload and test again and let me know |
| 03:07.03 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 03:07.04 | Scrye | now the digits work to the 1234 test |
| 03:07.27 | shmaltz | Scrye, you mean to say after taking out the t the problem is gone? |
| 03:07.45 | Scrye | yes except no voice works now |
| 03:08.10 | shmaltz | Scrye, then the problem might not be gone |
| 03:08.26 | shmaltz | Scrye, on the pb you had nat=yes, is it still natted? |
| 03:08.36 | shmaltz | did you add the canreinvite=no? |
| 03:08.40 | shmaltz | did you reload? |
| 03:09.38 | shmaltz | Scrye, you there? |
| 03:09.47 | Scrye | nothing is natted anymore |
| 03:09.47 | Scrye | and i issued 'reload' into asterisk -r |
| 03:10.03 | shmaltz | and no voice? |
| 03:10.15 | Scrye | my f1000g doesnt work behind nat for some reason |
| 03:10.30 | Scrye | so i took out all the nat by putting in a vpn to the asterisk server heh |
| 03:10.42 | Scrye | the voice is working now with the canreinvite=no |
| 03:10.51 | Scrye | now i can call my vm with it and it reckonizes the keys |
| 03:10.57 | Scrye | but phone -> phone the keys still dont work |
| 03:11.12 | shmaltz | Scrye, how do you know that the keys don't work? |
| 03:11.17 | shmaltz | does it still blank out? |
| 03:11.41 | shmaltz | ok, I got to go |
| 03:11.43 | shmaltz | cya |
| 03:11.45 | shmaltz | bye |
| 03:11.47 | shmaltz | gn |
| 03:12.49 | Scrye | bha |
| 03:14.29 | jgoo | ok, I am reading up on dialplan - how would I print something to console from dialplan? |
| 03:14.36 | infinity1 | noop |
| 03:14.54 | infinity1 | assuming you mean the * console, not the linux console |
| 03:15.04 | jgoo | noop I see, I guess that is kinda lik echo |
| 03:16.21 | jgoo | ok, inside a context, where I receive a call, how would I noop aparticular detail of that call? I would like to echo the //dialled// number |
| 03:18.38 | jgoo | aaah I guessI need a list of tings like $(DIALSTATUS) |
| 03:18.48 | jgoo | and others... |
| 03:20.08 | infinity1 | fuckin bored. anyone livin in sunnyvale? |
| 03:21.58 | BryanFE2 | hey all, I sort of don't believe my eyes, but google.com appears to be offline |
| 03:22.05 | BryanFE2 | anyone confirm? |
| 03:22.19 | infinity1 | works for me |
| 03:22.29 | Skram_Buffalo | works for me as well |
| 03:22.39 | Skram_Buffalo | able to ping from multiple locations |
| 03:22.44 | Skram_Buffalo | BryanFE2: traceroute |
| 03:23.06 | BryanFE2 | blockage within Cox Internet |
| 03:23.10 | BryanFE2 | cox.net |
| 03:23.19 | BryanFE2 | interesting |
| 03:23.47 | jgoo | adding at the start of the context >> exten => s,1,NoOp(Just a test ) |
| 03:23.51 | jgoo | didnt do anything though |
| 03:24.29 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 03:26.33 | jgoo | [TK]D-Fender - what does asterisk call the number that was used to dial a trunk? |
| 03:26.53 | jgoo | that is, if > 1 number reaches a trunk, how can I ask asterisk which number was dialled? |
| 03:27.04 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 03:33.32 | awk | CrashHD: the HD might have Crashed |
| 03:33.39 | awk | could be wrong |
| 03:34.35 | awk | ddos |
| 03:34.45 | awk | rrg |
| 03:35.02 | batphone | grr |
| 03:35.06 | batphone | <PROTECTED> |
| 03:37.39 | jgoo | hrm, asterisk1*CLI> I set verbose 1, and the cli stopped responding to all input |
| 03:38.07 | jgoo | I can change to diff terminal though on ctrl-alt |
| 03:40.38 | jgoo | when editing extensions.conf - (in configedit as it happens on trixbox) do I need to restart asterisk? |
| 03:40.59 | tempest1 | yes |
| 03:41.12 | tempest1 | just type "reload" in the asterisk terminal |
| 03:41.19 | tempest1 | I read that wrong, you don't need to restart it |
| 03:43.18 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:44.24 | *** join/#asterisk Echelo (i=Echelo@c-67-181-14-240.hsd1.ca.comcast.net) |
| 03:46.29 | jgoo | thanks tempest1 |
| 03:47.39 | *** join/#asterisk nerolab1 (n=Andrew@c-24-17-37-138.hsd1.wa.comcast.net) |
| 03:47.53 | nerolab1 | trying to recover a lost password on a cisco 7960 phone anybody willing to help? |
| 03:48.41 | jgoo | does noop dump to a logfile as well? this console fills up too much |
| 03:49.53 | *** join/#asterisk pigpen2 (n=mark@209.159.234.250) |
| 03:50.16 | *** join/#asterisk albertito (n=net@host76.201-252-17.telecom.net.ar) |
| 03:50.29 | *** join/#asterisk albertito (n=net@host76.201-252-17.telecom.net.ar) |
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| 03:53.42 | Scrye | good windows sip client? |
| 03:54.04 | jgoo | Scrrye, X-Lite works... |
| 03:54.46 | jgoo | on my outgoing - I get Executing Dial("SIP/200-0ee0", "ZAP/g0/9ANDHENUMBER|120|r") |
| 03:54.57 | jgoo | then, a little down |
| 03:55.10 | jgoo | Channel 0/1, span 1 got a hangup request |
| 03:55.29 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:55.32 | jgoo | does that mean the outgoing isn't striping off the 9? |
| 03:55.39 | jgoo | incoming works |
| 03:55.44 | jgoo | only one trunk specified |
| 03:56.20 | jgoo | correct zone loaded in zaptel |
| 04:00.53 | *** join/#asterisk Evanrude (n=david@ip70-185-153-117.lu.dl.cox.net) |
| 04:07.57 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 04:08.09 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
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| 04:12.42 | Dovid | if the GotoIfTime is false it goes to the next prioroty ? |
| 04:14.05 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 04:17.09 | jgoo | ok this I don't get (and also with man pages) |
| 04:17.23 | jgoo | Usage: DATABASE GET <family> <key> |
| 04:17.29 | jgoo | when really, what they means is |
| 04:18.22 | jgoo | Usage: database_get('family', 'key'); |
| 04:19.23 | russellb | what are you talking about? |
| 04:19.33 | jgoo | hrm maybe not, wtf. I was looking at perl docs and php code ;) |
| 04:19.43 | jgoo | you can tell I am a f(/("(") java programmer |
| 04:20.19 | jgoo | damnit. I just want to print the trunks numer, the channel number, anything that might be the number that was dialled by the end user. So I am using AGI |
| 04:20.42 | jgoo | debug() seems good, but i need a list of the variables that are known |
| 04:24.13 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 04:25.29 | *** join/#asterisk droops (n=root@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 04:25.33 | *** part/#asterisk albertito (n=net@host76.201-252-17.telecom.net.ar) |
| 04:28.53 | jgoo | ok, on my outgoing - I get Executing Dial("SIP/200-0ee0", "ZAP/g0/9ANDHENUMBER|120|r") |
| 04:29.04 | jgoo | but then, Channel 0/1, span 1 got a hangup request |
| 04:29.44 | jgoo | incoming calls are working, so I am guessing the outgoing is configured wrong? |
| 04:29.45 | wunderkin | it doesnt want to talk to heman |
| 04:29.57 | jgoo | :S |
| 04:30.34 | wunderkin | well for one use Zap.. umm hmm i dunno may need to see the debug for the rest |
| 04:30.50 | wunderkin | i dont know that much about t1s |
| 04:30.51 | jgoo | using zap, shall I pastebin this? |
| 04:31.18 | wunderkin | i meant it is customary at least to use Zap and not ZAP.. i dont know if it matters, probably not.. is it pri? |
| 04:31.33 | Qwell | it shouldn't matter |
| 04:33.14 | jgoo | bri |
| 04:33.19 | jgoo | and ZAP was defaulted |
| 04:35.07 | wunderkin | is there a bri debug? |
| 04:38.42 | clyrrad | I have a phone that uses call forwarding all the time - the phone internally never rings - this phone is also part of a Queue - How can i make it so that if you dial into a Q - and a Q member has their phone set to call forward - that it rings the call forward number instead of the phones extension? |
| 04:41.46 | clyrrad | can anyone help? |
| 04:42.33 | *** part/#asterisk Samoied (n=Samoied@201.21.216.149) |
| 04:44.08 | *** join/#asterisk WeirdM (n=weirdm@udp079073uds.hawaiiantel.net) |
| 04:44.36 | WeirdM | Is there a way to refresh the config from extensions.conf without restarting asterisk? |
| 04:44.45 | Nugget | extensions reload |
| 04:44.50 | WeirdM | cool thanks |
| 04:46.52 | Lyfe | so, is it bad when you see a line that says "Jul 26 23:48:54 NOTICE[20143]: chan_iax2.c:3144 iax2_read: I should never be called!" |
| 04:50.13 | bkw_ | yes |
| 04:50.29 | bkw_ | because iax2 doesn't have a read function |
| 04:50.33 | bkw_ | something is calling it when it shouldn't be |
| 04:50.41 | clyrrad | anyone know about my call forwarding question? |
| 04:51.07 | Lyfe | hmm.. well, i was playing with music on hold, and called WaitMusicOnHold, whild using an IAX softphone.. woudl that wdo it? |
| 04:52.00 | Lyfe | eww, i can't type. "would that do it?" |
| 05:01.57 | *** part/#asterisk nerolab1 (n=Andrew@c-24-17-37-138.hsd1.wa.comcast.net) |
| 05:03.01 | russellb | Lyfe: did you happen to use a call file or manager originate action by chance? |
| 05:05.10 | Lyfe | as i don't know what a call file is, and haven't done any manager-related stuff, i'm going to say no. |
| 05:05.20 | russellb | ok :) |
| 05:06.12 | Lyfe | literally, the dialplan for the extension said "SetMusicOnHold(default)", "WaitMusicOnHold(30)", "Hangup" |
| 05:07.05 | russellb | i don't think there is such a thing as WaitMusicOnHold |
| 05:07.23 | Lyfe | that'd be a good reason. |
| 05:07.44 | russellb | WaitExten(30|m(default)) |
| 05:07.46 | Damin | russellb: Yo.. |
| 05:07.49 | Lyfe | but, it doesn't spit out any errors about the function. |
| 05:08.10 | russellb | Damin: hey, didn't know if you'd still be up :) |
| 05:08.13 | Damin | russellb: Re: http://bugs.digium.com/view.php?id=7592 <- Nope.. no real time queues.. |
| 05:08.23 | russellb | ok, thank you |
| 05:08.26 | Damin | russellb: I was just on the phone w/ Kielhofner for like 2 hours.. :) |
| 05:08.46 | russellb | issuing any reloads or anything? |
| 05:09.12 | russellb | that you know of |
| 05:09.25 | Damin | russellb: Nope.. just sitting there with 4 queues defined, and 3-4 calls in the support queue. It just faults.. |
| 05:09.36 | russellb | alright, thanks. i'll keep looking |
| 05:10.15 | Damin | russellb: Let me write a little script to abuse it and I'll put some calls into our support queue.. |
| 05:10.32 | Damin | russellb: Maybe I can get you a couple more backtraces to look at.. |
| 05:10.53 | russellb | alright. i was thinking of writing a little script myself to try to reproduce it |
| 05:12.16 | jgoo | anyone using AGI php?? |
| 05:12.51 | awk | 0.o |
| 05:13.08 | *** join/#asterisk [-Thomas-] (n=thomask@Barrie-ppp108495.sympatico.ca) |
| 05:13.23 | awk | i get to see an ass-linux box tomorrow |
| 05:13.28 | awk | or some such |
| 05:13.41 | awk | i hope its as cool as everyone says |
| 05:14.18 | [-Thomas-] | can someone suggest a client I can use in windows XP that I can route calls to from my asterisk box? (If there is such a beast) I don't have any IP phones or ata's yet.. |
| 05:14.59 | *** join/#asterisk speekac (n=alwin@60.51.217.58) |
| 05:15.02 | speekac | hi |
| 05:15.16 | speekac | can someone help me with "queues" ? |
| 05:16.12 | Damin | russellb: If you want to help, dial 216-619-2000, exten 454 |
| 05:16.47 | Damin | russellb: while true do; asterisk -rx "show queueus"; done |
| 05:16.55 | russellb | ha |
| 05:17.01 | russellb | i'm probably going to bed soon ... |
| 05:17.07 | awk | that would kill my nptl box |
| 05:17.07 | Damin | russellb: Don't laugh! :) I'm doing that now.. :) |
| 05:17.28 | awk | while true; do yes;done |
| 05:18.10 | speekac | does anyone here know what is the "W" represents in W:0, C:6, A:2, SL:16.7% within 0s when we do "show queues" ? |
| 05:18.21 | russellb | Damin: no, i think it's a great idea |
| 05:18.56 | awk | while true;do uptime;done |
| 05:18.56 | *** join/#asterisk Frogdude (n=Frogdude@c-24-16-72-159.hsd1.mn.comcast.net) |
| 05:19.53 | jgoo | can I ask what DID stands for? |
| 05:19.57 | russellb | ~did |
| 05:19.59 | jbot | somebody said did was Direct Inward Dialing |
| 05:20.05 | Qwell | You just did (pardon the pun) |
| 05:20.09 | jgoo | aaah, I thought it was like DDI |
| 05:20.13 | awk | ~seen somebody |
| 05:20.23 | jbot | somebody <n=somebody@200.189.112.30> was last seen on IRC in channel #debian-br, 293d 10h 15m 41s ago, saying: 'como faço para adicionar uma impressora da rede do windows no debian'. |
| 05:20.23 | Qwell | ~ddi |
| 05:20.25 | jbot | i heard ddi is Direct Dialling Inward, URL: http://www.wilco-telephony.co.uk/did.html |
| 05:20.25 | jgoo | Qwell, nobody can pardon that pun |
| 05:20.32 | Qwell | jgoo: Then I've done my job |
| 05:20.48 | jgoo | so DID == DDI |
| 05:20.53 | Qwell | pretty much |
| 05:21.04 | jgoo | I love it when they make it simple... |
| 05:21.19 | jgoo | ok so I found the incoming number w00t. |
| 05:21.20 | speekac | somebody help me with AMI's "show queues" please? |
| 05:21.31 | jgoo | now I need to pass that to a query.. and I have no idea |
| 05:22.44 | jgoo | There is a script that interfaces sugarcrm with FOP. |
| 05:23.12 | jgoo | now, I am guessing it takes a trigger from the FOP (which I have no idea how that works... reloading from a serverside file? network connection? |
| 05:23.58 | jgoo | I am looking for, when a call comes in, to redirect it to the relevant SIP phone (soft) and have a page that immediately pulls up the relevant details |
| 05:24.44 | jgoo | anyone done anything like this? How does it hook in? I need a flash bit on the page or an applet that monitors the extension? |
| 05:25.23 | jgoo | does linphone applet so a good job? anyone use it? (ok lots of questions, butI found nothing on linphone applet) |
| 05:25.36 | wunderkin | applet? . o O ( [21:19] <jgoo> you can tell I am a f(/("(") java programmer ) |
| 05:25.38 | wunderkin | heh |
| 05:26.50 | jgoo | heh, you in australia? |
| 05:26.58 | jgoo | no, west coast US? |
| 05:27.06 | wunderkin | umm yes that was closer |
| 05:27.13 | jgoo | south america |
| 05:27.23 | jgoo | wait.. |
| 05:27.31 | wunderkin | cold! |
| 05:28.16 | jgoo | Canada? anyway, back on point. |
| 05:28.26 | jgoo | hooking an extensions monitor into the webpage |
| 05:28.32 | jgoo | so I can update it live |
| 05:29.00 | jgoo | I am sure there is a big demand for this, but I only found one thing to tie FOP to a Scrm page |
| 05:31.36 | *** join/#asterisk Revilo_ (n=local@port-212-202-168-18.dynamic.qsc.de) |
| 05:32.08 | Revilo_ | hello, wich ports must i open in my firewall to use asteris(client + server)? I use SIP |
| 05:32.24 | Revilo_ | 5060 with udp ok |
| 05:32.42 | Revilo_ | for rtp 5004 on both sites too |
| 05:33.01 | Revilo_ | And what still? |
| 05:33.08 | speekac | 5038 |
| 05:33.15 | Revilo_ | for? |
| 05:33.23 | speekac | the AMI |
| 05:34.06 | Revilo_ | what is ami |
| 05:34.20 | speekac | asterisk manager interface |
| 05:35.14 | Revilo_ | yes but this is not necessary? |
| 05:35.24 | Revilo_ | I need only the ports what MUST open |
| 05:35.37 | Revilo_ | must be open |
| 05:36.05 | Lyfe | anyone have any suggestions for why music on hold would start, and then stop right away? (no errors) |
| 05:36.10 | *** join/#asterisk ditesh|cassini (n=ditesh@60.51.217.61) |
| 05:36.37 | speekac | Revilo_ sorry i am not sure about the required open ports |
| 05:37.15 | speekac | does anyone here know what is the "W" represents in (W:0, C:6, A:2, SL:16.7% within 0s) when we do "show queues" thru the AMI? |
| 05:40.06 | znoG | speekac: waiting? (just a guess) |
| 05:40.21 | Revilo_ | znoG: Can you help me with the ports? |
| 05:40.27 | speekac | znoG: nope, it is not |
| 05:40.42 | speekac | znoG: i am still trying to figure it out |
| 05:41.19 | znoG | Revilo_: you need 5060 TCP/UDP and RTP ports (10000 -> 20000 by default) |
| 05:41.36 | Revilo_ | nothing more? |
| 05:41.42 | znoG | not AFAIK |
| 05:41.55 | Revilo_ | do you have a site? |
| 05:41.59 | Revilo_ | where can i read this? |
| 05:42.16 | znoG | i looked it up back when i had to open it up |
| 05:42.17 | znoG | look up voip-info.org, google |
| 05:42.20 | Revilo_ | ok thansk |
| 05:43.16 | Revilo_ | Ehm |
| 05:43.25 | *** join/#asterisk trivex (n=trivex@CPE00112f8785ab-CM000f9f50281e.cpe.net.cable.rogers.com) |
| 05:43.52 | Revilo_ | And for SDP and RTCP? |
| 05:43.56 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:44.08 | speekac | i couldn't find more relevant info on 'queues' at voip-info.org |
| 05:44.34 | Revilo_ | @ znoG? |
| 05:45.21 | znoG | Revilo_: i don't know that info. |
| 05:45.28 | Revilo_ | Hm |
| 05:47.20 | Revilo_ | Hm |
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| 05:49.23 | Revilo_ | Omg |
| 05:49.23 | Revilo_ | Flood! |
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| 06:15.41 | jgoo | Guys, in my dialplan, how to I wait for input? |
| 06:17.52 | Revilo_ | jgoo: Wich Ports must i Open for SDP and RTCP? |
| 06:17.56 | carrar | DISA |
| 06:18.08 | Revilo_ | or carrar ;) |
| 06:18.17 | jgoo | trixbox needs emacs ;_; |
| 06:18.22 | jgoo | can I yum it in? |
| 06:19.13 | hads | ~trixbox |
| 06:19.20 | jbot | somebody said trixbox was NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 06:19.54 | Revilo_ | Hm |
| 06:19.58 | Revilo_ | No Answer? |
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| 06:26.03 | carrar | Revilo, open up UDP |
| 06:26.53 | Revilo_ | What @ UDP? |
| 06:27.02 | Revilo_ | And what @ TCP? |
| 06:27.27 | carrar | I suggest reading the how it works before you think about opening it up so you know what you are getting yourself into |
| 06:27.48 | Revilo_ | I cant do this in the Moment |
| 06:27.54 | Revilo_ | I must write a docu |
| 06:28.03 | carrar | best research what you are writting about |
| 06:30.09 | Revilo_ | Hm |
| 06:30.13 | Revilo_ | No Site for that? |
| 06:32.04 | *** join/#asterisk niteowldave (i=niteowlO@203.82.162.38) |
| 06:33.22 | carrar | There's only like 4 billion web pages on those protocols |
| 06:33.44 | *** join/#asterisk niteowldave (i=niteowlO@203.82.162.38) |
| 06:34.25 | niteowldave | file: Can t.38 work from behind a nat firewall |
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| 06:42.28 | x86 | has anyone tried to use a modem over a G.711u channel? |
| 06:43.07 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 06:43.08 | x86 | i'm doing end-to-end G.711u across the internet, and I can fax with a success rate of about 99%, but can not get a modem to connect at all (even if trying to negotiate 14.4kbps speeds) |
| 06:43.44 | x86 | anything that I can do to make it work? |
| 06:43.56 | carrar | Implement QoS Ennd to End |
| 06:44.02 | x86 | it is |
| 06:44.10 | carrar | "across the internet"? |
| 06:44.12 | JonR800 | no |
| 06:44.14 | denon | you have QoS across the Internet? |
| 06:44.18 | x86 | PPP has LCP extensions anyway, which provide reliability |
| 06:44.20 | denon | huh, what backbone would that be? |
| 06:44.28 | x86 | denon: eh, QoS over VPN :P |
| 06:44.33 | denon | doesnt matter |
| 06:44.35 | denon | Internet == the suck |
| 06:44.36 | x86 | so not entirely end-to-end ;) |
| 06:44.39 | x86 | indeed |
| 06:44.42 | x86 | but faxing works fine |
| 06:44.44 | JonR800 | so not end to end at all. |
| 06:44.49 | JonR800 | faxing is a different beast |
| 06:44.54 | denon | faxing will work fine at 300bps |
| 06:45.01 | denon | faxing will almost never reach 14.4 |
| 06:45.03 | x86 | i do faxing at 14.4kbps |
| 06:45.04 | denon | especially on your voip |
| 06:45.10 | denon | unless you're doing t.38 |
| 06:45.14 | x86 | *nod* |
| 06:45.23 | x86 | t.38 pass-through with asterisk 1.2.10 |
| 06:45.26 | x86 | works great :) |
| 06:45.35 | x86 | 14.4kbps almost every time |
| 06:45.38 | denon | data/modem connections have no t.38 equivilant |
| 06:45.40 | denon | well, they do .. |
| 06:45.43 | denon | but that's another show |
| 06:45.48 | x86 | ? |
| 06:45.53 | denon | t.38 is enabling faxing to work well .. |
| 06:45.56 | denon | but it doesnt work for data |
| 06:46.00 | x86 | right |
| 06:46.12 | x86 | so it's not possible to use a modem over g.711u ? |
| 06:46.27 | denon | it is, in a controlled environement |
| 06:46.39 | x86 | hmm |
| 06:46.42 | denon | otherwise you're better off shipping the serial off via a tcp session, and doing serial over ip |
| 06:46.50 | denon | instead of a modem negotiation, and relying on acoustics |
| 06:46.57 | x86 | yeah, but the other end has to support it :( |
| 06:47.13 | x86 | both ends need to support it |
| 06:47.17 | Assid | umm... suppose you use a pc (modem+fax software) + pap2 -- ULAW (asterisk) --- [provider] -- FAX |
| 06:47.20 | Assid | will that work? |
| 06:47.22 | carrar | There is vast amounts of information over the internet tubes |
| 06:47.27 | denon | it will probably never work reliably, unless you can get everything on one backbone .. and even then it's doubtful |
| 06:47.49 | Revilo_ | Anyone here wich know, what ports must be open @ SIP (Server and Client)? SDP Port?/RTCP Port? RTP Port? SIP Port? |
| 06:47.53 | x86 | even if i force a lower bit rate? |
| 06:48.01 | denon | most likely |
| 06:48.10 | x86 | like, say, 9600bps ? |
| 06:48.13 | denon | you'd probably have the best luck around 4800 |
| 06:48.18 | denon | which is pretty impractical |
| 06:48.25 | Assid | would it be possible to do something like that? where you connect a fax using a pap2 ? |
| 06:48.25 | x86 | not really |
| 06:48.33 | denon | but .. even then, it really really really .. probably wont work |
| 06:48.37 | denon | I mean, it will work occasionally |
| 06:48.42 | x86 | user is only using a modem for dialing into a cisco console server ;) |
| 06:48.42 | denon | but I'd say 75% of the time it wont |
| 06:48.58 | x86 | crappy :( |
| 06:49.17 | denon | voip is designed to carry voices, t.38 to carry faxes.. |
| 06:49.26 | denon | none were designed to carry data converted to accoustics |
| 06:49.36 | x86 | someone needs to get on that ;) |
| 06:49.55 | carrar | Thats what Ethernet is for |
| 06:50.06 | x86 | i figured the LCP extensions on top of the PPP protocol would reliably handle re-transmits and what not |
| 06:50.29 | x86 | carrar: you can not dial into a console server remotely via ethernet bud ;) |
| 06:50.34 | Revilo_ | Anyone here wich know, what ports must be open @ SIP (Server and Client)? SDP Port?/RTCP Port? RTP Port? SIP Port? |
| 06:50.42 | carrar | You can if you buy the right products |
| 06:50.43 | Assid | denon: would it be possible to do the above fax using a pap2 using ulaw codec? |
| 06:50.44 | x86 | Revilo_: please stop repeating |
| 06:50.59 | Revilo_ | Hey good Morning x86, can you help me? |
| 06:51.14 | x86 | carrar: sure, a serial --> tcp box at each end, which we've already gone over :) |
| 06:51.19 | x86 | Revilo_: google.com can! |
| 06:51.32 | Revilo_ | i cant find it @google |
| 06:51.36 | Revilo_ | SIP and RTP Ok |
| 06:51.44 | Revilo_ | But what is with SDP and RTCP? |
| 06:51.46 | x86 | then you dont know how to use google ;) |
| 06:51.50 | Revilo_ | :-( |
| 06:52.12 | Assid | Revilo_: http://www.chebucto.ns.ca/~rakerman/port-table.html |
| 06:52.17 | Assid | or google for more |
| 06:52.30 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 06:53.11 | Assid | isnt thre some kind of sip software on its own which allows faxing? |
| 06:54.24 | Revilo_ | Assid: x+1 ok, so rtp is on 10k-20k so i must open 10000-10001? |
| 06:54.42 | Revilo_ | i the table is a number ~5k? |
| 06:54.46 | Revilo_ | What is now right? |
| 06:55.12 | Assid | your contradicting yourself |
| 06:55.21 | Assid | you just said 10K-20K |
| 06:55.36 | Assid | and you wanna open 10K-10K1 |
| 06:55.51 | Revilo_ | is this right or not? |
| 06:55.55 | Assid | read dude |
| 06:56.05 | Assid | else google.. this is well documented |
| 06:56.21 | Revilo_ | I must comple my docu in the next 2h |
| 06:56.25 | Revilo_ | Please help me :-( |
| 06:56.45 | Assid | do you know how to google? |
| 06:57.11 | Clausian | does anyone here have asterisk setup with voipdiscount? |
| 06:57.13 | carrar | So you want us to finish your paper because you did not do any research? |
| 06:57.16 | Assid | its very well documented.. so take an effrot and learn to use it |
| 06:57.39 | Assid | Clausian: well.. one of their products.. so yes |
| 06:57.51 | Assid | their website has a document on how to connect sip accounts |
| 06:58.20 | Clausian | yeah, the main part im confused about is connecting my softphone to asterisk |
| 06:58.34 | Assid | works the same way as any sip device.. |
| 06:58.36 | Clausian | i can get it to connect and hear monkeys, but i cant figure out how to make calls through voipdiscount with it |
| 06:58.43 | Revilo_ | assid: pls can you answer me without google? is my presumption with 10k1 right? |
| 06:58.54 | Assid | no |
| 06:59.07 | Clausian | i have a trunk [voipdiscount] setup, but what do i tell my softphone to connect to? |
| 06:59.45 | *** join/#asterisk daysmen3 (n=primus@host86-140-40-138.range86-140.btcentralplus.com) |
| 06:59.54 | Assid | Clausian: have you made a dialplan before? |
| 06:59.58 | Clausian | no |
| 07:00.03 | Assid | hear monkeyus? |
| 07:00.23 | Clausian | i can get the monkeys to work |
| 07:00.39 | Revilo_ | assid: what must i do now? |
| 07:01.00 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 07:01.16 | Assid | Revilo_: you learn to use a search engine called google |
| 07:01.29 | Assid | Clausian: you got monkeys working for you??? |
| 07:01.34 | Clausian | yes |
| 07:01.56 | carrar | I have some Monkey picked oolong tea |
| 07:02.15 | Clausian | you know what i mean? tt-monkeys |
| 07:02.31 | Assid | train it to learn asterisk dialplans.. and you can retire |
| 07:02.52 | Clausian | i understand now why ass is in your username |
| 07:03.01 | Assid | yeah.. i id'd your ass |
| 07:03.19 | Assid | give a stupid comment.. get a stupid answer |
| 07:03.29 | Clausian | i asked a question |
| 07:03.37 | Assid | you said you could hear monkeys |
| 07:03.47 | Clausian | the tt-monkeys sound playback |
| 07:04.00 | Revilo_ | Assid: I cant find the SDP Port |
| 07:04.20 | Assid | Revilo_: just search for sip firewall portforwarding |
| 07:04.31 | Revilo_ | thanks |
| 07:04.40 | carrar | How do I type that in? |
| 07:04.40 | Assid | cant believe im teaching him to use google |
| 07:04.45 | Clausian | my softphone will connect and i setup a simple extension to playback monkeys and hangup |
| 07:05.16 | Clausian | now, i have asterisk connecting succesfully to voipdiscount, but i dont know how to make the softphone place calls through that connection |
| 07:06.02 | Assid | how do you know asterisk is connecting successfully to voipdiscount? |
| 07:06.10 | speekac | does anyone here know what is the "W" represents in (W:0, C:6, A:2, SL:16.7% within 0s) when we do "show queues" thru the AMI? |
| 07:06.30 | Assid | Clausian: http://www.voip-info.org/wiki-Asterisk+cmd+Dial <-- that will show you how to make a call |
| 07:06.52 | Assid | use that to call through voipdiscount context |
| 07:07.28 | carrar | Revilo, I suggest you start over with reading TCP/IP Illistrated Vol 1, then reading about specific protocols |
| 07:08.00 | carrar | going in half ass is going to get you half ass results |
| 07:08.39 | Clausian | sip show registry |
| 07:08.47 | carrar | and nothing but 100% accuracy is acceptable |
| 07:08.48 | Clausian | it tells me its connected |
| 07:10.28 | Clausian | nevermind im an idiot |
| 07:10.36 | Clausian | i had allow=ulaw&glaw |
| 07:10.40 | Clausian | then disallow=all |
| 07:11.20 | Assid | one day .. there will be a glaw |
| 07:11.57 | carrar | word G |
| 07:12.08 | carrar | gangstah law |
| 07:12.14 | Assid | heh |
| 07:12.42 | x86 | one day there will be a dblaw... drive-by law |
| 07:12.55 | Assid | dammit.. there should be a sip software for faxing straightr from the pc |
| 07:13.10 | x86 | Assid: write some ;) |
| 07:13.27 | x86 | Assid: there is software to make a virtual fax modem over sip |
| 07:13.38 | Assid | if i could.. i would |
| 07:13.39 | Revilo_ | Assid, If i user nat=yes must i have then a STUN etc Server behind a nat? |
| 07:13.47 | x86 | Assid: then you can fax using standard fax software via the virtual modem |
| 07:14.44 | Assid | know the url by any chance x86? |
| 07:15.17 | Revilo_ | sorry |
| 07:15.18 | Assid | Revilo_: having a stun behind the nat defeats the purpose of the stun |
| 07:15.55 | Assid | nat=yes means the packets are re-routed by your internet ip instead of lan |
| 07:16.00 | Revilo_ | so i dont need a STUN uPNP or so server? |
| 07:16.04 | Revilo_ | ah |
| 07:16.20 | Revilo_ | so i dont need it |
| 07:16.22 | Revilo_ | right? |
| 07:16.37 | Assid | i dont know what you plannng to do.. how can i tell you if you need it or no |
| 07:16.43 | carrar | Might I suggest paying someone to write your paper |
| 07:17.18 | Clausian | Jul 27 17:16:42 NOTICE[16783]: chan_sip.c:3685 process_sdp: No compatible codecs! |
| 07:17.35 | x86 | Assid: why not setup a CUPS print-to-fax gateway? |
| 07:17.51 | niteowldave | x86: which ATAs are you using for t38 |
| 07:17.51 | Clausian | how do i tell what codec i need to get? |
| 07:17.53 | *** join/#asterisk docelm0 (n=Snake@55-65.126-70.tampabay.res.rr.com) |
| 07:18.12 | *** join/#asterisk JonR800 (n=jon@p1mp.org) |
| 07:18.26 | Assid | x86: out of my reach.. not sure what to do in there |
| 07:18.37 | *** join/#asterisk santiago (i=santiago@debian/developer/santiago) |
| 07:18.58 | x86 | niteowldave: i've been using grandstream HT386's just fine, i'm not 100% positive they are taking advantage of t.38 or not, but the t.38 pass-through module is loaded on asterisk |
| 07:19.26 | Clausian | where is asterisks log file located? |
| 07:19.38 | x86 | <PROTECTED> |
| 07:19.38 | carrar | <PROTECTED> |
| 07:19.40 | carrar | yeah |
| 07:19.51 | Assid | i wouldnt mind the the virtual modem solution really.. i just cant get how would it interface with sip |
| 07:20.24 | niteowldave | x86: thanks....do your atas live behind a nat |
| 07:20.28 | Assid | x86: you wouldnt know the url for that virtual modem would you? |
| 07:20.42 | Clausian | Jul 27 17:18:19 NOTICE[16783] chan_sip.c: No compatible codecs! |
| 07:20.48 | Clausian | thats not very helpful :| |
| 07:21.16 | x86 | Assid: hold on while I google it, since no one else knows how to search ;-) |
| 07:21.27 | e-ddie | Clausian: at least it tells you what the problem is |
| 07:21.51 | Clausian | yeah but not what codec i need |
| 07:22.08 | e-ddie | that's why you have manuals |
| 07:22.11 | Assid | x86: im trying.. cant seme to get the sip version.. i saw a windows one shareware.. but its not sip based |
| 07:22.27 | Clausian | what do you mean manuals? |
| 07:23.41 | Assid | i didnt know you could use & between codecs |
| 07:23.43 | Assid | i thought it was ; |
| 07:23.47 | Clausian | you cant :X |
| 07:23.57 | E-bola | Morning |
| 07:24.10 | hads|home | evening |
| 07:25.09 | *** join/#asterisk s0lid (n=jlq@58.69.210.140) |
| 07:26.24 | x86 | niteowldave: yep |
| 07:26.32 | x86 | niteowldave: and no, i dont use STUN at all |
| 07:27.05 | x86 | Assid: try http://www.vocal.com/ ? |
| 07:27.50 | Assid | http://sourceforge.net/projects/siptapi/ <-- here i just found this |
| 07:28.36 | Assid | would that work? |
| 07:28.54 | Assid | damn.. vocal not opening for me :( |
| 07:29.50 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 07:31.19 | Assid | x86: is something wrong with that ste? it just refuses to open for me |
| 07:33.58 | Assid | yep .. dead |
| 07:34.06 | Clausian | works for me |
| 07:34.34 | Assid | i tried even from my remote box. didnt work :( |
| 07:34.45 | Clausian | Jul 27 17:33:37 WARNING[16922]: chan_sip.c:2552 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 07:34.47 | Clausian | whats that mean? |
| 07:35.09 | Assid | you tryingg729 |
| 07:35.22 | Clausian | its set to allowed |
| 07:35.35 | Assid | your trying 729 and you dont have the codecs for it to transcode |
| 07:36.17 | Assid | 1 format is ulaw.. 1 format is 729 |
| 07:36.37 | Assid | voipdiscount i dont think support 729.. last time i tried.. they had some issues |
| 07:36.55 | Assid | although they mention on the site they do |
| 07:48.59 | Clausian | they dont |
| 07:49.10 | Clausian | well it wont work anyway |
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| 07:52.55 | nicox | do anybody know, is it possible to unload the wct4xxp kernel module and reload it without restarting Asterisk? |
| 07:53.26 | *** join/#asterisk vgster (n=vgster@217.78.147.238) |
| 07:55.30 | benjk | I doubt it |
| 07:55.59 | benjk | at the very least you'd have to unload the chan_zap module |
| 07:56.09 | benjk | and that might crash Asterisk |
| 07:59.24 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 07:59.46 | nicox | no, asterisk did not crash with an unload chan_zap.so |
| 08:00.14 | nicox | but if i try to rmmod wct4xxp i get an, the module is busy |
| 08:04.59 | *** join/#asterisk Pazzo (n=thomas@dialin-225136.rol.raiffeisen.net) |
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| 08:22.26 | nicox | anyone there who tried to restart wct4xxp without restart asterisk? |
| 08:23.02 | *** join/#asterisk clive- (n=pirch@dsl-145-52-134.telkomadsl.co.za) |
| 08:24.00 | clive- | has anyone had any expereinces getting reallly bad zaptel timming on a x100p on a xeon board ? |
| 08:26.23 | *** join/#asterisk shaZwaz (n=chatzill@203.81.222.122) |
| 08:27.37 | WeirdM | Is there a way to drop out of the voicemail module in to something else when a caller presses a number? |
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| 08:38.49 | nicox | <PROTECTED> |
| 08:44.53 | shaZwaz | nicox: u mean unload wct4xxp driver and load it again ? |
| 08:45.23 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 08:45.29 | shaZwaz | just do a rmmod wct4xxp and then again modprobe wct4xxp |
| 08:46.10 | hads|home | modprobe -r |
| 08:53.08 | s0lid | i tried to put modprobe wct4xxp on /etc/rc.local but it deosn't work |
| 08:53.16 | s0lid | im using centos 4.3 |
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| 09:06.07 | tengulre11 | Hi,all |
| 09:08.53 | *** join/#asterisk speekac (n=alwin@60.51.217.58) |
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| 09:14.53 | clive- | can anyone help me with a zapata.conf for a tdm400 card with no modules installed...can this work ??? |
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| 09:20.25 | clive- | any ideas? |
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| 09:22.45 | shaZwaz | no modules installed ? |
| 09:25.29 | clive- | shazwaz yes, it that possible? |
| 09:25.46 | clive- | I mean is it possible to get it working for zaptel timming purposes |
| 09:26.53 | shaZwaz | did u try ? |
| 09:27.07 | shaZwaz | just modprobe it |
| 09:28.41 | clive- | I am trying, it doesnt like an empty zaptel.conf, so I tried a zaptel.conf, but then it complains that it cant find modules |
| 09:31.35 | X-Gen | clive-, serves u right for using xeon boards, u shoulda use AMD ! feel the pain ! |
| 09:31.41 | X-Gen | hehehehe |
| 09:32.36 | clive- | X-gen:) |
| 09:37.20 | *** join/#asterisk Stephnie (i=Stephnie@u15157627.onlinehome-server.com) |
| 09:37.29 | Stephnie | hi |
| 09:37.40 | Stephnie | any one use vonage with asterisk ? |
| 09:40.13 | *** join/#asterisk _fa_ (i=faceoff@een.os3.kn.pl) |
| 09:42.42 | *** join/#asterisk my007ms (n=noor@217.139.224.195) |
| 09:43.44 | *** join/#asterisk adorah (n=Administ@87.68.145.244.cable.012.net.il) |
| 09:46.59 | *** join/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 09:48.39 | TypMic | Is there a way to get asterisk to bind to a specific address. Trying to get asterisk to only send packets across an OpenVPN interface. It works for incoming traffic from a networked host but doesn't for incoming phone-line connections |
| 09:50.25 | *** join/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.mn.comcast.net) |
| 09:50.57 | rikstah | hey, i'm in the states for a few months, and i was wondering if there is a free service that provides an incoming phone number did -> sip (dont need outgoing) ? |
| 09:51.57 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 09:56.56 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:58.13 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:01.02 | DarKnesS_WolF | dose asterisk has h263p " h263+ " codec ? |
| 10:04.09 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:06.16 | *** join/#asterisk enemy^x (n=enemy_x@c213-158-248-202.sdsl.no) |
| 10:06.42 | enemy^x | anyone know how I can return the call to the person transfering the call if busy? |
| 10:08.08 | *** join/#asterisk Ast001 (n=uros@212-200-207-245.adsl.sezampro.yu) |
| 10:08.45 | Ast001 | Hello Asterisk users |
| 10:11.28 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 10:17.20 | Ast001 | I have Asterisk PBX with ADSL 768/256 on point A and on point B CABLE 256/128.When I use XLite softphone man on point A can hear very bad voice quality. |
| 10:18.11 | Ast001 | Does anyone have problem like this ? |
| 10:21.30 | Ast001 | Is 128 kb/s enough for u-law and a-law codecs ? |
| 10:21.47 | *** join/#asterisk vlt (n=daniel@dslb-088-073-229-005.pools.arcor-ip.net) |
| 10:23.01 | adorah | those codecs consumes about 85/85Kbits each |
| 10:24.03 | vgster | hi, has anyone played with call detection for screen popping? |
| 10:24.58 | Ast001 | so 128 kb/s upload should be enough. |
| 10:25.22 | *** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net) |
| 10:26.26 | adorah | <Ast001>In theory. provided nothing else is consuming bandwidth..use GSM unstead from extension to pbx.. |
| 10:26.55 | adorah | =instead.. |
| 10:27.07 | dlynes_laptop | unstead sounds better |
| 10:27.12 | adorah | LOL |
| 10:27.40 | vlt | Hello, my asterisk is connected to my voip provider via one SIP channel on UDP port 5060 over an SDSL line and I want to priorize VoIP traffic. I have logged several phone calls and found out that only the initializing data is sent over udp:5060 (yes, that's why it's called S_I_P), and the phone traffic itself goes from 16,364 to 28,124, in another call from 14,742 to 4,160 and so on. How can I find out or define what port range is |
| 10:27.41 | vlt | <PROTECTED> |
| 10:28.09 | adorah | I have a problem of hoarse voice on IAX trunk when I call from a remot IAX client -gsm codec any idea where find the problem? |
| 10:28.22 | adorah | =remote.. |
| 10:28.24 | dlynes_laptop | vlt, the ports you're referring to is known as 'RTP' |
| 10:28.26 | dlynes_laptop | ~rtp |
| 10:28.28 | jbot | rtp is probably The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 10:28.51 | Ast001 | It works only with Eyebeam (g729 codec) |
| 10:29.17 | Ast001 | and I dont like to use proprietory codecs |
| 10:29.22 | dlynes_laptop | vlt, you'll need to examine your sip device configuration to determine what rtp ports they're using |
| 10:29.30 | adorah | g729 is good too.. |
| 10:29.45 | dlynes_laptop | adorah, hoarse voice? |
| 10:30.11 | adorah | as if the other side has a thore throat.. |
| 10:30.30 | dlynes_laptop | are you sure they don't? |
| 10:30.31 | Ast001 | I read that one call which use gsm729 is equal to 5 calls which use u-law or a-law in terms of CPU load |
| 10:31.01 | vlt | lynes_laptop: Does iptables know "rtp"? Can I add a rule to the mangle table that sais "--sport rtp"? |
| 10:31.16 | dlynes_laptop | Ast001, at your asterisk cli, type 'show translation' to see a more accurate calculation for your specific cpu |
| 10:31.33 | Ast001 | ok dlynes |
| 10:31.34 | adorah | <dlynes_laptop>If they do why r they sound clear when calling from local extension?:P |
| 10:31.59 | dlynes_laptop | adorah, try pinging their remote ip...what is the ping time? |
| 10:32.36 | dlynes_laptop | adorah, under normal circumstances, gsm will probably sound better than analog |
| 10:32.39 | adorah | <dlynes_laptop>the point is that when I call from the same client to the remote IAX ITSP it sounds all clear |
| 10:32.58 | dlynes_laptop | client -> remote iax == clear |
| 10:33.06 | dlynes_laptop | remote iax -> client == hoarse? |
| 10:33.06 | adorah | the provider uses iax/gsm.. |
| 10:33.42 | dlynes_laptop | yes, or no? |
| 10:34.02 | adorah | remote client>remote iax=clear remote client>asterisk>remote iax hoarse.. |
| 10:34.11 | vlt | lynes_laptop: My sip.conf has only one line matching "port": bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) |
| 10:34.25 | vlt | lynes_laptop: What file is for sip device configuration? |
| 10:34.44 | dlynes_laptop | vlt, i'm pretty sure it doesn't, but someone's probably got a netfilter module and/or script for it if you check google |
| 10:35.04 | dlynes_laptop | vlt, that's for sip traffic, and registry traffic |
| 10:35.08 | dlynes_laptop | vlt, that's not for rtp |
| 10:35.18 | dlynes_laptop | vlt, for rtp on the asterisk side, it's in rtp.conf |
| 10:35.45 | dlynes_laptop | vlt, for rtp on your sip device side, I wouldn't have a clue...it's different from one sip device to the next; some even go so far as to hardcode it |
| 10:35.53 | dlynes_laptop | adorah, ok |
| 10:36.08 | dlynes_laptop | adorah, let's do this: |
| 10:36.08 | vlt | lynes_laptop: Thank you. (Must have been blind ;-) |
| 10:36.21 | Assid | hrmm.. is there a way to default include a context? |
| 10:36.23 | dlynes_laptop | adorah, from remote client --> asterisk, do a ping |
| 10:36.44 | dlynes_laptop | adorah, also from asterisk --> remote iax do a ping |
| 10:36.51 | dlynes_laptop | adorah, let me know what the results are for both |
| 10:36.58 | vlt | lynes_laptop: The asterisk is on "my side", the clients too (in local LAN). I need shaping outbound only. |
| 10:37.09 | adorah | ok |
| 10:37.27 | dlynes_laptop | vlt, are you using outbound/inbound iax2 only? |
| 10:37.28 | adorah | from remote client to * 41ms |
| 10:37.42 | dlynes_laptop | vlt, or do you need your asterisk box to be able to send/receive sip from outside the lan as well? |
| 10:39.10 | vlt | lynes_laptop: No, all phone traffic from LAN goes to asterisk, and asterisk "forwards" it to my VoIP provider's asterisk using SIP. |
| 10:39.28 | dlynes_laptop | vlt, ok so no iax2 then |
| 10:39.29 | adorah | now I'll check from * to remote iax:18ms thos r pretty good results.. |
| 10:39.50 | dlynes_laptop | adorah, ok, try this now |
| 10:40.07 | dlynes_laptop | adorah, do a traceroute in both directions...see if you have any '*''s showing up |
| 10:40.26 | vlt | lynes_laptop: And vice versa: Phone calls from external PSTN are answered by asterisk and handed over to the clients in the LAN. |
| 10:41.19 | dlynes_laptop | vlt, yeah...so you'll need to handle udp 5060, and whatever udp whatever rtp ports you have defined in your rtp.conf file |
| 10:42.06 | vlt | lynes_laptop: Yes. With your help now I found out that thae range is 10,000 to 20,000. |
| 10:42.16 | dlynes_laptop | vlt, you shouldn't need to do any port forwarding as long as you're using nat=yes or nat=route, and canreinvite=no for the sip config for your upstream provider |
| 10:42.22 | adorah | <dlynes_laptop>how do I trace route in linux? |
| 10:42.36 | dlynes_laptop | adorah, use the '/sbin/traceroute' command |
| 10:42.45 | adorah | oh ok.. |
| 10:42.51 | dlynes_laptop | adorah, or maybe it's /usr/sbin/traceroute |
| 10:43.00 | dlynes_laptop | I have it in my path, so I don't pay attention |
| 10:43.38 | vlt | lynes_laptop: Mmh, didn't understand that. I don't want to do any FW. Just want to shape my outgoing DSL traffic ... |
| 10:43.47 | dlynes_laptop | and i'm not on linux right now, so I can't check |
| 10:44.04 | dlynes_laptop | vlt, ok, so just ignore what i said then...just assumed you were using a firewall :) |
| 10:44.42 | adorah | /usr/sbin is fine.. |
| 10:45.00 | dlynes_laptop | anyways |
| 10:45.04 | dlynes_laptop | i'm heading out shortly |
| 10:45.15 | dlynes_laptop | so if you have any other questions, ask them soon |
| 10:45.19 | *** part/#asterisk jgoo (n=foo@ppp134-168.adsl.forthnet.gr) |
| 10:45.22 | adorah | ok thx 4 now:) |
| 10:45.50 | dlynes_laptop | ok, on that note |
| 10:45.52 | dlynes_laptop | i'm out |
| 10:45.56 | dlynes_laptop | have a good night, peeps |
| 10:46.10 | adorah | It gives me an answer 'unknown host' only on linux machine..weured |
| 10:47.49 | Ast001 | Is it possible to connect ordinary phone to Asterisk and recieve and place calls (from and to phones connected conncected to post ) ? |
| 10:48.26 | E-bola | post? |
| 10:48.27 | Ast001 | and if it is do I need something more that E1 compatible card to do that ? |
| 10:48.45 | Ast001 | post - I think its PSTN on english |
| 10:49.09 | Ast001 | Publich Swich Telephone Network |
| 10:49.57 | Ast001 | Example my home phone to conect to Asterisk and to call someone else's home phone which is not connected to Asterisk |
| 10:50.36 | nettie | Hey guys, anyone have experience with polycoms phones and jitterbuffer please? |
| 10:50.50 | nettie | I'm wondering if default settings are fine on a pretty loaded link? |
| 10:51.24 | rikstah | http://pastebin.ca/102134 please can someone tell me why this very simple dial plan ${test} returns as nothing when i enter some digits |
| 10:51.47 | rikstah | im trying to transfer a call |
| 10:52.24 | DarKnesS_WolF | dose asterisk has h263p " h263+ " codec ? |
| 10:53.16 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 10:53.34 | Ast001 | ok people see you later.Bye |
| 10:53.37 | *** part/#asterisk Ast001 (n=uros@212-200-207-245.adsl.sezampro.yu) |
| 10:54.53 | Nobbie | hi |
| 10:56.09 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 10:56.19 | Nobbie | i'm haviong a problem with Pickup() application. it doesn't work on a transferred call and i'm assuming it's becuae the call was already Answer()'d. is there a workaround or solution ? |
| 11:06.02 | *** join/#asterisk stse (n=stse@muedsl-82-207-243-205.citykom.de) |
| 11:06.44 | stse | Hello! Are here German people who are using asterisk 1.2.9.1 to send SMS? |
| 11:11.53 | *** join/#asterisk rocket007 (n=sasaim@202.142.191.67) |
| 11:22.22 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 11:31.00 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:32.47 | daysmen3 | can gnome be updated by and the supplementary source as was the case of KDE (before they changed it that is) |
| 11:33.01 | daysmen3 | ...by adding the supplementary source |
| 11:38.40 | *** join/#asterisk dapper-daniel (n=daniel@pD9E005FE.dip0.t-ipconnect.de) |
| 11:39.14 | *** part/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.mn.comcast.net) |
| 11:40.01 | daysmen3 | sorry about that people - wrong peeps |
| 11:41.06 | dapper-daniel | hi! i have some problems with asterisk behind an nat gateway (FritzBox Fon) |
| 11:43.06 | dapper-daniel | i have absolutely no idea what went wrong but the server is not reachable via phone |
| 11:43.23 | rocket007 | dapper-daniel |
| 11:43.25 | dapper-daniel | sip debug tells me |
| 11:43.34 | rocket007 | dapper-daniel<--sip is buggy behind NAT |
| 11:45.07 | rocket007 | You trying to register SIP phones to an asterisks box, which behind NAT or the other way around? i.e phones are behind NAT |
| 11:45.21 | dapper-daniel | no no no |
| 11:45.33 | dapper-daniel | i try to bild a callback server |
| 11:46.03 | dapper-daniel | it should register 2 sip numers at my provider (1und1.de) |
| 11:46.36 | dapper-daniel | when i call one of these numers it should hnagup and callback |
| 11:47.10 | dapper-daniel | with the second numer i can do phone call after callback |
| 11:47.25 | dapper-daniel | do you understand? |
| 11:48.44 | rocket007 | dude I have never configured callbacks, but as FAR as my testing of SIP is concerned, i had to redo my firewalls in 7 locations just to run SIP behind FWs. So I opted for IAX2 |
| 11:49.44 | rocket007 | may be someone else more experience over here can jump in and help. |
| 11:50.13 | rocket007 | but since there is a provider in question I dont think you have much choice ?? |
| 11:50.28 | dapper-daniel | no i have to use SIP |
| 11:51.00 | rocket007 | I can see |
| 11:51.10 | dapper-daniel | i oppend the firewall and forwarded a few ports |
| 11:51.28 | dapper-daniel | asterisk registers the sip numers at the provider |
| 11:51.48 | dapper-daniel | but something is wrong.. i dont know.. |
| 11:52.22 | dapper-daniel | you cann see my configs in a german here: |
| 11:52.24 | dapper-daniel | http://www.ip-phone-forum.de/showthread.php?t=109702 |
| 11:52.51 | dapper-daniel | its a german thread but i think you could at least read the configs |
| 11:55.55 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:55.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:58.42 | stse | all right, is anyone here using Asterisk to send SMS? |
| 12:01.10 | carl0s- | gonna buy a g.729 license now. hope it works right |
| 12:03.47 | carl0s- | "Asterisk currently supports G.729 Annex A only". What's that mean then? "The G.729 codec works with all Digium hardware cards and on any processor." Well, I'm not bothered about the 'hardware card'. I want it so my upstream SIP provider can go through my Asterisk box over to my cisco phone and GSM gatway with g.729. that'll work won't it? |
| 12:05.59 | nettie | Hi guys, anyone know what's the maximum jitterbuffer value in ms for the Polycom Soundpoint 501 pleasE? |
| 12:09.57 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:14.07 | TypMic | Why is incoming legacy phone-line audio thru asterisk to a xten softphone so low (having to turn up the receive audio on the PC very loud) while outgoing audio is fine? Anyone know |
| 12:20.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.58 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:23.36 | carl0s- | meh. how long does it take to get a registration code for the g729 codec? I paid via paypal about 7 minutes ago ffs! |
| 12:23.38 | carl0s- | ;) |
| 12:26.44 | trelane_ | carl0s-, didn't you see where it said allow 3-6 weeks for delivery? |
| 12:27.09 | carl0s- | get stuffed! :D |
| 12:27.14 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 12:28.23 | *** join/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 12:29.09 | henk | Hi, are there any "cool" things i can do with the music on hold feature? Like having people skip numbers they don't like etc ? |
| 12:29.26 | e-ddie | yeah, you can play britney spears |
| 12:29.51 | e-ddie | it will make less customers wait in the queue |
| 12:29.59 | henk | I know that, but can i let them control that ? |
| 12:30.00 | trelane_ | e-ddie, because they'd kill themselves |
| 12:30.13 | trelane_ | henk, not that I'm aware of at present |
| 12:30.35 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 12:30.54 | henk | trelane, can the 'on hold' state ba an agi extention ? I might be able to get php agi to do it for me |
| 12:31.23 | trelane_ | not entirely sure, I've never looked at doing it |
| 12:31.28 | e-ddie | henk: yeah, just have a Playback saying "if you dont like our britney spears songs, simpy hang up" |
| 12:31.48 | trelane_ | e-ddie, please never mention her again |
| 12:31.53 | e-ddie | hehehe |
| 12:31.55 | e-ddie | i know you love her |
| 12:32.48 | henk | ha .. yes i get the point. I like britney a lot, but have you heard the song by paris hilton yet? that will really do the job ;) |
| 12:33.14 | e-ddie | no, i'm the kind of person who hangs up :D |
| 12:33.43 | henk | and the paris hilton song will make sure of that ;) |
| 12:34.13 | *** join/#asterisk s0lid (n=jlq@gr-153-4.eglobalreach.net) |
| 12:34.20 | henk | trelane, Having people play a phonegame should be nice also |
| 12:34.51 | e-ddie | except for when someone answers, when the customer is in the middle of a game |
| 12:35.02 | e-ddie | and the customer gets really pissed of :D |
| 12:35.04 | henk | "put me back on hold now!" |
| 12:35.28 | henk | yea... that whould be drefreshingly ifferent though |
| 12:35.53 | e-ddie | could've been a new concept to make the customers stay on the line longer |
| 12:36.21 | e-ddie | and actually make the support/customer care lines a service to pay for pr minute |
| 12:38.12 | r_marvin | i'm trying to setup a h323 channel in asterisk, but i can't figure out what's with the gatekeeper stuff, are there any howtos available? |
| 12:38.34 | r_marvin | i only worked with iax2 and sip until now |
| 12:38.41 | algorithmn | rule of thumb... if possable, don't use h323 |
| 12:39.13 | algorithmn | you can google the protocol.. and get specific stats on it... |
| 12:39.19 | r_marvin | i managed to dodge it until now :( |
| 12:39.31 | algorithmn | beyond that, asterisk has (at least had) shotty support for h323 and skinny |
| 12:39.43 | algorithmn | r_marvin, sorry to hear that |
| 12:39.56 | *** join/#asterisk vlt (n=daniel@dslb-088-073-227-095.pools.arcor-ip.net) |
| 12:40.07 | r_marvin | i got the asterisk-oh323 package from debian, and i'm staring at the config files |
| 12:40.11 | algorithmn | when you were to google h323... you don't need asterisk in there... |
| 12:40.22 | algorithmn | site:voip-info.org asterisk h323 |
| 12:40.29 | algorithmn | in google would be my first place to check |
| 12:40.38 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 12:43.10 | vlt | Hello. I forced my sip phones and my ATA with fax connected to use G711 ALAW when connecting to asterisk. But asterisk converts to ULAW when forwarding the call to my VoIP provider. How can I prevent asterisk from changing the used codec (I hope that my problems with faxing are solved then)? |
| 12:44.06 | trelane_ | vlt, remove ulaw from the allowed codecs in the block for your sip provider in sip.conf |
| 12:44.17 | trelane_ | disallow = all |
| 12:44.19 | trelane_ | allow = gsm |
| 12:47.55 | carl0s- | What time is it over in Digiumland? |
| 12:49.13 | trelane_ | 8:45am |
| 12:49.24 | carl0s- | oh ok |
| 12:49.25 | carl0s- | :) |
| 12:49.32 | trelane_ | though digiumland is a magical place itself, posessing neither time nor space |
| 12:49.39 | carl0s- | lol |
| 12:49.39 | vlt | relane_: Mmh, there's no line containing any codecs in sip.conf. |
| 12:49.45 | vlt | trelane_: Mmh, there's no line containing any codecs in sip.conf. |
| 12:50.25 | trelane_ | vlt, and there you have your first problem |
| 12:50.47 | vlt | trelane_: Believe me, it's not the first one! ;-) |
| 12:50.49 | trelane_ | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 12:50.55 | trelane_ | read |
| 12:51.09 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
| 12:52.03 | r_marvin | algorithmn: i went through the wiki already, but i am still confused about who initiates the channel and whatever documentation i found about the protocol is unclear, that's why i asked about other known good resources |
| 12:52.38 | r_marvin | i'll dig more on the wiki and google, of course, but any other help would be welcome :) |
| 12:53.17 | algorithmn | hmm... http://www.gnugk.org/h323-intro.html looks good |
| 12:53.35 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
| 12:53.50 | r_marvin | thanks |
| 12:54.01 | algorithmn | http://www.packetizer.com/voip/h323_vs_sip/ also |
| 12:54.11 | algorithmn | r_marvin, lemme know how it works out |
| 12:54.20 | r_marvin | i will |
| 12:54.22 | r_marvin | thnks |
| 12:54.25 | tzanger | [TK]D-Fender: around? |
| 12:57.37 | stse | \quit |
| 12:59.59 | carl0s- | algorithmn: that documents seems to greatly suggest that H.323 is a better protocol than SIP. How come there's very little support of H.323 nowadays? |
| 13:00.48 | algorithmn | i don't agree with it then |
| 13:00.55 | carl0s- | :) |
| 13:01.32 | carl0s- | It says most of the comparasons done to date have been biased, or simply blindly following previous biased comparasons. |
| 13:01.42 | vlt | trelane_: Ok I read it. And I found out that in my sip.conf there's only a section [general] containing context, port, nat and a "register =>" line for incoming calls, and then a section for each *client* phone in the local LAN. The SIP account for outgoing calls is not defined in sip.conf. There's an entry in extensions.conf that contains the data for my voip provider: "exten => _X.,2,Dial(SIP/${EXTEN}@****************,gtT)". Wher |
| 13:01.42 | vlt | e to put the "allow" lines now? |
| 13:04.10 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 13:04.16 | toerkeium | hello guys |
| 13:04.40 | toerkeium | does anyone know why I get this error, even when none is connected to the * server? handle_request_subscribe: Got SUBSCRIBE for extension 541152184829@internal from 190.55.25.64, but there is no hint for that extension |
| 13:07.22 | trelane_ | vlt, you've apparently missed some important bits, please reread? |
| 13:09.58 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 13:11.16 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:12.08 | trelane_ | ooh well |
| 13:13.13 | carl0s- | ISTR reading that VAD / Silence Suppression shouldn't be used with Asterisk. Is that still the case? My GSM gateway gives me the option to enable it or not. |
| 13:13.33 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 13:14.33 | vlt | trelane_: Mmh, I reread it. But can't find any hint. I'm trying to understand asterisk and need a liitle help now. I don't want you to configure my whole system, but I simply can't find out what's missing now. Can you help a liitle? |
| 13:14.41 | vlt | trelane_: *little |
| 13:15.33 | trelane_ | <trelane_> vlt, remove ulaw from the allowed codecs in the block for your sip provider in sip.conf |
| 13:15.33 | trelane_ | <trelane_> disallow = all |
| 13:15.33 | trelane_ | <trelane_> allow = gsm |
| 13:16.25 | vlt | trelane_: Ok, I understood that. I'm facing the problem here, that there in fact is NO entry for my sip provider in sip.conf ... |
| 13:16.44 | trelane_ | f'ing what? |
| 13:16.50 | vlt | trelane_: It seems to be done just by extensions.conf |
| 13:16.50 | trelane_ | who is your sip provider? |
| 13:16.59 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.59 | trelane_ | it's not |
| 13:17.15 | vlt | trelane_: voip.dus.net |
| 13:18.39 | vlt | trelane_: the "register" line in sip.conf's section [general] contains login data for my SIP account there. |
| 13:18.40 | carl0s- | vlt: have you posted your sip.conf to a pastebin? |
| 13:18.45 | trelane_ | don't speak german |
| 13:18.56 | trelane_ | vlt, but you need a corrosponding block in your sip.conf to provide context information |
| 13:19.08 | trelane_ | carl0s-, he doesn't have one to speak of |
| 13:19.21 | carl0s- | oh |
| 13:19.25 | trelane_ | where did you get that register line from? |
| 13:20.55 | vlt | trelane_: The register line: "register => 000387220***:passwd@voip.dus.net" |
| 13:21.30 | trelane_ | where did you get it from? a website? did you make it up? the elves? |
| 13:21.51 | vlt | trelane_: ;-) |
| 13:21.58 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:22.00 | vlt | trelane_: I got it from their phone support |
| 13:22.25 | carl0s- | hey benjk :) |
| 13:22.28 | vlt | trelane_: BTW it woks perfectly now. I just want to control what codecs are used ... |
| 13:22.35 | benjk | hi |
| 13:22.55 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 13:23.13 | carl0s- | hi! My WRAP 1e103 is on its way to me now :) |
| 13:24.48 | benjk | cool |
| 13:30.20 | vlt | trelane_: Again the significant line in extensions.conf: "exten => _X.,2,Dial(SIP/${EXTEN}@voip.dus.net,gtT)". I'm shure there's no corresponding entry in any other conf file. The line is as posted here, containing no password. |
| 13:31.33 | *** join/#asterisk henk (n=marius@s5593c2e9.adsl.wanadoo.nl) |
| 13:33.23 | vlt | trelane_: Just found one posted error: There's an addidional "120" (for timeout?): "exten => _X.,2,Dial(SIP/${EXTEN}@voip.dus.net,120,gtT)" |
| 13:33.27 | hmmhesays | bag |
| 13:33.29 | hmmhesays | bah |
| 13:35.17 | trelane_ | vlt, I'm sorry, it just doesn't work that way, you need a connection block below to specify things like codec, and contexts |
| 13:35.47 | Dr-Linux|work | question, how can i put multipal caller ID's in CallerID variable? is there anyway? |
| 13:36.14 | hmmhesays | so getting usb audio to work on this is proving to be a real biatch |
| 13:36.21 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.182) |
| 13:36.49 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 13:36.58 | [TK]D-Fender | Dr-Linux|work : ....huh?! |
| 13:36.58 | Dr-Linux|work | i want to allow few of cell number can make calls via asterisk. so i wanna filter on their caller IDs? |
| 13:37.41 | [TK]D-Fender | Dr-Linux|work : Still not making any sense.... slow down and reword it. |
| 13:38.20 | Dr-Linux|work | [TK]D-Fender, i saw you online at MSN so i came here to ask you ... |
| 13:39.31 | SplasPood | Dr-Linux|work: CID based verification is a bad idea (tm) |
| 13:39.52 | Dr-Linux|work | [TK]D-Fender, simply i want if some one dialed my asterisk PSTN number .. and dials 9 for outging .. so if his callerid is "6502515252" then he is allowed to make outgoing call |
| 13:39.53 | hmmhesays | ok should usb devices show up somewhere like /dev/usb? |
| 13:40.14 | ph|ber | anyone have the new polycom 601 software?? 1.6.7.. |
| 13:40.17 | Dr-Linux|work | otherwise, he get a message like "sorry, you are not allowed" |
| 13:41.07 | Dr-Linux|work | [TK]D-Fender, hopefully it can be done via Gotoif() application .. but i'm not sure how can i add multiptal numbers in the variable |
| 13:42.27 | Dr-Linux|work | [TK]D-Fender, like GotoIf($["${CALLERID(num)}" = "5001"]?5:2) |
| 13:42.55 | Dr-Linux|work | how can i add 10 favorite numbers instead of "5001" |
| 13:43.22 | [TK]D-Fender | Dr-Linux|work : dON'T USE CALLERID, USE A PASSWORD TO PROTECT IT. |
| 13:45.04 | *** join/#asterisk apocn (n=apo@201.229.137.190) |
| 13:45.07 | *** join/#asterisk s0lid (n=jlq@124.107.26.37) |
| 13:45.19 | Dr-Linux|work | [TK]D-Fender, i'm already using password method, |
| 13:45.39 | Dr-Linux|work | but i wanna use athentication on Caller ID |
| 13:45.44 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 13:46.00 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:46.22 | [TK]D-Fender | Dr-Linux|work : then just do 1 GotoIf for each |
| 13:46.31 | apocn | I've installed Asterisk 1.2.9.1 under FreeBSD. I'm trying to enable the cdr_addon_mysql, but I cannot find the module file. Any help? |
| 13:46.32 | Dr-Linux|work | [TK]D-Fender, so my question is simple here, How can i place multipal numbers in {CALLERID(num)} variable |
| 13:46.38 | [TK]D-Fender | Dr-Linux|work : But again, that's just not "A Good Idea (tm)" |
| 13:47.01 | [TK]D-Fender | Dr-Linux|work : That isn't a VARIABLE. Its a FUNCTION. And the answer is NO. |
| 13:47.37 | e-ddie | apocn: search for it on voip-info.org |
| 13:47.58 | apocn | e-ddie, I did, but I didnt found what Im looking for |
| 13:48.05 | Dr-Linux|work | [TK]D-Fender, i see, btw can i have only 5 numbers, so is it possible that i do "Gotoif for each"? |
| 13:48.07 | e-ddie | then try again |
| 13:48.34 | apocn | e-ddie, do you know how to do this? |
| 13:48.48 | e-ddie | no, but i know the info is there |
| 13:48.53 | apocn | he he... |
| 13:50.31 | [TK]D-Fender | Dr-Linux|work : Just do 1 GotoIf for each person. |
| 13:51.16 | apocn | bah, I found it by myself. They dont specify this in voip-info.org. on BSD asterisk-addons is located in a sepparated port. |
| 13:51.21 | tzanger | [TK]D-Fender: hey |
| 13:51.33 | tzanger | [TK]D-Fender: ever seen a polycom ip501 that refuses to have its timezone set? |
| 13:51.37 | tzanger | the time is right, but it's always GMT |
| 13:51.59 | tzanger | I've tried putting -5 in the phone itself, I've tried -5 and override dhcp in sip.cfg... it's always GMT |
| 13:52.18 | [TK]D-Fender | tzanger : it isnt' measures in hours in the phone, its in SECONDS. |
| 13:52.26 | jbalcomb | tzanger press and hold 4, 6, 8, and * |
| 13:52.27 | [TK]D-Fender | tzanger : Been there, done that :) |
| 13:52.51 | tzanger | [TK]D-Fender: on the phone itself it's hours, but sip.cfg is seconds? That would explain it! |
| 13:52.59 | [TK]D-Fender | tzanger : yup |
| 13:53.06 | tzanger | jbalcomb: I don't need to factory reset the phone I don't think |
| 13:53.50 | jbalcomb | tzanger: hey, you don't want to accept my help that fine! just dont come crying to me when nothing works the way you /think/ it should pal!! |
| 13:53.58 | jbalcomb | =) |
| 13:54.02 | tzanger | jbalcomb: hahaha no I'm not ignoring you or anything |
| 13:54.19 | tzanger | the entire config is from the FTP site though so a factory reset isn't gonna help |
| 13:54.24 | [TK]D-Fender | jbalcomb : Yeah, he HEARD you fine ;) |
| 13:54.25 | tzanger | unless there is something "locked" in the phone |
| 13:54.29 | jbalcomb | i think we should replace "reboot" with "factory reset" on the answer ball. |
| 13:55.14 | jbalcomb | tzanger: well, there is the my-newt possibility that you could clear the phones internal config just in case |
| 13:55.53 | file | this is creepy |
| 13:55.55 | jbalcomb | [TK]D-Fender: Hey, I'm just trying to help here damnit. |
| 13:56.11 | file | woke up, got the postal mail, came back... sat down... and my laptop started playing the Jenny 867-5309 song |
| 13:56.41 | jbalcomb | [TK]D-Fender: I think I'm starting to understand the call flow of my own phone system finally. ;) |
| 13:57.09 | jbalcomb | how you make it dark purple? |
| 13:57.09 | [TK]D-Fender | jbalcomb : So we take a left.. right? ;) |
| 13:57.24 | [TK]D-Fender | jbalcomb : Dye? |
| 13:57.34 | jbalcomb | oh wait, yeah, i see now. |
| 13:57.39 | jbalcomb | [TK]D-Fender: correct! |
| 13:58.32 | [TK]D-Fender | jbalcomb : More valuable directions for life : 2 wrongs don't make a right, but 2 rights make a LEFT :) |
| 13:58.38 | [TK]D-Fender | err, 3 |
| 13:59.01 | Alric | So make 3 wrongs? |
| 13:59.25 | rob0 | DIP LEG DYES. (I signed up for a Stanaphone phone number, that's what I got.) |
| 13:59.29 | tzanger | jbalcomb: true enough |
| 14:00.36 | jbalcomb | [^_-] |
| 14:00.45 | jbalcomb | (tm) |
| 14:01.47 | tzanger | there I rebooted the phone bu tI haven't got a web cam to see if the time's right |
| 14:01.55 | jbalcomb | i sooo want to make out with the lady that sits across from me |
| 14:01.59 | tzanger | jbalcomb: pic? |
| 14:02.22 | jbalcomb | 32 year old MILF |
| 14:02.35 | jbalcomb | blonde, plays volleyball and golf |
| 14:03.10 | *** join/#asterisk javar (n=javar@69.79.216.179) |
| 14:03.47 | tzanger | jbalcomb: nice |
| 14:03.58 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:04.29 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:04.55 | jbalcomb | ok, its officially tested. reload app_queue.so does NOT drop callers or members from your call queues |
| 14:05.37 | [TK]D-Fender | jbalcomb : Crossed your fingers and prepared to tell them the call loss was an "Act Of God"? ;) |
| 14:06.15 | jbalcomb | [TK]D-Fender: :) yeps. "Dunno, I'll look at the logs and see what happened." |
| 14:09.48 | jbalcomb | This is great information from Asterisk: "User disconnected from queue tesuto-queue when they almost made it" |
| 14:10.15 | jbalcomb | I'd like to /assume/ that means they hung up while they were first in line. |
| 14:11.07 | [TK]D-Fender | jbalcomb : yup |
| 14:11.21 | *** part/#asterisk TypMic (n=TypMic@outland.cmf.nrl.navy.mil) |
| 14:11.56 | [TK]D-Fender | jbalcomb : * is for overly optimistic. Should read "User disconnected from queue tesuto-queue when * was preparing to segfault for the third, no FOUTH time today" |
| 14:12.11 | [TK]D-Fender | :O |
| 14:12.17 | jbalcomb | ok, I have 'periodic-announce-frequency = 25' & 'periodic-announce = custum/tesuto-periodic' but its not showing even trying to play it. |
| 14:12.32 | jbalcomb | [TK]D-Fender: haha! suckers! |
| 14:12.41 | jbalcomb | [TK]D-Fender: you should submit a patch. |
| 14:13.03 | Mercestes | In the polycom admin guide, I noticed a reference to a "backup proxy server." Is there a specific syntax I use to specify more than one proxy per line? Or are they just saying to use a different proxy server per line? |
| 14:13.04 | [TK]D-Fender | jbalcomb : Probably because you are constantly DUMPING your caller out of the queue and it never reaches that point... |
| 14:13.28 | [TK]D-Fender | Mercestes : rep.1.server.2 |
| 14:13.32 | [TK]D-Fender | Mercestes : reg.1.server.2 |
| 14:14.04 | jbalcomb | [TK]D-Fender: naw, i made my own copy of the queue macro, changed the time to 300 instead of 45, and set up the periodic announce instead of yanking them out. |
| 14:14.14 | Mercestes | [TK]D-Fender: Your my hero. |
| 14:14.39 | jbalcomb | Mercestes: [TK]D-Fender is everyones Polycom hero. |
| 14:14.52 | jbalcomb | esoecially Polycom's |
| 14:16.50 | [TK]D-Fender | jbalcomb : you mis-spelled "custom" |
| 14:16.56 | jbalcomb | [TK]D-Fender: also, isnt the announce = file supposed to play that file to a members when they take a call? |
| 14:18.43 | hmmhesays | ~hmmhesays |
| 14:18.45 | jbot | i guess hmmhesays is not really here... |
| 14:18.49 | *** join/#asterisk c4t3l (n=c4t3l@69.15.174.114) |
| 14:19.10 | hmmhesays | openwrt is pissing me off today |
| 14:19.11 | hmmhesays | damnation |
| 14:19.52 | Assid | hrmm anyone managed to use real time to log the useragent of the sip account? |
| 14:20.10 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 14:20.34 | jbalcomb | [TK]D-Fender: strange but that works now. |
| 14:21.03 | *** join/#asterisk awi (n=awi@88.81.231.25) |
| 14:22.29 | jbalcomb | "If you would like to leave a message and have us call you back please press 1" |
| 14:22.40 | jbalcomb | or something like that... |
| 14:25.24 | [TK]D-Fender | jbalcomb : Yup. And REMOVE the timeout. |
| 14:25.38 | jbalcomb | [TK]D-Fender: Is there a way to make it so that 1 goes to voicemail but all the other keys just says invalid selection and leave them in the queue? |
| 14:25.52 | jbalcomb | [TK]D-Fender: the timeout on the queue() command? |
| 14:26.25 | [TK]D-Fender | jbalcomb : by default all other keys are ignored, and if you put something there you violate queue priority AGAIN because any valid key EXITS the queue. |
| 14:26.33 | [TK]D-Fender | jbalcomb : Yes, when you call Queue() |
| 14:26.47 | jbalcomb | [TK]D-Fender: i thought it defaults to 300? |
| 14:27.27 | jbalcomb | [TK]D-Fender: ok, well, i'll leave the invalid keys alone for now |
| 14:27.33 | [TK]D-Fender | jbalcomb : The moment you let them CHOOSE to exit you should remove any automated timeout. Wheen I want to talk to a damn human being you can be sure I'll wait in queue 20 mins for it if need be. Thats what my Polycom speakerphone is for! |
| 14:27.48 | [TK]D-Fender | jbalcomb : No... only YOU defaulted to 300 :0 |
| 14:28.01 | file | especially when they have good hold music, like Telus Mobility... |
| 14:28.18 | *** join/#asterisk kiong (i=Kiong@bb58-185-167-101.singnet.com.sg) |
| 14:28.45 | jbalcomb | 'timeout' sets the time in seconds that a call will wait in the queue before it is routed to the next priority in the dialplan. It defaults to 300 seconds (5 minutes). |
| 14:29.05 | kiong | hi all, i got x100p but when i try to run zttool it said alarm RED |
| 14:29.26 | jbalcomb | file: I'm using Groove Salad for my hold music. ;) |
| 14:30.28 | [TK]D-Fender | jbalcomb : Time to override |
| 14:30.39 | jbalcomb | [TK]D-Fender: "Voicemail(su854854)" I can change this to "Voicemail(sb854854)" to use the busy message instead right? |
| 14:30.44 | file | maybe tonight I'm gonna my luck, I can see that you want me |
| 14:30.48 | [TK]D-Fender | jbalcomb : yup |
| 14:30.55 | file | er try my luck rather |
| 14:31.00 | *** part/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
| 14:31.11 | jbalcomb | [TK]D-Fender: good cause right now it tells you we are closed if you press 1' |
| 14:31.34 | [TK]D-Fender | jbalcomb : Don't look at me.. I didn't set up that box! ;) |
| 14:31.35 | jbalcomb | [TK]D-Fender: the unavailable message is used for the DND |
| 14:31.42 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:31.55 | file | [TK]D-Fender: yes you did, you just don't remember! |
| 14:31.57 | jbalcomb | [TK]D-Fender: well, your PO was approved 4 months ago pal |
| 14:31.57 | file | they wiped your memory man |
| 14:31.58 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 14:32.00 | hmmhesays | good lord i'm going to brick this thing yet |
| 14:32.06 | [TK]D-Fender | jbalcomb : just have it BYPASS the prompts and play a fixed recording instead maybe. |
| 14:32.10 | file | hmmhesays: I thought you already did |
| 14:32.32 | E-bola | Anybody familiar with the spa922? MY phone lights up its display at interval without any activity |
| 14:32.33 | jbalcomb | [TK]D-Fender: i was thinking about doing that but this way i can tell them they have to record the prompts themselves for each queue |
| 14:32.41 | E-bola | its a bit anoying, is it when it reregisters or something? |
| 14:33.09 | jbalcomb | E-bola: i think it does that when it senses your going to waste someones time.. |
| 14:33.12 | buzzdeee | hi |
| 14:33.19 | jbalcomb | E-bola: is it doing it right now actually? ;) |
| 14:33.26 | *** join/#asterisk ChrisDE4 (n=ChrisDE@tmo-000-46.customers.d1-online.com) |
| 14:34.09 | buzzdeee | hi benjk, I tried to raise the timeout values, but it does not help anything, I have a lot of calls where the last two or one digits are missing |
| 14:34.19 | jbalcomb | [TK]D-Fender: i thout them things were call trawls? |
| 14:34.21 | [TK]D-Fender | jbalcomb : I guess I'd use the Busy for off-hours, and Unavail for "opt-out" personally. |
| 14:34.38 | E-bola | heh |
| 14:34.41 | jbalcomb | [TK]D-Fender: busy is for when we are busy.... |
| 14:34.57 | ChrisDE4 | hi ... any news or hints to bug http://bugs.digium.com/view.php?id=7546 ? |
| 14:35.15 | jbalcomb | buzzdeee: how are you sending DTMF? |
| 14:35.45 | [TK]D-Fender | jbalcomb : Neither are correct... spelling is off today... |
| 14:36.17 | *** join/#asterisk waglik (n=mwegrzyn@polpak.litex.pl) |
| 14:36.21 | Alric | ugh. accidentally did 'select * from cdr;' talk about console spam... |
| 14:36.33 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 14:37.37 | Mercestes | Alric: Good job. |
| 14:38.47 | waglik | hello! |
| 14:39.15 | buzzdeee | jbalcomb, I have nothing specially configured for DTMF |
| 14:39.18 | waglik | I have weird problem with my asterisk box connected to euroisdn with digium Te210P card |
| 14:40.03 | waglik | new asterisk (1.2.10) does not react to a hangup of incoming call |
| 14:40.31 | waglik | if the call wasn't pickup by dialed extension |
| 14:40.43 | waglik | everything worked ok under 1.2.5 |
| 14:40.54 | waglik | did anyone encounter anything similar? |
| 14:41.02 | *** part/#asterisk aldente (n=ed@pool-72-67-217-107.lsanca.fios.verizon.net) |
| 14:41.16 | jbalcomb | waglik: sorry, i don't running bleeding edge. |
| 14:41.17 | vader-- | is there any reason im having trouble using extension 0 in a menu? |
| 14:41.31 | jbalcomb | vader--: yes? |
| 14:41.35 | vader-- | im getting Jul 27 10:39:55 WARNING[21763]: pbx.c:4792 ast_add_extension2: Unable to register extension '0', priority 1 in 'main-incoming-menu', already in use |
| 14:41.58 | waglik | hmm |
| 14:42.03 | jbalcomb | vader--: cd /etc/asterisk |
| 14:42.10 | jbalcomb | vader--: grep exten => 0 * |
| 14:42.17 | waglik | so I'll try to come back to older asterisk |
| 14:43.24 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:43.33 | ChrisDE4 | hi again... isn't there anyone having trouble with 1.2.10? open channels? or bad speech quality...? |
| 14:43.49 | jbalcomb | [TK]D-Fender: can i set timeout to 0 for infinity? |
| 14:43.51 | ChrisDE4 | I can't believe that I'm the only one |
| 14:44.08 | waglik | ChrisDE4: I'm having problems with not detected hangups on 1.2.10 |
| 14:44.51 | ChrisDE4 | yes. right |
| 14:44.54 | jbalcomb | ChrisDE4: i'm running 1.2.5 and it is solid. |
| 14:45.14 | waglik | ChrisDE4: so zap channel stays open 'till someone answers it or they timeout |
| 14:45.30 | ChrisDE4 | right |
| 14:45.36 | ChrisDE4 | e.g.: sip1-ffm*CLI> show channel SIP/mysipid-0820 |
| 14:45.37 | ChrisDE4 | SIP/mysipid-0820 is not a known channel |
| 14:45.47 | jbalcomb | ChrisDE4 waglik: does that cause you to run out of lines for incoming or outgoing calls? |
| 14:45.47 | ChrisDE4 | and show channels shows that this channel is up |
| 14:46.02 | ChrisDE4 | no. asterisk crashes |
| 14:46.18 | ChrisDE4 | almost every night now |
| 14:46.58 | jbalcomb | [TK]D-Fender: IT WORKS!! Holy crap, my call queue works correctly and uses 1,000,000 less lines of config!! |
| 14:47.39 | waglik | ChrisDE4: hmm, I don't have that much traffic to run out of lines (I have pri connection) |
| 14:48.12 | jbalcomb | ChrisDE4 waglik: ah, gotcha. well, good luck to yaz. |
| 14:48.38 | waglik | ChrisDE4: :) |
| 14:48.49 | jbalcomb | ChrisDE4 waglik: you might try bugging file and Qwell. they seems to be connected to development somehow. |
| 14:49.33 | file | I will neither acknowledge or deny that |
| 14:50.19 | nortex | They are more the connected ;) |
| 14:50.27 | nortex | They are more then connected ;) |
| 14:50.46 | file | if you make sure you're connected, the writing's on the wall |
| 14:51.38 | MikeJ | like from pink floyd? |
| 14:52.02 | file | nah, this is from someone else |
| 14:52.04 | file | I forget whom |
| 14:52.20 | jbalcomb | stumble you might fall |
| 14:52.51 | E-bola | anybody have a spa922? |
| 14:52.57 | E-bola | they apear completely undocumented |
| 14:53.04 | E-bola | like what do the Line Key LED Pattern settings do? |
| 14:53.06 | ph|ber | anyone have the new firmware for polycom 601?? 1.6.7. |
| 14:53.26 | jbalcomb | Connected |
| 14:53.26 | jbalcomb | --Stereo MC's |
| 14:53.30 | trelane_ | file++ |
| 14:53.45 | file | moo |
| 14:55.11 | jbalcomb | [TK]D-Fender: Why am I doing "exten => s,3,SetMusicOnHold(tesuto)" when the queue.conf has a "musiconhold = tesuto"? |
| 14:56.03 | E-bola | heh it needs to be in ledscript |
| 14:56.03 | trelane_ | jbalcomb, because you misspelled "tiesto"? |
| 14:56.05 | E-bola | wonder what that is |
| 14:56.25 | jbalcomb | trelane: tesuto is japanese for test |
| 14:56.32 | EyeCue | hehe |
| 14:56.37 | EyeCue | innocente (tiesto mix) |
| 14:56.40 | EyeCue | for the win. |
| 14:56.53 | mut | I NEED MORE COW BELL! |
| 14:59.10 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.71) |
| 15:00.16 | trelane_ | EyeCue, well done! |
| 15:00.28 | *** part/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 15:00.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:05.46 | [TK]D-Fender | jbalcomb : Remove your repetitive redundancy ;) |
| 15:06.19 | jbalcomb | [TK]D-Fender: yeah, i did and it still works fine. woot. Now i just need these messages to say something intelligent and I'm done. |
| 15:06.37 | [TK]D-Fender | jbalcomb : to remove the timout, leave it blank (but use | to seperate it) |
| 15:06.53 | [TK]D-Fender | jbalcomb : "IQ not included" |
| 15:07.23 | jbalcomb | [TK]D-Fender: Yes, saddest disclaimer of my life. |
| 15:08.17 | [TK]D-Fender | Big Box Of Atoms (some assembly required) |
| 15:08.19 | jbalcomb | Should the busy message say something about high call volume? |
| 15:08.31 | jbalcomb | I'm definitely thinking it should appologize.. |
| 15:08.52 | [TK]D-Fender | jbalcomb : Why lie when the trust is your agents are slackers and jsut ignore calls? ;) |
| 15:09.14 | jbalcomb | [TK]D-Fender: Heidi Brown makes me lie. |
| 15:09.24 | [TK]D-Fender | jbalcomb : Who? |
| 15:09.30 | jbalcomb | [TK]D-Fender: the DOO |
| 15:09.32 | tzanger | [TK]D-Fender: probably that MILF he was referring to |
| 15:09.38 | tzanger | DOO? |
| 15:09.49 | jbalcomb | tzanger: I'd do the DOO too |
| 15:09.55 | tzanger | what's a DOO? |
| 15:09.57 | [TK]D-Fender | :O |
| 15:10.01 | jbalcomb | Director of Operations. |
| 15:10.11 | tzanger | ahh |
| 15:10.22 | jbalcomb | Sorry, I guess they don't have titles like that at blockbuster. |
| 15:10.40 | jbalcomb | hahahaha |
| 15:10.43 | tzanger | funny guy |
| 15:10.48 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 15:10.51 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:10.51 | *** join/#asterisk coppice (n=chatzill@127.166.17.210.dyn.pacific.net.hk) |
| 15:11.01 | rob0 | <== Director of daily operations |
| 15:11.21 | [TK]D-Fender | <- director of parking his butt in a barcolounger |
| 15:11.23 | jbalcomb | do you have a weekly, monthly, and yearly as well? |
| 15:11.44 | jbalcomb | rob0: hey wait, you're a DoDo? |
| 15:11.57 | rob0 | Drat, my secret is out |
| 15:12.49 | jbalcomb | big storm here, ill probably lose power soon |
| 15:14.34 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.161) |
| 15:15.18 | tzanger | jbalcomb: where's here? |
| 15:15.48 | jbalcomb | Cleveland, Ohio, USA |
| 15:16.08 | *** join/#asterisk Maan (n=mbsat@230.139.203.62.cust.bluewin.ch) |
| 15:16.12 | Maan | hello |
| 15:17.04 | eKo1 | tornado... |
| 15:18.34 | hmmhesays | i'm having a hell of a time getting usb audio to work on this wrt |
| 15:19.04 | *** join/#asterisk zrb (i=59202@selje.idi.ntnu.no) |
| 15:19.09 | [TK]D-Fender | hmmhesays : USB audio on a friggen el-cheapo router... you are SICK. |
| 15:19.18 | jbalcomb | [TK]D-Fender: http://pastebin.ca/102348 |
| 15:19.38 | *** part/#asterisk zrb (i=59202@selje.idi.ntnu.no) |
| 15:19.46 | jbalcomb | hmmhesays: Are you going to make it yell PACKET everytime is passes a packet? |
| 15:19.48 | [TK]D-Fender | jbalcomb : line 8 = bad |
| 15:20.06 | jbalcomb | [TK]D-Fender: i know, its noted in my statement. |
| 15:20.08 | [TK]D-Fender | jbalcomb : Heck, line 7 is useless too |
| 15:20.15 | jbalcomb | BRG |
| 15:20.17 | jbalcomb | BRB |
| 15:20.37 | rpm | is it safe to run a te410p card and a tdm400p with two fxo modules in a machine? |
| 15:21.48 | *** join/#asterisk steff (n=steff@26-25.252-81.static-ip.oleane.fr) |
| 15:21.52 | steff | hi all |
| 15:22.44 | [TK]D-Fender | jbalcomb : http://pastebin.ca/102355 |
| 15:22.47 | *** join/#asterisk cr0n (n=david@dsl-146-201-189.telkomadsl.co.za) |
| 15:22.51 | steff | any users of junghanns with zaphfc ? |
| 15:22.55 | [TK]D-Fender | rpm : Should be if your MB is sane |
| 15:24.48 | Maan | anybody here use a linksys/sipura spa-922 or spa-942? |
| 15:28.00 | syzygyBSD | got one in the other office |
| 15:28.32 | Maan | do you have the backlight that lights up by itself every 30 seconds or so? |
| 15:28.33 | syzygyBSD | oh, nm, it is a 941 |
| 15:28.46 | syzygyBSD | Maan: no, no problems like that |
| 15:28.49 | Maan | oh it doesn't have the backlight... |
| 15:29.01 | syzygyBSD | ya, sorry, I checked the version again |
| 15:30.05 | rpm | [TK]D-Fender: i had some problems with two tdm400p's a while back, it gave kernel panics.. so i was wondering if a te410p and tdm400p would do the same. i'll give it a try. |
| 15:30.58 | *** join/#asterisk steff (n=steff@26-25.252-81.static-ip.oleane.fr) |
| 15:35.39 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 15:36.02 | Jason99 | Hello, how would I make *67 block outbound name/number ? |
| 15:36.38 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:36.38 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 15:36.39 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 15:36.53 | smackus | should Monitor work on calls delivered to extensions from a queue? |
| 15:37.10 | Jason99 | I assume I have to set the CallerID() values, but I dont know what to set them too |
| 15:37.14 | smackus | I have extensions that are logged into a queue, and i have them set up to use Monitor. But I get no call recordings. |
| 15:38.05 | *** join/#asterisk enemy^x (n=enemy_x@c213-158-248-202.sdsl.no) |
| 15:38.46 | enemy^x | How can I return the call to the person transfering the call when the person in the other end is busy? |
| 15:38.51 | smackus | If I set Monitor on the Queue it's self it records, but it crashes the whole server. |
| 15:39.00 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:39.36 | enemy^x | Or even return denied to the person trying to transfer at once |
| 15:43.23 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:43.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:44.48 | _alex_mx_ | smackus, do you need to record all the calls in your queue? |
| 15:45.53 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.71) |
| 15:46.14 | smackus | yes... now, I know that in the agents.conf there is a recording piece. But where I am lost is when I record other calls that are not from the queue, ie just regular extensions.... I am also setting a variable for uniqueid. I don't know how, or if it is possible to do this from within the agents.conf. |
| 15:46.58 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402BDC8.pool.t-online.hu) |
| 15:47.30 | _alex_mx_ | smackus, we record all calls from some queues with monitor-format=wav (or whatever format you want) in queues.conf |
| 15:48.24 | salviadud | do you use mixmonitor? |
| 15:48.49 | salviadud | the only problem i see with mixmonitor is when i flash my phone, and put them on hold |
| 15:48.50 | smackus | _alex_mx_: ok. So is it possible that way to assign a file name with uniqueid? for example, here is how I do the other extensions; |
| 15:48.51 | smackus | exten => _1XXXNXXXXXX,1,Set(CALLFILENAME=${UNIQUEID}) |
| 15:48.51 | smackus | exten => _1XXXNXXXXXX,2,Monitor(wav,/recordings/${CALLFILENAME},m) |
| 15:49.11 | smackus | Can I get that same effect within the queues.conf? |
| 15:49.57 | _alex_mx_ | in the dialplan we set it with exten => s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/bosch/q700-${TIMESTAMP}-${CALLERIDNUM}) |
| 15:50.13 | _alex_mx_ | then transfer the call to the queue |
| 15:50.24 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 15:50.36 | _alex_mx_ | add the previous line to queues.conf and you're set |
| 15:50.49 | _alex_mx_ | the previous line being monitor-format= |
| 15:51.34 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 15:51.55 | _alex_mx_ | with that asterisk will initiate the recording you don't have to call monitor or mixmonitor |
| 15:52.35 | _alex_mx_ | oh and adjust SetVar to the new syntax >P |
| 15:52.45 | jbalcomb | [TK]D-Fender: What happens to a call if it can't join the queue? |
| 15:53.26 | _alex_mx_ | jbalcomb, why can't it join the queue? no agents? |
| 15:53.32 | jbalcomb | _alex_mx_: yes |
| 15:53.34 | Assid | hrmm anyone using broadvoice? |
| 15:53.51 | [TK]D-Fender | jbalcomb : Why would that happen? |
| 15:54.06 | jbalcomb | [TK]D-Fender: no queue members |
| 15:54.58 | [TK]D-Fender | jbalcomb : Fine, shove a VM before the hangup with an appology "Sorry our slackers have outdone themselves (ironic contradiction, no?) so leave a VM so they can ignore that TOO..." |
| 15:55.26 | _alex_mx_ | jbalcomb, it should do whatever you have as the next line in your dialplan after Queue() |
| 15:55.49 | jbalcomb | [TK]D-Fender: I like that. Can I have a NoOp in there that lets me know when someone can't join a queue? or perhaps I can search for a line in the log and file a report? |
| 15:56.24 | jbalcomb | [TK]D-Fender: Maybe instead of the VM there I should just Dial the DOO since shes the one that is convinced our queues aren't unmanned |
| 15:56.49 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
| 15:56.49 | [TK]D-Fender | jbalcomb : Make a .call file rigging the caller ID with somehting like "CSRs Lazy Again" or something. |
| 15:56.57 | [TK]D-Fender | jbalcomb : Maybe an automated e-mail.... |
| 15:57.09 | _alex_mx_ | jbalcomb, you could write an AGI that emails everyone and their dog, dump the call to the supervisor..... |
| 15:57.15 | websae | TK: check IM |
| 15:57.24 | jbalcomb | [TK]D-Fender: put the call on PA system? |
| 15:57.51 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 15:58.06 | jbalcomb | "The CRRs handling call queue $ARG1 have failed to take a call" |
| 15:58.08 | rene- | hey |
| 15:58.14 | vader-- | fender i got that php stuff working |
| 15:58.49 | *** join/#asterisk r_marvin (n=marvin@83.166.220.142) |
| 15:59.14 | rene- | i found newman telecom nvfaxdetect 1.0.6 for download from newman's web page, the software is dated may 2005, does anybody knows if that code builds correctly in current 1.2.9.1 asterisk systems?? |
| 15:59.17 | *** part/#asterisk Assid (i=assid@203.115.83.215) |
| 15:59.24 | jbalcomb | [TK]D-Fender: the funniest things is hearing "we don't have long hold times" and then "turn off the queue time announcement so no on knows how long our hold times are". |
| 15:59.32 | hmmhesays | so anyone in here know a lot about usb audio? |
| 15:59.54 | jbalcomb | hmmhesays: perhaps #linux? |
| 15:59.58 | [TK]D-Fender | jbalcomb : Everything is possible |
| 16:00.07 | hmmhesays | yeah |
| 16:00.13 | [TK]D-Fender | vader-- : ABOUT TIME! |
| 16:00.19 | hmmhesays | its ##linux |
| 16:00.26 | *** join/#asterisk vlt (n=daniel@dslb-088-073-227-095.pools.arcor-ip.net) |
| 16:00.28 | [TK]D-Fender | jbalcomb : FUD++ ! |
| 16:00.38 | jbalcomb | hmmhesays: i thought it just changes itself? just like #php |
| 16:01.06 | jbalcomb | [TK]D-Fender: yeah, that shit runs rampant around here. |
| 16:01.47 | jbalcomb | [TK]D-Fender: I'm very very tempted to set joinempty=yes |
| 16:02.16 | Vorondil | hi all, quick question: is there an easy way to get the record() application to save files with numbers that increment in the filenames using more than one digit. (e.g. - "exten => 800,n,Record(custom/%d.ulaw,2,,t)" only creates 0.ulaw, 1.ulaw, etc. I want 00.ulaw, 01.ulaw ... 09.ulaw, 10.ulaw, etc) |
| 16:02.46 | rene- | that autofill feature of 1.4 is cool |
| 16:03.13 | jbalcomb | Vorondil; that is a good question. i dont like numbers being out of order because people cant be bothered to buffer with zeros in thier code |
| 16:04.01 | smackus | if i make a change to the queues.conf, will a reload of asterisk activate those changes? or does asterisk need to be restarted. |
| 16:04.02 | salviadud | 1.4, will it ever come out? |
| 16:04.21 | jbalcomb | smackus: reload app_queue.so will do ya right |
| 16:04.32 | salviadud | we are in the month of Leo, the month of war |
| 16:04.39 | salviadud | and... no 1.4 yet |
| 16:04.44 | jbalcomb | smackus: also, it doesn't kick calls or agents so you are alright there |
| 16:05.00 | smackus | jbalcomb: thanks much. |
| 16:05.07 | jbalcomb | smackus: hey, np. |
| 16:05.10 | vlt | Hello. I forced asterisk in sip.conf's section [general] to only use ALAW codec. Now faxing outbound works fine. The problem is now: When a fax comes in "sip show channels" shows me both outbound and internal ALAW as the used codec. After a few seconds "NOTICE[5029]: chan_sip.c:3608 process_sdp: No compatible codecs!" appears and the internal SIP communication changes to ulaw which kills proper fax recieving. |
| 16:05.13 | vlt | Any idea why the codec is changed even when it's not allowed to? |
| 16:06.20 | vlt | *receiving |
| 16:06.39 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:06.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:06.55 | jbalcomb | vlt: where are you located? |
| 16:07.09 | smackus | jbalcomb: will the agents need to be logged out and back in? or will everything just start working when I make a change to a queue? |
| 16:07.31 | jbalcomb | smackus: everything still worked for me when i did it and my changes took effect |
| 16:07.44 | Vorondil | jbalcomb: so i have to make my dialplan has to deal with the ten's-place? |
| 16:07.53 | rene- | 1.4 should be released at astricon i think |
| 16:07.54 | Vorondil | (manually) |
| 16:07.56 | rene- | like 1.2 was |
| 16:08.13 | jbalcomb | Vorondil: as far as i know, yes |
| 16:08.23 | jbalcomb | vlt: where are you located? |
| 16:08.34 | [TK]D-Fender | jbalcomb : Ask yourself why there would be no agents, and is this something a guy will run back to his desk to correct.... |
| 16:08.58 | Vorondil | jbalcomb: *grumble, grumble* thanks :) |
| 16:09.16 | jbalcomb | [TK]D-Fender: i think i need a way to let them know that they are unmanned more than they think. |
| 16:09.33 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 16:09.38 | salviadud | where will the next astricon be at? |
| 16:09.40 | [TK]D-Fender | jbalcomb : You can monitor that without affecting call flow... |
| 16:09.44 | mog | dallas |
| 16:09.45 | jbalcomb | [TK]D-Fender: As I gather it, they would rather throw someone directly to voicemail than let them sit on hold for 15 minutes. |
| 16:09.58 | salviadud | mmmmm, that's pretty close |
| 16:10.01 | salviadud | you guys know when? |
| 16:10.10 | jbalcomb | [TK]D-Fender: which i don't think is unreasonable but why are we not taking every call? |
| 16:10.16 | carl0s- | How long does it normally take Digium to send you a G729 license code? |
| 16:10.26 | [TK]D-Fender | jbalcomb : They do all sorts of crazy things don't they? |
| 16:10.31 | jbalcomb | carl0s-: 6 - 12 months |
| 16:10.44 | jbalcomb | [TK]D-Fender: yeah, thats how we make our money. ;) |
| 16:10.46 | vlt | jbalcomb, Germany, The Old Europe |
| 16:11.07 | [TK]D-Fender | jbalcomb : don't forget that you are working on the most important feature right now... letting the CALLER decide to wait or not. I'd say leave it. |
| 16:11.21 | rene- | salviadud: october |
| 16:11.29 | rene- | salviadud: you are based in Mexico? |
| 16:11.34 | jbalcomb | vlt: you grep all your configs for ulaw to see if its listed anywhere? if not, restart asterisk just to make sure its not left over from something? |
| 16:12.09 | salviadud | rene-, yeah, I live here since I was 1 year old |
| 16:12.16 | jbalcomb | [TK]D-Fender: Is there some way I can monitor hold times in the queue and alert a manager if a call goes beyond five minutes? |
| 16:12.39 | rene- | salviadud: what part? |
| 16:12.54 | [TK]D-Fender | jbalcomb : yup. AMI. |
| 16:13.01 | salviadud | rene-, the northern part of mexico, Monterrey Nuevo Leon to be precise |
| 16:13.04 | [TK]D-Fender | jbalcomb : I have options like that for my Polycom MB's |
| 16:13.05 | rene- | cool |
| 16:13.16 | rene- | yeah astricon will be held real close to where you are |
| 16:13.22 | jbalcomb | [TK]D-Fender: What is "MB's"? |
| 16:13.31 | salviadud | is astricon a weekend? |
| 16:13.39 | rene- | no |
| 16:13.41 | [TK]D-Fender | MicroBrowser. I have live Queue stats on my phones. |
| 16:13.46 | salviadud | a whole week? |
| 16:13.50 | jbalcomb | What is the point of AstriCon? |
| 16:13.58 | rene- | meet people? |
| 16:13.59 | jbalcomb | [TK]D-Fender: Holy crap, gimmie!!! |
| 16:14.01 | vlt | jbalcomb, There's only one remaining ulaw line in iaxprov.conf. But I don't use IAX for connecting. I'll try restarting *. |
| 16:14.03 | rene- | buy stuff? |
| 16:14.10 | rene- | learn something? |
| 16:14.12 | salviadud | get laid with ladies that love to talk on the phone |
| 16:14.16 | rene- | haha |
| 16:14.26 | [TK]D-Fender | jbalcomb : I'll consider it part of my contract :) |
| 16:14.26 | jbalcomb | salviadud: Sweet! I'm in. See you in Dallas. |
| 16:14.27 | rene- | yeah well thats what astricon most sorely lacks |
| 16:14.33 | jbalcomb | [TK]D-Fender: that is sick |
| 16:14.47 | jbalcomb | [TK]D-Fender: Should that be covered by your Polycom contract? ;) |
| 16:14.57 | salviadud | come on! i know some ladies in the phone industry |
| 16:14.58 | [TK]D-Fender | jbalcomb : How so? Didn't say I'd charge for previous dev time, just installing and personalizing. |
| 16:14.59 | rene- | D-Fender sweet |
| 16:15.15 | salviadud | Vicky Crowder... |
| 16:15.22 | mut | clam chowder |
| 16:15.28 | jbalcomb | [TK]D-Fender: Well, you'd best get on it then because I think that would blow these peoples minds |
| 16:15.30 | salviadud | Vicky's secretary |
| 16:15.34 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:15.44 | rene- | salviadud: sure the ones at expocomm cant be counted as being in the phone industry, what can i say is that here in ecaudor the ladies at the call center are really hot |
| 16:16.03 | salviadud | pinche vato!!! |
| 16:16.12 | jbalcomb | besa mi mucho, mi amor |
| 16:16.14 | rene- | ya wey |
| 16:16.18 | rene- | heheh |
| 16:16.24 | salviadud | rene-, yeah man, i know what you're talking about |
| 16:16.26 | rene- | lleven al edgar se cae, como ringtone |
| 16:16.30 | vader-- | is there any way to group a bunch of numbers into a variable and then use that variable for the extension? |
| 16:16.44 | jbalcomb | vader--: yes |
| 16:16.53 | salviadud | rene-, that edgar kid was famous for quite a while... |
| 16:16.54 | vader-- | i have a group of main incomming numbers and i want to set it up like this exten => variable,1,blah |
| 16:17.04 | vader-- | jbalcomb how would i go about doing that? |
| 16:17.06 | rene- | he should be the focus of all conversations |
| 16:17.06 | rene- | now |
| 16:17.17 | carl0s- | jbalcomb: come on now, how long does it really take? I'm very surprised that it wasn't an automated system. |
| 16:17.21 | jbalcomb | vader--: *shrug* i code like ronald mcdonald |
| 16:17.25 | salviadud | guys, would you like to hear to some social engineering? |
| 16:17.33 | rene- | just kiddng |
| 16:17.34 | rene- | sure |
| 16:17.34 | salviadud | i got this 5 min wav file to prove it |
| 16:17.43 | rene- | heh |
| 16:17.45 | vlt | jbalcomb, Same behavior after restarting. I forget to mention before: There's another warning evereytime a fax is received. That was already there before I started reconfiguring codecs: "WARNING[5029]: chan_sip.c:3490 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38". The client is an ATA (Grandstream) with an analog fax connected. |
| 16:17.47 | jbalcomb | carl0s-: yeah, i dunno. digium sucks the pipe. |
| 16:18.12 | jbalcomb | vlt: the ATA is definitely set to alaw? |
| 16:18.28 | jbalcomb | carl0s-: I'm certain it won't take anywhere near that long though. |
| 16:18.34 | carl0s- | :) |
| 16:18.51 | jbalcomb | salviadud: put it on an 800 number so we can all listen |
| 16:18.54 | carl0s- | jbalcomb: I only ordered it four hours ago ! |
| 16:19.25 | vlt | jbalcomb, There's a web based config menu with a section "Preferred Codecs" and 7 select fields. All set to PCMA. |
| 16:19.29 | jbalcomb | carl0s-: haha.. yeah, relax guy. |
| 16:20.04 | carl0s- | jbalcomb: I was being semi-sarcastic. 4hrs is bloody ages when you're wanting to try something out. |
| 16:21.00 | salviadud | jbalcomb, i dunno how to do that :P yet |
| 16:21.17 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 16:21.22 | a1fa | hey |
| 16:21.28 | a1fa | i am trying to create an extension of Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 16:21.31 | a1fa | s-"ERROR" |
| 16:21.35 | a1fa | what error is this? |
| 16:21.41 | jbalcomb | vlt: "the codec information comes in the sdp |
| 16:21.42 | jbalcomb | parameters |
| 16:22.03 | salviadud | ok, check this out, i call this dude, and i tell him that i'm messing with his girlfriend |
| 16:22.03 | *** join/#asterisk adorah (n=Administ@87.68.145.244.cable.012.net.il) |
| 16:22.17 | salviadud | it's 1.7 megs |
| 16:22.19 | a1fa | exten => s- what? |
| 16:22.28 | salviadud | 128 kbps/8hz |
| 16:22.31 | a1fa | exten => s- what? for Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 16:22.32 | salviadud | wav |
| 16:22.45 | salviadud | it's in english |
| 16:23.22 | a1fa | i need to create an extension if user is unreachable |
| 16:23.39 | a1fa | and the reason why the user is unreachable is No route to destination |
| 16:24.08 | jbalcomb | a1fa: doesn't that normally mean the users phone is not registered? |
| 16:24.22 | jbalcomb | a1fa: or maybe it means the registration information is incorrect? |
| 16:25.10 | a1fa | yes |
| 16:25.48 | jbalcomb | a1fa: so if asterisk is already try to send the call how is it going to catch the error? |
| 16:25.59 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:26.16 | a1fa | after the fact |
| 16:26.29 | carl0s- | fuck it, i'm sick of waiting. I'll just try http://kvin.lv/pub/Linux/Asterisk/codec_g729-gcc-athlon-sse.so and switch over to the one I've paid for when they get round to giving me a license code. |
| 16:26.30 | jbalcomb | a1fa: what if you just tested the extension first, matched the error, and then sent the call elsewhere? |
| 16:26.32 | a1fa | if its unreachable, unregistred, then play |
| 16:26.39 | rene- | does anybody has a spare copy of nvlinedetect.c? i need one real quick |
| 16:26.41 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:26.41 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:26.54 | rene- | carl0 |
| 16:26.55 | a1fa | jbalcomb : same thing is done if the extension is BUSY ---> S-BUSY |
| 16:27.03 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:27.04 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 16:27.06 | carl0s- | rene-: yo |
| 16:27.07 | a1fa | no reason to test the extension first |
| 16:28.30 | a1fa | carl0s- : you pirate |
| 16:28.46 | jbalcomb | a1fa: yeah, i follow you there but i dont think it exists. testing the extension might let you do it manually. |
| 16:29.02 | salviadud | alright |
| 16:29.05 | carl0s- | oh be quiet! I've paid for the damn thing I just haven't been given the license code that I've paid for yet :) |
| 16:29.05 | salviadud | http://www.yousendit.com/transfer.php?action=download&ufid=C4E6F8E629271CC3 |
| 16:29.30 | salviadud | i kick ass on the phone |
| 16:30.23 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 16:30.33 | ph|ber | salviadud: what is that? |
| 16:30.40 | salviadud | wav file |
| 16:30.46 | *** join/#asterisk Splas (n=jwb@206.252.198.101) |
| 16:30.52 | salviadud | it is me, as hubert cumberdale |
| 16:31.07 | *** join/#asterisk Dunnly (n=Administ@206-248-137-231.dsl.teksavvy.com) |
| 16:31.12 | salviadud | just download it, you'll thank me later |
| 16:31.41 | a1fa | salviadud : you prank called people? |
| 16:31.43 | jbalcomb | [TK]D-Fender: ok, "joinempty = yes" & "leavewhenempty = no" are in place. When they approve my test queue they will be approving those options too, they just might no know that. |
| 16:32.05 | jbalcomb | a1fa: He lives in Mexico so there really isn't anything else to do. |
| 16:32.08 | salviadud | that last one, i did it for a girl |
| 16:32.13 | salviadud | a 16 year old girl... |
| 16:32.17 | salviadud | yum yum |
| 16:32.22 | *** join/#asterisk RoyK (n=roy@ti211310a080-15035.bb.online.no) |
| 16:32.29 | jbalcomb | salviadud: is that legal in Mexico? |
| 16:32.31 | carl0s- | Hubert Cumberdale |
| 16:32.46 | a1fa | salviadud : you got english accent? |
| 16:32.47 | carl0s- | sounds like Kenny really thinks its true :) |
| 16:32.48 | salviadud | who cares if it's legal, its freaking phone |
| 16:33.01 | salviadud | yeah, i do english accents if i want |
| 16:33.05 | jbalcomb | salviadud: I meant the 16 year old girl you DA |
| 16:33.11 | Mercestes | Your sick. |
| 16:33.16 | salviadud | i'm not sick |
| 16:33.21 | a1fa | que puta man? |
| 16:33.21 | salviadud | here's the story |
| 16:33.23 | carl0s- | not many of us English sound quite like that :D |
| 16:33.26 | a1fa | stop dude |
| 16:33.42 | salviadud | my sister's got this friend from san diego |
| 16:33.51 | a1fa | ok, and? |
| 16:33.51 | salviadud | and she likes some oddball from myspace |
| 16:34.15 | a1fa | i think your accent is hillarious |
| 16:34.16 | salviadud | so I told her, "that dude?!, come on, he's an idiot, i'm gonna prank him" |
| 16:34.26 | rene- | so does anybody has nv_linedetect.c coud they post it, no pirating here it is gpl stuff, i just need it real bad now |
| 16:34.33 | a1fa | so this dude is from san diego that you pranked ? |
| 16:34.33 | salviadud | alfa, thanx man :) |
| 16:34.40 | rene- | cant be downloaded, need to be requested via email |
| 16:34.41 | jbalcomb | rene-: I looked but I don't have it. |
| 16:34.55 | rene- | jbalcomb: thanks man |
| 16:34.58 | jbalcomb | no |
| 16:35.01 | Qwell[] | salviadud: Where's the text transcripts? |
| 16:35.03 | jbalcomb | np |
| 16:35.06 | salviadud | yeah, he's from san diego |
| 16:35.12 | a1fa | whats his number |
| 16:35.16 | a1fa | i want to call him |
| 16:35.25 | a1fa | I am Hurbert's cuzin from Finland |
| 16:35.39 | salviadud | i'd need to ask the girl, i don't remember the number |
| 16:35.45 | a1fa | look at your log |
| 16:35.49 | Qwell[] | What, you didn't record the dtmf? |
| 16:35.49 | a1fa | qwick |
| 16:36.06 | *** part/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 16:36.08 | a1fa | too bad |
| 16:36.08 | a1fa | :) |
| 16:36.25 | salviadud | besides, it's on another box, right now i'm at work |
| 16:36.33 | a1fa | so ssh to the box and give it up |
| 16:36.36 | carl0s- | yah |
| 16:36.48 | salviadud | come on! leave the poor dude alone |
| 16:36.51 | carl0s- | :D |
| 16:36.57 | a1fa | that guy sounds hilarious |
| 16:37.02 | a1fa | he sounds like a redneck |
| 16:37.04 | salviadud | he didn't expect a 30 something english man calling him |
| 16:37.05 | carl0s- | he sounded either stoned or braindead. |
| 16:37.13 | jbalcomb | [TK]D-Fender: * hangs up the call after thirty seconds now. You sure about no having anything in the timeout field? |
| 16:37.17 | a1fa | salviadud : how much did you pay for those two minutes? |
| 16:37.23 | salviadud | 0 |
| 16:37.28 | adorah | I have a bitchie problem. Calling from remote iax client to iax trunk I get a hoarse line..any idea where to fuid the problem? |
| 16:37.29 | salviadud | voipdiscount rocks |
| 16:37.46 | salviadud | alfa, I enjoy voip... hehehe |
| 16:37.58 | a1fa | like a mexican enjoys a corona |
| 16:38.00 | a1fa | ? |
| 16:38.14 | a1fa | s/enjoys/enjoying/ |
| 16:38.17 | salviadud | alfa, corona is not my favorite beer... |
| 16:38.21 | a1fa | ah |
| 16:38.22 | Mercestes | Dos Equis... |
| 16:38.26 | salviadud | alfa, but something like that |
| 16:38.26 | a1fa | same thing |
| 16:38.28 | a1fa | :P |
| 16:38.37 | a1fa | puta hermano, puta |
| 16:41.36 | *** join/#asterisk _alex_mx_ (n=_alex_mx@200.78.229.18) |
| 16:42.16 | a1fa | i love prenk calling mexicans |
| 16:42.21 | a1fa | s/prenk/prank/ |
| 16:42.27 | salviadud | now this one |
| 16:42.29 | salviadud | this one i like |
| 16:42.32 | salviadud | http://www.yousendit.com/transfer.php?action=download&ufid=86A9D18518FBDF18 |
| 16:42.40 | a1fa | lets hear it you crazy banditos |
| 16:43.10 | jbalcomb | wtf. "file.c:584 ast_readaudio_callback: Failed to write frame" |
| 16:43.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:43.53 | a1fa | lol |
| 16:43.55 | jbalcomb | [TK]D-Fender: Leaving the timeout empty or using 0 seems to default to 30 seconds where it promptly hangs up. what gives/ |
| 16:44.01 | salviadud | hehehe |
| 16:44.05 | a1fa | dude, who did you call now? |
| 16:44.13 | salviadud | rehab clinic |
| 16:44.16 | a1fa | where |
| 16:44.17 | jbalcomb | file Qwell[]: Is there a way to put a call in the queue forever? |
| 16:44.18 | salviadud | in florida |
| 16:44.35 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 16:44.39 | Qwell[] | jbalcomb: what, like make it ring a single person forever? |
| 16:44.41 | *** join/#asterisk CyberMad (n=cybermad@202.73.117.106) |
| 16:44.43 | Qwell[] | seems a bit...useless |
| 16:44.45 | salviadud | i like it, because i do a more "american" accent |
| 16:44.48 | carl0s- | if "show codecs" gives me : "256 (1 << 8) (0x100) audio g729 (G.729A)" then does this mean g.729 is loaded and ready? |
| 16:44.49 | Qwell[] | just Dial() without a timeout if tat's all you want |
| 16:45.05 | RoyK | carl0s-: show modules like 729 |
| 16:45.08 | jbalcomb | Qwell[]: no, just want the call in the queue forever, i.e. no time out |
| 16:45.27 | Qwell[] | don't set a timeout |
| 16:45.30 | RoyK | carl0s-: and show g729 |
| 16:45.30 | a1fa | crazy mexicans prank calling americans |
| 16:45.44 | carl0s- | RoyK: thanks. It does show up there. It says "use count: 0" though. |
| 16:45.46 | a1fa | 1) they take over our jobs 2) they run across the border 3) now they prank call us.. what is this world coming to |
| 16:45.46 | salviadud | hey, i'm an artist! |
| 16:45.56 | eKo1 | a1fa: hahaha |
| 16:46.01 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:46.05 | RoyK | carl0s-: that just means no channels is using it |
| 16:46.05 | jbalcomb | Qwell[]: Queue(${ARG5}||||) seems to default to 30 seconds |
| 16:46.17 | carl0s- | RoyK: so if I place a call and it shows, "1", then I'm laughing. |
| 16:46.23 | Qwell[] | I'm pretty sure it shouldn't be doing that |
| 16:46.31 | Qwell[] | I'd have to look though...gimme a sec |
| 16:46.32 | jbalcomb | Qwell[]: As does.. Queue(${ARG5}||||0) |
| 16:46.37 | [TK]D-Fender | jbalcomb : thats due to another setting in your queue |
| 16:46.41 | salviadud | asterisk is the best thing that has happened to me, i prank since the days of H323 |
| 16:46.42 | Qwell[] | 0 should be 0 |
| 16:46.44 | jbalcomb | [TK]D-Fender: what setting? |
| 16:46.46 | RoyK | carl0s-: it should also show the number of licenses |
| 16:46.49 | [TK]D-Fender | jbalcomb : PB it |
| 16:47.06 | salviadud | back when dialpad.com was free |
| 16:47.06 | jbalcomb | Qwell[]: well, sometimes 0 = off and sometimes 0 = inifinity |
| 16:47.13 | Qwell[] | 0 is..0 |
| 16:47.15 | carl0s- | royK: i'm using an unlicensed version based on the Intel IPP code, until Digium get round to sending me the license code that I paid for 4hrs ago. |
| 16:47.32 | [TK]D-Fender | jbalcomb : not 0.... you want [blank] |
| 16:47.44 | *** join/#asterisk chris-fn (n=chris@netblock-66-51-202-171.dslextreme.com) |
| 16:47.46 | jbalcomb | [TK]D-Fender: i tired both |
| 16:47.46 | RoyK | carl0s-: then I don't think show g729 is relevant. pastebin show modules like 729 |
| 16:47.52 | *** join/#asterisk BurnDown (i=Luft@bl5-49-16.dsl.telepac.pt) |
| 16:48.23 | jbalcomb | [TK]D-Fender http://pastebin.ca/102495 |
| 16:48.43 | carl0s- | RoyK: looks good to me: http://pastebin.ca/102496 |
| 16:49.26 | jbalcomb | [TK]D-Fender: http://pastebin.ca/102498 |
| 16:49.32 | chris-fn | okay guys im having trouble installing the zaptel library. the first question i have is: should i 'yum update' before i start installing. |
| 16:49.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:49.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:49.47 | RoyK | carl0s-: indeed |
| 16:49.50 | *** join/#asterisk dlynes_laptop (n=dlynes@zz212094.cipherkey.net) |
| 16:49.57 | salviadud | chris-fn, is that fedora core? |
| 16:50.03 | chris-fn | centos4 (yes) |
| 16:50.04 | RoyK | carl0s-: try to make a call and 'sip show modules' (if you use sip) |
| 16:50.24 | carl0s- | RoyK: the only problem is that this codec seems to be 1.2.x only. I'm also running svn-trunk for testing. Hopefully the Digium one will work with trunk as well. This one gives "missing mod_data for module xxxx.so" with trunk. |
| 16:50.28 | carl0s- | RoyK: ok 1 sec. |
| 16:50.42 | dlynes_laptop | good morning, jbalcomb |
| 16:51.13 | RoyK | good afternoon, all |
| 16:52.18 | enemy^x | I`m using blindtransfer to return a call to the originating point when busy, but I`m wondering where the callerid gets lost.... Ive read something about transfername but can't seem to find it anywhere. |
| 16:52.40 | *** join/#asterisk robl^ (n=robl@dsl093-025-218.hou1.dsl.speakeasy.net) |
| 16:52.55 | carl0s- | RoyK: sip show modules isn't a recognised command here.. |
| 16:54.56 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
| 16:57.15 | dlynes_laptop | carl0s-, that's because there is no such command |
| 16:57.29 | carl0s- | ok |
| 16:57.31 | RoyK | carl0s-: sorry. sip show channels |
| 16:58.17 | carl0s- | so anyway, I made a call, and tracing back through the log now, it looks like g.729 was used *in parts*, but I'm not sure what the end result was. there's lots of "g729 to alaw" and vice-versa going on. |
| 16:59.03 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:59.14 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 16:59.34 | jbalcomb | dlynes_laptop: good morning to you too sir |
| 17:01.10 | jbalcomb | [TK]D-Fender: You have timeout blank on your setup and it goes to the end of the universe and beyond |
| 17:01.37 | viler | Hello there, I'm trying to use "setinterfacevar=yes" in my queues.conf, but the system shows to me "Unknown keyword in queue setinterfacevar" why ? maybe asterisk version ? |
| 17:02.23 | jbalcomb | viler: I get the same thing for "eventmemberstatusoff = no" even though the sample config says its an option. |
| 17:02.38 | jbalcomb | viler: what version are you running? |
| 17:03.27 | viler | jbalcom: Asterisk 1.2.4 |
| 17:03.48 | jbalcomb | viler: did you look in the queues.conf.sample? I don't see that option in 1.2.5 |
| 17:05.12 | jbalcomb | viler: It's not an option in 1.2.1 either |
| 17:05.13 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:05.18 | viler | jbalcom: No, I am using an example that I found on Asterisk Site |
| 17:05.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:05.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:06.38 | jbalcomb | viler: Kinda looks like that is coming in 1.4 |
| 17:06.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:06.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:07.32 | carl0s- | hmm yep g.729 is working on Athlon XP. I just read that to be able to put someone on hold would require and additional g.729 license. |
| 17:08.27 | Assid | carl0s-: if your transcoding yes |
| 17:08.44 | Assid | or read/write .. unless its passthrough |
| 17:11.22 | *** join/#asterisk eBody (n=none@207.71.51.162) |
| 17:11.32 | carl0s- | So voicemail requires an extra license because one is being used for the call, and another is being used for the .wav/.mp3->g729 transcoding, at the same time? |
| 17:11.51 | eBody | how do i get rid of the static when two people are talking at the same time?? |
| 17:12.21 | ph|ber | carl0s-: yes. |
| 17:12.37 | ph|ber | carl0s-: what we do.. we use 729 on outgoing and incoming.. not for internal use. |
| 17:12.44 | ph|ber | thats stupid on a 10/100 network. |
| 17:12.56 | ph|ber | unless you have like 2342432 extensions all used at same time |
| 17:13.10 | ph|ber | just find a provider that supports 729. |
| 17:13.57 | carl0s- | yep. but wouldn't it keep CPU usage down if both my upstream provider and my local device were using g.729? rather than just using g.729 for upstream SIP provider and having Asterisk transcode that to u-law for the client? |
| 17:14.20 | [TK]D-Fender | carl0s- : Or you can switch your recording formats to G.729 globally as well as your MoH. |
| 17:14.25 | ph|ber | how many extensions are you talking about?? 1. 2? |
| 17:14.59 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:14.59 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 17:14.59 | carl0s- | ph|ber, one, on an Athlon XP 3200 :D but I'm just trying to understand the principle |
| 17:15.10 | Dovid | is this normal ? |
| 17:15.19 | ph|ber | carl0s-: i would not worry about it. |
| 17:15.21 | carl0s- | Dovid: is what normal? |
| 17:15.32 | ph|ber | use the 729 for outbound and inbound from a provider |
| 17:15.58 | Dovid | i have one macro that calls another and sets a variable with SET when the second macro is done it goes back to the first macro and the variable comes up as empty |
| 17:16.05 | carl0s- | [TK]D-Fender: If I was to do that (switch the whole * system and all media to g.729), then where would the g.729 codec license be being used? For recording new voicemail and playing back existing, or what? |
| 17:16.21 | Dovid | it seems i can only pass values with maros forward and not backwards |
| 17:16.42 | ph|ber | 729 gets used for everything, if you had 1 call, and 1 going to voicemail, you would need 2 codecs. |
| 17:16.51 | ph|ber | or else it would be ulaw |
| 17:17.17 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 17:17.38 | robl^ | yo , bkw_ |
| 17:17.42 | dlynes_laptop | Does anyone know if asterisk is capable of performing a 'link'? I'm not sure what the telco terminology is for it, but 'link' is the term Nortel systems give to it |
| 17:18.27 | robl^ | dlynes_laptop: flash to an external line.. i.e. centrex control? |
| 17:18.28 | dlynes_laptop | It's basically a transfer, where it relinquishes the phone line when doing the transfer |
| 17:18.35 | carl0s- | I'll wait for this free g.729 codec to be compiled for svn-trunk then. I've just paid $10 to use the codec whilst I learn and play about with this at home. I'm not paying $10 again. I already bought one of their TDM04B cards. |
| 17:18.45 | dlynes_laptop | but it's on analog lines, not centrex lines |
| 17:19.05 | [TK]D-Fender | carl0s- : It would only rarely be used for missing bits and pieces. other alls from non-g729 enpoints, etc. |
| 17:19.24 | dlynes_laptop | robl^, is that the one you're thinking of, too? |
| 17:19.31 | viler | jbalcomb: If I do not have the setinterfacevar variable, what can I do to know the member of queue that answers the call ? |
| 17:20.10 | [TK]D-Fender | dlynes_laptop : Centrex = analog. Just that the telco offers a few more analog accessable features on it. |
| 17:20.11 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 17:20.13 | smackus | could someone double check my work? Is this correct for using two pri's in one group? http://pastebin.ca/102533 |
| 17:20.47 | dlynes_laptop | [TK]D-Fender, i thought centrex was digital? When I hook a centrex line up to my butt set, I don't get dial tone |
| 17:21.08 | dlynes_laptop | [TK]D-Fender, i have to hook a meridian digital handset up to it to get dial tone |
| 17:21.45 | robl^ | dlynes_laptop: a "link" on a Norstar is just a flash. Centrex is analog from the telco but has features like call transfer, forwarding, etc. |
| 17:22.08 | carl0s- | [TK]D-Fender: That contradicts what pb|ber just said. So if I convert all media, and tell * to record voicemail in g.729 etc, would I still need a g.729 codec? What's the benefit of converting everything? Does it mean only one codec is needed per call rather than the additional ones needed when doing voicemail * MoH? |
| 17:22.25 | dlynes_laptop | robl^, so the 'centrex' lines where you need to hook a nortel digital handset up to, are not really called centrex then? They're called something else? |
| 17:23.12 | robl^ | dlynes_laptop: Centrex is a service from the phone company.. offered on analog lines.. |
| 17:23.16 | bkw_ | ho robl^ |
| 17:23.18 | bkw_ | er yo |
| 17:23.21 | *** join/#asterisk pnk (i=pnk@pdpc/supporter/active/pnk) |
| 17:23.23 | bkw_ | didn't mean to call you a ho :P |
| 17:23.31 | dlynes_laptop | robl^, so what's the digital service they offer called then? |
| 17:23.34 | robl^ | I ain't no ho! at least not today. ;_ |
| 17:23.36 | carl0s- | "If you record Voicemails in g729 (which is possible with * by using format_g729.c) and all your phones used g729, you would not need a license, however you would need a license to access the Voicemail from PSTN!". hmmm. I see. Why would I still need one to access the voicemail from PSTN? |
| 17:23.47 | dlynes_laptop | robl^, and how would I perform this 'link' on an analog line? no idea? |
| 17:23.49 | pnk | Can I use a 5v PCI card in a standard i386 pc or should I get the 3..3v ? |
| 17:24.15 | dlynes_laptop | pnk, 3.3V...5V is usually only available in certain rack-mount systems |
| 17:24.28 | coppice | wrong way around |
| 17:24.39 | coppice | most PCs only have 5V slots |
| 17:24.42 | dlynes_laptop | coppice, oops...thought 3.3V was the factory standard |
| 17:24.47 | robl^ | dlynes_laptop: show application flash |
| 17:25.04 | dlynes_laptop | coppice, because the new miniPCI standard only supports 3.3V |
| 17:25.29 | coppice | speeds above 33MHz are only 3.3V too |
| 17:25.31 | pnk | Then Ill get the 5V |
| 17:25.57 | *** join/#asterisk gabb0 (n=gabb0@131.202.90.23) |
| 17:26.01 | Zodiacal | anyone know how to set a polycom 601's reg.x.linekeys labels so that they can say somthing to the effect of "line 1", "Line 2", etc? im using reg.1.linekeys="6" so that it repeats the main line for all 6 line apearence keys.. it alows new calls to ring down to the next line apearance button if the prevous button had a call allready.. |
| 17:26.02 | [TK]D-Fender | carl0s- : You'd need the G.729 codec license for anything that does NOT get translated. Might be needed for MeetMe for example... not sure overall, but your load would be significanly reduced. |
| 17:26.02 | gabb0 | hey everyone |
| 17:26.04 | viler | Does anyone know if Is possible to know the queue member that finally answers a call ? |
| 17:26.07 | Zodiacal | right now it just repeates the label over and over.. |
| 17:26.37 | [TK]D-Fender | Zodiacal : It isn't a seperate "line", its a single line, with multiple CHANNELS. |
| 17:26.46 | [TK]D-Fender | Zodiacal : Quick answer = NO |
| 17:26.47 | watchy | PCI 2.2 allows for 66 MHz signalling (requires 3.3 volt signalling) (peak transfer rate of 503 MB/s) |
| 17:26.47 | watchy | PCI 2.3 permitted use of 3.3 volt and universal keying, but did not support 5 volt keyed add in cards. |
| 17:26.47 | watchy | PCI 3.0 is the final official standard of the bus, completely removing 5 volt support |
| 17:26.49 | watchy | oops |
| 17:26.59 | gabb0 | can someone point in the right direction to make mpg123 work with amd64 (redhat). I saw a post for a patch but it's no longer there. |
| 17:27.01 | dlynes_laptop | robl^, it doesn't seem to give details on what flash applications it supports, and how to select between them |
| 17:27.10 | watchy | i think 5v pci is depreciated' |
| 17:27.15 | [TK]D-Fender | Zodiacal : Also its wastefui to have 6 line keys for a reg like that... seriously... how many calls can 1 human being shuffle? |
| 17:27.27 | dlynes_laptop | robl^, or can you only have one flash application on your analog line? |
| 17:27.37 | [TK]D-Fender | gabb0 : Forget MPG123 and use Native MoH |
| 17:27.51 | [TK]D-Fender | viler : Yes, its in the Queue log file. |
| 17:27.59 | gabb0 | [TK]D-Fender, how do I do that |
| 17:28.14 | [TK]D-Fender | gabb0 : Go look it up on the Wiki, on in THEBOOK |
| 17:28.14 | Zodiacal | tkd-fender a lot |
| 17:28.16 | [TK]D-Fender | ~docs |
| 17:28.23 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:28.23 | [TK]D-Fender | ~book |
| 17:28.24 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:28.27 | robl^ | dlynes_laptop: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Flash |
| 17:28.32 | Zodiacal | tkd-fender they also flog and beat the receptionists here often too |
| 17:28.45 | [TK]D-Fender | Zodiacal : Sounds kinky ;) |
| 17:28.58 | [TK]D-Fender | Zodiacal : IVR should seriously reduce receptionist load.... |
| 17:29.12 | pnk | So I should get the 3.3v version then? |
| 17:29.13 | Zodiacal | they think IVR's arn't personal |
| 17:29.14 | [TK]D-Fender | Zodiacal : Your coproate practices are counterproductive. |
| 17:29.30 | dlynes_laptop | robl^, ah...cool |
| 17:29.37 | [TK]D-Fender | Zodiacal : IVR's only suck when they PREVENT you from getting a human being. |
| 17:29.38 | dlynes_laptop | robl^, thanks....that's exactly what I needed |
| 17:29.54 | pnk | Please do answer me I have this big project and I need to know the correct hardware asap. |
| 17:30.01 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 17:30.03 | pnk | 3.3 or 5v cards? |
| 17:30.08 | robl^ | dlynes_laptop: cash, master card, visa or american express? ;) |
| 17:30.10 | dlynes_laptop | Also, does anyone know if call forward when busy will only call forward a maximum of one call? |
| 17:30.19 | dlynes_laptop | robl^, heh |
| 17:30.26 | viler | [TK]D-Fender: I need to execute an Agi when the call finally arrive to member queue. |
| 17:30.35 | Zodiacal | tkd-fender they need to allways speak with the same person, usualy. and most of the time the lines are that busy |
| 17:30.49 | watchy | i think 3volt pci is standard |
| 17:30.50 | Zodiacal | well maybe i can make the IVR answer if the line is busy |
| 17:30.51 | Zodiacal | hrm |
| 17:30.59 | Zodiacal | well thats just as the same as "please hold" |
| 17:31.06 | dlynes_laptop | watchy, well, i got trounced by coppice when i said that |
| 17:31.26 | watchy | lemme go grab a pci card from out front my store and see what it says |
| 17:31.41 | robl^ | or if it is busy, forward to the local bowling alley? |
| 17:31.57 | dlynes_laptop | watchy, most pci cards (except pci cards from digium) will work in both 3.3V and 5.0V slots |
| 17:32.34 | dlynes_laptop | hey Dr-Linux|work |
| 17:32.39 | coppice | the only machine made which has 32 bit slots with 5V power was one Dell box. almost everything you buy today with 32 bit slows is PCI 2.1 or 2.2. It has 5V power and 3.3V signalling |
| 17:32.45 | [TK]D-Fender | Zodiacal : Everyone gets a personal Queue :) |
| 17:32.53 | Dr-Linux|work | hhmm... :S |
| 17:33.11 | pnk | coppice: And that means I have to use 3.3v cards? |
| 17:33.16 | coppice | sorry, the only machine made which has 32 bit slots with 3.3V power |
| 17:33.18 | Zodiacal | tkd-fender and if the queue is empty, it will not even say anything but directly forward the call? |
| 17:33.37 | Zodiacal | tkd-fender is there a way for ip hard phones to know how many are waiting in a queue? |
| 17:33.39 | pnk | Ok, so if I buy a 5v card - which kind of server will I need to use it? |
| 17:33.47 | [TK]D-Fender | Zodiacal : Things are possible, but I'd sooner do it in dialplan.... |
| 17:33.53 | coppice | any box with a 32 bit slot |
| 17:34.16 | clive- | coppice, the intel nob-hill server board has 3.3V 32bit slots too |
| 17:34.17 | [TK]D-Fender | Zodiacal : Queueing extensions is messy.... |
| 17:34.25 | dlynes_laptop | coppice, i had a cheap piece of crap with two pci-x slots; both pci-x slots were the voltage that wasn't common |
| 17:34.30 | clive- | which really pissed me off:) |
| 17:34.38 | dlynes_laptop | coppice, so i guess it's more than just one dell box |
| 17:34.49 | coppice | pci-x is always 3.3V, but its always 64 bit too |
| 17:35.12 | dlynes_laptop | coppice, but 32-bit cards still work in it |
| 17:35.19 | clive- | when will theyrelease a pci-x T/E1 board? |
| 17:35.19 | nestar | Zodiacal: i don't think so |
| 17:35.24 | coppice | clive-: i didn;t know about that board - so, there are two :-) |
| 17:35.26 | gabb0 | [TK]D-Fender, thanks. simple to setup and works. thanks again. on last question, is this less system intensive than using mpg123 |
| 17:35.39 | dlynes_laptop | clive-, do you mean pci-x, or pci-express? |
| 17:35.56 | dlynes_laptop | clive-, pci-x is a 64-bit version of pci; pci-express is something completely different |
| 17:36.04 | robl^ | what about a VLBus version of a T1 card? ;-) |
| 17:36.14 | coppice | nope. pci-x is a faster version of pci |
| 17:36.20 | nestar | hehe |
| 17:36.23 | watchy | pci-x is 64bit |
| 17:36.29 | *** join/#asterisk Jimerson (n=Jim@197-dcmp9.goeaston.net) |
| 17:36.31 | nestar | i was really hoping for a MCA t1 card. |
| 17:36.32 | carl0s- | .. and 133Mhz |
| 17:36.41 | nestar | that's what i'm waiting for |
| 17:36.51 | clive- | I mean pci-express rather:) |
| 17:37.11 | dlynes_laptop | coppice, i don't know...someone on here told me my pci-x slots were 64-bit extended pci slots |
| 17:37.14 | *** join/#asterisk MUD (n=Administ@206-248-137-231.dsl.teksavvy.com) |
| 17:37.23 | Qwell[] | dlynes_laptop: I think some are 64 bit |
| 17:37.34 | dlynes_laptop | coppice, i'd never seen the stupid slots before |
| 17:37.34 | [TK]D-Fender | gabb0 : Definately, fewer processes, etc. |
| 17:37.34 | clive- | whichever, I am confused with all these slots, but the new motheboards have less and less pci slots |
| 17:37.50 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 17:37.54 | coppice | yet few PCI-E cards are available |
| 17:37.57 | smackus | in queues.conf, I am understanding this correctly? If i use monitor-format = wav, this will record any queue calls answered by an agent, and will create a wav file named using the uniqueid? |
| 17:37.59 | dhill | all incoming calls default to [incoming], yes? |
| 17:38.05 | dlynes_laptop | clive-, i've got an old compaq p75 with six pci slots sharing address space with 6 isa slots...you want to buy it? |
| 17:38.05 | Qwell[] | coppice: there is a bitchin pci-e 16x raid card |
| 17:38.18 | Qwell[] | several 1x NICs |
| 17:38.25 | Qwell[] | I think I saw a 4x quad NIC |
| 17:38.32 | tzanger | yeah they exist |
| 17:38.36 | carl0s- | coppice: starting to see some SATA PCI-E RAID cards now, and some U320 ones too. Also some cheap Silicon Image PCI-E RAID1 cards. |
| 17:38.54 | Qwell[] | the bandwidth is definitely there |
| 17:38.58 | clive- | dlynes lol....cool, I can fir in a couple of "linejack" cards....remember those |
| 17:39.04 | coppice | yeah, but as a percentage of add in cards its pretty low. |
| 17:39.05 | rob0 | My RAID card is PCI-X ... 3ware 9550SX-4LP. |
| 17:39.12 | Qwell[] | coppice: give it time |
| 17:39.13 | tzanger | clive-: I have a pair of those :-) |
| 17:39.16 | dhill | [incoming] |
| 17:39.16 | dhill | exten => s,1,Answer() |
| 17:39.16 | dhill | exten => s,2,Playback(tt-monkeys) |
| 17:39.16 | dhill | exten => s,3,Hangup() |
| 17:39.20 | dhill | why doesn't that work? |
| 17:39.25 | dlynes_laptop | clive-, you mean quickjack? or whatever they were called? |
| 17:39.29 | Qwell[] | dhill: what does the CLI say? |
| 17:39.31 | dhill | if i replace s with 1001 |
| 17:39.33 | Qwell[] | phonejack? |
| 17:39.35 | dhill | it works just fine |
| 17:39.40 | Qwell[] | dhill: s isn't a catchall |
| 17:39.41 | dhill | CLI says nothing |
| 17:39.45 | coppice | giving it time is not much of an option with the motherboard slots disappearing so fast |
| 17:39.52 | dhill | hmm |
| 17:39.53 | smackus | dhill: do you have an extension to use this macro? |
| 17:39.55 | nestar | T1 CARDS FOR VESA LOCAL BUS!!! |
| 17:40.02 | Qwell[] | coppice: yeah, MB manufs are going a little faster than the card manufs |
| 17:40.12 | Qwell[] | but, that's good...makes them hurry it up |
| 17:40.12 | clive- | quickjack/linejack...I have gone through my share of telephony cards |
| 17:40.42 | dlynes_laptop | clive-, the one that's supported natively by the linux kernel, right? |
| 17:40.45 | smackus | dhill: what extension are you dialing to test this? |
| 17:40.48 | dhill | i thought 's' was automatically executed |
| 17:40.56 | dhill | smackus: i am dialing a full phone number |
| 17:40.57 | carl0s- | Order Date: 07/27/2006 Order Status: PROCESSING |
| 17:41.01 | dhill | so if i put in |
| 17:41.04 | carl0s- | What time is it in Digiumland now then? |
| 17:41.05 | dlynes_laptop | anyways |
| 17:41.06 | dlynes_laptop | gotta run |
| 17:41.07 | dlynes_laptop | laters |
| 17:41.10 | dhill | the 9 digits, it works |
| 17:41.14 | file | carl0s-: 12:41PM |
| 17:41.18 | rob0 | carl0s-: 12:41 |
| 17:41.25 | carl0s- | d'oh ;) |
| 17:41.29 | clive- | dlynes, hahaha.... tpgether with all my other junk old cards I have |
| 17:41.30 | dlynes_laptop | carl0s-, 12:40, CST |
| 17:41.47 | smackus | dhill: it works with 9 digits? but what are you dialing when it does not work? |
| 17:41.51 | dlynes_laptop | carl0s-, On Thursday, July 27 |
| 17:42.02 | carl0s- | 6hrs behind me then |
| 17:42.24 | dhill | hold |
| 17:42.27 | dhill | i have an idea :) |
| 17:42.40 | smackus | any one able to help me with my recording queued calls question |
| 17:43.02 | rob0 | +1-256-764-2963 exten. 8463 (TIME) |
| 17:44.00 | carl0s- | rob0: that'd cost a pretty penny from here. We do have a speaking clock though. Sponsored by Accurist (still.). That reminds me. Isn't there an in-built speaking clock with Asterisk? |
| 17:44.08 | smackus | in queues.conf, I am understanding this correctly? If i use monitor-format = wav, this will record any queue calls answered by an agent, and will create a wav file named using the uniqueid? |
| 17:44.09 | carl0s- | asterisk.. accurist.. blah |
| 17:44.31 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 17:46.53 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.static.sasknet.sk.ca) |
| 17:48.00 | Tall-guy | !seen |
| 17:48.03 | Tall-guy | hmmm |
| 17:48.05 | Tall-guy | ~seen |
| 17:48.13 | enemy^x | I`m using blindtransfer to return a call to the originating point when busy, but I`m wondering where the callerid gets lost.... Ive read something about transfername but can't seem to find it anywhere. |
| 17:48.37 | Tall-guy | ~seen Strom |
| 17:48.40 | jbot | strom <i=storm@jihad.carbomb.org> was last seen on IRC in channel #debian, 154d 20h 20m 9s ago, saying: 'dunno'. |
| 17:49.18 | [TK]D-Fender | lol @ Strom's ID :) |
| 17:49.22 | Dovid | TK: is this a bug ? i have Macro1 that calls Macro2, Macro2 sets a value to a variable (with SET cmd) when Macro2 ends and it goes back to Macro1 Macro1 see's the value of the variable that I set in Macro2 as blank |
| 17:49.23 | Tall-guy | :) |
| 17:49.26 | Tall-guy | yeah, kinda funny aint it |
| 17:49.32 | rob0 | oh ... I could put it up on IAXtel or something |
| 17:49.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.71) |
| 17:49.44 | [TK]D-Fender | Dovid : Show me. |
| 17:49.45 | rob0 | carl0s-: yes, it's SayUnixTime() |
| 17:49.52 | file | yay IAXtel |
| 17:49.52 | Tall-guy | ok, who was I talking to last week in this channel about "warbling" audio on my zap lines? |
| 17:49.54 | Dovid | ok |
| 17:50.01 | Dovid | gona pastebin |
| 17:50.02 | brad_mssw | iaxtel is dead, no ? |
| 17:50.07 | file | nope, IAXtel is alive |
| 17:50.22 | brad_mssw | since when ? |
| 17:50.29 | brad_mssw | last time i tried, nadda |
| 17:50.31 | file | since a month or two ago |
| 17:50.34 | Tall-guy | ~seen storm |
| 17:50.36 | jbot | storm <i=joel@200.213.44.81> was last seen on IRC in channel #debian-br, 158d 4h 19m 34s ago, saying: 'Koolosh, resolveu????'. |
| 17:50.41 | *** join/#asterisk daysmen3_ (n=primus@host86-140-40-138.range86-140.btcentralplus.com) |
| 17:50.55 | Tall-guy | hmmm.....I wonder if "seen a guy with an st in his name" will work? |
| 17:51.07 | brad_mssw | hmm, no updated news on iaxtel's site |
| 17:51.17 | file | brad_mssw: I'm working on improving it more first |
| 17:51.24 | rob0 | IAXtel works for me |
| 17:51.32 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 17:51.39 | brad_mssw | ah, cool |
| 17:52.47 | arkonadev | hey do you guys know what to set callerid equal to in the zapata.conf to get inbound callerid's to show up? |
| 17:54.04 | *** join/#asterisk s0lid (n=jlq@61.28.166.67) |
| 17:54.43 | [TK]D-Fender | arkonadev : you DON'T set it. use "usecallerid=yes", "callerid=asreceived". |
| 17:55.08 | rob0 | carl0s-: IAXtel 1-700-764-2963 exten. 8463 (TIME) |
| 17:55.27 | arkonadev | yeah when i set callerid=asreceived it shows the callerid that i am calling not the callerid of the phone i am using to call |
| 17:55.50 | jbalcomb | [TK]D-Fender: ok, I set the timeout to 86400 minutes. |
| 17:56.06 | [TK]D-Fender | jbalcomb : Woohoo! |
| 17:56.14 | jbalcomb | [TK]D-Fender: err.. seconds i mean |
| 17:56.28 | brad_mssw | hmm, my iaxtel registration actually succeeded ... cool |
| 17:56.36 | jbalcomb | [TK]D-Fender: i figure nobody will really want to be on hold for more than a day |
| 17:56.37 | *** join/#asterisk Ark_Molt (n=chatzill@65.203.186.131) |
| 17:56.52 | *** join/#asterisk signuts (n=signuts@sig.triton.net) |
| 17:56.59 | [TK]D-Fender | jbalcomb : So they can literally sit around waiting all day for service when they call you! ;) |
| 17:57.06 | Tall-guy | hey [td]d-fender: we were talkin yesterday about the poly ip501 vs linksys sipura 942......what makes the poly your preferred choice? |
| 17:57.24 | signuts | How do I recover from an AGI not being available when asterisk calls AGI(agi://127.0.0.1) |
| 17:58.00 | [TK]D-Fender | Tall-guy : Poly = superior mfg/audio quality & configurability. Considerably better speakerphone & LCD use, and its bigger as well. |
| 17:58.38 | Tall-guy | tkd-fender, works for me :) |
| 17:58.48 | file | brad_mssw: it's pretty insane if you turn iax2 debug on |
| 17:58.48 | [TK]D-Fender | Tall-guy : Also easier to deploy multple phones... |
| 17:59.09 | *** join/#asterisk linlin (n=linlin@c-67-184-229-183.hsd1.il.comcast.net) |
| 18:00.07 | clive- | signuts good question, does it go to the "t" extension |
| 18:00.30 | jbalcomb | [TK]D-Fender: You wanna email me the stuff on using the AMI to check my queues? |
| 18:00.33 | Tall-guy | tkd-fender: well good, I ordered a poly yesterday :) |
| 18:01.13 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 18:02.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:02.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:03.34 | *** join/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 18:03.56 | enemy^x | when u have ${var1} and u do ${var1:1:3} then it should return the first 3 chars in var1 right? |
| 18:04.07 | Qwell[] | enemy^x: no |
| 18:04.10 | Qwell[] | chars 2, 3, and 4 |
| 18:04.20 | Qwell[] | (it's 0 based) |
| 18:05.01 | *** join/#asterisk [Airwolf] (n=airwolf@dsl54026FAB.pool.t-online.hu) |
| 18:05.04 | Qwell[] | You probably want :0:3 |
| 18:05.25 | enemy^x | exten => s,n,noop($[${CALLERID}) works but exten => s,n,noop($[${CALLERID:0:3}) does not work... ? |
| 18:05.42 | Qwell[] | You need to close your $[ |
| 18:05.48 | Qwell[] | but, like...that would just be 1 |
| 18:05.51 | Qwell[] | every time |
| 18:05.55 | Qwell[] | So, remove it entirely |
| 18:06.16 | enemy^x | argh..... |
| 18:06.16 | enemy^x | hehe |
| 18:06.18 | enemy^x | thank you |
| 18:06.24 | enemy^x | been messing around with it too long |
| 18:06.25 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 18:07.05 | carl0s- | What might "Hold by RFC" mean in my misc. SIP settings on my SIP gateway device? |
| 18:07.20 | brad_mssw | file: what codec does it require ? I get 'unable to negotiate codec' responses when I try to make a call |
| 18:07.35 | file | gsm |
| 18:08.04 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:08.04 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 18:08.07 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 18:08.39 | *** part/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 18:09.00 | brad_mssw | file: hah, I had everything except gsm enabled |
| 18:09.09 | file | silly |
| 18:09.21 | jbalcomb | carl0s-: i perhaps its hold via asterisk rather than by the device? Perhaps RFC for rfc2833? |
| 18:09.35 | *** join/#asterisk folder (n=carl0s@compsup.demon.co.uk) |
| 18:09.47 | *** join/#asterisk motu (n=motu@192.165.166.143) |
| 18:14.05 | folder | Anybody got any tips for reducing background noise on my GSM calls? (SIP GSM box -> cellphone). Is this just natural GSM network noise or can I do something to help it? |
| 18:15.18 | clive- | what does this mean :? "ast_indicate: Unable to handle indication 8" |
| 18:18.31 | hmmhesays | bah why is my nfs mount not working |
| 18:18.38 | hmmhesays | it should be open to mount from anywhere |
| 18:18.44 | *** join/#asterisk brad6254 (n=brad6254@pool-72-72-199-176.altnpa.east.verizon.net) |
| 18:19.01 | salviadud | is it true that sun microsystems invented NFS? |
| 18:19.25 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:19.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:19.43 | brad6254 | I'm having trouble getting asterisk to recognize the transer codes. When I dial the transfer code, sometimes it works, most times it doesn't. Any suggestions? |
| 18:20.05 | hmmhesays | salviadud: thats what I read |
| 18:20.16 | hmmhesays | i can't get my wrt to mount my nfs share though |
| 18:20.17 | hmmhesays | damnit |
| 18:20.18 | Tall-guy | does anyone know where and how often the archive of this channel is stored? |
| 18:20.19 | salviadud | maybe if you used samba |
| 18:20.32 | salviadud | bash.org |
| 18:20.46 | folder | "have you ever wondered what's the overhead (IP/UDP/RTP headers) of sending 50 packets per second (most codecs have 20ms frames)? The answer is 16 kbps just for headers" |
| 18:20.49 | folder | that's interesting. |
| 18:20.56 | salviadud | </justkidding> |
| 18:21.26 | salviadud | Tall-guy, why would you want lines and lines of crazy banter and extensions? |
| 18:21.51 | *** part/#asterisk brad6254 (n=brad6254@pool-72-72-199-176.altnpa.east.verizon.net) |
| 18:22.50 | Tall-guy | salviadud: cause i forgot who I was talkling to about sumthin |
| 18:23.12 | salviadud | yeah, you could just grep yourself easily |
| 18:23.24 | salviadud | Tall-guy, i dunno dude... sorry |
| 18:23.58 | clive- | anyone know abt thiese "indication" messages ? |
| 18:24.53 | coppice | folder: makes a farce of trying to change from 8kbps G.729 to 5.3kbps G.723.1, eh? :-) |
| 18:26.15 | StewLG | I have a Sangoma A200 card on my POTS line. It works fine, but I was hoping to have the calls stay entirely off analog, so I got a local DID and forwarded calls to it. That works really well, actually, sounds very professional and clearer. But there's a hitch. When call forwarding goes through, it rings the POTS line briefly, and the Sangoma card goes into action, and answers the dialtone. Then I get two extensions ringing, one of which is t |
| 18:26.17 | *** join/#asterisk boch (n=root@201.216.241.97) |
| 18:26.23 | boch | sup dudes |
| 18:26.38 | StewLG | Can I change the pickup time for the POTS trunk? Or do something else about this problem? |
| 18:26.56 | nestar | StewLG: remove the answer, or insert a wait command |
| 18:27.05 | nestar | that's the simple answer |
| 18:27.19 | StewLG | Where would I put the wait command? |
| 18:27.27 | nestar | before the Answer |
| 18:27.43 | StewLG | I'll reveal my ignorance: which .conf file? |
| 18:27.50 | nestar | extensions.conf |
| 18:27.53 | folder | coppice: it does a bit yeah. There doesn't seem to be much point. Anything below say 32kbps would do me though. I have 256kbps upstream for the moment, and I can't always be sure that I'll have a good clean 64kbps available, hence looking for something a bit smaller at least. |
| 18:28.01 | StewLG | I'll try that. Thank you! |
| 18:28.15 | nestar | i'm assuming your POTS calls goto a specific context, say "inbound" or something |
| 18:28.19 | boch | is it possible to make calls via a cellphone connected through serial rs232 to an * box ? |
| 18:28.21 | nestar | to s,1 in inbound |
| 18:28.50 | nestar | put a Wait(3) in there |
| 18:28.54 | nestar | or similar |
| 18:29.04 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
| 18:29.10 | nestar | Wait(x) where x is the number of seconds you want to wait before answering |
| 18:29.21 | StewLG | Nestar: I'm digging in the conf. I'm trying to follow along.. |
| 18:29.50 | folder | boch: nah. What you can do though is get one of those cellphone->fxs adapters. They turn your cellphone into an analogue phone line, complete with dialtone. Tie that into an fxo card on your * box and that'll do. Alternatively do what I did and buy a GSM <-> SIP gateway for $170 from www.portech.com.tw |
| 18:30.13 | nestar | StewLG: in my zapata.conf file, i have my X100P going to "context=inbound-analog |
| 18:30.39 | TommyTheKid | I have two (fairly stupid) questions I was hoping someone could help me with... first one: Why does "Spawn extension" always exit *non-zero* .. that indicates to me that there was an error.. all that happened was one side or the other hung up. Am I reading too much into that? |
| 18:30.46 | StewLG | It's from-pstn for me in my zapata.conf. |
| 18:30.57 | nestar | ok, so go into extensions.conf |
| 18:31.01 | nestar | and look for from-pstn |
| 18:31.33 | StewLG | Is a 10 line paste considered a flood? |
| 18:31.38 | StewLG | I don't want to be rude. |
| 18:31.44 | carl0s- | StewLG: I would have thought so |
| 18:31.45 | boch | carl0s-: $170 how many channels ? |
| 18:31.47 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 18:31.56 | nestar | pastebin.ca |
| 18:32.22 | carl0s- | boch: single channel. bought it from ebay user portech_888 - someone called Rebecca from Portech. Shipped straight from Taiwan in under a week for $40. |
| 18:32.23 | StewLG | Nestar: oh wow. Great! |
| 18:32.43 | boch | carl0s-: great, thanks |
| 18:33.03 | *** join/#asterisk AJaymn (i=AJaymn@70.59.126.198) |
| 18:33.05 | carl0s- | boch: no prob. works well-ish for me so far. still tweaking things but I'm a newb anyway |
| 18:33.12 | StewLG | nestar: http://pastebin.ca/102604 |
| 18:33.43 | carl0s- | boch: I sent messages to 3 or 4 other people who'd bought them (through the sellers feedback) and they were all happy.. going to buy more etc. |
| 18:34.17 | *** join/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 18:34.41 | nestar | StewLG: what's in extensions_custom? |
| 18:35.26 | CyberMad | is there someone here from indonesia? |
| 18:35.33 | StewLG | Nestar: http://pastebin.ca/102606 |
| 18:35.44 | jbroome | ~fxo |
| 18:35.45 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 18:36.07 | jbroome | i can't keep all my channel bots straight. :) |
| 18:36.16 | nestar | StewLG: you've got some pretty serious macros and other crap going on here... |
| 18:36.26 | StewLG | nestar: It's a TrixBox install. |
| 18:36.37 | Dovid | TK: can i post my problem with variables being sent back and forth between macros ? |
| 18:36.47 | StewLG | I know some of your guys turn up your noses at that -- I'm just getting started. |
| 18:37.19 | StewLG | I wanted to get something working first, and then consider if I wanted to strip it back to the metal and build it up from scratch. |
| 18:37.40 | nestar | StewLG: I'm not really sure. I could spend 30 minutes just figuring out how the call flow works with this thing. |
| 18:37.56 | StewLG | Well, I don't feel so bad for being confused. |
| 18:38.07 | StewLG | Thanks for having a look at it. |
| 18:38.18 | nestar | StewLG: Despite the fact that zapata.conf says "from-pstn", i get the feeling that this from-zaptel context has something to do with it. |
| 18:38.37 | Dovid | TK: This works fine: http://pastebin.ca/102611 |
| 18:38.51 | nestar | StewLG: you can try and add a wait comment to the s, portion of the from-zaptel context |
| 18:38.58 | Dovid | TK: having issues with this one: http://pastebin.ca/102608 |
| 18:39.00 | nestar | make it s,1,Wait(3) |
| 18:39.11 | StewLG | Nestar: Ok, hold on. |
| 18:39.22 | enemy^x | is it possible for me to return result codes to my users, like for example 486 for busy in extensions.conf? |
| 18:40.02 | Dovid | anyone here can help me with passing variables between macros ? |
| 18:41.48 | StewLG | Nestar: Nope.. still answering.. I'll try a huge delay. |
| 18:42.30 | Dovid | anyone ? |
| 18:42.35 | nestar | may not be in the right place. watch the console as the call comes in, see where the calls go |
| 18:42.41 | nestar | set your debug to level 3 |
| 18:42.44 | StewLG | Ok, I'll do that. |
| 18:42.55 | nestar | it should show the you Goto |
| 18:42.58 | nestar | 's and Macros |
| 18:43.55 | StewLG | Yeah, I can see both calls. |
| 18:44.18 | carl0s- | My SIP gateway keeps not accepting calls. Then I give up, and try again and it works. The "sip show channels" says "Last Message: Init: INVITE" for the SIP gateway. any ideas? |
| 18:44.30 | StewLG | "- Executing Goto("Zap/1-1", "ext-group|1|1") in new stack" |
| 18:44.40 | StewLG | Lots of traffic about Zap/1-1 on the second call. |
| 18:45.35 | CunningPike | Dovid: What's the issue? |
| 18:45.55 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
| 18:46.03 | CunningPike | wb, dlynes_office |
| 18:46.06 | carl0s- | Could having the sip-timeout mismatched between the gateway and Asterisk cause a problem? Asterisk is to to 120sec normal, 300sec maximum. Gateway is just set to 300sec. |
| 18:46.30 | dlynes_office | heya cp |
| 18:46.34 | CunningPike | Sup? |
| 18:46.42 | rene- | in the originate manager event there is an option to set variables, how can i retrieve those values? |
| 18:46.54 | StewLG | Nestar: Here's the output http://pastebin.ca/102621 |
| 18:47.32 | *** join/#asterisk onefastsubaru (n=onefasts@adsl-70-250-63-70.dsl.hstntx.swbell.net) |
| 18:48.53 | Dovid | CunningPike: I have 2 marcos |
| 18:48.55 | Tall-guy | cunning: move all your beers out of the server room |
| 18:49.44 | Dovid | the first macro goes to the secon macro, the second maco sets a variable when it goes back to the first variable it dosent see it, this is with a complicated macro with a simple one it works fine. here are the two paste bins |
| 18:49.55 | Dovid | CunningPike: This works fine: http://pastebin.ca/102611 |
| 18:50.06 | Dovid | CunningPike: having issues with this one: http://pastebin.ca/102608 |
| 18:50.18 | nestar | StewLG: looking |
| 18:50.52 | CunningPike | Tall-guy: Heh heh |
| 18:50.59 | StewLG | Nestar: Thank you. |
| 18:51.12 | CunningPike | Dovid: Hang on while I take a look |
| 18:51.55 | Dovid | CunningPike: thanks |
| 18:52.45 | Bullseye_Network | Is there anyone here that can help me with a segfault, I have a backtrace. |
| 18:53.38 | eKo1 | Bullseye_Network: what does the backtrace say? |
| 18:53.44 | Bullseye_Network | When using native musiconhold and I have the agents login the audio gets really garbled for the musicon hold If I switch from native mp3 player to mpg123 and do a moh reload the system dumps. backtrace: http://pastebin.ca/102557 |
| 18:54.13 | *** join/#asterisk bofh42 (n=bofh42@p5482A74A.dip0.t-ipconnect.de) |
| 18:54.19 | Dovid | CunningPike: I think it is a bug when you use M option in dial, |
| 18:54.36 | Tall-guy | arko: usercallerid=yes callerid=asreceived hidecallerid=no (did I miss anything?) |
| 18:54.36 | [TK]D-Fender | Dovid : Your proble is due to something you didn't mention... its in a macro called by DIAL <- |
| 18:54.39 | StewLG | Nestar: It might be relevant to know that I only have one "inbound route" configured in the FreePBX GUI. |
| 18:54.47 | StewLG | It's a catch -all. |
| 18:54.52 | Tall-guy | wow, that was a lagged response from me (ignore!) |
| 18:55.04 | [TK]D-Fender | Dovid : That occurs in another channel completely and hence the variables are local to that channel. |
| 18:55.56 | CunningPike | Dovid: I concur - we have variable passing working fine for regular 'context hopping', but I'm wondering if variables behave the same way in a macro called from Dial() with the M option |
| 18:56.00 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 18:56.04 | nestar | StewLG: I'm not sure, the commands that it's executing, I don't see in your extensions.conf.. |
| 18:56.09 | Dovid | CunningPike: Look at what TK said |
| 18:56.26 | CunningPike | Dovid: Right - makes sense |
| 18:57.18 | CunningPike | Dovid: Can you rewrite what you're trying to do? |
| 18:57.19 | [TK]D-Fender | Dovid : Use ASTDB for maintaining end-to-end continuity and pass the family/key as a value to the macro. |
| 18:57.25 | Dovid | TK: Is there any work around ? Or a way to pass a var from one channel to the next just for this call ? (i can set global for reasons) |
| 18:57.41 | [TK]D-Fender | Dovid : See above |
| 18:58.06 | Dovid | [TK]D-Fender: I did that but the next time a person calls the variable is set and it looks at the last entry |
| 18:58.38 | StewLG | Nestar: Thanks for trying. |
| 18:58.43 | Dovid | end-to-end meaning ? Set(DB(var-indb)=2)) ? |
| 19:00.03 | [TK]D-Fender | Dovid : Better. LIke I said, use ASTDB to maintain continuity. |
| 19:00.37 | Dovid | [TK]D-Fender: I did that but the next time a person calls the variable is set and it looks at the last entry |
| 19:01.02 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 19:01.05 | [TK]D-Fender | Dovid : I never said to use the SAME family/key for every call did I? Make it unique to the call. |
| 19:01.16 | rene- | D-Fender, what about variables that are set using originate? i am sending a line like Variable: a=123 but i dont that it shows up in new callerid or new channel events |
| 19:01.17 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 19:01.22 | [TK]D-Fender | Dovid : In case you missed it there is a subtle hint on what to use to do this... |
| 19:01.31 | *** join/#asterisk s0lid (n=jlq@202.71.179.140) |
| 19:01.34 | rene- | i meant: but i dont see showing up |
| 19:01.43 | [TK]D-Fender | rene- : Don't know originate unfortunately. |
| 19:01.44 | rene- | dont see it showing up |
| 19:01.52 | rene- | np |
| 19:02.15 | rene- | i could use the astdb or the matter right? |
| 19:02.19 | rene- | for the matter |
| 19:02.30 | Dovid | [TK]D-Fender: I am lost. if i make a unique for ASTDB in the seond macro how will the first one know which one to pull up from astdb ? |
| 19:03.30 | Dovid | i am sorry for being such a neeeeeeeeeew be |
| 19:03.42 | carl0s- | I thought jitterbuffer was only used for FXO stuff, not SIP ? I'm getting : "***[JB LOG]*** fixed jitterbuffer created on channel Local/07766087677@fr" |
| 19:04.13 | Dovid | [TK]D-Fender: I think i may have figured it out. one sec |
| 19:04.34 | coppice | have you ever seen an FXO that jittered? |
| 19:04.41 | StewLG | I just added and removed a trunk in FreePBX, and did nothing else -- and now it's "all circuits are busy now". Sheesh. |
| 19:05.39 | carl0s- | coppice: umm. ? dunno. anyhoo, the jitterbuffer parameters are in zapata.conf, that's zaptel which is umm.. hardware lines (ok, not specifically FXO, but line-interface cards anyway). Does this still apply to SIP calls coming in from my upstream SIP provider? |
| 19:06.13 | [TK]D-Fender | Dovid : You pass the family key as a PARAMETER in your macro. |
| 19:06.38 | Dovid | whats a famlily key ? |
| 19:06.43 | carl0s- | Actually, I don't have any jitterbuffer parameters. Those that I saw were in zapata.conf.sample. |
| 19:06.49 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:06.52 | Dovid | i think will generate a variable pass it down |
| 19:06.58 | [TK]D-Fender | Dovid : Fine, go read up on AstDB before you wonder how it all works! |
| 19:07.02 | Dovid | lol |
| 19:07.02 | Dovid | ok |
| 19:07.15 | [hC] | timmeh! |
| 19:07.21 | Dovid | haha |
| 19:07.26 | [TK]D-Fender | Dovid : And you may find variables survive when passed as args not globals. |
| 19:07.37 | Dovid | ok |
| 19:07.45 | Dovid | its ok. give me the beatin i deserveit |
| 19:08.17 | carl0s- | "Asterisk currently has a jitterbuffer in chan_iax2, as well as in chan_zap. There is no jitterbuffer in the RTP-based channels." |
| 19:08.29 | carl0s- | how come asterisk says it's setting up a jitterbuffer on my sip -> sip calls then? |
| 19:08.49 | *** join/#asterisk linlin (n=linlin@c-67-184-229-183.hsd1.il.comcast.net) |
| 19:09.15 | dlynes_office | CunningPike: not much...just haven't been on much lately because a sprinkler burst in our house |
| 19:09.45 | [TK]D-Fender | dlynes_office : Try putting them out on your lawn.. far safer... |
| 19:10.00 | dlynes_office | [TK]D-Fender: fire sprinklers, not garden sprinklers |
| 19:10.04 | [hC] | I had a sprinkler burst this morning when i got up |
| 19:10.07 | [hC] | thank got it was aimed into the toiler! |
| 19:10.10 | [hC] | toilet, even |
| 19:10.20 | dlynes_office | heh |
| 19:10.41 | CunningPike | dlynes_office: Ya - bummer |
| 19:10.56 | CunningPike | carl0s-: What version of asterisk are you running? |
| 19:11.02 | [TK]D-Fender | dlynes_office : Of all stupidity, we have a fire sprinkler directly over our server rack. No we don't have to worry about them burning as they DROWN! |
| 19:11.09 | carl0s- | CunningPike: 1.2.9.1 |
| 19:11.13 | dlynes_office | heh |
| 19:11.34 | dlynes_office | Well, this one was on the wall directly across the room from the cable modem and router :P |
| 19:11.40 | dlynes_office | so guess what got hit with water first? |
| 19:11.44 | CunningPike | carl0s-: No patches? I ask, because there is a SIP jitterbuffer in trunk |
| 19:12.10 | *** join/#asterisk Ralesk (n=ralesk@nilus-1815.adsl.datanet.hu) |
| 19:12.26 | carl0s- | CunningPike: well, it's a trixbox install, so perhaps they patched it in. I've just noticed the SIP jitterbuffer patch at http://bugs.digium.com/view.php?id=3854. |
| 19:12.40 | carl0s- | CunningPike: can I tweak/alter the parameters? |
| 19:12.42 | *** part/#asterisk Ralesk (n=ralesk@nilus-1815.adsl.datanet.hu) |
| 19:12.54 | *** join/#asterisk arkonadev (n=chatzill@65.203.186.131) |
| 19:13.00 | CunningPike | carl0s-: No idea, I'm afraid - no clue about Trixbox |
| 19:13.05 | hmmhesays | anyone ever mount an nfs on a wrt? |
| 19:13.06 | CunningPike | ~lart trixbox |
| 19:13.17 | syzygyBSD | ono! |
| 19:13.52 | carl0s- | well, I have compiled svn-trunk from sauce, and have that running in parallel, but as the open-source g.729 doesn't work with trunk, I'm back to running the trixbox installed 1.2.9.1 while I play with g.729. |
| 19:14.07 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 19:15.59 | onefastsubaru | anyone on here familiar with the new TE412P T1/E1 cards from Digium and the WCT4xxp module? |
| 19:16.29 | *** join/#asterisk Ark_Molt (n=chatzill@65.203.186.131) |
| 19:17.21 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 19:17.47 | Ark_Molt | Why if i set callerid=asrecieved in zapata.conf can't i get the caller ID of an incoming call, just the number that i originally dialed off my cell phone? |
| 19:17.55 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.51) |
| 19:20.25 | carl0s- | Good greif. The license code for my G.729 codec just came through. |
| 19:20.57 | *** join/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 19:21.08 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:21.13 | generalhan | whats up all !? |
| 19:22.16 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-179-213.bflony.east.verizon.net) |
| 19:22.51 | generalhan | im trying to find the easiest way to import my Master.csv to my new DB ... i went here: http://www.voip-info.org/wiki/view/Asterisk+CDR+csv+mysql+import : and followed the instructions but when i run the php script it only imports like 10 lines at a time. does anyone have a better solution ??? |
| 19:24.52 | *** join/#asterisk groditi (n=wroditi@38.113.5.165) |
| 19:25.30 | *** join/#asterisk backblue (n=moo@87-196-79-145.net.novis.pt) |
| 19:25.59 | Ark_Molt | Im having some callerID issues with incoming calls from cell phones... callerid=asreceived in zapata.conf but is always displaying the number i'm calling.. not calling from.. anyone have any suggestions please? |
| 19:26.10 | arkonadev | come on arkmolt everyone knows the answer to that |
| 19:26.54 | groditi | Hi, I am trying to hack res_features.c to display the park address on the phone display rather than 'say' it. but i'm a little over my head. anyone know where i could find some docs forthis kidna thing |
| 19:26.59 | Zodiacal | generalhan can you just import it using phpMyAdmin? |
| 19:27.41 | generalhan | Zodiacal: thought of that too .. but with phpmyadmin you are limited to only a 2MB file for importing ... Master.csv = 95MB |
| 19:27.50 | Zodiacal | oic |
| 19:28.05 | Zodiacal | well 2MB's is better than 10 records :P |
| 19:28.09 | Zodiacal | u could chop it up i guess |
| 19:28.20 | Zodiacal | but thats not very elegant |
| 19:28.42 | generalhan | well this site has a very good script i just have to figure out how to make it work for me |
| 19:28.58 | groditi | what about mysql-import? |
| 19:29.06 | generalhan | i figured it out ... it checks the last date on the DB and only imports whats after that. |
| 19:29.15 | generalhan | i need to take out that check some how to import everything |
| 19:30.09 | Zodiacal | generalhan its the second to last if statment |
| 19:30.26 | Zodiacal | if(strtotime($end) > $lasttimestamp) { // we found a new record so add it to the DB |
| 19:30.35 | Zodiacal | just change that to a loop i guess |
| 19:30.55 | generalhan | i figured i would just put a line in that says $lasttimestamp = |
| 19:31.37 | *** part/#asterisk StewLG (i=user@216-99-218-126.dsl.aracnet.com) |
| 19:31.44 | Zodiacal | there ya go then u can even make that variable a parameter to your script er somthin |
| 19:31.50 | generalhan | but $lasttimestamp = 1154028648 before its run .. so i need to figure out what that representaion of the date would be fore a year ago ! |
| 19:32.54 | *** join/#asterisk MikeJ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:33.38 | Zodiacal | is it seconds passed since like 1970 or somthin? |
| 19:33.46 | generalhan | haha really ?? |
| 19:34.07 | generalhan | so ill just make it = to 0 and it will import everything |
| 19:34.23 | noname32 | groditi, stocker |
| 19:34.24 | SuPrSluG | hello all |
| 19:34.31 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-13.getnet.net) |
| 19:34.42 | Zodiacal | generalhan ya that would work |
| 19:35.00 | SuPrSluG | anyone work with trixbox? sure is a whacky way of doing asterisk! |
| 19:35.34 | SuPrSluG | they should call it MacroBox |
| 19:36.55 | dlynes_office | SuPrSluG: /join #freepbx |
| 19:38.05 | generalhan | Zodiacal: It must have worked ... i ran the script and its been running for a good little bit now, usually it would only take about 30 seconds to import 10 records ! |
| 19:38.21 | SuPrSluG | i have. i'm used to doing the normal asterisk. just ranting i guess |
| 19:38.44 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:39.15 | *** join/#asterisk okdo (n=goldenol@static-71-121-225-7.sttlwa.dsl-w.verizon.net) |
| 19:39.20 | okdo | hi |
| 19:39.38 | *** join/#asterisk linlin (n=linlin@c-67-184-229-183.hsd1.il.comcast.net) |
| 19:40.23 | dlynes_office | SuPrSluG: so don't use it then |
| 19:40.46 | *** join/#asterisk megasquid (n=Mike_Jon@63.119.244.2) |
| 19:40.55 | *** join/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 19:41.08 | Dovid | what does this mean ? |
| 19:41.10 | Dovid | unexpected TOK_EQ, expecting |
| 19:41.10 | Dovid | TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; |
| 19:41.15 | megasquid | does asterisk interpret voicexml files? |
| 19:41.21 | okdo | question --- I have one server passing a call to another server behind NAT --- when the call comes in the phone rings but disconnects within 2 rings anything I can do? |
| 19:41.22 | Qwell[] | DaveHope: means you messed up on your ael |
| 19:41.28 | SuPrSluG | but the crm's nice |
| 19:41.35 | dlynes_office | Dovid: it means you've got a syntax error in your extensions.conf or extensions.ael |
| 19:41.57 | dlynes_office | Dovid: more than likely you've got a space where one doesn't belong...the asterisk expression parser doesn't handle whitespace very well |
| 19:42.10 | Dovid | dlynes_office: i know that, i want to specifci what ir dosent like cause its still working even with the errors |
| 19:42.16 | Dovid | ah ok |
| 19:42.53 | Kernel-Kris | im trying to compile the zaptel drivers for a debian 2.6.8-2-686-smp kernel and it fails, any suggestions ... this is the error im getting make[1]: Entering directory `/usr/src/kernel-headers-2.6.8-2-686-smp' |
| 19:42.53 | Kernel-Kris | make[1]: Makefile: No such file or directory |
| 19:42.53 | Kernel-Kris | make[1]: *** No rule to make target `Makefile'. Stop. |
| 19:42.54 | Kernel-Kris | make[1]: Leaving directory `/usr/src/kernel-headers-2.6.8-2-686-smp' |
| 19:42.54 | Kernel-Kris | make: *** [linux26] Error 2 |
| 19:42.54 | dlynes_office | Dovid: you're out of luck on that...its expression parser is not terribly verbose about specifying where errors are |
| 19:43.10 | generalhan | Kernel-Kris: please use pastebin |
| 19:43.52 | Dovid | i was using quotes |
| 19:44.06 | Kernel-Kris | generalhan: sorry i thought it would add it all to one line |
| 19:44.32 | groditi | anyone here familiar with the park internals? |
| 19:45.08 | *** part/#asterisk onefastsubaru (n=onefasts@adsl-70-250-63-70.dsl.hstntx.swbell.net) |
| 19:45.23 | *** join/#asterisk vlt (n=daniel@dslb-088-073-227-095.pools.arcor-ip.net) |
| 19:45.45 | groditi | im trying to change the way the park extension is handled from it being played back over the phone to a message being sent to the phone to display it in the screen. i can do it via an execl() call |
| 19:46.22 | groditi | but i need to know the IP of the phone, and none of the structs passed to the park function seem to know it |
| 19:47.34 | okdo | do I need to mpa anything besides 5060 for SIP Server behind NAT? |
| 19:48.21 | Kernel-Kris | anyone else have problems compiling zaptel for debian |
| 19:49.09 | backblue | okdo: y |
| 19:49.19 | backblue | and udp rtp ports too. |
| 19:49.36 | backblue | but you never know which port will be used, by the phone |
| 19:49.50 | backblue | so you should have snat to the server |
| 19:50.15 | *** join/#asterisk PoWeRKiLL (n=powerkil@85.64.221.116.dynamic.barak-online.net) |
| 19:51.48 | *** join/#asterisk RoyK (n=roy@host-81-191-115-203.bluecom.no) |
| 19:52.04 | Dovid | dlyned_office: I am noticing that I am only getting those errors when the varible dosent exist |
| 19:52.48 | Dovid | sometimes it isnt created |
| 19:52.58 | dlynes_office | Dovid: so you need to set that variable to a known value when it doesn't exist |
| 19:53.18 | Dovid | thanks |
| 19:53.42 | bkw_ | if you do (${VAR}) it won't error |
| 19:53.51 | bkw_ | like if it doesn't exist |
| 19:55.18 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 19:55.21 | *** part/#asterisk Kernel-Kris (n=kkirklan@lfkn-fw.angelinacounty.net) |
| 19:56.44 | *** join/#asterisk dhill (i=dhill@fog.mindcry.org) |
| 19:59.11 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 19:59.48 | groditi | hmm i dont seem to be having much luck here. anyone here know the source at all? particularly res_features ? |
| 19:59.53 | groditi | or know ast_sendtext? |
| 20:03.52 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 20:06.28 | *** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
| 20:08.57 | sharp | is the IAXy powerful enough to plug in to my phone line and have the house running on it? |
| 20:09.52 | [TK]D-Fender | sharp : I'd bet on low REN support |
| 20:10.16 | sharp | REN? |
| 20:10.26 | Dovid | TK: I figured it out. First time I am writing sucha complex system |
| 20:10.29 | MikeJ | stimpy |
| 20:11.18 | [TK]D-Fender | MikeJ : You useless bloated sack of protoplasm!!!! |
| 20:11.23 | sharp | i don't have any phones with actual ringers... |
| 20:11.40 | clyrrad | I have a phone that uses call forwarding all the time - the phone internally never rings when you call its extension because its forwarded - this phone is also part of a Queue - How can i make it so that if you dial into a Q - and that Q member has their phone set to call forward - that it rings the call forward number instead of the phones? My problem is the forward works only when you dial by extensoin - but not b |
| 20:12.15 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 20:15.20 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.74.28) |
| 20:17.37 | CunningPike | sharp: The IAXy is an FXO device - you plug a phone into it, not a phone line. If you plug a phone line into it, you'll let the factory-installed smoke out the first time someone calls you |
| 20:18.11 | dlynes_office | CunningPike: but that's part of the magic show |
| 20:18.19 | CunningPike | :) |
| 20:19.11 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 20:19.14 | CunningPike | clyrrad: Are you forwarding on the phone, or in asterisk? |
| 20:19.36 | clyrrad | Asterisk |
| 20:19.46 | sharp | CunningPike, i want to plug it into the copper in my house |
| 20:19.56 | sharp | CunningPike, to provide service to the phones in my house |
| 20:20.26 | sharp | REN is 5, that should do fine... |
| 20:20.32 | clyrrad | CunningPike - asterisk takes care of the call forwarding when you press a persons extension - I just does not if you reach that extension comming from a queue.... |
| 20:21.02 | CunningPike | clyrrad: Right - probably because the queue is calling a SIP device, not an extension |
| 20:21.15 | CunningPike | sharp: Sorry - misunderstood your question |
| 20:21.36 | clyrrad | CunningPike - yes it set to agent=SIP/XXX |
| 20:21.38 | dlynes_office | sharp: you want a sipura 3000, not an iaxy |
| 20:21.42 | clyrrad | is that wrong? |
| 20:21.55 | dlynes_office | sharp: it's called a voip gateway, not a voip ata |
| 20:22.06 | CunningPike | clyrrad: Not wrong per se, but it bypasses the extension-based forwarding in asterisk |
| 20:22.11 | dlynes_office | sharp: the iaxy is a voip ata |
| 20:22.21 | clyrrad | CunningPike - what is the way to correct this? |
| 20:22.28 | sharp | dlynes_office, why's that? i dont want a FXO interface |
| 20:22.53 | dlynes_office | sharp: you said you wanted to plug into the copper in your house and provide service to the phones in your house |
| 20:22.54 | CunningPike | clyrrad: If you want agents to be able to take calls from different phones, try getting them to log in, rather than defining SIP UAs as agents |
| 20:23.14 | dlynes_office | sharp: you didn't specify whether the copper was the pairs running to the phones or the phones running to the telco |
| 20:23.18 | Bullseye_Network | How can i figure out what is causing my memory leak? at 10am i have 124m used memory, and now im at 1.35G used.... |
| 20:23.20 | dlynes_office | erm lines running to the telco |
| 20:23.23 | CunningPike | dlynes_office: He means into an FXS line that splits into multiple phones |
| 20:23.30 | sharp | dlynes_office, the copper is running to the phones |
| 20:23.36 | *** join/#asterisk gaspiz (n=gaspiz@86.34.6.164) |
| 20:23.54 | dlynes_office | sharp: yeah...so iaxy, sipura 2000, sipura 2002, Linksys PAP2-NA |
| 20:23.55 | clyrrad | No I do not want agents taking calls on different phones, each employee has thier own extension, and some extensions are part of the Queues, I just want that if an extension has call forwarding enabled that it will forward the call instead of ringing the extensions phone.... |
| 20:24.19 | dlynes_office | sharp: and grandstream handytone 486(?) are all ata's |
| 20:24.24 | dlynes_office | sharp: they'll all do what you're looking for |
| 20:24.36 | CunningPike | clyrrad: Then use phone-based call forwarding |
| 20:24.38 | gaspiz | hi, seems the people in the queue are hearing a "beep" while they are in an active call and another call is coming in. Any ideas why? |
| 20:24.54 | clyrrad | CunningPike - is there not a way to have Asterisk do this? |
| 20:25.00 | sharp | dlynes_office, i think i want an IAXy, iirc doesn't the sipura have FXO also? |
| 20:25.01 | CunningPike | gaspiz: Yes, it's called 'Call Waiting' |
| 20:25.12 | dlynes_office | Bullseye_Network: 1.2.7.1 or 1.2.9.1? |
| 20:25.28 | CunningPike | clyrrad: Not the way you have it set up, no |
| 20:25.30 | gaspiz | CunningPike: where do I remove the beep? |
| 20:25.46 | CunningPike | gaspiz: It's a function of the set - what type of phone are you using |
| 20:25.49 | dlynes_office | sharp: the sipura 3000 is a gateway; the 2000, 2002, pap2-na, handytone 486, cisco ata-186 are all analog terminal adapters (ata's) |
| 20:25.53 | clyrrad | for some reason on our older version of asterisk it was working like that..... |
| 20:26.04 | clyrrad | agents never had to logon |
| 20:26.08 | gaspiz | CunningPike: SPA 941 |
| 20:26.27 | CunningPike | gaspiz: Don't no then - I only know Polycoms |
| 20:26.40 | sharp | ok well i don't need a gateway... 1 ata will do fine i think |
| 20:26.40 | CunningPike | s/no/know/ |
| 20:26.58 | gaspiz | :) |
| 20:27.04 | dlynes_office | gaspiz: because you've got asterisk set to ring the phone, even when it's busy |
| 20:27.13 | dlynes_office | gaspiz: also, you have call waiting enabled on it |
| 20:27.29 | dlynes_office | gaspiz: in your sip.conf, add incominglimit=1 |
| 20:27.30 | clyrrad | CunningPike - thanks for trying... I will try again later on |
| 20:27.41 | dlynes_office | gaspiz: for that particular extension |
| 20:27.48 | CunningPike | dlynes_office: More likely that he has the Sipura equivalent of calls per line set to more than 1 |
| 20:28.13 | gaspiz | dlynes_office: thanks |
| 20:28.17 | dlynes_office | CunningPike: yeah, but if you make the change in sip.conf, then it's a no-brainer...you don't need to worry about adjusting the sipura unit |
| 20:28.31 | dlynes_office | CunningPike: and then you don't have to care how to adjust it on the sipura unit |
| 20:29.01 | CunningPike | dlynes_office: True....... good point |
| 20:31.55 | hmmhesays | bah this thing is giving me hell |
| 20:32.14 | dlynes_office | hmmhesays: you have a low karma score |
| 20:32.26 | MikeJ | heh... |
| 20:32.37 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.71) |
| 20:32.39 | MikeJ | do you have to be nice to have good karma? |
| 20:32.51 | hmmhesays | so I nfs mounted this toolchain on my wrt box |
| 20:32.55 | dlynes_office | no, but good karma is essential to taming asterisk :P |
| 20:32.56 | hmmhesays | but I can't execute gcc, blah |
| 20:33.14 | anthm | you back to that again? =p |
| 20:33.21 | hmmhesays | anthm yep |
| 20:33.27 | dlynes_office | hmmhesays: maybe because gcc is compiled for an intel box, not whatever crappy cpu you're got on your crappy wrt? |
| 20:33.37 | hmmhesays | dlynes_office: funny |
| 20:33.47 | hmmhesays | anyway I'm using the mipsel toolchain off uclibc.org |
| 20:33.53 | dlynes_office | ah |
| 20:34.07 | TommyTheKid | Why does spawn extension (log lines in the console) say it exits "non-zero" .. is hanging up not considered a valid way to end a call? |
| 20:34.13 | anthm | did you chroot to it ? |
| 20:34.16 | hmmhesays | [root@fc50 roto]# usr/bin/gcc -v |
| 20:34.16 | hmmhesays | -bash: usr/bin/gcc: cannot execute binary file |
| 20:34.28 | CunningPike | TommyTheKid: AFAIK, it's always non-zero - never seen a zero one |
| 20:34.33 | dlynes_office | anthm: obviously not :) |
| 20:34.41 | hmmhesays | thats from the wrt |
| 20:34.48 | TommyTheKid | CunningPike: me either, I just wondered, non-zero seems to indicate an error to me :) |
| 20:34.51 | hmmhesays | why would I have to chroot to it to execute? |
| 20:34.57 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 20:35.04 | anthm | cos the paths would be messed up |
| 20:35.08 | TommyTheKid | hmmhesays: file usr/bin/gcc .. |
| 20:35.11 | dlynes_office | hmmhesays: it needs access to the shared libraries, ... |
| 20:35.13 | anthm | you need to chroot to the env it's a whole distro |
| 20:35.17 | hmmhesays | for version? |
| 20:35.25 | CunningPike | TommyTheKid: Things like 'ERROR:' indicate errors to me....... ;) |
| 20:35.36 | TommyTheKid | hehe |
| 20:35.46 | anthm | chroot /wherever/you/mounted /bin/bash or some shit |
| 20:35.59 | Zodiacal | anyone know of a way to know if a incoming call is really a parked call that timed out calling the user back? |
| 20:36.04 | anthm | or wherever bash is related to the point |
| 20:36.16 | Zodiacal | would a good work around be for me to append a number to the callerid saying its a parked call? |
| 20:36.18 | TommyTheKid | Does the WRT with the two phone ports on it work as like FXS ports in Ast? |
| 20:36.40 | Zodiacal | number = string & number.. i.e. Park 1 |
| 20:37.04 | hmmhesays | ahh got it |
| 20:37.13 | CunningPike | Zodiacal: That would work..... |
| 20:37.15 | hmmhesays | i'm gettin rpciod hanging a lot her eon my nfs mout |
| 20:37.17 | hmmhesays | it is pissing me off |
| 20:37.42 | CunningPike | hmmhesays: You need a new keybard |
| 20:37.47 | Zodiacal | cunningpike do you know off hand how parked calls callerid works? does the callerid get reset when it rings back? |
| 20:38.00 | hmmhesays | heh I don't think so |
| 20:38.05 | CunningPike | Zodiacal: Let me try...... |
| 20:38.10 | hmmhesays | ahh yes, chrooting helped |
| 20:38.37 | Zodiacal | cunningpike i guess i could just find the 70 ext and change the callerid string right there |
| 20:38.46 | Zodiacal | and increment the counter i..e park 1 2 3 etc.. |
| 20:39.10 | CunningPike | Zodiacal: Possibly - I have a call parked right now - let's see what happens |
| 20:39.45 | anthm | you are still posibly gonna hit some bs in the res_crypto and some other crap unless you find some binary packages of whatever libs they need so beware |
| 20:40.02 | anthm | you can start by commenting out any files it can't compile |
| 20:40.24 | anthm | and the resulting binary is going to be dependant on the libc from that root |
| 20:40.28 | TommyTheKid | doesn't res_crypto just uses openssl ? |
| 20:40.43 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.71) |
| 20:40.44 | Nugget | res_crypto uses dual-rounds of rot13. |
| 20:40.48 | TommyTheKid | guh, I need a new keyboard too.. or a new interface between the chair and keyboard |
| 20:41.02 | hmmhesays | what is res_crypto used for? |
| 20:41.09 | eKo1 | TommyTheKid: may i recommend the kkh |
| 20:41.14 | anthm | timy wrt mipsel build image doesnt come with all that stuff |
| 20:41.15 | eKo1 | err, hhk |
| 20:41.25 | hmmhesays | cause if I can just ditch it that would be cool |
| 20:41.34 | hmmhesays | and I need chan_oss in there otherwise I would just use the package |
| 20:41.45 | TommyTheKid | eKo1: you may, but I am not that rich.. will stick with the Sun Type 7 for now :) |
| 20:41.55 | anthm | ditch anything it wont compile |
| 20:42.15 | anthm | even if you have to edit the makefile and take the obj out as a first try |
| 20:43.20 | anthm | you will also run into some TZ crap when it starts calling gettimeofday a ton |
| 20:43.49 | hmmhesays | so what do I do about the libc problem |
| 20:43.59 | anthm | bring it with =D |
| 20:44.22 | anthm | put theier libc in it's own dir and put it at the beginning of LD_LIBRARY_PATH |
| 20:44.56 | hmmhesays | so when whenever what I compile looks for it it will find it first |
| 20:45.04 | hmmhesays | instead of openwrt's libc |
| 20:45.21 | anthm | or you could always try and merge that image with the flash disk so it boots that distro instead |
| 20:45.23 | CunningPike | Zodiacal: Sorry - cocked it up. 105 seconds to go now......... |
| 20:45.43 | anthm | only when you prefix the shell command with the LD_ line |
| 20:45.50 | Zodiacal | cunningpike i'll be here :) |
| 20:45.55 | CunningPike | ;) |
| 20:46.07 | Zodiacal | just say my nick it beeps me :P |
| 20:46.10 | Zodiacal | cunningpike |
| 20:46.13 | hmmhesays | i got no serial cable on this thing I can't brick it |
| 20:46.38 | hmmhesays | I wonder what would happend if i just try compiling chan_oss |
| 20:46.52 | hmmhesays | and use the headers from the kamikaze package |
| 20:47.04 | *** join/#asterisk salviadud (n=dude@201.135.3.2) |
| 20:47.27 | salviadud | hmmm, what you pondering? |
| 20:47.46 | Qwell[] | he's pondering going kamikaze |
| 20:47.49 | CunningPike | Zodiacal: Ya - it retains the existing CID - but you could change that before parking the call |
| 20:47.51 | hmmhesays | yeah pretty much |
| 20:48.06 | hmmhesays | if i'm using the same version of asterisk as the packaged one that should work |
| 20:48.12 | Zodiacal | cunningpike trying to find that place now :P |
| 20:48.26 | CunningPike | Zodiacal: OK |
| 20:48.29 | salviadud | Zodiacal, what's your sign dude? |
| 20:48.54 | CunningPike | salviadud: Wrong chat room - try AOL ;) |
| 20:49.10 | hmmhesays | no wayz LOLERZ |
| 20:49.21 | salviadud | ah come on |
| 20:49.24 | Zodiacal | salviadud Exit Only, im straight. |
| 20:49.42 | hmmhesays | smoke break |
| 20:50.05 | salviadud | you guys get way too stressed, you should prank call like me |
| 20:50.09 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.99) |
| 20:50.22 | eKo1 | or prank chat rather |
| 20:51.04 | TommyTheKid | like in a boat? |
| 20:51.20 | salviadud | stress is all emotional, bunch of crybabies |
| 20:54.47 | hmmhesays | so where are those mexican hotties you promised me salviadud |
| 20:55.08 | salviadud | www.fotolog.com/las_antreras |
| 20:55.16 | salviadud | the one with the big hooters |
| 20:56.05 | salviadud | i know her from the call center i used to work in |
| 20:56.16 | salviadud | she's a capricorn, she's crazy |
| 20:56.34 | salviadud | if you ever come down here, easy to hook you up with those ladies |
| 20:57.56 | dlynes_office | CunningPike: you read the georgia strait? |
| 20:58.16 | CunningPike | dlynes_office: Nope - why? |
| 20:58.24 | dlynes_office | CunningPike: well, do you like mexican food? |
| 20:58.46 | dlynes_office | CunningPike: like tapas? |
| 20:59.20 | CunningPike | dlynes_office: Sure |
| 20:59.36 | dlynes_office | CunningPike: Are you familiar with the restaurant, 'Santa Teresita's'? |
| 21:00.35 | *** join/#asterisk TrevorSHarrison (n=trevorsh@24-49-36-218-st.chvlva.adelphia.net) |
| 21:00.37 | Zodiacal | cunningpike i can't seem to figure out how parking works , its like a magic ext :) |
| 21:01.07 | dlynes_office | Zodiacal: features.conf |
| 21:01.20 | CunningPike | dlynes_office: No |
| 21:01.25 | Zodiacal | dlynes_office yeah.. what im trying to do is append a string to parked calls callerid |
| 21:01.31 | Zodiacal | dlynes_office know where i would do that off hand? |
| 21:01.42 | dlynes_office | Zodiacal: you'd have to change some code in res_features.c |
| 21:01.51 | Zodiacal | dlynes_office oic |
| 21:02.18 | Zodiacal | would it be a major thing? |
| 21:03.59 | *** join/#asterisk mocker (n=ks@in.kansas.but.not.a.republi.cn) |
| 21:04.02 | CunningPike | Zodiacal: We're using Polycoms and they pass parked calls into an extension called callpark, so it's a lot easier that way |
| 21:04.46 | Zodiacal | my polycom doesn't seem to let me use the softkey to park calls.. i was just going to create a speeddial for ext 70 |
| 21:04.57 | Zodiacal | i read that polycom's softkey park doesn't work with asterisk |
| 21:04.57 | Zodiacal | :/ |
| 21:05.19 | Zodiacal | cunningpike are you using the park softkey? |
| 21:05.38 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 21:06.05 | CunningPike | Zodiacal: Yes, we are. I'll pm our dialplan entry |
| 21:06.15 | Zodiacal | okie |
| 21:06.39 | Mercestes | ok....I've compiled * with + debug I set verbose 99, set debug 99, and debug level 99, and I still get the "check debug for more info" stuff. *where* is the debug?? |
| 21:06.46 | Zodiacal | when i press the park softkey the phone just presents a blinking cursor like its waiting for me to type somthing |
| 21:07.57 | CunningPike | Zodiacal: exten => callpark,1,ParkAndAnnounce(pbx-transfer:PARKED|120|SIP/${DIALEDPEERNUMBER}|internal,${DIALEDPEERNUMBER},1) |
| 21:08.26 | Zodiacal | and u didn't change anything with yer polycom configs? |
| 21:08.52 | *** join/#asterisk daysmen3 (n=primus@host86-140-209-46.range86-140.btcentralplus.com) |
| 21:08.57 | CunningPike | Put that in your dialplan so it's in the same context as the current call. Then, when you get the blank cursor, enter anything you like and press 'Park' again |
| 21:09.02 | *** join/#asterisk manholas (n=manholas@163.Red-217-125-180.dynamicIP.rima-tde.net) |
| 21:10.06 | dlynes_office | Zodiacal: you can put some settings your in your features.conf file to modify the default behaviour of parked calls, too |
| 21:10.29 | dlynes_office | Zodiacal: that's so that they'll drop into a specific context, timeout after a different period of time and so on and so forth |
| 21:10.38 | Zodiacal | cunningpike thank you! i will give it a try in just a sec.. |
| 21:10.46 | CunningPike | OK |
| 21:11.17 | CunningPike | Zodiacal: Yes - all this assumes that you have something sensible in features.conf for parked calls |
| 21:11.29 | Zodiacal | dlynes_office yeah it seems like [parkedcalls] is that context. maybe i can set the callerid from in there? |
| 21:12.02 | Zodiacal | cunningpike so that allows the user to choose the park # slot? |
| 21:12.17 | CunningPike | Zodiacal: Set it in the callpark extension - exten => callpark,1,Set(CallerID(Name))...... etc |
| 21:12.27 | CunningPike | Zodiacal: Nope - it selects it for you |
| 21:12.50 | CunningPike | Zodiacal: And then dials you back and announces it |
| 21:13.43 | dlynes_office | Zodiacal: maybe, but cunningpike would be a better person to ask |
| 21:13.59 | dlynes_office | Zodiacal: I just recently started using call parking...never had a need for it before |
| 21:14.19 | Zodiacal | okie i go play with it all now... |
| 21:14.24 | Zodiacal | Thanks again guys! |
| 21:14.34 | CunningPike | Zodiacal: No problem |
| 21:14.37 | dlynes_office | Zodiacal: but, you can still try playing with the c code |
| 21:14.54 | dlynes_office | Zodiacal: you're welcome to try that if you're really bored and you need something to do |
| 21:15.05 | Zodiacal | dlynes_office i am curious about it, i havn't alter asterisk code before.. but probably as a last resort :) |
| 21:15.07 | CunningPike | dlynes_office: We're about to experiment with changing away from call parking to simulating a key system style hold functionality |
| 21:15.29 | CunningPike | dlynes_office: Our users don't like call parking for some reason |
| 21:15.37 | dlynes_office | CunningPike: you mean place the call on hold and then pick it up from a different location? |
| 21:15.43 | CunningPike | dlynes_office: Exactly |
| 21:15.47 | Zodiacal | cunningpike shared line appearnces |
| 21:15.53 | Zodiacal | didn't think * could do that yets |
| 21:16.03 | dlynes_office | Zodiacal: yeah...that's exactly what he's talking about |
| 21:16.12 | dlynes_office | Zodiacal: 1.4 will apparently partially support it |
| 21:16.37 | Zodiacal | our users don't like parking either, thats why i thought appending a string telling them that its a parked call calling them back would be helpfull. but a key system or shared line apearnces is what they really want :) |
| 21:16.38 | CunningPike | Zodiacal: Bridged lines, to be pedantic |
| 21:16.55 | Zodiacal | very interesting... |
| 21:20.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:27.38 | *** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do) |
| 21:28.43 | paolob-parroquia | Hi guys! I have a asterisk central working. All is ok, but the CID: the sipura spa-3000 doesn't pass the it to asterisk. Any hint? |
| 21:30.47 | eKo1 | paolob-parroquia: run 'sip debug peer <spa-3000 peer name>' on the cli and check if the cid is being passed. |
| 21:31.54 | paolob-parroquia | eKo1, let me see |
| 21:33.18 | paolob-parroquia | eKo1, where should I see it? the debug gave me too many lines!! |
| 21:33.37 | eKo1 | Can you read SIP? |
| 21:34.01 | paolob-parroquia | eKo1, .... |
| 21:38.40 | *** join/#asterisk jayma777 (n=me@adsl-66-142-188-62.dsl.rcsntx.swbell.net) |
| 21:38.57 | paolob-parroquia | eKo1, the only significant thing I see is a "Call-ID: 3e3dc6d72d4f0c9a66a5de2d6f7eca09@10.152.58.9" line |
| 21:39.08 | paolob-parroquia | why does it pass that weird string? |
| 21:39.48 | eKo1 | That is an ID used to identify the call. It has nothing to do with CID. |
| 21:40.09 | eKo1 | Since I'm not an expert in SIP, I can't tell you what to look for. You're going to have to read up on it. |
| 21:40.31 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 21:41.15 | jayma777 | Hello All... |
| 21:41.20 | c4t3l | howdy |
| 21:41.22 | jayma777 | Need some best practice info |
| 21:41.29 | c4t3l | shoot |
| 21:41.40 | jayma777 | I've walked into a bad situation... and trying to make it better |
| 21:41.52 | jayma777 | Client has 11 analog lines (all separate) |
| 21:41.59 | jayma777 | wants to combine them.. |
| 21:42.11 | TommyTheKid | is there any sizing information for conferencing with meetme (or something else?) |
| 21:42.31 | jayma777 | no |
| 21:42.49 | jayma777 | I'm more concerned on the Hardware side... |
| 21:42.58 | jayma777 | How to get the lines into the box... |
| 21:43.02 | TommyTheKid | jayma777: are they willing to go digital, I think 10-12 analog lines was about the breaking point to moving to a T1 at one of my customers |
| 21:43.09 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 21:43.11 | jayma777 | these are all (currently) rj11 |
| 21:43.33 | jayma777 | well... the problem with that is... currently he has all you can eat long distance one price... |
| 21:43.47 | jayma777 | I can't seem to find that from any vendor in a T1 config |
| 21:43.49 | jayma777 | so.. |
| 21:44.03 | jayma777 | i'd like to combine what he has into an asterisk box... |
| 21:44.09 | jayma777 | and have it manage the lines.. |
| 21:44.21 | jayma777 | now... is that possible? |
| 21:44.36 | eKo1 | Yes |
| 21:44.37 | twisted[asteria] | get a channel bank with fxo ports |
| 21:44.44 | twisted[asteria] | and a t1 card. |
| 21:44.44 | TommyTheKid | yeah, that |
| 21:44.56 | jayma777 | that's what I was thinking |
| 21:45.00 | TommyTheKid | then they have the option of moving to a digital line in the future too :) |
| 21:45.07 | twisted[asteria] | yepyep |
| 21:45.08 | eKo1 | or an 11 port fxo gateway |
| 21:45.13 | twisted[asteria] | EW |
| 21:45.30 | jayma777 | very kewl... |
| 21:45.38 | jayma777 | just needed a sanity check... |
| 21:45.47 | jayma777 | I'll probably be back once I've got the HW |
| 21:45.48 | dhill | re/quit |
| 21:45.56 | jayma777 | thanks guys |
| 21:49.14 | carl0s- | I find this most frustrating. About 50% of my calls are being connected/answered/forwarded properly. The rest of the time the line just rings. My outbound gateway just doesn't seem to respond to Asterisk's SIP INVITE. Then I try again and it still doesn't work, then I leave it a bit and try again and it works fine. |
| 21:49.45 | carl0s- | Hardly any use in a 'production' environmeny |
| 21:49.48 | carl0s- | environment |
| 21:50.15 | eKo1 | outbound fxo gateway? |
| 21:50.39 | carl0s- | nah, outbound GSM gateway. |
| 21:52.56 | *** part/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 21:53.21 | *** part/#asterisk mitcheloc (n=mitchelo@020.131-62-69.ftth.swbr.surewest.net) |
| 21:58.30 | CunningPike | An Astribank might work for jayma777, too...... |
| 22:01.10 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:06.45 | *** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca) |
| 22:08.38 | *** join/#asterisk eBody_ (n=ehernand@207.71.51.162) |
| 22:08.56 | eBody_ | anyone know about cleaning up static using a tdm2400 card?? |
| 22:10.26 | eBody_ | would adjusting the txgain and rxgain in zapata.conf cause static across POTS lines? |
| 22:12.42 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:13.29 | carl0s- | I'm still seeing " Init: INVITE" under sip show channels for the gateway. 1 in 3 calls are connecting :( |
| 22:14.13 | carl0s- | and after I cancel that call, I can see it changes to "Rx: REGISTER". Then I try again and it works. |
| 22:14.58 | carl0s- | and after I cancel that call, I can see it changes to "Rx: REGISTER". Then I try again and it works. |
| 22:15.11 | carl0s- | oops. thought I was in the Asterisk CLI there.. repeating my command. |
| 22:15.27 | jbalcomb | Anyone have a copy of the Watchguard VPN Manager? |
| 22:16.52 | Nugget | "a copy"? |
| 22:20.10 | eBody_ | anyone know why caller id info wouldn't be coming through the POTS lines? |
| 22:20.48 | carl0s- | eBody_: not enabled with the telco, card which doesn't support type of CID used. Configuration files.. where are you located? |
| 22:21.03 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 22:21.29 | eBody_ | texas |
| 22:21.32 | eBody_ | we have caller id |
| 22:22.02 | carl0s- | hmm. you've got the usecallerid=yes set ? |
| 22:22.07 | eBody_ | yes :) |
| 22:22.11 | carl0s- | :D |
| 22:22.14 | carl0s- | dunno then :) |
| 22:22.27 | eBody_ | in zapata.conf right? |
| 22:22.27 | mocker | Are there any decent iax2 softphones? |
| 22:22.33 | carl0s- | you can tell the zaptel driver which type of caller id you have.. might be worth looking at that |
| 22:22.46 | eBody_ | oh yeah, where? in a config? |
| 22:23.16 | carl0s- | hmm. dunno. 1 sec |
| 22:23.41 | *** join/#asterisk crochat (i=crochat@84-74-151-250.dclient.hispeed.ch) |
| 22:24.07 | Nugget | I'm a symlink! |
| 22:24.33 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) |
| 22:24.53 | carl0s- | eBody_: well, it's cidsignalling= but I'm not sure if that's only specific to a particular patch which was written for the UK CID system. |
| 22:24.55 | jarrod | hmm... when matching CDR against International billing whats the best method since there is no standard on country code length? |
| 22:25.26 | eBody_ | in zapata.conf again? |
| 22:26.08 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 22:26.15 | foo | I don't need an asterisk server online, do I? |
| 22:26.20 | foo | Err, I mean... |
| 22:26.32 | foo | I don't need it onsight, right? |
| 22:26.36 | carl0s- | eBody_: yeah. have a look at http://wiki.sangoma.com/ast-original-zapata for some of the parameters |
| 22:26.59 | carl0s- | eBody_: to be honest, I would have thought the defaults would be correct for you, being in America and whatnot. |
| 22:28.17 | *** part/#asterisk viler (i=1000@200.114.70.228) |
| 22:29.21 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:29.33 | *** join/#asterisk mxmasster (n=Max@adsl-71-130-205-20.dsl.irvnca.pacbell.net) |
| 22:29.51 | *** join/#asterisk mivck (i=1000@200.114.70.228) |
| 22:30.14 | mxmasster | hi all |
| 22:31.26 | mxmasster | i want to configure a queue to use for a follow/find me feature, i want to queue to simultaneously ring x phone numbers that i will configure, i do not want the queue to ever remove an agent from the queue, and when one of the numbers picks up i would like the person to be prompted to press a digit to complete the call |
| 22:31.35 | mxmasster | is there a good howto/example for this? |
| 22:32.33 | eBody_ | carl0s-, thanks!! |
| 22:34.26 | eBody_ | what does DTMF stand for? |
| 22:35.19 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 22:36.19 | carl0s- | erm |
| 22:36.24 | carl0s- | dual tone muti frequency I think |
| 22:36.35 | *** join/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 22:36.41 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:36.46 | *** part/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 22:36.47 | carl0s- | it means tones anyway.. as in tone-dialling. the tones are made up of two frequencies played simultaneously |
| 22:37.28 | eBody_ | is there anyway to overcome the two different technologies, analog and sip?? |
| 22:37.39 | eBody_ | i need to send DTMF info via sip for our internal voice mail |
| 22:38.42 | *** join/#asterisk mitcheloc (n=mitchelo@020.131-62-69.ftth.swbr.surewest.net) |
| 22:39.30 | carl0s- | I don't understand exactly. You can still send tones over a SIP connection. If you're on your IP Phone.. say Snom/Cisco/Grandstream, and chatting to someone, or you've called up an automated system, you can still 'press 1 for billing' or 'press 2 for voicemail' and 'enter your voicemail password' by pressing the keys, thereby sending the DTMF tones, over the SIP link. |
| 22:39.36 | *** part/#asterisk mitcheloc (n=mitchelo@020.131-62-69.ftth.swbr.surewest.net) |
| 22:40.12 | eBody_ | hmmm.......... |
| 22:40.20 | eBody_ | lemme try again then. i wasn't working via SIP info |
| 22:40.37 | Qwell[] | eBody_: just make sure both devices match for the dtmf mode |
| 22:40.42 | carl0s- | just not sure exactly what you're getting at.. what you're trying to do. |
| 22:41.47 | eBody_ | i have these SIP extensions |
| 22:42.07 | carl0s- | as in.. phones? |
| 22:42.12 | eBody_ | in the phones config i set it to send DTMF via SIP info or else we can't interact with our voice mail system |
| 22:42.14 | eBody_ | yes. |
| 22:42.35 | eBody_ | but when we call an external automated system the buttons won't work |
| 22:42.54 | carl0s- | so when you call your voicemail system from the SIP phone, you're getting the voicemail system saying "press this" and "press that", but the tones aren't getting through when you press the keys? |
| 22:43.18 | eBody_ | if i have it set to send DTMF via SIP info it will work. |
| 22:43.28 | carl0s- | oh right. |
| 22:43.37 | eBody_ | but with that same setting external automated attendents won't work. |
| 22:43.51 | carl0s- | that's beyond my knowledge already then 'cause I didn't realise you could do that :D I'm new. |
| 22:44.14 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:44.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:44.38 | carl0s- | So you mean the SIP phone is in effect telling the asterisk box to generate the tones for you? rather than them being generated by the handset itself. |
| 22:44.58 | eBody_ | no these setting i'm talking about are on the phone itself. |
| 22:45.05 | eBody_ | Grandstream gpx-2000 |
| 22:48.30 | carl0s- | "DTMF via SIP INFO: Some IVR systems may need DTMF events signalled via SIP INFO messages, this can be enabled here." |
| 22:48.34 | carl0s- | cool. |
| 22:48.45 | carl0s- | so what's your internal voicemail system then - the one that works? |
| 22:50.56 | eBody_ | yeah |
| 22:51.03 | carl0s- | "Users of Grandstream phones should use Info for the Asterisk voicemail; the other methods won't work." |
| 22:51.04 | *** join/#asterisk rene- (n=rene-@gea-gye-internet.telconet.net) |
| 22:51.12 | eBody_ | lol |
| 22:51.27 | eBody_ | but it's the other systems, the older ones, that now won't work using SIP |
| 22:51.28 | eBody_ | lol |
| 22:51.43 | eBody_ | so if i call my bank who is using an older analog system. the buttons won't work |
| 22:51.45 | eBody_ | i'm stuck. |
| 22:51.58 | rene- | what sort of variables does he AMI originate action sets? i mean how do i retrieve them? |
| 22:52.00 | carl0s- | Can you change to "Inband"? I suspect 'Inband' is what you want for normal tones, e.g. what you're trying to acheive. But the information i've just read says that you won't be able to use Asterisk voicemail then, as you must use INFO mode. |
| 22:52.57 | eBody_ | where? can this be sent? |
| 22:52.59 | eBody_ | set^ |
| 22:53.24 | eBody_ | dang, so it's really one or the other!? |
| 22:53.32 | carl0s- | within the Grandstreams configuration. It looks though like it's either one or the other. Unless of course you could get Asterisk to generate the actual tones on your behalf, when it receives the INFO messages. |
| 22:53.41 | carl0s- | Surely that must be possible |
| 22:53.57 | eBody_ | where that's where i'm at. |
| 22:54.42 | *** join/#asterisk crochat (i=crochat@84-74-151-250.dclient.hispeed.ch) |
| 22:56.46 | carl0s- | I wonder if DTMF Relay is the term here. Where are all the experts when you need them :) |
| 22:57.12 | file | what's up and what'cha need? |
| 22:57.15 | carl0s- | :D |
| 22:57.54 | carl0s- | this chap has a Grandstream phone which only works with Asterisk voicemail with DTMF mode set to "via SIP INFO messages". But then the key-tones aren't being sent when he calls an external automated system, like his bank. |
| 22:58.27 | eBody_ | carl0s-, i'm soo sorry. my boss in an idiot and was giving me wrong info |
| 22:58.33 | eBody_ | everything works via SIP info :) |
| 22:58.38 | carl0s- | d'oh |
| 22:58.40 | carl0s- | :D |
| 22:58.49 | rene- | bosses being idiots where have i heard about that |
| 22:58.50 | carl0s- | then everything's good :D |
| 22:58.54 | eBody_ | ur good though. |
| 22:59.01 | Qwell[] | it automatically passed DTMF through to the other channel |
| 22:59.10 | Qwell[] | it "just works" |
| 22:59.13 | eBody_ | now i'm trying to figure out these custom ring tones on this GXP-2000 |
| 22:59.14 | eBody_ | lol |
| 22:59.15 | Qwell[] | s/passed/passes/ |
| 22:59.37 | carl0s- | Qwell[]: thats cool. I didn't even know DTMF was sent as 'info'. I thought the tones were just generated by the phone ('inband'?) up until 2 mins ago. |
| 22:59.52 | Qwell[] | there is inband, info, and rfc2833(?) |
| 22:59.55 | file | SIP has... a few ways |
| 23:00.00 | carl0s- | neat |
| 23:00.03 | file | there's 5 I know of |
| 23:00.09 | Qwell[] | 5? |
| 23:00.20 | file | inband, info, rfc2833, notify, and an XML encoded one |
| 23:00.22 | Qwell[] | besides b0rked cisco non-rfc dtmf? |
| 23:00.51 | carl0s- | You'd think Cisco should get things fairly right wouldn't you.. |
| 23:00.52 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 23:02.27 | rene- | if iam originating with AMI and i set some variables where do they go? and how can i retrieve them again using the manager interfase |
| 23:02.42 | carl0s- | I wonder if increasing the SIP registration expiration from 5mins to 24hrs will help with my gateway not accepting 60% of the calls that Asterisk tries to put through it. |
| 23:02.52 | brockj49464_home | OK I have a wildcard z101p in red alarm, what can I do to fix it? |
| 23:03.07 | carl0s- | brockj49464_home: I think that means no line detected. |
| 23:03.18 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-69-129.dsl.peoril.sbcglobal.net) |
| 23:03.20 | rene- | and also i need to set a fixed callerid in my originate events otherwise my PSTN provider (E1) wont take the call that is shitty also |
| 23:03.45 | carl0s- | brockj49464_home: X101P though, yeah? |
| 23:05.23 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 23:07.14 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 23:08.09 | brockj49464_home | carl0s: thx. |
| 23:08.11 | MACscr | anyone here use the grandstream gxp 2000? |
| 23:08.19 | eBody_ | yup :) |
| 23:08.33 | eBody_ | MACscr, i'm working with some right now. |
| 23:08.38 | eBody_ | maybe we can help each other. |
| 23:08.44 | knarfly | MACscr: BT-200 here |
| 23:08.45 | MACscr | i just got mine and i upgraded the firmware, but when i click on advanced settings, i get an error |
| 23:08.48 | MACscr | Access Denied Content-Type: text/html;charset=iso-8859-1 Server: Grandstream/1.10 |
| 23:09.14 | MACscr | i got the same error before the update as well |
| 23:09.15 | carl0s- | ooh. I just found forums.digium.com . Is this /the/ place to start asking my newbie questions? Or are there busier/better forums? |
| 23:09.42 | MACscr | i have a remote asterisk box that i need to figure out how to connect to as well =P |
| 23:09.45 | dan42 | anyone heard of voicemail odbc storage messing up and creating HUGE files (2G+)? it keeps taking my system down |
| 23:09.55 | carl0s- | MACscr: ssh? |
| 23:10.05 | Qwell[] | dan42: using an x100p? |
| 23:10.07 | MACscr | no, i mean with my phone |
| 23:10.10 | carl0s- | oh :) |
| 23:10.13 | MACscr | =P |
| 23:10.17 | knarfly | carl0s-: go ahead...we'll either help you or flame you... |
| 23:10.24 | carl0s- | tee hee :D |
| 23:10.29 | dan42 | Qwell[]: not sure what an x100p is |
| 23:10.39 | Qwell[] | analog single port, generic POS fxo... |
| 23:10.41 | rene- | does anybody know? about the little variables i set with originate where did they go? |
| 23:10.48 | eBody_ | dang MACscr my phone at least works :) srry |
| 23:10.59 | MACscr | lol, lucky. mine is ganked up |
| 23:11.19 | MACscr | i got it to sync as far as time with my asterisk box though =P |
| 23:11.40 | rene- | i am sending an Variable: var1=something |
| 23:11.40 | rene- | <PROTECTED> |
| 23:11.56 | dan42 | Qwell[]: no.. using asterisk voicemail app.. and sip.. |
| 23:12.08 | Qwell[] | dan42: sip provider? |
| 23:12.09 | dan42 | with odbc message storage |
| 23:12.52 | dan42 | Qwell[]: can you clue me in on what this has to do with voicemail app creating 2g+ files in peoples directories? |
| 23:12.54 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:12.54 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:13.12 | Qwell[] | how are calls getting into app_voicemail? |
| 23:13.57 | dan42 | Qwell[]: sip from home user devices direct to us or incoming calls through any number of sip providers |
| 23:14.10 | Qwell[] | okay, so no analog? |
| 23:14.16 | dan42 | nope |
| 23:14.18 | dan42 | all sip |
| 23:14.27 | Qwell[] | What version of *, and do you have any extra patches applied? |
| 23:15.08 | Qwell[] | and are you doing anything silly, like having a conference call voicemail? |
| 23:16.09 | dan42 | 1.2.10 .. only patch was from 7416 |
| 23:16.36 | MACscr | lol, got into advanced config |
| 23:16.37 | dan42 | its doing odbc message storage and realtime |
| 23:17.05 | carl0s- | MACscr: what did you have to do? Put a password in? :) |
| 23:17.05 | Qwell[] | I have a feeling your patch b0rked it :p |
| 23:17.07 | knarfly | Anyone running FreeBSD w/* 1.2.10.1? |
| 23:17.21 | MACscr | lol, how was i supposed to know their were two different passwords |
| 23:17.23 | MACscr | =P |
| 23:17.23 | dan42 | Qwell[]: did you look at the patch? |
| 23:17.28 | Qwell[] | no |
| 23:17.32 | carl0s- | MACscr: lol :D |
| 23:17.48 | Qwell[] | Does it happen without that patch? |
| 23:17.54 | dan42 | Qwell[]: it adds a single non-functional fstat |
| 23:17.57 | eBody_ | what would be the best vocoder: PCMU, PCMA, G.723.1, G.729A/B, or GSM |
| 23:18.02 | dan42 | it doesnt work without |
| 23:18.44 | Qwell[] | dan42: not even if you don't hangup during vm-intro? |
| 23:19.00 | Qwell[] | or, unavail rather |
| 23:19.37 | dan42 | Qwell[]: theres no way to test that since the system doesnt wrok without the patch |
| 23:19.37 | MACscr | So Carlos, am i supposed to add my asterisk server info to one of the Accounts pages? |
| 23:19.56 | Qwell[] | dan42: okay, remove the unavail file. Does voicemail with odbc storage work, or no? |
| 23:20.06 | dan42 | Qwell[]: its happened 3 times over 24 hrs that these huge files have filled the drive |
| 23:20.25 | dan42 | Qwell[]: then all of a sudden the drive is full and i need to kill asterisk |
| 23:21.40 | Qwell[] | either you're giving me wrong info now, or your bug report has wrong info... |
| 23:22.20 | dan42 | Qwell[]: you lost me |
| 23:22.42 | dan42 | Qwell[]: you asked me what patches i had, i told you.. that bug report is not about the problem im having now |
| 23:22.44 | Qwell[] | remove your patch. remove the u when dialing voicemail. Does odbc work? |
| 23:22.56 | carl0s- | MACscr: Never used any phone except the Csico 7960 I have here, but yeah. You need to add probably: username/register name, password, sip proxy [ast-ip] |
| 23:23.40 | dan42 | Qwell[]: no, youre not listening.. in 24hrs its created 2G+ files 3 times.. otherwise, it works.. thousands of people checking their voicemail and such.. then all of a sudden the drive is full and theres a 2G+ file someplace |
| 23:24.09 | dan42 | Qwell[]: i cant remove the patch and see if it happens because then the system wont work most of the time |
| 23:24.11 | MACscr | ok, im going to have to take a nap first |
| 23:24.57 | Qwell[] | Here are the facts, as they stand. |
| 23:25.05 | Qwell[] | Unless you can test with a completely untested patch, we cannot help you. |
| 23:25.11 | Qwell[] | erm, unpatch system :p |
| 23:25.34 | dan42 | Qwell[]: thats bullshit because no one would help me figure out the problem that caused me to do the stupid patch |
| 23:25.49 | dan42 | Qwell[]: cant have it both ways |
| 23:26.12 | dan42 | Qwell[]: did you look at the patch? its a single useless fstat that doesnt mean anything |
| 23:26.36 | Qwell[] | No, I cannot look at that patch, unless you have a disclaimer on file |
| 23:26.40 | *** join/#asterisk _Tristan_ (n=Tristan@A.Condom.Made.My.Firewall.Pr0tected.us) |
| 23:26.55 | Qwell[] | ahh, faxed already |
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| 23:27.59 | dan42 | Qwell[] |
| 23:28.42 | dan42 | clearly my patch isnt a real patch.. and does nothing to fix the real problem.. but since i couldnt get any assistance anyplace and it made the proble go away, its the best that i could do.. i spent hours getting that far |
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| 23:29.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:30.02 | Qwell[] | dan42: /j #asterisk-bugs |
| 23:30.16 | Bullseye_Network | it seems the memory problems (leaks) have brought asterisk to a hault. Down to 260m free. |
| 23:31.21 | dan42 | Qwell[]: i spent days in -dev and -bugs trying to get help.. posted to the list.. the best i could get was "learn to be a better programmer" |
| 23:31.58 | Bullseye_Network | The system is dialing numbers but never connects them to the phones |
| 23:32.11 | dan42 | Qwell[]: pretty much the same thing that happened with 7552 |
| 23:35.11 | dan42 | Qwell[]: you know what.. im sorry man.. im just stressing on you.. my system is exploding on me and i cant seem to get any help with anything ever.. and im not a good enough programmer to figure it out myself.. sorry about that |
| 23:35.39 | Qwell[] | I've given you steps you can take to test. |
| 23:36.19 | tlow | Call digium if you are desperate, they can troubleshoot almost anything for a small fee. |
| 23:36.21 | Qwell[] | If you have to work around the other problem some other way, you'll need to do that, but, unless you have an unpatched system, it's VERY hard to test |
| 23:36.23 | dan42 | Qwell[]: youve given me an impossible suggestion which only frustrated me more |
| 23:36.30 | Qwell[] | dan42: remove the u option from voicemail |
| 23:36.49 | Qwell[] | from reading the bug, that would be a valid workaround |
| 23:37.27 | dan42 | removing the u would be a valid workaround to it creating random 2G+ files? |
| 23:37.42 | Qwell[] | No, to it dying when hanging up while playing the unavail sound |
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| 23:38.40 | dan42 | Qwell[]: which would then make all my customers unavailable messages stop working.. creating another nightmare |
| 23:38.40 | Qwell[] | dan42: weigh your options |
| 23:38.52 | Qwell[] | customers mad because a feature is missing, or things breaking completely? |
| 23:39.29 | dan42 | Qwell[]: so you figure my fstat is causing the 2G+ files? |
| 23:39.37 | Qwell[] | maybe, yes |
| 23:46.11 | fgwaller | Does anyone here have a clue if I can override the hangup of a ZAP channel, so that the call is not going to be hung up when some Agent accidentially hangsup his phone? |
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