| 00:03.20 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 00:03.59 | Druken | anyone get an unsolicited email from some uno communications out of ny, ny ? |
| 00:04.36 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:05.05 | *** join/#asterisk Pageus (n=FreePBX0@ip70-190-19-6.ph.ph.cox.net) |
| 00:05.11 | r_evolution | the spammers have you in their grasp Druken. |
| 00:05.33 | Druken | pfft, no shit |
| 00:05.38 | r_evolution | fux. :( |
| 00:05.41 | Pageus | evening all |
| 00:05.46 | Druken | i get so much god damn spam it's not funny |
| 00:06.18 | r_evolution | time for a new e-mail address |
| 00:06.20 | RES2 | SpamAssassin is your friend. |
| 00:06.27 | r_evolution | and if you stop browsing all those porn sites... |
| 00:06.33 | r_evolution | you might get a little less :) |
| 00:07.02 | *** join/#asterisk shaynes (n=shayne@chic01-104.221.digitalpath.net) |
| 00:07.05 | shaynes | Hello again! |
| 00:07.18 | r_evolution | such an exuberant fellow |
| 00:07.18 | Druken | uhmm... sorry, have never used my business email on any pron sites.... |
| 00:07.22 | Druken | :) |
| 00:07.27 | shaynes | What is the variable for the CID number of an incoming call to be used in extensions.conf? |
| 00:07.30 | Druken | i have a seperate account for those :) hehe |
| 00:07.30 | ManxPower | We finished our first cable run in the underground conduit today. All is good. |
| 00:07.48 | ManxPower | shaynes, README.variables in /path/to/src/asterisk/doc |
| 00:07.54 | shaynes | ManxPower: Thanks! |
| 00:08.09 | shaynes | ManxPower: I am setting up a TTS engine that will BLOW Festival out of the water. |
| 00:08.22 | ManxPower | shaynes, Cepstral. |
| 00:08.41 | Druken | i don't think it's called cepstral anymore... no ? |
| 00:09.01 | ManxPower | Druken, the binary file is called something different. |
| 00:10.00 | shaynes | ManxPower: No. It uses the AT&T labs very impressive TTS engine to generate a very human like voice. I have written code to that will take text input, query AT&Ts servers, receive the TTS WAV, reformat it for Asterisk and send it back to Asterisk for playback. -- Right now it will take a zip code and return weather forecasts in realtime. |
| 00:10.00 | Druken | i thought it was purchased or something like that... and the name changed... |
| 00:10.26 | ManxPower | shaynes, Ah, yes that. Totally against the usage agreement, of course. |
| 00:10.33 | ManxPower | Druken, maybe it has. |
| 00:10.39 | rpm | can i bridge two channels together through the AMI interface? |
| 00:10.50 | shaynes | ManxPower: Not for personal and non-commercial use. |
| 00:10.57 | ManxPower | rpm, See Flash Operator Panel |
| 00:11.08 | Druken | why can i see that taking a while... |
| 00:11.19 | ManxPower | shaynes, where does it say that? |
| 00:11.30 | *** part/#asterisk RES2 (n=RES@chello213047231029.tirol.surfer.at) |
| 00:12.10 | shaynes | ManxPower: On their website: at the bottom: http://public.research.att.com/~ttsweb/tts/demo.php |
| 00:12.24 | ManxPower | One of my arch nemesis is going to be written up by the MIS director. Today is a good day. |
| 00:13.11 | r_evolution | haha... Arch Nemesis Manx? |
| 00:13.45 | blitzrage | ManxPower: lol |
| 00:14.16 | ManxPower | shaynes, you didn't read the FULL list of restrictions. "Direct access to the CGI scripts is not permitted." "Building or prototyping a software package using our audio rather than recording your own recorded prompts is not OK. " See: http://public.research.att.com/~ttsweb/tts/faq.php#WebPolicy |
| 00:14.37 | r_evolution | yikes! |
| 00:14.37 | shaynes | Maxxed: I did read it. that's the thing |
| 00:14.40 | Aurs | .......May 9 00:32:50 WARNING[4085]: loader.c:325 __load_resource: libmysqlclient.so.15: cannot open shared object file: No such file or directory |
| 00:14.45 | r_evolution | He just didn't care!! |
| 00:14.45 | shaynes | ManxPower: I did read it, that's the thing. |
| 00:14.53 | r_evolution | shaynes says... FUCK YOUR LAWS! |
| 00:15.04 | shaynes | r_evolution: to a point, yes1 |
| 00:15.08 | r_evolution | laws/rules/etc |
| 00:15.17 | r_evolution | yeah i used to say much the same... |
| 00:15.24 | r_evolution | it stopped working when i got arrested :-\ |
| 00:15.32 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:15.34 | shaynes | ManxPower: It's for personal use and technically I am not using their CGI script directly. --- software package? -- No, it's a script. I am sure I will sleep at night. |
| 00:15.51 | shaynes | r_evolution: I am sure I will not get arrested for this. |
| 00:16.04 | shaynes | r_evolution: ...and as I said "to a point." |
| 00:16.12 | TheFeds | You're nicked, shaynes |
| 00:16.16 | r_evolution | nah... i doubt you will either... so long as you're not doing it on a large level where you're making a lot of money |
| 00:16.17 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-243.apple.com) |
| 00:16.20 | TheFeds | stealing wav files! |
| 00:16.21 | shaynes | TheFeds: Damn ... damn you all! |
| 00:16.23 | ManxPower | blitzrage, this "person" did three factory resets on the new Polycom phone. The MIS director goes there, does one factory reset and the phone starts working. This is also the same person that placed signs on the cableing in the NOC, pulling out the network cable from the router. Then blames me for both problems. |
| 00:16.23 | TheFeds | You're going DOWN! |
| 00:17.00 | blitzrage | ManxPower: heh -- people are stupd |
| 00:17.15 | shaynes | blitzrage: Are you referring to me? |
| 00:17.25 | r_evolution | nah shaynes... read manx |
| 00:17.36 | blitzrage | shaynes: well... not originally I didn't |
| 00:17.42 | shaynes | blitzrage: ha ha |
| 00:17.48 | shaynes | blitzrage: supid, no. blind, yes. |
| 00:18.08 | blitzrage | heh |
| 00:18.11 | blitzrage | I do that too :) |
| 00:18.14 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 00:18.14 | ManxPower | blitzrage, this is the same person that can't seem to do transfers on polycom phones with more than a %50 success rate. |
| 00:18.23 | blitzrage | lol |
| 00:18.32 | blitzrage | ok -- playoff hockey time! |
| 00:18.34 | fjean | hello all, how r u tonight |
| 00:19.32 | Druken | ManxPower: some people just can't use certain things.... |
| 00:19.36 | r_evolution | we r great when u r n0t TyPiNg LiEk that |
| 00:19.45 | r_evolution | like keyboards Druken? |
| 00:19.54 | fjean | I was wondering, anyone was able to authenticate an asterisk SIP account using SER ? I need some help doing that.. |
| 00:19.58 | Druken | r_evolution: perhaps... |
| 00:20.21 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 00:21.00 | *** join/#asterisk mugawump (n=bbentley@adsl-068-209-173-175.sip.int.bellsouth.net) |
| 00:21.41 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 00:21.45 | r_evolution | i need to learn how to use my bed :( |
| 00:22.00 | Druken | my bed got good use today.... hehehe |
| 00:22.07 | MooingLemur | only if the bed supports IAX |
| 00:22.14 | gandhijee | would it be best to make the kernel non-preemptive for asterisk? or does it not care? |
| 00:22.43 | MooingLemur | I don't think it'd care that much. Maybe it'd make a difference with dozens or hundreds of calls |
| 00:22.52 | Druken | gandhijee: don't want to have premature disconnections? :) |
| 00:23.23 | gandhijee | well this is my first "big" box |
| 00:23.35 | gandhijee | i've been doing like 20 calls with a standard preemptive kernel |
| 00:24.10 | gandhijee | but was wondering if i should turn it off, the "big" box is hopefully gonna handle around 60 or 70 calls hopefully |
| 00:24.30 | ManxPower | gandhijee, no, Big Boxes handle 200 - 300 calls at one time. |
| 00:24.40 | X-Rob | that's a little box |
| 00:24.41 | docelm0 | haha.. my boxes do 300+ |
| 00:24.42 | X-Rob | well |
| 00:24.45 | X-Rob | medium box |
| 00:24.48 | docelm0 | thats my office PBX |
| 00:24.49 | docelm0 | :) |
| 00:25.12 | gandhijee | well its bigger for me |
| 00:25.17 | r_evolution | what do you define as big box docel? |
| 00:25.43 | docelm0 | qwells box that does 2.5k |
| 00:25.46 | r_evolution | because im trying to figure out exactly how far I can push the ones I have here before something explodes |
| 00:25.56 | gandhijee | X-Rob: you have preemptive kernel or not? |
| 00:26.06 | X-Rob | just leave it all standard. |
| 00:26.09 | r_evolution | i mean spec wise... |
| 00:26.14 | gandhijee | ok |
| 00:26.15 | docelm0 | Im running dual Xeon's w/ 2GB and running 600 easy |
| 00:26.22 | docelm0 | then this 255 issuse poped up |
| 00:26.24 | r_evolution | clock speed? |
| 00:26.26 | gandhijee | wow. my box is prolly over kill then |
| 00:26.28 | docelm0 | 800 |
| 00:26.36 | gandhijee | for what i am using it for |
| 00:26.36 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 00:26.42 | r_evolution | k... whats the back end clock? |
| 00:26.49 | docelm0 | thats 600 w/ transcoding.. :) |
| 00:26.54 | r_evolution | im assuming you mean fsb = 800 |
| 00:26.56 | docelm0 | and its pushing just fine |
| 00:26.57 | docelm0 | yes |
| 00:27.03 | docelm0 | not sure the back side.. |
| 00:27.14 | docelm0 | I just call dell and say I want server bla and they send it |
| 00:27.21 | r_evolution | cool :) |
| 00:27.51 | Druken | docelm0: tell them you want server blah, delivered to my address, billed to yours |
| 00:27.54 | docelm0 | my one office I am running Dual Core Xeon w/ 4GB and tested with sipp to 1100 calls |
| 00:27.54 | r_evolution | the server i'm using for the customer platform is a dual 3.2 xeon 800fsb 4GB RAM dual 160GB scsi |
| 00:28.14 | r_evolution | and they've be pestering the shit out of me for the capacity |
| 00:28.15 | docelm0 | Im running SATA II |
| 00:28.28 | r_evolution | i say... push it till something explodes |
| 00:28.43 | docelm0 | You can probably get 600 easy |
| 00:29.03 | shaynes | quit |
| 00:29.19 | gandhijee | how many calls do you guys think a Pentium D 3.0 w/ 2 gigs of RAM and a SATA drive can push? |
| 00:31.33 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 00:31.33 | docelm0 | 150 maybe.. |
| 00:31.46 | docelm0 | Transcoing? 80's maybe depending on codec used |
| 00:32.24 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 00:42.17 | tzanger | ManxPower: where's your stdexten macro these days? |
| 00:42.23 | tzanger | fnords.org/~eric's blank |
| 00:46.06 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:52.04 | *** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 00:54.56 | *** join/#asterisk juice (n=juice@mo-71-50-26-181.dhcp.sprint-hsd.net) |
| 00:55.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:56.56 | ManxPower | tzanger, in my dialplan 8-) |
| 00:57.01 | ManxPower | tzanger, want a copy? |
| 00:59.47 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 01:00.35 | Druken | anyone got copies of regenesis episodes? perhaps from a tivo ? |
| 01:03.06 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 01:03.59 | tzanger | ManxPower: yes, please |
| 01:04.12 | ManxPower | tzanger, standby .. be about 10 mins. |
| 01:04.41 | tzanger | ok |
| 01:05.22 | ManxPower | tzanger, by tomorrow I should be routing SSH over the PSTN modem instead of DirecWay |
| 01:05.48 | docelm0 | WOO! |
| 01:05.51 | docelm0 | or something |
| 01:10.32 | rpm | there has got to be a way to take a call off hold after you have transferred it to a hold extension.. is there anyway to re-bridge calls via ami? |
| 01:10.41 | tzanger | ManxPower: nice. that'll take your latency down quite a bit I imagine |
| 01:11.03 | ManxPower | tzanger, reduce it by at least 2/3 |
| 01:12.01 | *** join/#asterisk loud (i=ariel@cypher.punk.net) |
| 01:13.37 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-243.apple.com) |
| 01:15.14 | ManxPower | OK, everyone, here is the world famous ManxPower Macro(std-exten): http://www.fnords.org/~eric/std-exten.txt |
| 01:16.48 | *** join/#asterisk keyhack (n=keyhack@c-24-60-209-35.hsd1.ma.comcast.net) |
| 01:16.58 | keyhack | Anyone here used Java with Asterisk before? |
| 01:19.02 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 01:22.00 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-243.apple.com) |
| 01:22.37 | austinnichols101 | keyhack: yes, with the manager interface |
| 01:23.19 | *** join/#asterisk kruz123 (n=higsadin@69.73.127.92) |
| 01:23.35 | kruz123 | sup guys, are u all idolers? |
| 01:23.51 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-243.apple.com) |
| 01:24.06 | Nivex | i don't worship idols, I just idle. |
| 01:24.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:24.24 | kruz123 | lol sorry for sp |
| 01:24.27 | kruz123 | its been a long day. |
| 01:24.40 | kruz123 | any of you guys work at digium? like in huntsville? |
| 01:24.41 | Nivex | I'll forgive you... this time. :) |
| 01:24.49 | kruz123 | yay! thank you ;] |
| 01:26.01 | kruz123 | anyone in huntsville? where there based? |
| 01:26.14 | keyhack | austinnichols101: Did you use Asterisk-Java library? |
| 01:26.21 | kruz123 | i thought one of my friends that works there might idle in here but, maybe not |
| 01:26.52 | kruz123 | matt b, the software developer? |
| 01:28.09 | austinnichols101 | keyhack: yes |
| 01:28.32 | keyhack | austinnichols101: What was your opinion of it? I was going to use it in lieu of Microsoft Speech Server (and all that mess) |
| 01:28.44 | keyhack | austinnichols101: I need to do basic automated calling with user interaction/verification |
| 01:29.10 | austinnichols101 | keyhack: I didn't have any problems with it at all. Basically kept me from having to deal with the low-level ip communications between my app and the manager. |
| 01:29.32 | austinnichols101 | keyhack: I could have written that part myself, but why if there's already a library that works ok |
| 01:29.49 | syzygybsd | I am getting static on zap lines when there is more then 4 active calls, what reasons could there be? There are 0 irq misses, and CPU usage is only 10% |
| 01:30.02 | keyhack | austinnichols101: Yeah, seems like its kept up well, latest release was back in Nov/Dec 2005, and I could expand upon it if necessary. Kinda sucks, kinda wanted basic speech reco |
| 01:32.51 | austinnichols101 | keyhack: it's not really designed for anything like speech reco |
| 01:33.08 | keyhack | austinnichols101: Yeah, so I'll have to use DTMF reco instead |
| 01:33.20 | keyhack | austinnichols101: Maybe one day Asterisk will have built in speech reco |
| 01:35.01 | kruz123 | anyone here live in huntsville alabama?? |
| 01:35.10 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 01:37.39 | austinnichols101 | keyhack: I thought there was something for speech already... |
| 01:38.23 | keyhack | austinnichols101: What do you mean? |
| 01:38.47 | austinnichols101 | keyhack: sphinx |
| 01:39.03 | ManxPower | brain. hurt. cisco. dial demand routing. route maps. brain. hurt. |
| 01:39.21 | kruz123 | cisco=good for awesome |
| 01:39.29 | kruz123 | ccna=better for awesome |
| 01:39.37 | kruz123 | is CCNA worth pursueing do u guys think? |
| 01:40.10 | keyhack | austinnichols101: Yeah, but how do I get sphinx to work with Asterisk-Java? |
| 01:40.46 | austinnichols101 | keyhack: they're two separate things |
| 01:40.56 | keyhack | austinnichols101: Yeah... but I need them to exist as one |
| 01:41.03 | austinnichols101 | keyhack: not sure what you're trying to accomplish |
| 01:41.07 | keyhack | austinnichols101: I need to place VoIP calls, and have a sort of IVR system |
| 01:41.21 | keyhack | austinnichols101: Allow the user to either interact with my program via voice or via DTMF |
| 01:41.36 | austinnichols101 | keyhack: would be something more like sphinx + fastAGI out to your external app |
| 01:42.35 | keyhack | austinnichols101: fastAGI? |
| 01:42.50 | *** join/#asterisk JSabines (i=JSabines@dsl-201-129-80-49.prod-infinitum.com.mx) |
| 01:43.20 | austinnichols101 | keyhack: there are several solutions that let you call external applications from asterisk |
| 01:43.49 | austinnichols101 | so you do all of your voice detection using sphinx and then use one of those solutions when you need to go out for data based on the detected speech |
| 01:43.49 | keyhack | austinnichols101: Well, it looks like Asterisk-Java will register my Java app as an AGI on the server and call the method when a call is answered |
| 01:44.11 | kruz123 | hey keyhack and austin, dont mean to butt in, but whats the best programming lanuage to start with??? |
| 01:44.22 | austinnichols101 | kruz123: basic :) |
| 01:44.30 | kruz123 | im really enjoying watching this convo being a pbx'r myself :D |
| 01:44.32 | syzygybsd | how can I tell the model of a Zaptel card I have installed? |
| 01:44.38 | keyhack | kruz123, I started with QBasic |
| 01:45.00 | austinnichols101 | 20 GOTO 10 |
| 01:45.02 | kruz123 | hmm |
| 01:45.04 | keyhack | austinnichols101: I'm not sure how I could pass the audio and whatnot into Sphinx and have all the interaction happen from that, etc. |
| 01:45.16 | kruz123 | thanx austin, continue with keyhack, im liking this :D |
| 01:45.28 | austinnichols101 | k |
| 01:45.34 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 01:46.04 | austinnichols101 | keyhack: I still can't quite visualize the whole app but I think you're close. I haven't read through the sphinx stuff so I'm not sure how the integration works. |
| 01:46.20 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 01:46.44 | fjean | hi, anybody use SER to send calls to Asterisk here ? |
| 01:46.49 | keyhack | austinnichols101: I mean, on the basic level, I see that the Asterisk-Java AGI stuff lets me say words, phrases, play music, wait for certain key values, etc. Which is fine, but I'd eventually like to offer speech reco instead of basic DTMF reco, but I don't think the AGI can integrate into the AGI stuff |
| 01:48.46 | keyhack | I mean the AGI integrate with the Sphinx stuff |
| 01:48.55 | MooingLemur | ooh, sphinx |
| 01:49.01 | MooingLemur | I've been meaning to look at that |
| 01:49.50 | keyhack | austinnichols101: However, look at : http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+AGI&source=54 |
| 01:49.53 | MooingLemur | syzygybsd: lspci -v? |
| 01:50.03 | keyhack | austinnichols101: And I quote: "The EAGI extension will let you receive sound from the channel into your application. It will not let you send sound. EAGI is intended to allow you to write a script that passes sound to an external application - such as the Sphinx speech-to-text/speech recognition application (as the example script included with asterisk does). Your EAGI application must also listen for a text response. In the case of |
| 01:50.31 | austinnichols101 | keyhack: sounds like WAY too much fun |
| 01:50.38 | keyhack | austinnichols101: I'm trying to absorb it all |
| 01:50.46 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 01:50.51 | syzygybsd | thanks MooingLemur |
| 01:51.09 | keyhack | austinnichols101: It doesnt seem to have much more info other than that paragraph |
| 01:51.38 | austinnichols101 | keyhack: yeah, welcome to the bleeding edge :) |
| 01:53.24 | keyhack | austinnichols101: You running 1.2? I don't see the "included sample script" |
| 01:54.10 | keyhack | austinnichols101: http://www.voip-info.org/wiki/view/Sphinx |
| 01:54.55 | keyhack | austinnichols101: http://turnkey-solution.com/asterisk-sphinx.html |
| 01:55.06 | austinnichols101 | saw that one |
| 01:55.11 | austinnichols101 | looks rough |
| 01:56.06 | keyhack | yeah, overkill |
| 01:56.12 | keyhack | the Asterisk docs made it sound so easy |
| 01:56.13 | keyhack | lol |
| 01:56.19 | austinnichols101 | keyhack: /var/lib/asterisk/agi-bin/eagi-sphinx-test |
| 01:56.43 | austinnichols101 | and /usr/src/asterisk/agi/ |
| 01:56.48 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-243.apple.com) |
| 01:58.08 | Qwell | how many bytes are there (including overhead) in a 20ms ulaw frame? |
| 01:58.46 | austinnichols101 | a lot |
| 01:59.00 | keyhack | austinnichols101: Hmm, I'll have to look into this a lot more |
| 01:59.16 | austinnichols101 | keyhack: yes - sounds like it's a non-trivial task |
| 02:00.25 | keyhack | austinnichols101: Yeah, looks like you basically save the WAV file, and then call Sphinx to read it in and return the string |
| 02:00.45 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 02:02.36 | Zodiacal | anyone know why my sound card seems to stop working with other apps after someone uses the paging function in asterisk? e.g. "play filename.wav" stops working after i use the paging features of asterisk. a reboot lets "play" work again |
| 02:02.51 | Zodiacal | asterisk is setup to use oss, if i used alsa , would that fix it? |
| 02:03.12 | Zodiacal | or does asterisks CLI have a way to play a sound file? |
| 02:03.24 | Zodiacal | maybe with .call files or somthin |
| 02:03.46 | keyhack | austinnichols101: Well, I gotta get going, thanks for the heads up on some of the stuff, big help |
| 02:04.06 | *** join/#asterisk rushowr (n=none@cpe-24-210-49-134.columbus.res.rr.com) |
| 02:04.25 | rushowr | hello all! couple of questions for anyone interested in helpin' out |
| 02:04.29 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 02:04.33 | keyhack | austinnichols101: Quick question (I haven't read any of the Manager API stuff or anything yet), but is there a way for my manager code that places the original call to know if the call line was busy, unanswered, etc.? I need to be able to handle scenarios where the call was never answered or could not be completed as dialed |
| 02:06.09 | rushowr | anyone know how you could find out when a call starts ringing (an outbound call, currently SIP based) from within the dialplan? For CDR and provider performance recording purposes |
| 02:07.18 | zwelch | does anyone know what the status of the app_conference module is against 1.2? I am using gentoo, and it shows that module is incompatible with the 1.2 install; does that mean it's now in -addons or something else? |
| 02:07.27 | keyhack | austinnichols101: Alright, I gotta get going, thanks |
| 02:08.54 | rushowr | nobody? |
| 02:08.55 | keyhack | austinnichols101: Leave me a privmsg if you want, I'll be back in an hour or so |
| 02:09.01 | kruz123 | rushowe: hey |
| 02:09.06 | austinnichols101 | keyhac: no idea on your question |
| 02:09.12 | rushowr | kurz123 - yes? |
| 02:09.21 | rushowr | (austin) thx anyway |
| 02:09.35 | kruz123 | rushowr: let me look a lil bit here. |
| 02:09.40 | rushowr | k :) |
| 02:10.52 | *** join/#asterisk downunder33 (n=robert@219.95.168.191) |
| 02:11.47 | *** join/#asterisk NativePHP (n=ajb@c-69-249-119-114.hsd1.nj.comcast.net) |
| 02:11.49 | downunder33 | hi. Is anyone here using asterisk in combination with ldap? |
| 02:12.47 | downunder33 | interested to know under which scenario(s) you find ldap integration useful. thx. |
| 02:13.56 | *** part/#asterisk rushowr (n=none@cpe-24-210-49-134.columbus.res.rr.com) |
| 02:17.55 | fjean | hi hi, anybody using SER with asterisk ? |
| 02:20.34 | Hmmhesays | yes |
| 02:21.34 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 02:21.42 | fjean | hmmmhesays: hi, how a re you....you might be able to help me then...I am looking for a way to send calls from SER to asterisk (2 diff. boxes) in a "secure" way |
| 02:21.49 | SwK | lotta people use ser with asterisk then |
| 02:22.17 | Hmmhesays | unfortunately i'm headed out |
| 02:22.27 | Agrajag- | gday. im a noob and trying to setup SIP. i've edited my sip.conf and added a user similar to the example at the bottom of http://www.digium.com/en/docs/asterisk_handbook/sip.conf.html. when i start asterisk and do "sip show users" it doesn't display anything however. what am i missing? |
| 02:22.34 | fjean | hmmmmhesays: ok, no prob |
| 02:23.52 | *** join/#asterisk rushowr (n=none@cpe-24-210-49-134.columbus.res.rr.com) |
| 02:24.03 | fjean | from what I read we can't authenticate |
| 02:24.37 | rushowr | anyone here know how one might make queries to a postgresql database from the dialplan? |
| 02:24.50 | fjean | bu there must be a way to do it without leaving everything open, right ? |
| 02:26.45 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 02:27.26 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 02:28.44 | *** join/#asterisk kph100 (n=kph100@206-248-132-2.dsl.teksavvy.com) |
| 02:35.44 | *** part/#asterisk rushowr (n=none@cpe-24-210-49-134.columbus.res.rr.com) |
| 02:36.54 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 02:37.40 | ManxPower | I guess it would be good if my default route didn't go thru the tunnel |
| 02:39.44 | *** join/#asterisk ramo (n=ramo@59.92.130.217) |
| 02:45.23 | *** part/#asterisk downunder33 (n=robert@219.95.168.191) |
| 02:46.42 | *** join/#asterisk juice (n=juice@mo-71-50-26-181.dhcp.sprint-hsd.net) |
| 02:47.43 | *** join/#asterisk michaelo (n=michaelo@adsl-153-12-12.gsp.bellsouth.net) |
| 03:01.14 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:01.14 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.7.1 Released! (April 13, 2006), Upgrade from 1.2.7 only necessary if you use app_page -=- http://www.asterisk.org/ -=- AMP/FreePBX/Asterisk@Home users should join #freepbx for support |
| 03:06.40 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 03:13.03 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 03:13.06 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-60.cybersurf.com) |
| 03:16.48 | *** join/#asterisk voiper (n=voiper@c-68-38-69-242.hsd1.nj.comcast.net) |
| 03:16.58 | voiper | Hi |
| 03:18.37 | [TK]D-Fender | SHHHH!!! You'll wake the crickets! |
| 03:18.51 | *** join/#asterisk WhoDaMan (n=DaMan@161.91.171.66.subscriber.vzavenue.net) |
| 03:19.27 | WhoDaMan | /msg voiper yo! |
| 03:20.06 | voiper | did anyone connected asterisk with yate ? |
| 03:21.10 | blitzrage | why? |
| 03:21.18 | voiper | when yate is sending 487 asterisk is ignoring that |
| 03:23.57 | ManxPower | voiper, What is 487, and define "ignore" |
| 03:24.36 | voiper | SIP 487 Cancel / Terminate request |
| 03:25.03 | ManxPower | what is the value of DIALSTATUS? |
| 03:25.28 | voiper | how would i know that ? |
| 03:25.59 | ManxPower | um, in the priority after Dial( put a Noop(DIALSTATUS=${DIALSTATUS}) |
| 03:26.18 | voiper | let me try that |
| 03:26.19 | voiper | thanks |
| 03:27.08 | ManxPower | It's basic Dial debugging stuff. |
| 03:33.16 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
| 03:36.00 | voiper | ManxPower, I am getting this back from yate SIP/2.0 487 EndedByConnectFail but it is not falling into next priority until the timeout has been reached. When it goes to next priority I am getting a dialstatus "NOANSWER" |
| 03:36.51 | ManxPower | voiper, odd |
| 03:44.00 | WhoDaMan | 487 seems to be a pretty common cause code |
| 03:44.29 | WhoDaMan | from the docs, it seems that you'd get a 487 if the user hangs up *before* asterisk does an ANSWER() |
| 03:45.50 | voiper | so do you think yate is sending a wrong code for connect failure |
| 03:46.13 | WhoDaMan | cant say |
| 03:46.46 | distortion | 487 is many times near end hangup |
| 03:47.07 | WhoDaMan | either way, I'd think * would recognize the 487 (error?) and go to the next state in the callflow |
| 03:47.13 | distortion | asterisk and yate work fine together, ive connected to many yate endpoints |
| 03:48.03 | distortion | callflow? * -> yate? yate -> *? |
| 03:48.29 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-57-73.dsl.irvnca.pacbell.net) |
| 03:48.31 | *** join/#asterisk willcampos123 (n=willcamp@adsl-11-96-212.mia.bellsouth.net) |
| 03:48.35 | WhoDaMan | I think its SIP -> * -> Yate -> H.323 |
| 03:49.03 | distortion | yeah that works fine... but 487's come from the calling end (most the time) |
| 03:49.18 | WhoDaMan | correct |
| 03:49.26 | WhoDaMan | I didn't give the full callflow |
| 03:49.32 | willcampos123 | Hello, anyone knows if you make 1 incomming call to asterisk and 2 outgoing, if the cdr unique id changes on the incomming call? |
| 03:49.41 | WhoDaMan | SIP -> * -> Yate -> H.323 -> Quintum |
| 03:49.52 | WhoDaMan | so the 487 is coming back from the quintum |
| 03:50.05 | distortion | what the frak |
| 03:50.14 | WhoDaMan | which Yate happily bounces back to * |
| 03:50.38 | WhoDaMan | but * doesn't seem to like the 487 for whatever reason |
| 03:50.58 | WhoDaMan | it wouldn't go to the next priority as voiper mentioned |
| 03:51.25 | distortion | ah, well manx was precise- asterisk doesnt look at cause codes to advance |
| 03:52.08 | distortion | it looks at the dialstatus, so that would be your best bet |
| 03:52.46 | distortion | i think it considers a 487 a normal call clearing |
| 03:53.09 | WhoDaMan | so in this case, the DIALSTATUS being "NOANSWER" what would the callflow (grammer) look like? |
| 03:53.16 | distortion | err, you need to cause map that in yate |
| 03:53.30 | WhoDaMan | I see |
| 03:53.39 | distortion | or find out what the quintum is sending back, because it most definately is not a 487 since its h323 |
| 03:54.13 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 03:54.14 | voiper | quintum is sending back EndedByConnectFail an h323 code |
| 03:54.29 | distortion | do a tethereal and find the q931 cause |
| 03:54.39 | WhoDaMan | I wanted to stay away from getting my hands dirty with H.323 |
| 03:54.53 | WhoDaMan | thought mucking with * would be the easier / more plausible route |
| 03:55.13 | distortion | well, you need to map the h323 cause received to a 503 if you can, asterisk will definately advance on that |
| 03:55.33 | *** join/#asterisk stillbourne (n=stillbou@c-24-9-8-59.hsd1.co.comcast.net) |
| 03:55.34 | zwelch | can someone give me some tips for debugging iax rsa authentication failures? i can't see what exactly is going wrong from the messages that I am getting |
| 03:55.47 | WhoDaMan | I see what you're saying |
| 03:55.55 | distortion | 487 is considered a "normal call clearing" so * thinks its a ring no answer |
| 03:56.04 | WhoDaMan | makes sens |
| 03:56.06 | WhoDaMan | *sense |
| 03:57.12 | WhoDaMan | its going to be a long and fun night :) |
| 03:57.25 | zwelch | i have two 1.2 * servers, and i'm trying to establish iax peering between them; i created new keys and think that I have them set up correctly, but the registrations don't work no matter what permutation of settings i try |
| 03:57.25 | WhoDaMan | thanks for your helpful pointers, distortion |
| 03:57.29 | WhoDaMan | I appreciate it |
| 03:57.55 | distortion | np, just return the favor someday |
| 03:58.12 | WhoDaMan | will try my best |
| 03:58.58 | voiper | disortion can that be done using a conf file in yate ? (the mapping) |
| 04:00.18 | *** join/#asterisk Altair256 (n=Altair25@tn-greenback1a-30.rhmdky.adelphia.net) |
| 04:02.43 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 04:03.20 | *** join/#asterisk trbldwine (n=trbldwin@71.194.161.170) |
| 04:03.29 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 04:03.41 | pauldy | http://www.sfgate.com/c/pictures/2006/05/08/sp_giants0121df.jpg |
| 04:03.56 | pauldy | just something I though asterisk fans would find ammusing |
| 04:07.42 | CunningPike | Can someone refresh my memory of how to get a queue to always try members in a particular order? |
| 04:14.16 | blitzrage | CunningPike: stategy= |
| 04:14.53 | CunningPike | blitzrage: I think penalty is more like it - none of the strategies always start with agent 1 |
| 04:14.54 | blitzrage | and you probably want to tie in some sort of weight with the agents |
| 04:15.01 | blitzrage | yah |
| 04:15.07 | blitzrage | there you go -- you already knew the answer :) |
| 04:15.08 | CunningPike | blitzrage: Thanks :) |
| 04:15.35 | blitzrage | note to all: don't use applicationmaps |
| 04:15.46 | blitzrage | it will segfault your system |
| 04:17.41 | *** join/#asterisk nain (n=nain@202.59.90.182) |
| 04:17.50 | nain | Hi Every body |
| 04:18.59 | nain | Sending to 209.132.204.50 : 5060 (NAT) |
| 04:18.59 | nain | Found no matching peer or user for '209.132.204.50:5060' |
| 04:19.27 | nain | I have created a user for this Host but still Asterisk is not picking up call from this user....? |
| 04:21.11 | blitzrage | hi doctor nick |
| 04:22.04 | blitzrage | users are matched using the name in the From: header of SIP against the name in square brackets [my_user]. If you want to match on IP address, you use type=peer |
| 04:23.28 | nain | blitzrage: thanks, let me try it... |
| 04:23.43 | Strom_C | so here's a dumb question to which the answer is probably "no": is there a provider that offers geographic U.S. DIDs and unlimited inbound calling either without a restriction on the number of concurrent calls you can have, or a restriction along the lines of 8+ calls? |
| 04:24.07 | blitzrage | no |
| 04:24.23 | blitzrage | concurrent calls is almost always what you get billed on |
| 04:24.29 | blitzrage | especially on unlimited plans |
| 04:24.42 | blitzrage | can get away with it on per-minute plans because you're paying for each concurrent call |
| 04:24.58 | dlynes_ | Just curious |
| 04:25.08 | dlynes_ | What's the difference between asterisk and asterisk-netsec? |
| 04:25.15 | nain | blitzrage: it's working now. but if want to match by host ip they why type=peer, as peer is used for dialing not to dial us.... ? |
| 04:25.21 | blitzrage | netsec is for the Ranch Networks device |
| 04:25.22 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:25.34 | blitzrage | allows asterisk to dynamically open and close ports on the firewall and other cool things |
| 04:25.38 | Strom_C | yeah - voicepulse connect restricts to four calls, and I was hoping that I might get lucky :) |
| 04:25.43 | dlynes_ | ah...ok...so not terribly useful for most people then? |
| 04:25.53 | dlynes_ | unless of course you've got that special device... |
| 04:26.04 | blitzrage | nain: SIP is wierd -- you have to remember that rule you have in your head has 101 exceptions |
| 04:26.12 | blitzrage | dlynes_: exactly |
| 04:26.20 | dlynes_ | ok, thanks |
| 04:26.43 | blitzrage | nain: users are for inbound on name, peer is for outbound and inbound on IP address, friend is for outbound and inbound, matching first on name, then IP address |
| 04:26.50 | blitzrage | roughly* |
| 04:27.59 | dlynes_ | btw...if an autoattendant file starts playing, but you can't hear it, and everything else works just fine, what would the problem be? |
| 04:28.09 | blitzrage | NAT |
| 04:28.15 | nain | blitzrage: That's very good info. but what if i set the type=friend, then how asterisk will match host weather by IP or by name ? |
| 04:28.19 | dlynes_ | but conversation works just fine |
| 04:28.27 | blitzrage | nain: read my sentence again |
| 04:28.44 | dlynes_ | and it's iax, not sip |
| 04:28.47 | nain | blitzrage: got it :) |
| 04:28.49 | blitzrage | dlynes_: blank file, volume too low, can't transcode |
| 04:29.06 | blitzrage | any number of issues I suppose |
| 04:29.07 | dlynes_ | nope..no transcoding errors either, and the file was working just fine on saturday |
| 04:29.37 | dlynes_ | It stopped working after the machine started having trouble trying to bring an x100p card online |
| 04:30.10 | blitzrage | bad timing, corrupted sectors on the HD |
| 04:30.19 | dlynes_ | the x100p card was sharing an interrupt with a network controller, which zaptel and asterisk didn't seem to have a problem with before, and then as soon as I load up an smbfs driver, whammo, all hell breaks loose |
| 04:30.35 | dlynes_ | and nothing's worked right since |
| 04:30.40 | dlynes_ | even without smbfs loaded |
| 04:30.48 | pauldy | samba is pretty hard on network through put |
| 04:31.00 | dlynes_ | like i said...even without smbfs loaded |
| 04:31.30 | pauldy | is nmbd running? |
| 04:31.31 | dlynes_ | all this weird shizzit seems to happen to me, but doesn't seem to happen to anyone else :( |
| 04:31.38 | dlynes_ | pauldy: nope...don't run samba |
| 04:33.21 | blitzrage | don't run stuff on your asterisk box but asterisk |
| 04:33.29 | blitzrage | or crazy weird shit happens |
| 04:33.40 | dlynes_ | none of that shit's running on the asterisk box |
| 04:34.05 | dlynes_ | only smbfs was, and it never did mount the external share, anyways...now i don't even have the driver loaded |
| 04:34.10 | pauldy | YMMV |
| 04:34.27 | dlynes_ | Was just trying to make it easier for the customer to update their moh and autoattendant files |
| 04:35.26 | blitzrage | again -- don't run shit on your asterisk box but asterisk |
| 04:35.47 | Strom_C | dlynes_: introduce them to winscp :) |
| 04:36.06 | dlynes_ | Strom_C: yeah, but then i have to set up keys and everything for them |
| 04:36.24 | Strom_C | no, just set up a user account |
| 04:36.27 | Strom_C | password it |
| 04:36.28 | dlynes_ | and give them sorta full access |
| 04:36.29 | Strom_C | bang |
| 04:36.37 | Strom_C | no |
| 04:36.39 | blitzrage | winscp? pfffft -- pscp rulez :) |
| 04:36.51 | Strom_C | make the folder writable by that user account |
| 04:37.02 | Strom_C | symlink it off the home directory |
| 04:37.04 | dlynes_ | Yeah, but you can't set it up so that local users can get in with passwords, and remote passwords must use keys |
| 04:37.14 | dlynes_ | s/passwords/users |
| 04:37.18 | blitzrage | night all |
| 04:37.30 | pauldy | night blitzrage |
| 04:37.33 | dlynes_ | It's either all one way, or all the other way |
| 04:38.06 | pauldy | welp mine works so I"m out too |
| 04:39.00 | *** join/#asterisk gursikh (n=guriskh1@adsl-209-30-246-160.dsl.hstntx.swbell.net) |
| 04:39.59 | *** join/#asterisk drowe_ (n=david@CPE-139-168-201-6.sa.bigpond.net.au) |
| 04:44.00 | *** join/#asterisk ckwall (n=ckwall@c-67-161-244-209.hsd1.ut.comcast.net) |
| 04:51.41 | *** join/#asterisk flynux (i=16vclez@cl-8.bru-01.be.sixxs.net) |
| 04:55.24 | *** part/#asterisk WhoDaMan (n=DaMan@161.91.171.66.subscriber.vzavenue.net) |
| 04:58.54 | CunningPike | dlynes: We're using samba without any problems |
| 05:00.14 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 05:00.16 | *** join/#asterisk d-tech (n=dtc@72.245.233.107) |
| 05:00.51 | CunningPike | Anyhoo, as Winnie the Pooh said rather stickily in Rabbit's Howse: I mutht be going now |
| 05:00.55 | CunningPike | Later |
| 05:02.44 | nain | blitzrage: first problem solved. would you plz tell me that the type=peer and type=user works in same manner for incoming calls for SIP as well as h323 ???? |
| 05:03.37 | nain | bcz I tried it in same way but call is not being routed to particular context in chan_h323. instead it fall back to default context..... |
| 05:04.14 | nain | Starting H323/ip$202.59.XXX.XXX:2950/7379 at default,51632XXXXX,1 failed so falling back to exten 's' |
| 05:04.49 | *** join/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 05:05.28 | *** join/#asterisk TimRiker (n=timr@pdpc/supporter/bronze/TimRiker) |
| 05:05.38 | dlynes_ | nain: he left |
| 05:05.56 | nain | dlynes_: hmmmm |
| 05:06.11 | nain | dlynes_: could you plz advise me.... |
| 05:06.26 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 05:06.28 | dlynes_ | nain: he left 30 minutes ago |
| 05:06.49 | nain | dlynes_: Ok np.... Can you solve my problem.... |
| 05:06.53 | dlynes_ | nain: i wouldn't have a clue about h323...I need to learn that yet, myself |
| 05:07.08 | dlynes_ | nain: Probably going to be setting up a connection to an h323 gatekeeper later this week |
| 05:07.17 | nain | dlynes_: no problem... thanks.... |
| 05:07.32 | nain | is there any body else who can solve the problem here... |
| 05:07.39 | dlynes_ | nain: but it sounds like you're having a connection problem |
| 05:07.50 | dlynes_ | nain: i.e. whatever info you're using to connect is not correct |
| 05:08.25 | nain | dlynes_: no this is not connection problem, call is falling back to default context and i include the particular context in default it works but this is not good for security reason... |
| 05:08.40 | dlynes_ | nain: obviously :) |
| 05:08.52 | dlynes_ | nain: I don't have anything going into default |
| 05:09.00 | dlynes_ | nain: not even default :) |
| 05:09.00 | nain | Actually call should match the host=IP and should fall back to particular context instead of default but it's not .. |
| 05:09.50 | nain | it's not a particularl h323 problem it's the peer, user and host configration problem |
| 05:10.21 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 05:10.50 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 05:12.11 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 05:15.04 | carrar | anyone use the Linksys SPA-941 |
| 05:19.45 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 05:22.11 | jake1932 | Tim - h263 is video - isn't it? |
| 05:22.57 | TimRiker | I think so, yes. |
| 05:23.33 | jake1932 | what do you mean by , "you need a sip acct"? do you have an asterisk server? |
| 05:23.43 | TimRiker | if I have 2 phones that will do h263 but need to find each other, can asterisk handle that? ie: will asterisk care about the codec required? |
| 05:23.59 | TimRiker | I don't have an asterisk server setup at this point. |
| 05:24.24 | TimRiker | though I suppose I should just set one up. :) I'm wondering how likely is is to work. |
| 05:25.20 | jake1932 | i just googled it and found there was a patch at one time - maybe it's been implemented by now |
| 05:26.05 | jake1932 | http://www.voip-info.org/wiki-Asterisk+video |
| 05:26.11 | jake1932 | looks like it supports h263 |
| 05:28.06 | TimRiker | http://en.wikipedia.org/wiki/H263 has some info. |
| 05:28.22 | jake1932 | yep - good luck with it |
| 05:30.40 | TimRiker | heh. so does asterisk have to support it in order for two devices using it to talk to each other? ie: using asterisk as the SIP server? |
| 05:31.45 | jake1932 | i've never used it - but it appears to |
| 05:33.05 | zobia | hello everyone |
| 05:34.18 | zobia | please see my problem , this copy is from Aasterisk CLI |
| 05:34.19 | zobia | http://pastebin.com/706879 |
| 05:36.40 | zobia | does anyone knows about "No translator path error " |
| 05:38.27 | *** join/#asterisk lorinc (n=ang@caracas-1888.adsl.interware.hu) |
| 05:39.24 | jake1932 | zobia: looks like you're trying to translate from Zap to 256 (whatever that is). What are your two endpoints? |
| 05:40.09 | zobia | sip phone |
| 05:40.23 | zobia | soft phone |
| 05:40.52 | zobia | I am not quit understand this error mean. i never account this error before. |
| 05:41.11 | zobia | since i change a new card |
| 05:42.19 | jake1932 | it would help to know what 256 is |
| 05:42.48 | zobia | yes. that's what i am wondering also |
| 05:45.05 | jake1932 | oh |
| 05:45.18 | jake1932 | what soft phone are you using? |
| 05:45.48 | jake1932 | or - even better - what codec is your softphone set to use? |
| 05:46.13 | zobia | softphone is xlite |
| 05:46.43 | zobia | sorry. using eye beam |
| 05:46.52 | jake1932 | make sure you have G.711 or ulaw checked. |
| 05:47.01 | jake1932 | you can unchecked everything else |
| 05:47.40 | jake1932 | also, make sure in the peer entry in asterisk you have "disallow=all"l and "allow=ulaw" |
| 05:47.50 | zobia | yes. i check. i check g711 ulaw and g711 alaw |
| 05:48.10 | jake1932 | besides the error - what are the symptoms? |
| 05:48.26 | zobia | just can not dial out any number |
| 05:48.34 | jake1932 | with either phone? |
| 05:48.37 | zobia | when dial the console will show that error |
| 05:50.03 | zobia | dial internal extension has not problem. just dial outbound 10 digits number failed |
| 05:50.16 | jake1932 | from either phone? |
| 05:50.33 | zobia | yes none of the phone can dial out |
| 05:51.33 | zwelch | TimRiker: yes; asterisk can serve as a bridge between two sip devices, though you might want something focused on SIP (e.g. openser). |
| 05:53.12 | jake1932 | zobia: it looks like a codec compatibility error. if you call between the two phones and do a sip show channels - you should see which codecs are in use |
| 05:54.00 | zobia | ok |
| 05:54.20 | TimRiker | zwelch: sup man? long time no see. headed to ols this year? I'm probably going to miss it. |
| 05:54.24 | zobia | let me check. |
| 05:55.19 | zwelch | TimRiker: not gunna make it this year; haven't had the resources to make it happen |
| 05:55.46 | zobia | jake1932 dial between sip phones showes using ulaw |
| 05:55.52 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 05:56.58 | jake1932 | and when you try to dial with the zap card, it can't translate? |
| 05:57.40 | zobia | yes. can not translate |
| 05:59.43 | *** join/#asterisk MGSsancho (n=user@adsl-67-126-128-145.dsl.irvnca.pacbell.net) |
| 06:01.33 | jake1932 | zobia: sorry can't be of more help. I don't think ulaw is 256. sound like you got a strange codec issue |
| 06:02.30 | zobia | ok. no problem, thanks a lot jake1932 |
| 06:03.47 | jake1932 | zobia: just did a show codecs |
| 06:03.55 | jake1932 | looks like g729 is 256 |
| 06:04.28 | jake1932 | looks like your phones are trying to talk g729 and I'm assuming you have no g729 licenses |
| 06:04.50 | jake1932 | that would explain pass thru working and not being able to translate |
| 06:04.58 | zobia | capabilities = 68 |
| 06:04.58 | zobia | format = 256 |
| 06:05.52 | jake1932 | 68 is 64 + 4 (slin and ulaw) |
| 06:05.55 | Altair256 | you could try setting your sip phones to g711 to test if this is the issue |
| 06:06.35 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 06:07.35 | harlequin516 | what versions of asterisk support the GET OPTION command ? |
| 06:07.41 | *** join/#asterisk Pageus (n=FreePBX5@ip70-190-19-6.ph.ph.cox.net) |
| 06:07.53 | jake1932 | in other words - your zap channel will natively take slin and ulaw, and your passing it a g729 which will require translation (and a g729 license for that) |
| 06:09.05 | jake1932 | arlight - gotta get some rest. but good luck zobia |
| 06:09.14 | Altair256 | same here... later guys |
| 06:11.45 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.10) |
| 06:18.06 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 06:19.18 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 06:23.03 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 06:24.16 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 06:29.09 | harlequin516 | Hmm... Why does gentoo portage only have one ebuild for Asterisk 1.2 ? |
| 06:30.44 | harlequin516 | Hmm... Why does gentoo portage only have one ebuild for Asterisk 1.2 ? |
| 06:40.07 | OloBola | nuf is a e'nuf, my fone is broken |
| 06:48.25 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:50.35 | *** join/#asterisk parag7732 (n=root@de2-b15916.alshamil.net.ae) |
| 06:50.39 | parag7732 | Anybody able to successfully implemented CALL BACK feature... |
| 06:50.48 | parag7732 | in freepbx |
| 06:51.08 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 06:51.51 | *** join/#asterisk KeX_WorX (n=chris@ng1.kurtkrenn.com) |
| 06:51.53 | KeX_WorX | hi |
| 06:51.58 | parag7732 | Anybody able to successfully implemented CALL BACK feature... |
| 06:52.17 | KeX_WorX | parag7732, if it is only asterisk intern, jep |
| 06:52.40 | KeX_WorX | is it possible to do playback, but don't answer the call ? |
| 06:52.43 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 06:52.47 | KeX_WorX | if i call from asterisk out? |
| 06:53.28 | KeX_WorX | if it is a fixed line number i wanna play a note, if it is an internet call, i wanna play another note, but don't answer the call yet |
| 06:53.33 | KeX_WorX | is this possible ? |
| 06:54.59 | KeX_WorX | I've read that the playback app has the option 'noanswer', but there is no differenc in specifieng this argument or not |
| 06:55.27 | *** part/#asterisk parag7732 (n=root@de2-b15916.alshamil.net.ae) |
| 07:00.55 | *** join/#asterisk Cresl1n (n=matt@myskin.iet.unipi.it) |
| 07:15.32 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:15.33 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:23.02 | dlynes | KeX_WorX: yes, it's possible |
| 07:23.59 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:25.00 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 07:25.06 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:25.27 | *** join/#asterisk crich1999 (n=crich@myskin.iet.unipi.it) |
| 07:28.58 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 07:29.21 | *** join/#asterisk rKR245 (n=ravi@office.callsat-telecom.com) |
| 07:30.00 | KeX_WorX | dlynes, how would i do that? |
| 07:30.36 | KeX_WorX | dlynes, I tried this: exten => _Z.,1,Playback(festnetzanruf,noanswer) |
| 07:30.49 | KeX_WorX | dlynes, and also exten => _Z.,1,Playback(festnetzanruf,skip) |
| 07:30.59 | dlynes | KeX_WorX: one sec...i just know it can be done...never done it...need to look up the documentation |
| 07:31.05 | dlynes | KeX_WorX: but i think it was the noanswer option |
| 07:31.22 | rKR245 | dlynes:how are you |
| 07:31.38 | rKR245 | after so long time |
| 07:31.41 | dlynes | rKR245: good...and you? |
| 07:31.46 | dlynes | rKR245: who are you? |
| 07:31.49 | dlynes | don't remember you |
| 07:31.49 | KeX_WorX | the first playes the soundfile, but answeres the channel imidiatly. the second doesn't answers but doesn't plays the soundfile |
| 07:31.54 | rKR245 | iam fine thank you |
| 07:32.16 | KeX_WorX | as the doc describes |
| 07:32.18 | rKR245 | but you helped me a lot before months so iremebered you |
| 07:32.36 | KeX_WorX | i tried with a zap and a sip channel |
| 07:32.41 | dlynes | ah..c.ouldn't have been months |
| 07:32.52 | dlynes | i've only been frequenting this channel for about 2 weeks now |
| 07:33.15 | rKR245 | no ? may be in freepbx |
| 07:33.24 | dlynes | nope...don't use freepbx |
| 07:33.39 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 07:33.40 | dlynes | don't have any plans to, either |
| 07:33.48 | rKR245 | o.k fair |
| 07:34.12 | rKR245 | but i know you since 8 weeks |
| 07:34.23 | rKR245 | any way |
| 07:34.33 | dlynes | maybe |
| 07:34.40 | dlynes | but i've only been here for about two weeks |
| 07:34.45 | dlynes | i've been using asterisk much longer, though |
| 07:35.05 | dlynes | KeX_WorX: yeah...it'll probably only work on zap channels |
| 07:35.13 | rKR245 | i know you because you helped me on asterisk sip channels |
| 07:35.17 | dlynes | KeX_WorX: did you try the noanswer option on a zap channel? |
| 07:35.20 | dlynes | could be |
| 07:35.27 | dlynes | that's what i predominantly work with on asterisk |
| 07:35.32 | rKR245 | how to create sip extensions |
| 07:35.35 | *** join/#asterisk littlejohn (n=little@host57-76.pool8711.interbusiness.it) |
| 07:35.42 | dlynes | maybe |
| 07:35.47 | dlynes | but i help so many people |
| 07:35.49 | rKR245 | i still rememberd that |
| 07:35.53 | dlynes | that it's difficult to remember everyone |
| 07:36.02 | rKR245 | ofcourse you do |
| 07:36.35 | rKR245 | dlynes now iam running my asterisk fine |
| 07:36.44 | dlynes | cool |
| 07:36.53 | rKR245 | and SER too |
| 07:36.59 | dlynes | beautiful |
| 07:37.08 | dlynes | I have yet to run SER, myself |
| 07:37.16 | dlynes | I want to run it |
| 07:37.20 | rKR245 | now iam using SER as proxy ,registrar |
| 07:37.24 | dlynes | Just haven't had the time to learn it and set it up yet |
| 07:37.27 | rKR245 | its easy to run |
| 07:37.38 | dlynes | yeah, but i've had other priorities |
| 07:37.43 | rKR245 | no you can do it in 15 min.. |
| 07:37.43 | dlynes | including writing a billing system |
| 07:37.47 | rKR245 | no. |
| 07:37.59 | dlynes | that may very well be |
| 07:38.00 | rKR245 | just to test proxy and registration |
| 07:38.06 | dlynes | but i don't want to run it on my asterisk box, either |
| 07:38.12 | dlynes | i want to run it on a separate server |
| 07:38.18 | rKR245 | yes its better |
| 07:38.21 | dlynes | and that requires a reformat, reinstall, repartition, ... |
| 07:38.24 | dlynes | that all takes time |
| 07:38.38 | rKR245 | due to this iam running my asterisk at port 5065 |
| 07:39.06 | rKR245 | because both SER and ASTERISK default run at 5060 |
| 07:39.16 | dlynes | Yeah, i've already got three spare computers waiting in the wings at the colo, for when I ramp up to a more distributed system |
| 07:39.24 | dlynes | I just haven't had the time to finish the job |
| 07:39.45 | rKR245 | do you have mediaproxy in asterisk |
| 07:39.54 | dlynes | no idea wtf that even is |
| 07:40.03 | KeX_WorX | dlynes, I tried on zap channels and on sip channels |
| 07:40.20 | KeX_WorX | dlynes, with the same result: asterisk answers the channel and playes the file |
| 07:40.31 | dlynes | How do you know that asterisk has answered the channel? |
| 07:41.28 | rKR245 | any way i just installed a mediaproxy including rtp proxy in my SER |
| 07:41.32 | Mystiq | rKR245: Asterisk includes a "mediaproxy".. by default it wants to have control over the voice/rtp channels |
| 07:41.44 | rKR245 | ohhh: |
| 07:41.53 | KeX_WorX | dlynes, I see it on the CLI and the timer on the phone starts |
| 07:41.55 | rKR245 | so how we can activate |
| 07:42.16 | dlynes | KeX_WorX: maybe it only works on iax channels then |
| 07:42.17 | Mystiq | rKR245: default activated, plus you have to specify canreinvite=no in the sip.conf |
| 07:42.18 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 07:42.26 | rKR245 | o.k |
| 07:43.02 | dlynes | Mystiq: and that's just the regular proxy....what's the difference between that proxy and a "outbound proxy"? |
| 07:43.12 | rKR245 | so then there is no need for me to again installing media proxy in SER i just forward calls to asterisk |
| 07:44.41 | Mystiq | dlynes: most of the time outbound proxy is used on borders.. for example, you are on an internal network (with alot of other ip phones), and then you put an additional proxy on the border (intranet/internet) that sends/receives all the traffic |
| 07:45.17 | dlynes | Mystiq: ah...so you can use that outbound proxy for sip trunking, then? |
| 07:46.03 | Mystiq | yes, but most devices don't send "all" sip traffic to that proxy (register, etc) |
| 07:47.00 | dlynes | ah...so sip messages are still sent via the regular proxy |
| 07:47.08 | dlynes | and rtp is all sent via the outbound proxy? |
| 07:47.12 | Mystiq | no no |
| 07:47.23 | Mystiq | rtp is never sent through the proxy |
| 07:47.37 | dlynes | mediaproxy, or outbound proxy? |
| 07:47.44 | Mystiq | mediaproxy |
| 07:47.48 | Mystiq | in SER's case |
| 07:47.55 | Mystiq | but mediaproxy is another application/daemon |
| 07:48.02 | dlynes | ah...but we usually set asterisk up to be the media proxy |
| 07:48.03 | Mystiq | is not builtin into SER |
| 07:48.08 | Mystiq | yes, exactly |
| 07:48.48 | Mystiq | and if you would add an additional outbound proxy, it could also be asterisk |
| 07:48.55 | Mystiq | but then on the internal network |
| 07:48.55 | dlynes | oh |
| 07:49.28 | Mystiq | but i've never used the outbound proxy config |
| 07:49.37 | Mystiq | because most cases can be solved other ways |
| 07:50.11 | rKR245 | i think just bu using DHCP |
| 07:50.15 | dlynes | yeah...was just curious |
| 07:50.35 | dlynes | i keep seeing that outbound proxy mentioned in phones all the time, but i didn't know what the heck the damned thing was |
| 07:50.48 | dlynes | it was just irritating me more than anything that i didn't know what it was :) |
| 07:51.18 | dlynes | I've only ever had a problem with NAT twice...and even then, both times I eventually solved the problem |
| 07:51.37 | dlynes | both times it was because of crappy linksys routers |
| 07:53.07 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:53.36 | *** join/#asterisk Assid (n=assid@203.115.64.11) |
| 07:53.50 | Assid | woopies |
| 07:54.22 | Assid | hows everybuddy |
| 07:54.35 | rKR245 | how we can solve NAT problems by using mediaproxy? |
| 07:54.39 | *** join/#asterisk littlejohn (n=little@host57-76.pool8711.interbusiness.it) |
| 07:54.43 | Assid | mediaproxy ? |
| 07:54.55 | Assid | qu'est que c'est ? |
| 07:55.12 | Assid | im guessing you mean stun ? |
| 07:55.18 | *** join/#asterisk CKGL (n=Cglob@202.8.86.162) |
| 07:55.26 | dlynes | Assid: nope |
| 07:55.46 | dlynes | Assid: regular proxy, not outbound proxy, and not stun server |
| 07:56.01 | CKGL | Hi, I can dial using "Zap/g2/${EXTEN}" for zap channels |
| 07:56.14 | CKGL | How to do it with sip? |
| 07:56.24 | dlynes | SIP/${EXTEN} |
| 07:56.44 | CKGL | hmm, I mean a group of sip extensions |
| 07:56.53 | evilbuny | SIP/${EXTEN}@VSP |
| 07:57.03 | dlynes | SIP/100&SIP/101&SIP/102&SIP/103? |
| 07:57.03 | CKGL | like if one is busy, it will then go to other sip ext |
| 07:57.12 | dlynes | oh |
| 07:57.28 | evilbuny | CKGL: have a look at my doco site www.asterisk.net.au |
| 07:57.31 | evilbuny | covers fail over |
| 07:57.41 | CKGL | evilbuny: ok |
| 07:57.44 | dlynes | CKGL: You need to examine the ${DIALSTATUS} variable |
| 07:57.47 | *** join/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 07:57.53 | dlynes | CKGL: and branch appropriately |
| 07:58.24 | dlynes | I'm sure there's probably a better way of doing it, but... |
| 07:58.44 | CKGL | dlynes: isn't there a direct way doing it? =) |
| 07:59.04 | CKGL | dlynes: that's what I'm looking for |
| 07:59.22 | dlynes | CKGL: Well, SIP is not bundled like pri channels are |
| 08:00.02 | evilbuny | CKGL: the example on my site does status checking |
| 08:00.34 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:01.32 | CKGL | dlynes,evilbuny: what about "callgroup" parameter? |
| 08:01.42 | dlynes | evilbuny: yeah...that's what i was explaining to him, too |
| 08:01.55 | dlynes | CKGL: do you see a callgroup parameter in sip.conf? |
| 08:01.59 | dlynes | I don't |
| 08:02.18 | CKGL | yes, http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+callgroup |
| 08:03.19 | dlynes | I guess SIP/g1/${EXTEN} then |
| 08:03.37 | CKGL | probably |
| 08:03.55 | dlynes | I never actually tried using it in sip |
| 08:04.06 | dlynes | Most of our customers are typical keysystem users |
| 08:04.15 | dlynes | So when they hit line 1, they want it to go out on line 1 |
| 08:04.22 | dlynes | Same for when they hit line 2 |
| 08:04.42 | CKGL | I'm using this to serve IVR users |
| 08:05.00 | dlynes | So I've got it set to go out on line 1 if they ask for line 1, but if line 1 is busy and it's not a 911 call, go out on line 2 |
| 08:05.10 | dlynes | otherwise, hang up the caller on line 1, and make the 911 call |
| 08:05.13 | CKGL | who want to talk with professional consultants, not a specific one |
| 08:05.35 | dlynes | oh |
| 08:05.41 | dlynes | You want call queues then, right? |
| 08:05.42 | CKGL | they may consult any one that's available on SIP extensions |
| 08:06.01 | dlynes | not call groups |
| 08:06.13 | CKGL | dlynes: yeah, if all are busy then I will need call queues |
| 08:06.22 | dlynes | Yeah, but the thing is |
| 08:06.34 | dlynes | Call queues can still handle it even when all phones aren't busy |
| 08:06.50 | dlynes | It's just that if nobody's busy, the call never waits in the queue |
| 08:07.29 | CKGL | dlynes: probably that's what I'm looking for |
| 08:07.41 | dlynes | CKGL: Yeah...take a look at agents and queues |
| 08:07.51 | dlynes | CKGL: It's more flexible than what you're looking at, anyways |
| 08:07.53 | CKGL | dlynes: thanks |
| 08:08.04 | dlynes | CKGL: You can define how it determines who gets the next call, too |
| 08:08.07 | *** join/#asterisk debaser (i=debaser@mindsplit.net) |
| 08:08.08 | CKGL | dlynes: it also works with SIP, right? |
| 08:08.19 | dlynes | CKGL: it works on all phones |
| 08:08.34 | dlynes | CKGL: i.e. any channel type asterisk supports |
| 08:08.36 | CKGL | dlynes: okay |
| 08:08.39 | Mystiq | "Callgroups are not intended to call a group of phones" |
| 08:08.55 | dlynes | Yeah, they're intended to call out on a group of trunk lines |
| 08:09.18 | CKGL | dlynes: I used grouping to handle people from my ZAP channels |
| 08:09.33 | dlynes | CKGL: Yeah, but that's not the intended use for it |
| 08:09.51 | CKGL | dlynes: what a shame! =) |
| 08:10.21 | dlynes | CKGL: anyways...the queues and agents have i think three different algorithms you can use, to determine who gets the next caller |
| 08:10.25 | dlynes | CKGL: it's much better |
| 08:11.05 | dlynes | CKGL: and it has the added benefit that if everyone's busy, it'll put the caller on hold and play them musak |
| 08:11.48 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 08:12.14 | dlynes | Mystiq: btw...do you happen to know what the difference is between a 'group' and a 'callgroup'? |
| 08:12.18 | CKGL | dlynes: are these included in asterisk package or it's something else I have to install? |
| 08:12.25 | dlynes | CKGL: it's included |
| 08:12.29 | dlynes | CKGL: it's one of the modules |
| 08:12.54 | dlynes | CKGL: there's also one to zap those nasty telemarketers |
| 08:13.07 | dlynes | CKGL: and one for blacklisted callers to let thme know they're unwanted |
| 08:13.11 | dlynes | CKGL: all kinds of cool stuff |
| 08:13.22 | CKGL | dlynes: okay, will look at it now |
| 08:18.48 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:20.39 | Zeeek | good morning |
| 08:20.40 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-83.cybersurf.com) |
| 08:20.49 | dlynes | mornin' |
| 08:20.57 | Zeeek | oops, staff meeting! |
| 08:24.14 | pif | you forgot the cover sheet on these TPS reports! |
| 08:25.17 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 08:33.05 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
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| 08:40.12 | *** part/#asterisk achandra (n=achandra@12.44.122.130) |
| 08:40.27 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 08:50.56 | *** join/#asterisk magic_1 (n=quinton@wbs-196-2-105-97.wbs.co.za) |
| 08:51.39 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
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| 08:54.23 | *** join/#asterisk GoofBall (n=jsadler@s233-68-208.nap.wideopenwest.com) |
| 08:54.54 | GoofBall | Hi all.... |
| 08:55.23 | GoofBall | I'm seeing a wierd behavior on my ZAP ports when SIP calls are attempted to them... |
| 08:57.36 | GoofBall | Anyone care to help me debug it? |
| 08:58.01 | GoofBall | This is a behavior that just started -- all worked fine last week. No changes made to the config or OS, so I'm befuttled... |
| 09:00.14 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:02.55 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:06.22 | Zeeek | what is the behaviour? |
| 09:07.02 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 09:07.15 | GoofBall | I'm getting a "Call Progress" indication after the zap port is siezed and digits outpulsed. |
| 09:08.11 | GoofBall | I looked at the chan_sip code and found that indication is only sent if the RTP session has not been established at the time the signaling message is being sent. |
| 09:10.18 | GoofBall | Thing is the RTP session is established shortly afterward, but no indication is sent updating the state. |
| 09:11.48 | Zeeek | i can't help but maybe someone else is awake |
| 09:11.56 | GoofBall | thanks... |
| 09:12.53 | rKR245 | dlynes:are you there? |
| 09:14.29 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:16.56 | key2 | GoofBall, do you use a rtp proxy ? |
| 09:18.51 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
| 09:19.39 | achandra | hello..what effect does link aggregration on switch and ethernet bonding have on asterisk QoS ( ie jitter, rtp stream, etc?) |
| 09:21.51 | *** part/#asterisk GoofBall (n=jsadler@s233-68-208.nap.wideopenwest.com) |
| 09:22.28 | dlynes | ? |
| 09:24.08 | syle | anyone know what dbsecret field is in iax realtime? |
| 09:24.24 | Zeeek | it's a secret, no one knows! |
| 09:26.24 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 09:29.30 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:30.32 | zwelch | does anyone know how to add an AMR codec to asterisk? |
| 09:30.43 | zwelch | http://www.vovida.org/applications/downloads/AMR/ |
| 09:41.55 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:42.39 | littleball | hello, i have E1 lines connect to operators. But one server only have 4 E1 lines. How to scale the services? |
| 09:42.46 | littleball | eg., adding more servers |
| 09:42.47 | *** join/#asterisk stevedl (n=steve_dl@eth87.tas.adsl.internode.on.net) |
| 09:43.03 | *** join/#asterisk shiznatix (n=shiznati@213-35-237-37-dsl.end.estpak.ee) |
| 09:43.06 | shiznatix | mornin everyone |
| 09:45.37 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 09:49.49 | nettie | hey guys, anyone know if a SIP plug-in for outlook is available please? opensource or commercial? |
| 09:57.04 | shiznatix | I have a question. I am using WaitExten() but if i call from my cell phone to the zapata line and I enter '333' as the extension somtimes it will be say that I entered '333333' or '33' as my extension or some other random number of times. |
| 09:57.15 | shiznatix | is there some way to fix this problem? |
| 10:00.51 | *** join/#asterisk many (i=many@krikkit.ukeer.de) |
| 10:01.03 | Assid | check out timeout(response) and timeout(digit) |
| 10:01.05 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 10:01.14 | many | heya. |
| 10:01.21 | Assid | but then im not sure if zapata would behave anydifferently |
| 10:01.40 | Assid | biab |
| 10:03.10 | scanna | hi all, can someone explain me the difference between wrapuptime in agents.conf and wrapuptime configured for each queue? |
| 10:03.27 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 10:09.35 | scanna | :( |
| 10:14.36 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 10:15.19 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 10:20.44 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 10:20.49 | faberk64 | hi |
| 10:21.11 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 10:21.15 | faberk64 | I'm need to setup "early audio" on my sistem |
| 10:21.30 | faberk64 | I'm trying with this http://pastebin.com/707077 |
| 10:22.32 | faberk64 | but what append is that the guest is not billed(that's ok) but cannot ear nothing... no audio |
| 10:23.17 | faberk64 | I need that guests listen the file "advise", before talk to me |
| 10:23.24 | faberk64 | what's wrong? |
| 10:24.42 | faberk64 | where I'm wrong? |
| 10:25.05 | Zeeek | file is still in the air ? |
| 10:25.24 | file | faberk64: you just didn't give enough information... like what technology are you trying to do early media on |
| 10:25.24 | faberk64 | yes of course |
| 10:25.42 | faberk64 | is a gsm audio file |
| 10:25.52 | faberk64 | like all the others into * |
| 10:26.26 | faberk64 | I'm tryed also with other, default files coming with *. |
| 10:26.32 | faberk64 | but nothing change |
| 10:27.23 | faberk64 | I'm trying to do early media on a PRI E1 channel |
| 10:28.07 | shiznatix | How can I change the priority number back to 1 but instead of it looping back in that context keep going down the list? |
| 10:32.34 | *** join/#asterisk suma (n=suma@cm145.gamma29.maxonline.com.sg) |
| 10:33.41 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 10:34.17 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 10:34.58 | mut | man |
| 10:35.03 | mut | i am so pissssssssssssed |
| 10:35.20 | mut | i think i'm going to have to stick with that damned as5350 |
| 10:35.28 | qdk | mut: you shouldnt hold it so long then. :-) |
| 10:35.45 | mut | cause the digium card couldn't echo cancel |
| 10:35.53 | mut | the sangoma card can echo cancel but it won't work right |
| 10:36.05 | *** join/#asterisk blue9 (n=chatzill@host213-123-130-180.in-addr.btopenworld.com) |
| 10:36.59 | blue9 | Anyone with any knowledge of using Asterisk in a business available for me t |
| 10:37.01 | darkskiez | mut: whats wrong with the digiums echo can borad? |
| 10:37.04 | blue9 | *to pick their brains? |
| 10:37.22 | mut | darkskiez: dunno |
| 10:37.26 | mut | never tried it |
| 10:37.40 | mut | havn't heard good things tho |
| 10:37.45 | darkskiez | shame |
| 10:38.06 | darkskiez | how come you get so much echo then? |
| 10:38.25 | mut | so far from the CO maybe |
| 10:38.27 | mut | i dunno |
| 10:38.41 | mut | it's bad tho |
| 10:44.12 | blue9 | Ah, change of tack... anyone used any phones by Snom with * ? |
| 10:45.12 | mut | wonder what time this sangoma guy gets in his office |
| 10:49.56 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 10:51.42 | *** join/#asterisk UlbabraB (n=caplaz@host-84-222-44-13.cust-adsl.tiscali.it) |
| 10:54.28 | *** join/#asterisk ZX81 (n=ZX81@213-140-22-78.fastres.net) |
| 10:56.06 | Dr-Linux | question, i have 4 analog lines attached to my TDM400P FXO, i wanna route each line to differet extension, can i do that? |
| 10:56.18 | ZX81 | yes |
| 10:56.24 | ZX81 | put a different context line |
| 10:56.28 | ZX81 | for each channel |
| 10:56.31 | ZX81 | in zapata.conf |
| 10:56.39 | ZX81 | and then send them from the context to the same one |
| 10:56.46 | ZX81 | but with different extensions |
| 10:56.47 | ZX81 | :D |
| 10:57.28 | blue9 | Anyone know if I can configure a snom 360 to show the status of my ISDN lines, rather than my internal phones? |
| 11:06.06 | Dr-Linux | can anyone answer me question? |
| 11:06.08 | Dr-Linux | question, i have 4 analog lines attached to my TDM400P FXO, i wanna route each line to differet extension, can i do that? |
| 11:06.24 | blue9 | ZX81 just answered you, Dr Linux. |
| 11:08.05 | Dr-Linux | blue9: yeah, but his answer was not complete. |
| 11:08.20 | Dr-Linux | blue9: what can i define in zapata.conf |
| 11:08.23 | *** join/#asterisk magic_1 (n=quinton@wbs-196-2-110-87.wbs.co.za) |
| 11:08.33 | blue9 | A context line for each channel? |
| 11:09.27 | Dr-Linux | blue9: if you know, can you explain a bit how can i do that? |
| 11:09.35 | blue9 | I don't, I'm afraid. |
| 11:09.39 | blue9 | Try google? |
| 11:10.10 | Dr-Linux | blue9: yes i tried google but no luck |
| 11:10.25 | blue9 | Sorry, can't help any more than that then. |
| 11:10.46 | Dr-Linux | ok |
| 11:12.16 | rKR245 | any body using asterisk as gateway? |
| 11:12.23 | mog_home | nope |
| 11:12.24 | mog_home | never |
| 11:12.34 | qdk | rKR245: gateway to and from what? |
| 11:12.45 | rKR245 | gateway to pstn |
| 11:12.57 | rKR245 | from SER |
| 11:13.02 | qdk | rKR245: sure... lots. |
| 11:13.25 | qdk | rKR245: oh, dont know anything about OpenSER. |
| 11:13.34 | rKR245 | qdk,how you configure your ser |
| 11:13.42 | mog_home | yeah lots of people do |
| 11:13.45 | qdk | rKR245: i dont. :-D |
| 11:13.59 | rKR245 | that means ser.cfg |
| 11:14.03 | qdk | rKR245: but you could use SIp or maybe IAX if OpenSER supports it. |
| 11:14.26 | rKR245 | SERis only for sip |
| 11:15.17 | rKR245 | mog_home ,can you tell me what you did in you ser.cfg to forward pstn calls to asterisk |
| 11:15.32 | qdk | rKR245: ok, then you could use SIP to transport calls to your asterisk when the transport it to PSTN. |
| 11:16.06 | qdk | rKR245: its there a OpenSER forum for that? |
| 11:16.26 | rKR245 | exactly qdk but i just need a piece of c-base coding to write in SER and aswell as in asterisk |
| 11:17.16 | qdk | rKR245: that seems strange and unlikely to my knowledge. |
| 11:17.41 | rKR245 | qdk ser or openser both give docs about only proxies registars serweb and lots but only thing lacking is how to connect ser with asterisk |
| 11:18.25 | rKR245 | do you got me now? |
| 11:18.58 | qdk | rKR245: doesnt SER have a rfc compliant implementation of SIP? |
| 11:19.06 | rKR245 | it has |
| 11:19.35 | qdk | rKR245: then i dont see the problem with SER and Asterisk interconnecting. |
| 11:20.25 | rKR245 | can you tell me how |
| 11:20.31 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 11:21.57 | Ahrimanes | anyone tried having a line disconnected on a snom 190, when a second call arrives? |
| 11:22.21 | rKR245 | qdk my ser is working fine and even with rtpproxy ,just i want to use ser as proxy registrar redirecting and for forwarding sip calls with out asterisk and i just need is SER must forward pstn calls to asterisk because i want to connect asterisk to pstn gateway |
| 11:23.12 | rKR245 | qdk , do you have idea regarding this issue |
| 11:26.50 | qdk | rKR245: yes, but i dont know anything about SER configuration, but if it were an asterisk <-> asterisk i would make an IAX trunk (you could use SIP here) and setup my extentions to use that trunk when to call is for a PSTN phone (or just non local known number) |
| 11:26.59 | shiznatix | Does anyone know how I can use fax detection on asterisk? I need to use the 'fax' extension but if it is not a fax I want it to dial a extension |
| 11:27.08 | shiznatix | but right now the 'fax' extension is not working at all |
| 11:27.58 | qdk | shiznatix: you can detect a faxcall in asterisk have a look at that. |
| 11:28.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:29.12 | shiznatix | qdk, yes but how do I detect the faxcall? I heard it was faxdetect=yes but that does not work |
| 11:29.54 | *** join/#asterisk faberk64 (n=faberk@213.199.15.249) |
| 11:30.08 | rKR245 | qdk,thaks i will try now |
| 11:31.18 | qdk | shiznatix: http://www.voip-info.org/wiki-Asterisk+fax <- take a look at that... i dont use asterisk for/with faxing. |
| 11:31.32 | *** join/#asterisk azeteg (n=azeteg@c110.brewhouse.se) |
| 11:32.52 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 11:32.56 | azeteg | anyone knows how to setup asterisk SIP to talk to H323 gatekeeper and route all calls and accounts through taht? |
| 11:34.01 | qdk | azeteg: logically enough SIP doesnt talk with H323. |
| 11:34.50 | qdk | azeteg: but you wanna do SIP <-> Asterisk <-> H323 <-> H323-gateway? |
| 11:35.55 | azeteg | correct |
| 11:36.33 | azeteg | I have an ISP which has switched to new voip platform, and their voip platform doesn'twork as it should with our phones yet |
| 11:36.39 | azeteg | thats 70 phones |
| 11:36.47 | azeteg | which are right now behaving like bullshit |
| 11:37.05 | azeteg | so I figured I make an asterisk setup with SIP, and forward to h323 gatekeeper |
| 11:37.07 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 11:37.08 | PakiPenguin | noon |
| 11:37.18 | azeteg | the asterisk is up running fine with phones |
| 11:37.25 | qdk | azeteg: Ok, i guess many in here knows how to do that, but that doesnt help you if you expect them to spoonfeed you. |
| 11:38.01 | azeteg | I'm reading all docs I can, but just that little h323 conf part is hard to find |
| 11:38.20 | azeteg | if anyone cares to give some pointers to a destperate man, I would be very grateful |
| 11:38.22 | qdk | azeteg: ok, then you just need the H323 channel and then update your dialplan to make us of it. |
| 11:38.54 | azeteg | I need to get openh323 and pwlib for that= |
| 11:38.55 | qdk | azeteg: ok, thats a much better question. |
| 11:38.56 | azeteg | ? |
| 11:40.13 | qdk | azeteg: http://www.voip-info.org/wiki-Asterisk+H323+channels <- have you read that and its followup? |
| 11:41.01 | azeteg | I read some, just reading more |
| 11:42.06 | azeteg | reading AstRecipes right now |
| 11:43.17 | qdk | azeteg: ok, i know very little about H323, cox its the way of old carriers... Im a new bread. :-D |
| 11:43.30 | tzanger | a new bread? So what, like some 7-grain variety? |
| 11:43.42 | PakiPenguin | :) |
| 11:43.44 | PakiPenguin | lol |
| 11:46.05 | qdk | breed* :-D |
| 11:46.38 | qdk | 30% fibers |
| 11:46.40 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 11:49.26 | azeteg | I know its old shit |
| 11:49.31 | azeteg | but this isp is mad |
| 11:49.38 | azeteg | I think I'll do my own solution completely |
| 11:50.00 | azeteg | 70 phones that haven't been able to call or receive calls correctly for 4 days |
| 11:50.11 | azeteg | in a migration that should have taken under 1 minute |
| 11:50.24 | azeteg | smells like lawsuit to me |
| 11:52.30 | qdk | azeteg: we just converted to SS7 signaling with also was a pain due to the company we interconnect with. :-( |
| 11:55.29 | azeteg | I'll tell you - the main problem is the swissvoice ip10s phone and its h323 image |
| 11:55.40 | azeteg | and a cisco 7301 router somewhere in the isps net |
| 11:56.00 | azeteg | when the 7301 routes to the swissvoices - it sometimes sends packets through the wrong GRE tunnel |
| 11:56.04 | azeteg | for no reason at all |
| 11:56.17 | azeteg | so packets are lost - h245 negotiation especially |
| 11:56.30 | azeteg | it is a malfunction in the IOS software of the 7301 |
| 11:56.45 | azeteg | cisco are looking into itd |
| 11:57.25 | azeteg | current workaround is to tunnel h245 through h225 |
| 11:57.30 | azeteg | but we have some errors still |
| 11:58.03 | azeteg | so I just say skip it - and lets run our own asterisk that talks to their h343 gatekeepeer |
| 11:58.42 | *** join/#asterisk lorinc (n=ang@caracas-2120.adsl.interware.hu) |
| 12:03.16 | shiznatix | I am having trouble with using asterisk as a voice/fax switch. it works but it does not work until I sent the Hangup command |
| 12:04.51 | shiznatix | I have: http://pastebin.com/707198 but if I uncomment those lines then it does not go to the 'fax' extension until it is too late |
| 12:06.48 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 12:07.39 | magic_1 | what command can i type to kill all actie calls |
| 12:10.29 | magic_1 | sorry for that i got sorted |
| 12:18.02 | Kyler | I'm still trying to get Asterisk to transfer a SIP call and get out of the way. I'm using a simple Dial() command, the call is originated and terminated at the same host, and ulaw is used for both channels. I'm picking through the SIP debug output. |
| 12:19.13 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 12:20.08 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.40 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 12:22.52 | Kyler | ...and yes, I have "canreinvite=yes" for both the incoming and outgoing SIP peers. |
| 12:22.54 | Ariel_ | Morning everyone |
| 12:23.10 | darkskiez | Kyler: show appliction transfer |
| 12:23.50 | darkskiez | Kyler: might help |
| 12:23.55 | darkskiez | Kyler: not used it myselg |
| 12:24.03 | Kyler | But shouldn't Dial() do that anyway? |
| 12:24.11 | Ariel_ | humm last I knew for transfer to work you need asterisk to be in the mix. and you need canreinvite=no |
| 12:24.29 | darkskiez | Kyler: the reinvite causes the rtp stream to go direct, but asterisk still maintains the call setup teardown |
| 12:25.21 | Kyler | darkskiez: Oh...hmmm...I *hope* the RTP stream isn't going direct right now. There's way too much latency. How can I tell? |
| 12:25.42 | jake1932 | rtp debug |
| 12:25.51 | darkskiez | or tcpdump if u want to be sure too |
| 12:26.32 | RoyK | ethereal is nice |
| 12:26.39 | darkskiez | ethereal is lovely |
| 12:26.44 | Kyler | Ah ha! Yes, the RTP stream does seem to be going direct. Thanks for the tip! |
| 12:27.14 | Kyler | I'm getting a heck of a lot of latency for all of this happening at the provider. |
| 12:27.39 | darkskiez | out of curiosity, what provider/? |
| 12:27.47 | Kyler | Telesthetic |
| 12:29.23 | Ahrimanes | how long does a ring take? 5 seconds? |
| 12:29.39 | RoyK | Ahrimanes: varies. check indications.conf |
| 12:29.54 | RoyK | Ahrimanes: in norway, a ring is 1 second ring and 4 seconds pause |
| 12:30.21 | Ariel_ | In the US standard ring is 4.3 sec's so we us 5 as a round figure. |
| 12:31.32 | Ahrimanes | ah thanks |
| 12:31.34 | RoyK | Ahrimanes: 425/1000,0/4000 seems to be quite standard in europe |
| 12:31.52 | Ahrimanes | RoyK: 425/1000 ? |
| 12:32.13 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 12:33.27 | RoyK | iirc that means 425Hz for 1000ms |
| 12:33.38 | RoyK | then 4000ms silence |
| 12:33.45 | RoyK | Ahrimanes: see indications.conf |
| 12:33.57 | Ahrimanes | ah ok |
| 12:33.59 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 12:39.38 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 12:41.04 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 12:43.33 | AsteriskAlbania | asterisk and eyebeam |
| 12:43.41 | AsteriskAlbania | what do I neeed in asterisk for suport |
| 12:43.53 | AsteriskAlbania | for h623 codec |
| 12:44.10 | AsteriskAlbania | i have videosupport=yes |
| 12:44.33 | AsteriskAlbania | allow=h623 |
| 12:44.36 | AsteriskAlbania | allow=h623p |
| 12:44.46 | AsteriskAlbania | in sip.conf |
| 12:46.20 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 12:46.28 | Druken | morning everyone |
| 12:47.53 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 12:49.33 | Hmmhesays | hello |
| 12:49.47 | fjean | hi there |
| 12:49.54 | tzanger | morning Druken |
| 12:51.05 | AsteriskAlbania | what do I neeed in asterisk to suport video codec h623 h623+ |
| 12:51.39 | mut | no idear |
| 12:51.42 | mut | never used it |
| 12:52.43 | Druken | how is tzanger today ? |
| 12:52.48 | qdk | guess it wasnt that important. |
| 12:53.45 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-34-61.dsl.tul2ok.sbcglobal.net) |
| 12:54.46 | shiznatix | I am having trouble with using asterisk as a voice/fax switch. it works but it does not work until I sent the Hangup command |
| 12:54.53 | shiznatix | I have: http://pastebin.com/707198 but if I uncomment those lines then it does not go to the 'fax' extension until it is too late |
| 12:55.23 | *** join/#asterisk sflie (i=soulfly@anduin.net) |
| 12:59.17 | sflie | Hello, anyone here who can help me out with queue in asterisk? |
| 13:00.48 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 13:07.14 | Ironhand | how do I figure out whether a particular kernel would support running asterisk with realtime priority? |
| 13:08.37 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 13:09.52 | blue9 | Ironhand: Suck it and see. |
| 13:10.20 | *** join/#asterisk mog_home (n=mogorman@myskin.iet.unipi.it) |
| 13:10.55 | *** join/#asterisk clive- (n=pirch@dsl-146-64-134.telkomadsl.co.za) |
| 13:11.05 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 13:11.12 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 13:11.17 | Katty | morning. |
| 13:11.27 | littleball | hello, is it possible to connect asterisk to VoIP provider through SIP or H323? |
| 13:11.42 | magic_1 | to sip yes |
| 13:12.15 | littleball | how is h323? |
| 13:12.39 | magic_1 | i am sure that u can do it with H232 as well, i havent tried to yet havent needed to as yet |
| 13:12.59 | littleball | thanks. magic_1 |
| 13:13.23 | clive- | littleball stick to sip |
| 13:13.42 | littleball | ya. i try to. but some voip provider only support h3223 |
| 13:13.43 | jake1932 | mew Katty |
| 13:14.07 | magic_1 | who is the providee |
| 13:14.14 | magic_1 | i meant provider LOL |
| 13:14.32 | littleball | what is LOL? |
| 13:14.43 | blue9 | Laugh out Loud |
| 13:15.24 | Katty | jake1932: mew. |
| 13:15.26 | littleball | OK. :-). actually, i am thinking what is the difference ser express and asterisk if SIP is used |
| 13:15.32 | littleball | any comment? |
| 13:15.45 | littleball | i read some documents about ser express.... |
| 13:16.18 | magic_1 | ser express ? |
| 13:16.18 | clive- | ser is a sip proxy asterisk is a pbx |
| 13:16.20 | Katty | i think file is the SER person around here |
| 13:17.12 | littleball | it seems their functions are duplicated. at least some of the functions.. i already used asterisk e1 line connect to providers... but now try to use sip |
| 13:18.21 | *** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk) |
| 13:18.26 | magic_1 | hhmmm |
| 13:18.44 | scanna | hi |
| 13:19.30 | scanna | can someone explain me the difference between wrapuptime in agents.conf and queues.conf? |
| 13:19.58 | scanna | it seems to work only the one in queues.conf... |
| 13:20.20 | Hmmhesays | I use SER in a very simple fashion |
| 13:20.42 | Katty | Hmmhesays: YOU |
| 13:20.49 | Hmmhesays | me? |
| 13:20.51 | Katty | Hmmhesays: haven't talked to me in awhile |
| 13:20.53 | Katty | Hmmhesays: you snob. |
| 13:20.58 | Hmmhesays | been bizzay |
| 13:21.04 | Hmmhesays | got 2 huge projects going right now |
| 13:21.07 | Katty | excuses, excuses. |
| 13:21.11 | MikeJ[Laptop] | I make katty uncomfortable |
| 13:21.15 | Katty | MikeJ[Laptop]: you do. |
| 13:21.22 | MikeJ[Laptop] | by just being |
| 13:21.29 | Katty | MikeJ[Laptop]: yes. |
| 13:21.39 | Katty | MikeJ[Laptop]: are you proud of that? |
| 13:21.40 | Hmmhesays | MikeJ[Laptop]: was it you that was telling me about using sipsak for voicemail in an asterisk/ser mash |
| 13:21.41 | MikeJ[Laptop] | I got new bookshelves today |
| 13:22.02 | MikeJ[Laptop] | Hmmhesays, ummmm |
| 13:22.05 | MikeJ[Laptop] | don't think so |
| 13:22.06 | Hmmhesays | *voicemail notify's i should say |
| 13:22.16 | Hmmhesays | I can't remember who I was talking to about that |
| 13:22.21 | qdk | SER is quite populare in here today. |
| 13:22.29 | r_evolution | I guess that's why your name would be hmm he says... |
| 13:22.33 | MikeJ[Laptop] | it's a nice proxy |
| 13:22.52 | r_evolution | i havent messed with SER yet... i think that'll be my next little toy |
| 13:22.53 | Hmmhesays | I'm using external notify with sipsak to generate the notify message |
| 13:23.04 | clive- | from the little i know, ser has quite a few nice new features |
| 13:23.28 | clive- | its also known to be rock solidly stable |
| 13:23.32 | littleball | Hmmhesays, can you share your experience? Especially, how to scale the asterisk boxes (clustering more asterisk boxes) |
| 13:23.43 | r_evolution | yeah I read a bit about it yesterday... but not really enough to say I understand |
| 13:24.12 | littleball | how to combine ser with *? |
| 13:24.17 | Hmmhesays | i'm using it as a basic redirect server, using dns srv records to distribute calls amongst asterisk boxes |
| 13:24.54 | littleball | Hmmhesays, so basically, your clients are SIP phone users, right? |
| 13:25.01 | Hmmhesays | yes |
| 13:25.12 | littleball | do you enable authenticate in the ser? |
| 13:25.16 | Hmmhesays | no |
| 13:25.17 | littleball | authentication |
| 13:25.17 | Katty | and those who like to party, too. |
| 13:25.25 | Katty | Hmmhesays: do you consider them clients? |
| 13:25.38 | Hmmhesays | people who like to party? |
| 13:25.48 | littleball | that is to say, any users can spoil the contacts in ser, right? |
| 13:25.51 | Katty | your dedicated fans. |
| 13:26.03 | Hmmhesays | they're just fans |
| 13:26.09 | Katty | kay |
| 13:26.23 | Hmmhesays | littleball, most of this is on a private controlled network |
| 13:26.29 | Katty | my dad still gets fan mail. it's kinda weird. |
| 13:26.34 | Hmmhesays | haha |
| 13:26.47 | Hmmhesays | i started learning the solo's to sweet home alabama last night, ouch |
| 13:27.07 | Hmmhesays | feels like my wrist is going to fall off |
| 13:27.26 | Katty | which one? |
| 13:27.32 | Katty | left one? |
| 13:27.33 | Hmmhesays | left |
| 13:27.35 | mut | slit it too many times? |
| 13:27.40 | Katty | Hmmhesays: yeah it'll do that. |
| 13:27.51 | Katty | Hmmhesays: just wait till your hands start bleeding ;) |
| 13:27.54 | Hmmhesays | its not all that hard of a tune, just stuff i'm really not used to playing |
| 13:28.08 | *** part/#asterisk jcims (n=jcims@cpe-24-210-60-100.columbus.res.rr.com) |
| 13:28.16 | Hmmhesays | a lot of slightly off beat solo'ing |
| 13:28.53 | Katty | you'll get it. |
| 13:29.06 | littleball | Hmmhesays, how can you have such big volume in private network? |
| 13:29.08 | Katty | just takes some determination, a good ear, and a whole lotta perfectionism. |
| 13:29.26 | Katty | sometimes i have to play stuff for hours in a row until i'm happy with it. |
| 13:29.35 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:31.36 | *** join/#asterisk magic_1 (n=quinton@wbs-196-2-110-87.wbs.co.za) |
| 13:31.39 | Hmmhesays | littleball: the company i'm doing this for owns 400 locations, and the data networks at each |
| 13:32.50 | Hmmhesays | Katty: yeah I know it, I use guitar pro when i'm learning tunes and they have a looping trainer that increases the speed every loop |
| 13:33.48 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.118.Dial1.SanJose1.Level3.net) |
| 13:34.02 | Katty | Hmmhesays: can you play by ear? |
| 13:34.26 | Hmmhesays | yeah rhythm stuff, i usually grab the guitar pro file for the solo's though |
| 13:34.35 | Katty | ah |
| 13:34.41 | *** join/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 13:34.48 | Hmmhesays | just because that is the most awesome learning program ever written |
| 13:34.50 | Katty | that's half the battle down then...once you know exactly how the solo goes. |
| 13:34.52 | imperfect- | Anyone know how I can ring 2 SIP channels at once? |
| 13:34.55 | *** part/#asterisk blue9 (n=chatzill@host213-123-130-180.in-addr.btopenworld.com) |
| 13:35.11 | Katty | SIP/foo&/SIP/wocka |
| 13:35.12 | Hmmhesays | dial(tech/host1&tech/host2) |
| 13:35.20 | imperfect- | sweet! |
| 13:35.21 | imperfect- | thanks |
| 13:35.22 | imperfect- | ;) |
| 13:35.25 | imperfect- | i knew it had to be something simple |
| 13:35.26 | imperfect- | thank you |
| 13:35.34 | Hmmhesays | Katty: its not just for guitar either |
| 13:35.39 | Katty | Hmmhesays: no? |
| 13:35.59 | Hmmhesays | I'll show you a screen shot |
| 13:36.01 | Katty | k |
| 13:37.42 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:38.04 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:38.24 | jake1932 | anyone good with q.931 coding? I'm trying to send a route select message (granted this is not asterisk specific - so i'm reaching a bit) |
| 13:38.49 | *** join/#asterisk Schwuk (n=Schwuk@84.12.166.117) |
| 13:43.02 | imperfect- | last tech/ |
| 13:51.10 | *** join/#asterisk pulss (n=pulkk@81.10.35.247) |
| 13:51.15 | pulss | hello |
| 13:51.40 | pulss | I have a major "clicking" problem between a tdm400 card and sip calls |
| 13:51.45 | pulss | can anyone help? |
| 13:52.36 | darkskiez | pulss: use your free tech support ticket you get with the card. |
| 13:53.52 | pulss | the thing is I am not in the US. Can I use emails for that? |
| 13:54.05 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 13:55.03 | wasim | pulss: yes |
| 13:56.16 | pulss | thanks guys |
| 13:56.29 | pulss | meanwhile, is there any help you can provide me here? |
| 13:57.08 | wasim | check your power supply |
| 13:57.19 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-185-148.lsanca.fios.verizon.net) |
| 13:57.26 | *** join/#asterisk milestone (n=buddy@p54A7BFE4.dip0.t-ipconnect.de) |
| 13:57.54 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 13:57.55 | milestone | hi all |
| 13:58.26 | pulss | well i did. as a matter of fact, i tested the same exact configuration and setup and even the hardware cards on 2 totally differnt machines. one with a 1GHZ C3 process, the other with a p4 2.4GHZ with APIC support |
| 13:58.35 | pulss | and I get the same exact clicks |
| 13:59.06 | pulss | I only get the clicks when the bridge between the tdm and the sip call start |
| 13:59.13 | milestone | i am a complete newbie, and been searching the web on how to setup asterisk to be a sip gateway using a hisax_fcpcipnp: Fritz!Card PCI/PCIv2/PnP ISDN driver v0.0.1 ISDN Card in Germany |
| 13:59.23 | milestone | does anyone have a good howto for that? |
| 13:59.31 | pulss | when I do calls between an fxo and an fxs ports, i get super clear audio |
| 14:07.12 | *** join/#asterisk ddn_ (n=Daniel@200.84.67.165) |
| 14:07.18 | ddn_ | hi all |
| 14:07.32 | ddn_ | where can I find a tutorial on VoIP |
| 14:08.02 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 14:08.51 | jake1932 | voip-info.org has plenty of info (including tutorials) |
| 14:09.12 | milestone | jake1932: including answers to my questions? |
| 14:09.32 | brif8 | anyone tried the sip_ping.pl from "VoIP Hacks" it works to the * server but fails with an alarm to snom IP Phone ? both on same LAN and subnet |
| 14:09.44 | r_evolution | dammit... my eye is twitching :( |
| 14:09.53 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 14:09.59 | jake1932 | milestone: quite possibly |
| 14:10.28 | milestone | asterisk -vvvgc is giving me May 9 16:04:26 WARNING[24578]: loader.c:440 load_modules: Loading module chan_features.so failed! --- What The ****? |
| 14:10.38 | milestone | what is that module? |
| 14:11.43 | *** join/#asterisk magic_1 (n=quinton@wbs-196-2-101-116.wbs.co.za) |
| 14:15.02 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:15.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:16.19 | ddn_ | jake1932, hey ty |
| 14:16.36 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 14:18.58 | jaybuffet | hello.... our phone company is coming into the office to convince us to stay with them and not go voip... we have about 30 people in our company and about 8 on the phone at any given time... i belive we have a partial t1... how long would it take to set up an asterisk system, how much would it cost (approx. mid range equip) and would asterisk be a good solution for us? |
| 14:19.12 | mut | my god |
| 14:19.19 | jaybuffet | :-] |
| 14:19.20 | mut | these sangoma cards must work great for everyone else but me |
| 14:19.32 | mut | i've been on the phone with the ONE tech support guy they have for 40 minutes now |
| 14:19.37 | mut | hasn't gotten another call yet |
| 14:19.38 | *** join/#asterisk Juggie (i=Juggie@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 14:20.32 | mut | i have such bad luck |
| 14:22.19 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 14:22.43 | *** join/#asterisk camelon (n=chiardon@200.71.58.39) |
| 14:22.50 | camelon | Hello!!! |
| 14:23.03 | milestone | camelon: hello |
| 14:23.12 | milestone | is it me you're looking for ;) |
| 14:23.27 | camelon | I hope that!! |
| 14:23.44 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:23.45 | brif8 | jaybuffet: depends on your linux skills assuming you have a spare linux machine about 1-2 hours, costs +/- $ 500 |
| 14:23.49 | milestone | i can see it in your eyes |
| 14:24.41 | r_evolution | asterisk is always a good solution :-D |
| 14:25.23 | camelon | I used to have a frecuent problem with some extensions that suddenly are giving the busy messege! What I can do to overcome this embarrising situation? TIA |
| 14:25.37 | r_evolution | wow... here's a scary one... the guy here who builds web-sites in FrontPage wants to be my backup for administration of the two * servers here that run the VoIP platform for the customers |
| 14:25.39 | ddn_ | jake1932, is it hard to set a VoIP server? |
| 14:25.48 | *** part/#asterisk clive- (n=pirch@dsl-146-64-134.telkomadsl.co.za) |
| 14:25.52 | camelon | milestone . . perhaps you? |
| 14:26.29 | milestone | camelon: what is your setup? |
| 14:26.42 | camelon | and . . . wich could be the best alternative to put the extensions up? |
| 14:27.24 | jaybuffet | brif8: linux skills = good at following directions... no spare machine would need to purchase.. how beefy of a machine would i need ? |
| 14:27.27 | ManxPower | ~thebook |
| 14:27.29 | jbot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 14:28.00 | camelon | milestone . . 2 E1s . . . 2 (T1) channel banks . . . and 2 IP extensions thet never have this situation |
| 14:28.08 | r_evolution | yeah that's what i gave him Manx |
| 14:28.28 | brif8 | jaybuffet: for 8 people will you use IP phones as well or the stand phones ? |
| 14:28.36 | ManxPower | Source Forge's CVS servers SUCK |
| 14:29.19 | jaybuffet | brif8: mix of stand and IP phones... 8 simultaneous |
| 14:29.28 | imperfect- | anyone know why NoOP(Caller id is ${CALLERID}) doesn't work? |
| 14:29.36 | milestone | camelon: dunno |
| 14:30.03 | camelon | milestone . . .sure? |
| 14:30.33 | brif8 | jaybuffet: how many though IF you went just IP phones then you would need a smaller machine, if you had a high number of std. phones then you would need a bigger machine to handle the codec work between VoIP/* and the std phone |
| 14:30.37 | milestone | camelon: jupp |
| 14:30.49 | camelon | happppppp brffff!! |
| 14:31.14 | mut | HE HAD NO IDEA WHAT WAS WRONG?! |
| 14:31.20 | mut | AHHHHHHHHHHHHHHHHHHHHHH |
| 14:31.25 | mut | on the phone for 40 minutes |
| 14:31.36 | mut | and the only sangoma tech support guy there is has no idea whats wrong with this |
| 14:31.38 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-63.claranet.co.uk) |
| 14:31.47 | jpabuyer | do you guys use an IDE to program asterisk? |
| 14:32.29 | jaybuffet | brif8: i would say 7 on std and 1 on ip at any given time (in the beginning) |
| 14:32.47 | r_evolution | hey mut... |
| 14:32.51 | r_evolution | how does that make you feel? |
| 14:32.53 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:33.01 | r_evolution | When even Tech Support for the product can't help you? :-D |
| 14:33.03 | mut | like all voip companys hardware sucks |
| 14:33.11 | mut | except cisco |
| 14:33.16 | mut | i have to go back to my old cisco setup |
| 14:33.21 | mut | and eat this $2500 card |
| 14:33.38 | brif8 | jaybuffet: where are you located ? |
| 14:33.50 | jaybuffet | brif8: US |
| 14:34.01 | ddn_ | mut, a VoIP server has to have a hired company in the US? |
| 14:34.09 | brif8 | jaybuffet: I can see that se.rr.com where ? |
| 14:34.22 | jaybuffet | brif8: tampa, fl |
| 14:34.23 | mut | ddn_? |
| 14:34.27 | littleball | hello |
| 14:34.50 | brif8 | jaybuffet: I'm coming to Tampa on Thursday, if you want I can stop by and discuss it ? |
| 14:35.09 | ddn_ | mut, totally new at VoIP. I understand have to set a server that connects to a service company. Am I right? |
| 14:35.13 | littleball | i want to config my asterisk sip.conf so that my asterisk connect to sip provider. how to configure the authentication ? |
| 14:35.34 | ddn_ | mut, provider company yes |
| 14:35.44 | r_evolution | poor mut :) |
| 14:35.49 | ddn_ | mut, that happens when I am new. |
| 14:35.58 | r_evolution | hey littleball... you should just register it to your provider :) |
| 14:36.04 | mut | ddn_: don't but any t1 cards |
| 14:36.08 | qdk | littleball: http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
| 14:36.10 | mut | thats my advice |
| 14:36.15 | mut | buy |
| 14:36.21 | camelon | I used to have a frecuent problem with some extensions that suddenly are giving the busy messege! What I can do to overcome this? |
| 14:36.41 | littleball | r_evolution, register=>1234@mysipprovider.com/12222 |
| 14:36.45 | littleball | right/ |
| 14:36.46 | littleball | ? |
| 14:37.09 | littleball | then how to define the codec? |
| 14:37.10 | qdk | camelon: hangup the phone. :-P |
| 14:37.37 | qdk | littleball: http://www.voip-info.org/wiki-Asterisk+config+sip.conf <- search for codec |
| 14:37.41 | r_evolution | not exactly :) |
| 14:37.45 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.224.Dial1.SanJose1.Level3.net) |
| 14:38.03 | *** part/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 14:38.09 | r_evolution | register => SIP:PASSWORD@PROVIDER |
| 14:38.23 | r_evolution | then you've got to have somewhere for the sip to come into in your extensions file |
| 14:38.39 | r_evolution | best for you to follow qdk's advice :) |
| 14:38.49 | r_evolution | click the link... it's a page that explains what you're trying to do |
| 14:38.55 | camelon | qdk . . . it use to happen with all the extensions nothing related with the hard phone manipulation!! |
| 14:39.04 | littleball | thanks |
| 14:39.27 | qdk | camelon: maybe the connection to the sip is bad/down? |
| 14:39.27 | r_evolution | if you still have questions after reading that... then come back and ask... people will be glad to help you |
| 14:39.30 | littleball | i am reading handbook-draft now actually. it is not clear. |
| 14:39.43 | r_evolution | hey littleball... just read the book |
| 14:39.44 | nahirean | someone dcc me cure_hangover.c plz |
| 14:39.50 | r_evolution | it'll give you a better start :) |
| 14:39.52 | r_evolution | then read the wiki |
| 14:39.58 | r_evolution | then just play with it |
| 14:40.29 | r_evolution | hey camelon... why not try checking the peer in the CLI? |
| 14:40.34 | qdk | littleball: feel free to ask for elaboration of a specifik part of the documentation. |
| 14:40.53 | mut | and i'm having credit card processor problems too |
| 14:40.55 | camelon | qdk they aren't sip extensions . .use to happen with the zap extensions . .never with the SIPs |
| 14:41.12 | r_evolution | are you mut? You mean sending it over SIP? |
| 14:41.17 | mut | no |
| 14:41.22 | r_evolution | oh |
| 14:41.27 | mut | i mean the credit card company isn't processing it right |
| 14:41.30 | r_evolution | oh |
| 14:41.31 | r_evolution | haha |
| 14:41.34 | mut | and now it's not working AT ALL |
| 14:41.40 | r_evolution | your day is just sucking so far, huh? |
| 14:41.47 | mut | fuking rediculous |
| 14:42.02 | mut | these people are going to get bitched at so bad |
| 14:42.11 | qdk | camelon: not sip extentions? what do you mean? extentions are not bound to any particular tech. |
| 14:42.15 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:43.05 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 14:43.41 | r_evolution | hey mut... if it makes you feel any better |
| 14:43.55 | r_evolution | the guy who wants/thinks/intends to be my backup for the * boxes here |
| 14:44.02 | r_evolution | is the guy who builds web-sites in FrontPage :) |
| 14:44.12 | littleball | qdk, example, in the handbook-draft, it saild that my asterisk can register itself with another sip server under general secion of the sip.conf. Also, it said that we can define "peer" type in "entity section" and "peer" refers to sip provider. |
| 14:44.20 | r_evolution | i gave him the book... and said that'll take you about a day right? he said... no man... it'll take me 3 or 4 months |
| 14:44.25 | ManxPower | We assign the SIP userid/password of each of our SIP devices to be the MAC of the device. This forces us to NOT think of extensions=user=device. |
| 14:44.32 | mut | heh |
| 14:44.34 | camelon | qdk . .2E1s . . .2 (t1) chjannel bank . . .2 ATAs |
| 14:44.36 | r_evolution | :-D |
| 14:44.43 | mut | my hand is shaking |
| 14:44.46 | mut | i'm too ma |
| 14:44.47 | mut | dd |
| 14:44.51 | mut | >:| |
| 14:45.03 | r_evolution | so basically... management thinks the box i run... is about the equiv of a front-page site :) |
| 14:45.14 | r_evolution | guess what? asterisk has no pretty clicky buttons to generate shitty code that only half-works |
| 14:45.16 | r_evolution | :-D |
| 14:45.23 | Lino` | :D |
| 14:45.25 | ManxPower | *grumble* I need an affordable bulletproof day/night security camera. |
| 14:45.30 | r_evolution | haha... my eye is twitching... so let's go drink together |
| 14:45.32 | Lino` | except you're using AMP of course |
| 14:45.39 | r_evolution | nah |
| 14:45.52 | Lino` | but thats stupid |
| 14:45.53 | Lino` | ;) |
| 14:45.59 | r_evolution | well... like i said... |
| 14:46.04 | Lino` | dont speak of drinking, already had a few beers |
| 14:46.04 | r_evolution | this is the guy who builds web-sites in front page |
| 14:46.05 | Lino` | :-P |
| 14:46.17 | r_evolution | he's going to be my back-up :-D |
| 14:46.41 | Lino` | lol |
| 14:46.54 | qdk | littleball: yes, that sounds just about right... i only use IAX between asterisk so i dont know the specifics, but the link should help you a lot. |
| 14:46.55 | r_evolution | translation : if they piss me off to the point where i walk out... they want the guy who can't even use DreamWeaver to be my backup |
| 14:46.56 | Lino` | * TheFrontpageGuy sees the extensions.conf for the first time |
| 14:47.04 | Lino` | <TheFrontpageGuy> Holy Smoke! |
| 14:47.08 | Lino` | * TheFrontpageGuy dies |
| 14:47.14 | r_evolution | noooooo kidding |
| 14:47.20 | r_evolution | esp. the way i've got things setup ;x |
| 14:47.35 | Lino` | fronpage users even die when they see normal HTML |
| 14:47.35 | littleball | qdk. thanks. i thnk i can clear it myself. |
| 14:47.37 | r_evolution | fuck y0 comments! it was a bitch to put in... it damn well be a bitch to understand! |
| 14:47.44 | qdk | camelon: do you mix T1 and E1? |
| 14:47.47 | Lino` | <JohnDoeFrontpageUser> wtf is HTML? |
| 14:48.03 | r_evolution | Ach Tee Em El? Is that a new program from Microsoft? |
| 14:48.10 | Lino` | must be frontpage-related |
| 14:48.15 | Lino` | like a service pack or something *gg* |
| 14:48.29 | r_evolution | heh... im amused... |
| 14:48.34 | r_evolution | esp. b/c the book was maybe |
| 14:48.36 | r_evolution | step one :) |
| 14:48.39 | r_evolution | on this switch :-D |
| 14:48.48 | Lino` | who needs a book about frontpage? |
| 14:48.51 | qdk | camelon: have you done any systematic testing? to eliminate to possibility of a (semi)broken ATA and such. |
| 14:48.54 | ManxPower | Maybe a gun would be better than a security camera. The rednecks have done these things to our mailbox: shot, ran over, dragged it down the road, and finally stole it. |
| 14:49.11 | Lino` | :D |
| 14:49.22 | Lino` | actually you dont even need locks in the doors |
| 14:49.33 | Lino` | just a good ol' rifle |
| 14:49.39 | Lino` | and a string |
| 14:49.47 | Lino` | someone opens the door and booom |
| 14:50.04 | r_evolution | im talking about the * book lino |
| 14:50.08 | Lino` | oh |
| 14:50.09 | Lino` | ok |
| 14:50.11 | r_evolution | haha where do you live Manx? |
| 14:50.18 | Lino` | sounds like texas *g* |
| 14:50.19 | Hmmhesays | what happens when grandma stops by with a suprised cake |
| 14:50.38 | ManxPower | r_evolution, top of a mountian in North Central Alabama |
| 14:50.42 | camelon | qdk . .yeppp it use to hapen only with some zap channels!! |
| 14:50.44 | Lino` | ok |
| 14:50.45 | r_evolution | He said it'll take him about 3 -4 months to read that |
| 14:50.50 | r_evolution | ahhhh manx. |
| 14:50.56 | Lino` | as long as the KKK stays away, everything is fine i guess |
| 14:50.56 | r_evolution | hey hmm... when grandma stops by with the cake |
| 14:51.00 | r_evolution | you get your inheritance |
| 14:51.00 | r_evolution | :-D |
| 14:51.17 | ManxPower | We suspect it's young men. So shooting up the truck, which is prolly their father's, would do most everything we need. |
| 14:51.33 | Lino` | *g* |
| 14:51.36 | r_evolution | well... |
| 14:51.38 | r_evolution | my advice? |
| 14:51.42 | r_evolution | dig a DEEP DEEP fucking hole |
| 14:51.47 | ManxPower | along with a video tape of them destroying a mailbox, which is a federal offence. |
| 14:51.47 | Lino` | just be sure to use pretty bad bullets |
| 14:51.48 | r_evolution | fill it with concrete |
| 14:51.57 | r_evolution | put a solid steel pole IN said hole... |
| 14:52.05 | r_evolution | also filled with concrete... |
| 14:52.16 | r_evolution | mount mailbox with steel reinforcements on either side. |
| 14:52.19 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:52.21 | r_evolution | let them hit that :-D |
| 14:52.29 | r_evolution | vehicle = total loss on that one |
| 14:52.32 | qdk | camelon: it sounds like a very unstable system... have you gone through log of both asterisk and the system? |
| 14:52.35 | Lino` | oh you have video footage? |
| 14:52.51 | ManxPower | Lino`, not yet. But the conduit has been buried. |
| 14:53.05 | Lino` | *g* |
| 14:53.11 | Lino` | so right |
| 14:53.15 | Lino` | i remember now |
| 14:53.20 | ManxPower | the gate is about 900ft from the main house. |
| 14:53.25 | Lino` | USPS has a monopoly on the USPS mailboxes right? |
| 14:53.32 | Lino` | and they are like property of usps |
| 14:53.39 | ManxPower | Lino`, correct. |
| 14:53.41 | Lino` | kk |
| 14:53.49 | Lino` | i'm from germany, so i dont know the us-laws |
| 14:53.57 | ManxPower | even though you have to buy the mailbox. |
| 14:54.00 | Lino` | but i remember that UPS is not allowed to use the mailboxes |
| 14:54.12 | elvisthedj|work | For all those concerned, you'll be happy to know that I got my 7940 firmware upgraded after a mere 5 months |
| 14:54.14 | Lino` | at least the ones with USPS on them |
| 14:54.16 | ManxPower | Lino`, ONLY USPS is allowed to use USPS mailboxes. |
| 14:54.21 | Lino` | yeah |
| 14:54.21 | Lino` | ;) |
| 14:54.47 | Lino` | in germany the "Deutsche Post AG" is the only company allowed to deliver mail (letters) at all |
| 14:54.56 | Lino` | except for bicycle couriers |
| 14:55.17 | coppice | which is sneaky, when they own DHL |
| 14:55.21 | ManxPower | Lino`, that is *technically* true here. Only USPS is allowed to deliver 1st class letters. |
| 14:55.26 | Lino` | :D |
| 14:55.30 | Lino` | DHL is something different |
| 14:55.35 | Lino` | because DHL does parcels |
| 14:55.35 | ManxPower | but anyone can deliver "urgent" letters. |
| 14:55.38 | Lino` | :D |
| 14:55.46 | Lino` | yeah, you can still use FedEx |
| 14:55.49 | Lino` | just like intel does it |
| 14:55.55 | Lino` | but they are fricken expensive |
| 14:55.58 | elvisthedj|work | my dad went to prison for delivering a letter |
| 14:56.04 | Lino` | where? |
| 14:56.24 | elvisthedj|work | in my mind |
| 14:56.36 | Lino` | they have prisons in there? ^^ |
| 14:56.46 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 14:56.56 | Lino` | its not that bad here, you pay a fine and thats it |
| 14:56.57 | Lino` | :;D |
| 14:57.03 | camelon | qdk . .yeeepppp . . but without any good clue!!! |
| 14:57.32 | qdk | camelon: ok, im out of ideas. sorry... ang gotta go. |
| 14:57.35 | qdk | and* |
| 14:57.54 | camelon | qdk . . .TIA |
| 14:59.01 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:59.17 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 15:00.24 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 15:01.10 | stack_ | Does anyone have the voicemail indicator working on a Polycom phone? I can't find how to get it to work |
| 15:01.30 | ManxPower | stack_, it works by default. |
| 15:01.36 | rpm | is anyone here in calgary? |
| 15:01.50 | ManxPower | in the sip.conf [devicesection] mailbox=voicemailbox@voicemailcontext |
| 15:01.56 | stack_ | ManxPower: well, I've done something that negates that :) |
| 15:02.11 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 15:02.53 | stack_ | ManxPower: ah I see, I didn't set that up |
| 15:03.22 | ManxPower | stack_, Yeah, voicemail inidcator doesn't usually work if you don't set it up. |
| 15:03.35 | [TK]D-Fender | stack_ : It's buried not-so-deep in th FM ;) |
| 15:03.44 | mikefoo | anyone know of a way to transcode/convert 711 to 729? |
| 15:03.46 | stack_ | ManxPower, I was going through all of the Polycom config files :) |
| 15:03.48 | mikefoo | even possible? |
| 15:04.04 | ManxPower | mikefoo, Yes. You purchase the G729 codec license from Digium. |
| 15:04.58 | mikefoo | only $10? |
| 15:05.21 | stoffell | hm, what can the ACD softkeys be used for on the Polycom 501 ? |
| 15:07.37 | pif | can one combine auth + extension info in a SIP Dial string? |
| 15:07.57 | pif | I can't find the synthax for that |
| 15:08.06 | *** join/#asterisk tdonahue (n=tdonahue@www.vonworldwide.com) |
| 15:08.50 | tdonahue | i'm having some svn problems... is this correct to get the test-this-branch branch? "svn checkout http://svn.digium.com/svn/asterisk/team/oej/test-this-branch asterisk" |
| 15:11.02 | [TK]D-Fender | stoffell : not yet |
| 15:11.32 | *** join/#asterisk fugitivo (n=ajf@201.255.176.12) |
| 15:11.34 | fugitivo | hello |
| 15:12.23 | tdonahue | bah, nevermind i broke my dns resolver on that box |
| 15:12.27 | sevard | [TK]D-Fender: HIGH FIVE! ALRIGHT! |
| 15:13.16 | stoffell | [TK]D-Fender, too bad :) |
| 15:14.20 | *** join/#asterisk JackEStorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 15:15.02 | brodiem | Is there a better text-to-speech app then Festival? I don't need to be able to dynamically create the speech with an app, just a util to generate WAVs |
| 15:15.37 | *** part/#asterisk milestone (n=buddy@p54A7BFE4.dip0.t-ipconnect.de) |
| 15:15.57 | MikeJ[Laptop] | heh |
| 15:16.38 | *** join/#asterisk holaaa (n=holaa@85.137.83.66) |
| 15:16.48 | myiagy | brodiem festival has an app called text2wave that does that.. |
| 15:16.52 | ManxPower | brodiem, Cepstral |
| 15:17.12 | *** join/#asterisk cstomi (n=chatzill@22-36.adsl.etel.hu) |
| 15:17.17 | ManxPower | There are others. Cepstral is the most affordable for decent TTS |
| 15:17.29 | brodiem | myiagy yeah I know, I just didn't like the speech output of festival and wanted to know if there was something better |
| 15:17.49 | Faithful | Anyone got Asterisk@Home receiving vaxes and converting to email? |
| 15:17.52 | brodiem | ManxPower thanks, I know I stumbled on their site before and couldn't remember who it was :) |
| 15:18.35 | coppice | i thought all vaxes were converted to scrap these days :-) |
| 15:19.00 | jsharp | Nah, I've got one in my basement. |
| 15:19.19 | Faithful | Ok I made a typo... |
| 15:19.34 | holaaa | Using TDM400. My dialplan answers the call, but if caller o calling do nothing until the system disconnects, the phone call does not hang up or hangs up minutes later. should I place a exten => t, Hangup? would it solve the problen? where exactly in my dialplan? |
| 15:20.31 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 15:20.41 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 15:20.43 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 15:20.48 | Hmmhesays | so i'm testing this sip/skype bridge |
| 15:20.55 | Hmmhesays | anyone on skype right now that can call me? |
| 15:21.10 | sevard | Does anyone know if you can ControlPlayback an entire directory without having to entere in each file? like ControlPlayback(/some/directory,4000,#,*,8,0) |
| 15:21.25 | sevard | *playback every file in a directory |
| 15:21.39 | sevard | does * allow wild cards? it'd seem silly if it didn't :) |
| 15:23.33 | Hmmhesays | sevard |
| 15:23.35 | Hmmhesays | you on skype |
| 15:23.36 | Faithful | Hmmhesays: is that software? |
| 15:23.50 | sevard | i suppose i could write a script to list the directory and then loop with the variable in there |
| 15:23.51 | Hmmhesays | yeah |
| 15:24.05 | sevard | Hmmhesays: ha, if only my hardware could support that POS program. I'm running on 233s man |
| 15:24.13 | Hmmhesays | bwhahaah |
| 15:24.27 | sevard | dumpster diving for life, brother. |
| 15:24.28 | Hmmhesays | i got it hooked up to my asterisk right now |
| 15:24.37 | sevard | i didn't think that was possible |
| 15:24.45 | Hmmhesays | oh it is |
| 15:24.48 | sevard | heh |
| 15:24.56 | sevard | give me a sip trunk and i'll test it |
| 15:25.13 | Hmmhesays | i need to test incoming from the skype network |
| 15:25.16 | Hmmhesays | outgoing works fine |
| 15:25.29 | sevard | well give me a sip trunk on the skype network ;) |
| 15:25.37 | Hmmhesays | no. |
| 15:25.40 | sevard | haha |
| 15:26.22 | sevard | are you just piping audio or what are you doing |
| 15:26.34 | Hmmhesays | basically, skype allows for external control |
| 15:26.53 | Hmmhesays | so you have a sip client attached to the skype client |
| 15:28.16 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 15:28.38 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 15:28.39 | DeeJay[2] | hi |
| 15:28.49 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 15:28.52 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 15:28.53 | Micetto | hi |
| 15:28.56 | Micetto | ^_^ |
| 15:29.14 | holaaa | Using TDM400. My dialplan answers the call, but if caller o calling do nothing until the system disconnects, the phone call does not hang up or hangs up minutes later. should I place a exten => t, Hangup? would it solve the problen? where exactly in my dialplan? |
| 15:29.16 | Micetto | I have a problem with queue :) |
| 15:29.23 | Micetto | can someone help me ? |
| 15:29.32 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 15:29.36 | a1fa | yo yo yo |
| 15:29.37 | a1fa | ;P |
| 15:29.41 | sevard | Hmmhesays: put up another skype client and give me a sip trunk |
| 15:29.42 | a1fa | anybody using enum lookups? |
| 15:29.57 | sevard | Hmmhesays: just don't give me skypeout/in or something silly like that |
| 15:29.59 | Hmmhesays | sevard no. |
| 15:30.10 | sevard | why not bizzilch |
| 15:30.11 | a1fa | sevard : he just sayd <Hmmhesays> so you have a sip client attached to the skype client |
| 15:30.12 | Hmmhesays | that would be pain in the @$$ |
| 15:30.24 | a1fa | s/sayd/said/ |
| 15:30.25 | sevard | you're a pain in the ass for not doing it |
| 15:30.25 | DeeJay[2] | Suppose you have a secretary which receives a phone call for you... she transfer the call but is first talking to you to ask you if you want to take the call. While this moment, the transferred person is on music on hold.... when you accept the transfer, the secretary "really" make the communication and tell: You are now in communication... and then she leaves the communication while keeping the communication between you and the client... |
| 15:30.30 | DeeJay[2] | How do we call this kind of transfer?? |
| 15:30.37 | DeeJay[2] | We want to achieve it with polycoms and asterisk |
| 15:30.41 | Hmmhesays | attended transfer |
| 15:30.59 | a1fa | DeeJay[2] attended transfer |
| 15:31.07 | a1fa | ie. not blind transfer |
| 15:31.16 | a1fa | blind transfer is when you just kick somebody off |
| 15:31.20 | sevard | DeeJay[2]: get the call, flash, dial the person you want to transfer to, say DUDE LOLZ CALL KAY, hang up |
| 15:31.58 | DeeJay[2] | we would like to call only once your cell phone... |
| 15:32.04 | a1fa | ? |
| 15:32.08 | a1fa | wtf |
| 15:32.16 | sevard | i think he's using an online translator |
| 15:32.20 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
| 15:32.28 | DeeJay[2] | err ;) |
| 15:32.30 | DeeJay[2] | Lol.... |
| 15:32.43 | sevard | this guy got mad at me once |
| 15:32.43 | DeeJay[2] | I mean... we don't want to have to call your cell phone twice... |
| 15:32.45 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 15:32.51 | DeeJay[2] | Nor having to ask you to call back someone... |
| 15:32.53 | sevard | "fuck you child of bitch, what do you think about!?" |
| 15:33.20 | a1fa | DeeJay[2] : ur an idiot |
| 15:33.27 | DeeJay[2] | .... |
| 15:33.27 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 15:33.28 | a1fa | he just explained it to you how to make a transfer |
| 15:33.30 | sevard | DeeJay[2]: you get a call in you want to put it on hold, get somebody on the phone that you're transfering to and say "call, mang" and transfer, right? |
| 15:33.50 | DeeJay[2] | yeah...transferring..but also saying: You are now in communication |
| 15:34.00 | DeeJay[2] | so that both of you don't wait the other to say: "Hello" |
| 15:34.03 | a1fa | ok |
| 15:34.07 | a1fa | so flash again |
| 15:34.08 | DeeJay[2] | Because customers tends to still wait even if the music on hold stops.. |
| 15:34.08 | a1fa | and hang up |
| 15:34.15 | a1fa | ok |
| 15:34.32 | a1fa | so get a call, say hold, flash, dial your master, talk, flash back, say YOU GUYS SUCK |
| 15:34.33 | a1fa | and hung up |
| 15:34.39 | sevard | DeeJay[2]: do you want * to say "you are now in communication" or do you want the secretary to say it |
| 15:34.47 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 15:35.34 | sevard | YOU GUYS FRICKEN SUCK OKAY, YOU ARE NOW IN COMMUNICATION |
| 15:35.41 | r_evolution | man. |
| 15:35.42 | r_evolution | see |
| 15:35.47 | r_evolution | this is why i live here |
| 15:35.59 | r_evolution | because you guys crack me up :-D |
| 15:36.25 | sevard | paypal plz |
| 15:36.38 | salviadud | haha |
| 15:36.48 | salviadud | flashing with asterisk is cool |
| 15:36.58 | r_evolution | don't spend it all in one place :) |
| 15:37.00 | Faithful | Hmmhesays: what's your skyp id |
| 15:37.02 | *** join/#asterisk ckwall (n=ckwall@63.149.122.94) |
| 15:37.04 | Hmmhesays | hmmhesays |
| 15:37.14 | sevard | Hmmhesays: give me a sip trunk |
| 15:37.16 | sevard | bizzilch |
| 15:37.18 | znoG | question: if I do a Dial(${EXTEN}@foo) and I have a [foo] section in sip.conf with IP info, username, secret, etc.. will it use the username/secret to auth to the remote server? |
| 15:37.21 | Hmmhesays | stfu n00b |
| 15:37.25 | salviadud | somebody called me the other day, they were from the phone company, so i told them... please hold while i transfer you to brasil |
| 15:37.27 | sevard | lollercaust |
| 15:37.37 | salviadud | needles to say, they don't know portuguese |
| 15:37.38 | sevard | salviadud: hahahaha |
| 15:37.48 | [TK]D-Fender | znoG : that will get you nowhere.... |
| 15:38.03 | [TK]D-Fender | znoG : You need to specifiy the technology first |
| 15:38.09 | sevard | [TK]D-Fender: fancy pants dialplan? |
| 15:38.13 | ckwall | I am confused by a few things... I have asterisk running just fine, I am placing and receiving calls across my t1. But I never set anything up in SIP.conf for my users. how is this working? shouldnt I have had to have an entry for every phone in sip.conf? |
| 15:38.16 | znoG | [TK]D-Fender: sorry, i meant SIP/${EXTEN}@foo) |
| 15:38.52 | *** join/#asterisk kph100 (n=kph100@206-248-130-182.dsl.teksavvy.com) |
| 15:38.56 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 15:39.09 | ckwall | I am using the polycom soundpoint ip 501 |
| 15:39.29 | r_evolution | hahah @ X-Gen |
| 15:39.38 | gandhijee | what's the file to change to change the echo canceller zaptel uses? |
| 15:39.40 | Faithful | Hmmhesays: "Unknown" problem |
| 15:39.43 | af_ | ah. |
| 15:39.52 | Hmmhesays | try it one more time Faithful |
| 15:39.55 | Hmmhesays | i was on an echo test |
| 15:40.15 | ckwall | it was my understanding that for each phone connection, i should have a username and password to make them connect. |
| 15:40.24 | [TK]D-Fender | znoG : yes if you set up the auth, but you should use it like Dial(SIP/foo/${EXTEN}) |
| 15:40.32 | ckwall | all i did was plug them in and specify the server on the phone and they started working. |
| 15:40.47 | [TK]D-Fender | ckwall : Yes, you need an entry for each phone |
| 15:41.05 | ckwall | Fender: How is this working without it? |
| 15:42.01 | Hmmhesays | Faithul: i think I fixed it |
| 15:42.47 | Faithful | "Reason Unknown" |
| 15:43.05 | ckwall | sorry, I meant to ask that of TK |
| 15:43.36 | *** join/#asterisk Assid (n=assid@203.115.83.213) |
| 15:43.59 | znoG | [TK]D-Fender: in a scenario where i have 2 asterisk boxes, and they need to call each other (ie. phone1 -> asterisk box 1 -> internet -> asterisk box 2 -> phone2) and vice versa, would I need to create 2 entries on each asterisk box? (one for box1->box2 and box2->box1 and same on the other end) |
| 15:44.00 | Faithful | Hmmhesays: try calling me revelator310 |
| 15:44.02 | Assid | err.. is there a way to reduce the time it waits before it jumps to the next priority |
| 15:44.07 | Faithful | oops |
| 15:44.10 | Hmmhesays | i know my outbound works |
| 15:44.16 | Faithful | revelator319 |
| 15:44.24 | Assid | like im waiting for it to roll over to the next outgoing provider.. but its taking too long to rollover |
| 15:44.34 | Faithful | yeah but I haven't used it for so long I don't know it works at all... |
| 15:45.41 | Hmmhesays | hahaa |
| 15:45.49 | Hmmhesays | hold on i have create an extensions for you |
| 15:46.58 | *** join/#asterisk Carp1 (i=Carp1@ip-204-97-151-191.modem.logical.net) |
| 15:47.10 | Hmmhesays | can't connect |
| 15:47.25 | sevard | [TK]D-Fender: you there |
| 15:47.37 | Hmmhesays | sevard stfu you n00b |
| 15:47.43 | Faithful | Hmmhesays: |
| 15:47.48 | Hmmhesays | haha |
| 15:47.50 | Faithful | try againg |
| 15:48.02 | *** join/#asterisk gursikh (n=guriskh1@158.135.7.70) |
| 15:48.42 | gandhijee | what file do i edit to change the echo canceller zaptel uses? |
| 15:48.46 | gandhijee | anybody know |
| 15:49.36 | Hmmhesays | faithful i got the incoming call but it got routed wrong |
| 15:50.03 | Hmmhesays | hang up and try again |
| 15:50.42 | [TK]D-Fender | znoG : Many ways you can do it. Look on the WIKI under "dual servers" |
| 15:50.47 | Hmmhesays | ok hang up |
| 15:50.48 | [TK]D-Fender | sevard : barely |
| 15:50.51 | Hmmhesays | one more time i'll try to fix this route |
| 15:51.01 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 15:51.14 | sevard | [TK]D-Fender: remember when we were talking about that special hunt group |
| 15:51.17 | sevard | yesterday |
| 15:51.17 | [TK]D-Fender | ckwall : I have no idea yet, I'd have to see your config |
| 15:51.20 | Hmmhesays | aight hit it |
| 15:51.27 | [TK]D-Fender | sevard : Ok, remembering now... |
| 15:51.31 | sevard | :) |
| 15:51.41 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:52.30 | Hmmhesays | one more time Faithful |
| 15:52.45 | *** join/#asterisk oej (n=oej@myskin.iet.unipi.it) |
| 15:53.14 | Faithful | User not online |
| 15:53.46 | Hmmhesays | bah |
| 15:53.48 | Hmmhesays | i don't get it |
| 15:53.50 | [TK]D-Fender | znoG : I might suggest setting it up with one side treating the other like a straight SIP phone, and the other registering like you would to an ITSP. Make sure not to fix the callerid and the diaplan enty that sends calls over would modify the callerid of each sides users so that you know how to send the call back. |
| 15:54.04 | sevard | [TK]D-Fender: you said you'd show me an example of that special hunt group |
| 15:54.26 | Faithful | Hmmhesays: have you got normal skype... just check we can talk |
| 15:54.51 | Hmmhesays | this is a sip routing issue |
| 15:55.18 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 15:55.48 | sevard | apple nipple monkey |
| 15:55.50 | sevard | apple nipple poo |
| 15:55.58 | r_evolution | well |
| 15:56.03 | r_evolution | isnt that a motherfucker :) |
| 15:56.08 | paryl | i'm getting a strange problem on a new installation with polarity reversal... http://pastebin.ca/54557 |
| 15:56.13 | [TK]D-Fender | sevard : Yeah, outside work hours :) |
| 15:56.22 | sevard | [TK]D-Fender: :P! all up in yo grill! |
| 15:56.28 | [TK]D-Fender | sevard : You might have a chance in about 5 hours |
| 15:56.32 | Hmmhesays | ringing |
| 15:56.35 | [TK]D-Fender | or in9 |
| 15:56.47 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 15:57.25 | paryl | anyone know what could be causing that? |
| 15:58.00 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 15:58.49 | jake1932 | anyone doing cti with avaya? |
| 15:59.16 | darkskiez | [TK]D-Fender: whats the special hunt group ? |
| 15:59.58 | [TK]D-Fender | darkskiez : nothing special, just a smart way to do dial-plan entries for varios styles. |
| 16:00.21 | GerbilWrk | I've got an issue where the queue rings all of the agents assigned to it, but after it rings for a few minutes, it starts seeing Agents as unavailable and stops ringing their phones. Anyone know how to turn that off so that it just keeps ringing? |
| 16:00.27 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) |
| 16:01.06 | [TK]D-Fender | GerbilWrk : What kind of agents do you use? |
| 16:01.32 | GerbilWrk | statically assigned sip agents |
| 16:01.51 | ckwall | ok, I have been toying aorund trying to make caller id work... I have posted the parts of the files I am working with, can someone see what I am doing incorrectly? http://pastebin.ca/54559 |
| 16:02.09 | [TK]D-Fender | GerbilWrk : set the auto-logout option to no. |
| 16:02.44 | GerbilWrk | would that be in agents.conf or queues.conf? |
| 16:02.45 | [TK]D-Fender | ckwall : Ok, you have not yet grasped how to properly set up SIP phones with *. |
| 16:02.56 | [TK]D-Fender | ckwall : What time zone are you in? |
| 16:03.17 | ckwall | mountain |
| 16:03.47 | ckwall | thank you for noticing that I havn't figured this out :-D |
| 16:03.50 | [TK]D-Fender | ckwall : Good. I can help you after 5pm EST. |
| 16:04.08 | [TK]D-Fender | ckwall : And we'll get you up and running the right way. |
| 16:04.15 | ckwall | awesome, thanks. |
| 16:04.22 | [TK]D-Fender | ckwall : np. |
| 16:04.48 | r_evolution | and he'll only charge you a case of box wine and two 16 yr old azn prostitutes :) |
| 16:04.50 | [TK]D-Fender | ok, lunch time... |
| 16:04.58 | r_evolution | peace out TK :) |
| 16:05.19 | sevard | grr |
| 16:05.32 | sevard | what if i want the queue to have music on hold but agents waiting for a call don't want to hear moh |
| 16:06.01 | salviadud | they can suck on a lemon |
| 16:06.05 | Faithful | with the TDM400 I only need the power connector if I am running FXS modules right? |
| 16:06.52 | salviadud | sevard, you might want to add more classes, and they can pick their fav music |
| 16:07.27 | sevard | salviadud: i don't know how to detonate classes for agents |
| 16:07.40 | znoG | [TK]D-Fender: my main problem is that when i call box 1 from box 2, it says that it failed to authenticate user "asterisk" <sip:foo@mydomain.com> |
| 16:07.52 | salviadud | me neither, we'd both be learnin' something new |
| 16:07.56 | znoG | [TK]D-Fender: but I did clearly specify fromuser and username in the [foo] section |
| 16:08.11 | znoG | [TK]D-Fender: so i've no idea why it's trying to auth as user asterisk |
| 16:08.27 | jake1932 | anyone familiar with avaya cti want to make a quick $200? |
| 16:08.39 | sevard | salviadud: high five mofo |
| 16:08.55 | salviadud | sevard, you the man |
| 16:09.00 | sevard | ALRIGHT! |
| 16:09.34 | sevard | salviadud: found it |
| 16:10.04 | sevard | salviadud: in agents.conf under the [section] for your agents musiconhold => mohclass |
| 16:10.05 | salviadud | sevard, you found it on the wiki? |
| 16:10.13 | sevard | i'm adding an agent's calss with an empty moh dir |
| 16:10.19 | sevard | yeah |
| 16:10.31 | salviadud | there ya go, it wasn't that hard... |
| 16:10.59 | salviadud | and if one of your agents is a country music fan |
| 16:11.07 | salviadud | you con do a dir with a bunch of redneck songs |
| 16:11.25 | salviadud | can |
| 16:11.34 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:11.59 | *** join/#asterisk saftsack (n=saftsack@p54A7F4CE.dip.t-dialin.net) |
| 16:12.00 | *** join/#asterisk AsteriskAddict (n=speedy@r172h230.dixie-net.com) |
| 16:12.07 | sevard | salviadud: or drive them insane with jpop |
| 16:12.24 | *** join/#asterisk p1p (i=pip@64.200.16.25) |
| 16:12.50 | jsharp | Snoop Dogg as MOH. |
| 16:13.10 | p1p | Anyone here have any exp setting up the expansion module on polycom spip601's? |
| 16:13.13 | *** part/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:13.16 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:13.41 | *** join/#asterisk samourai1 (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma) |
| 16:13.45 | gandhijee | Faithful: right |
| 16:14.03 | gandhijee | p1p: i have one, i'm going to be setting it up today |
| 16:14.07 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 16:14.09 | *** part/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:14.44 | *** join/#asterisk bufh (n=user@ip216-239-71-75.vif.net) |
| 16:14.45 | p1p | gandhi:everything that ive read says that in order to monitor lines you need to add that ext to the contact list and select "monitor buddy" but I dont see this option anywhere |
| 16:14.48 | bufh | hello |
| 16:14.53 | p1p | gandhi:any idea? |
| 16:15.05 | bufh | wow, there *is* much more people here :) |
| 16:15.10 | a1fa | [TK]D-Fender : sup playar |
| 16:15.19 | *** join/#asterisk websae (n=websae@h69-129-251-26.69-129.unk.tds.net) |
| 16:15.19 | gandhijee | p1p: how are you tryin to modify it? through the XML file or the phone? |
| 16:15.19 | a1fa | "playar", btw :) |
| 16:16.24 | *** join/#asterisk ToyMan (n=stuq@74-32-62-42.dsl1.mdl.ny.frontiernet.net) |
| 16:16.43 | znoG | [TK]D-Fender: how does peer matching work? i mean, i'm calling box1 from box2 via SIP (with user/pass info). One would think that would automatically determine which [section] it will use in sip.conf |
| 16:16.50 | p1p | gandhi: Ive got provisioning setup but I was trying to add users through the phone interface, should I just be editing the 000000000-contacts xml? |
| 16:17.12 | *** join/#asterisk runa (n=martin@200.123.150.237) |
| 16:17.13 | bufh | good day, i've a little question, if someone can help me (or tell me where to look in the manual), i'd like ton know how to "intercept" a call ringing on a ringgroup or on a extension from another extention |
| 16:17.24 | [TK]D-Fender | znoG : After 5pm EST I'll be available for you |
| 16:17.36 | [TK]D-Fender | a1fa : Still breathing... |
| 16:17.36 | gandhijee | p1p: yeah |
| 16:17.40 | runa | hey :) what should I use to connect a gsm cell phone to asterisk? |
| 16:17.46 | gandhijee | p1p: the phone interface is really limited. |
| 16:18.05 | *** part/#asterisk Schwuk (n=Schwuk@84.12.166.117) |
| 16:18.15 | [TK]D-Fender | p1p :You need to enable presences support in sip.cfg to get the option in your contact list. |
| 16:18.17 | gandhijee | p1p: kris has a base xml file you can use as a base, kriskompanies.com |
| 16:18.37 | p1p | gandhi: thanks, that was pretty stupid of me I suppose considering im using provisioning because of how bad the phone interface/web int are |
| 16:18.57 | redondos | Can asterisk recognize voice commands for selecting IVR menu entries? In spanish? |
| 16:19.17 | [TK]D-Fender | gandhijee : You really have to be careful about taking someone elses config as one built for the wrong SIP version can lock up your phone. |
| 16:19.22 | salviadud | chinga tu madre, and then, you get tranfered |
| 16:19.24 | a1fa | redondos : si |
| 16:19.30 | a1fa | redondos : cabron |
| 16:19.32 | salviadud | chupame el pito, and then, hook flash |
| 16:19.34 | a1fa | redondos : pendejo |
| 16:19.44 | a1fa | chupa mi vergita, .. |
| 16:19.45 | a1fa | :P |
| 16:19.47 | [TK]D-Fender | redondos : * does not have speech recognition of its own, and Sphinx is not great. |
| 16:19.56 | gandhijee | Fender: really? |
| 16:20.04 | redondos | [TK]D-Fender: So I should use sphinx, that's all we got? |
| 16:20.14 | [TK]D-Fender | gandhijee : Yes. 1.5.x and 1.6.2 do NOT mix. |
| 16:20.17 | a1fa | they need to merge voice changer patch |
| 16:20.31 | [TK]D-Fender | redondos : That and others, but AFAIK only Sphinx is free, and its not easy..... |
| 16:21.52 | redondos | Ouch, ok. |
| 16:22.51 | gandhijee | oh, i didn't know there was a new firmware out |
| 16:24.19 | websae | anyone here worked much with an asterisk --> fax gateway? |
| 16:24.26 | websae | one that emails pdfs to you.. |
| 16:24.45 | [TK]D-Fender | gandhijee : You never know what revision may be in use. |
| 16:25.13 | [TK]D-Fender | websae : I use SpanDSP for PRI -> emailing of faxes |
| 16:25.36 | gandhijee | i see |
| 16:25.58 | websae | I wanted to just take a g711u fax and turn it into pdf file for email |
| 16:26.30 | p1p | ghandi: heres the million $ question, I enabled presence monitoring in sip.cfg and now I can see the presence of all my contacts but they dont show on the expansion module. What did I miss? |
| 16:26.33 | *** part/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 16:26.46 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 16:27.15 | paryl | is there a way to retreive the last record from a group, instead of the first? |
| 16:27.18 | gandhijee | no idea, like i said, i have one and am going to be setting it up today |
| 16:27.19 | gandhijee | i just got it |
| 16:27.19 | [TK]D-Fender | p1p : make sure your speed dial index is right, and understand there is a FLAW in that you can only buddy watch 7 people before it gets buggy. |
| 16:27.26 | websae | p1p: are you using polycome sidecar? |
| 16:27.29 | stack_ | websae: I've been using Asterisk with Hylafax and iaxmodem... it works great |
| 16:28.14 | p1p | websae: yes |
| 16:28.40 | websae | i heard there were issues |
| 16:28.43 | p1p | fender: I heard they were patching that soon, any idea if thats true? |
| 16:28.58 | p1p | fender: also what do you mean make sure the index is "right"? |
| 16:29.22 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 16:30.06 | [TK]D-Fender | p1p : Polycom is supposed to remove their atrificial limitation shortly, and Asterisk is due to support SIP-B (the "normal" way of supported shared lines) for 1.4 this summer |
| 16:30.23 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.224.Dial1.SanJose1.Level3.net) |
| 16:30.29 | [TK]D-Fender | p1p : make sure they are in order numerically or they won't show up on your sidecar at all. |
| 16:30.40 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 16:32.27 | *** part/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 16:32.55 | *** join/#asterisk Lino` (n=Lino@i577BC646.versanet.de) |
| 16:33.28 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 16:38.00 | salviadud | has icall.com been hacked yet? |
| 16:38.12 | salviadud | it would be kickass if we could |
| 16:38.56 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:39.26 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 16:39.44 | mut | http://www.palore.com |
| 16:39.53 | b0xii | is there a quick and easy way to block a single phone number? (i'm using aah2.8) |
| 16:40.57 | mut | sure, first exten in your dialplan |
| 16:41.19 | mut | exten => 9065551021,1,hangup |
| 16:41.41 | b0xii | thank you |
| 16:41.48 | [TK]D-Fender | mut : Ummm... very not what he had in mind I'm sure, and assumes a lot.. |
| 16:42.37 | mut | heh |
| 16:42.49 | salviadud | hahaha |
| 16:42.54 | salviadud | aah suxxors |
| 16:42.54 | mut | he never specified incoming or outgoing |
| 16:43.00 | b0xii | incoming |
| 16:43.23 | salviadud | i hate it, i only use it because they make me at work |
| 16:43.29 | [TK]D-Fender | mut : On on what :) or who's number, or ANYTHING :) |
| 16:43.56 | shiznatix | if i have a GoToIfTime thing and I want to go to a certain context on ONLY mondays and thursdays how do I do this? |
| 16:44.06 | p1p | salvia: A@H is pretty awesome, what do you have against it? |
| 16:44.17 | shiznatix | I know that I can say monday through thursday by doing: mon-thurs but how do I say ONLY mondays and thursday? |
| 16:44.27 | [TK]D-Fender | shiznatix : read the doc's on GotoIfTime... it says how to specify by days.... |
| 16:44.35 | runa | anyone? what's the best way to connect a gsm phone to asterisk? (I don't have the phone yet) using bluetooth+cellphone or with some kind of gsm2eth adapter? |
| 16:44.55 | shiznatix | [TK]D-Fender, i have looked but it does not specify how to do 2 days that are not sequential |
| 16:45.01 | b0xii | i can just slap the above line in from-pstn if i use that, correct? |
| 16:45.10 | [TK]D-Fender | shiznatix : So do it in 2 steps! |
| 16:45.14 | [TK]D-Fender | shiznatix : duh! |
| 16:45.17 | [TK]D-Fender | ;) |
| 16:45.51 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:45.54 | [TK]D-Fender | b0xii : I would advise against that, and note the next time you do a "commit" of any changes you'll blow away any mod like this you try to do. |
| 16:46.04 | [TK]D-Fender | b0xii : And please do read the channel topic. |
| 16:46.18 | shiznatix | [TK]D-Fender, so are you saying there is no way to do it when it is non sequential? i am writing a script to make it do everything automatically so it would be easier to do this without doing a 1000 lines |
| 16:47.04 | b0xii | [TK]D-Fender, alrighty... should've read the topic |
| 16:47.31 | salviadud | comon linux question: how do i check out the "current" rules on iptables? |
| 16:47.54 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 16:48.21 | *** join/#asterisk p1p (i=pip@64.200.16.25) |
| 16:48.22 | [TK]D-Fender | shiznatix : how complicated do you really need to make things? |
| 16:48.50 | kph100 | ne1 knows of a list of did providers? |
| 16:48.59 | docelmo | for what? |
| 16:49.16 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 16:50.35 | shiznatix | [TK]D-Fender, as simple as possible, that is why if I could do it all on one line it would be a lot easier for me |
| 16:50.42 | gursikh | kph100: there is a decent one at the voip wiki, and another on the aussie wiki |
| 16:50.53 | *** join/#asterisk p1p (i=tjcomp91@64.200.16.100) |
| 16:50.58 | *** join/#asterisk blackgecko (n=blackgec@201.152.98.35) |
| 16:51.03 | Hmmhesays | I hate conference calls with n00bs |
| 16:51.41 | sevard | Hmmhesays: what's up with you today |
| 16:51.50 | salviadud | i once was a n00b |
| 16:51.57 | sevard | you still are |
| 16:52.00 | docelmo | what do you mean once? |
| 16:52.02 | Hmmhesays | we all were, but messenger is much better than waisting my time on the phone |
| 16:52.32 | salviadud | the phone is a lot more real hugh? |
| 16:52.46 | camelon | I used to have a frecuent problem with some extensions that suddenly are giving the busy messege! What I can do to overcome this? |
| 16:52.48 | salviadud | you can FEEL the n00bness |
| 16:53.05 | Hmmhesays | so the skype api in linux seems pretty straightforward, I think it might be possible to interface asterisk with the linux skype client |
| 16:53.27 | docelmo | there goes the damn neighborhood |
| 16:53.41 | ManxPower | camelon, Fix the problem. |
| 16:53.52 | [TK]D-Fender | shiznatix : Not certain, read the instructions again. |
| 16:54.05 | sevard | this fax machine went off and now my teeth hurt |
| 16:54.13 | sevard | explain THAT fox tv |
| 16:54.28 | salviadud | your teeth? |
| 16:54.36 | salviadud | dude. did you bite it? |
| 16:54.40 | sevard | back ones, taste like i ate metal |
| 16:55.06 | sevard | it's aliens |
| 16:56.04 | *** join/#asterisk frk2 (n=kvirc@202.141.251.102) |
| 16:56.06 | frk2 | guys |
| 16:56.11 | [TK]D-Fender | sevard : Clearly its causing interference with your dental implant transmitter ;) |
| 16:56.13 | frk2 | i need insight |
| 16:56.29 | sevard | frk2: enlightenment is obtained by meditation |
| 16:56.36 | frk2 | dudes |
| 16:56.37 | frk2 | please |
| 16:56.38 | [TK]D-Fender | frk2 : try a mirror |
| 16:56.39 | frk2 | enlighten me |
| 16:56.40 | Hmmhesays | and hot freaky sex |
| 16:56.44 | frk2 | okay |
| 16:56.50 | frk2 | heres the deal over which im pulling my hair out |
| 16:56.51 | bufh | i've another question, when someone transfert a call (say the extension 101 transert the call "C" to the extension 102), the name which appear on the phone 102 is the name of 101, it should be the name of "C" once 101 hangoff, how could i do that ? |
| 16:56.56 | [TK]D-Fender | TALK! |
| 16:57.03 | frk2 | Case is that of a Grandstream GXP 2000 |
| 16:57.11 | frk2 | 4 GXPs |
| 16:57.21 | frk2 | I put them at client A, they all hang like shit |
| 16:57.29 | [TK]D-Fender | bufh : You need to do a BLIND transfer, not a consultative one. |
| 16:57.30 | frk2 | i put them at my office- they work awesome |
| 16:57.44 | frk2 | i put them at client B, they work awesome |
| 16:57.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:57.45 | [TK]D-Fender | frk2 : NAT hate... |
| 16:58.17 | frk2 | the hanging is totally blind to call load or firmware revision |
| 16:58.22 | frk2 | phone outputs jack shit on the syslog |
| 16:58.24 | frk2 | WTF |
| 16:59.02 | frk2 | seriously- what COULD be the issue? |
| 16:59.21 | frk2 | the only thing i can think of is power issues at the client |
| 17:00.15 | bufh | [tk]-fender < is BLIND an option for Transfert() or it is to the "phone" 101 to use a functionality of the phone to do a blind transfert ? |
| 17:00.23 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 17:00.27 | frk2 | Any insight guys? |
| 17:01.04 | ManxPower | frk2, you, of course, have qualify=yes and nat=yes for each of the NATed devices. Asterisk, is of course, on a public IP address. Also "hang like shit" is not a tecthnical term. |
| 17:01.42 | *** join/#asterisk glm2k (n=glm@rrcs-24-199-11-46.west.biz.rr.com) |
| 17:01.44 | camelon | ManxPower . . .how the problem could be fixed . . some guide please? |
| 17:01.45 | ManxPower | "frk2 the hanging is totally blind to call load or firmware revision" <-- I assume that english is a second language for you. |
| 17:01.56 | ManxPower | camelon, which problem? |
| 17:02.20 | camelon | I used to have a frecuent problem with some extensions that suddenly are giving the busy messege! What I can do to overcome this embarrising situation? TIA |
| 17:02.32 | nahirean | hahaha |
| 17:02.42 | [TK]D-Fender | bufh : It should eb an option on the phone. |
| 17:02.42 | ManxPower | camelon, there are 400 billion reasons this could be happening. |
| 17:02.55 | bufh | [tk]d-fender < i'll have a look on the phone-manual, thank you |
| 17:03.00 | ManxPower | [TK]D-Fender, I think he used "blind" to mean "doesn't depend on" |
| 17:03.18 | camelon | ManxPower . . .wher I can begin to look? |
| 17:03.20 | [TK]D-Fender | ManxPower : No, he's asking about my use of the term "blind transfer" |
| 17:03.41 | ManxPower | camelon, "sip show peers" when this is happening would be a good place to look. |
| 17:03.48 | bufh | yes i was, but i don't see "blind transfert" anywhere on my manual |
| 17:03.49 | bufh | hum |
| 17:04.12 | bufh | damn :( even google give few answers about that topic |
| 17:04.14 | coppice | blind transfers are made with a braille keypad |
| 17:04.17 | frk2 | Manxpower- yes it is. Sorry about that |
| 17:04.18 | ManxPower | bufh, "blind transfer", means "transfer the call to an extension and do NOT talk to the destination person first" |
| 17:04.21 | camelon | ManxPoer . . .bu the problem only happen with zap extensions no with sip!! |
| 17:04.23 | bufh | ahh |
| 17:04.32 | bufh | manxpower you are helping thank |
| 17:04.33 | bufh | hum |
| 17:04.48 | ManxPower | "supervised aka consultative transfers means transfer the call to an extensions, but DO talk to the destination person before completing the transfer" |
| 17:05.03 | bufh | but can't we configure asterisk to "refresh" the text sent on the destination phone with the name of the caller ? |
| 17:05.08 | jake1932 | aka attended |
| 17:05.15 | ManxPower | camelon, See, you have already wasted some of my time be not saying it was happening with Zap. |
| 17:05.24 | ManxPower | camelon, threewaycalling=no in zapata.conf |
| 17:05.32 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 17:05.43 | ManxPower | your users are hanging up and then picking up the phone too fast and asterisk thinks it's a 3-way call or a transfer. |
| 17:05.59 | frk2 | I guess nobody can provide me any insight. |
| 17:06.00 | bufh | i'll see if i can punish my users |
| 17:06.00 | camelon | ManxPower . . .sorry!! |
| 17:06.02 | frk2 | sigh |
| 17:06.09 | salviadud | punish? |
| 17:06.12 | salviadud | how? |
| 17:06.21 | ManxPower | users need to be punished. |
| 17:06.27 | jpabuyer | hahaha |
| 17:06.37 | bufh | poking them with a pockingstick |
| 17:06.41 | bufh | or so |
| 17:06.45 | salviadud | well, if the users are ladies |
| 17:06.48 | jake1932 | put a shocker on the hangup button - they press to fast - it shocks |
| 17:06.54 | salviadud | and the punishment comes from my gonzo |
| 17:06.57 | salviadud | i agree |
| 17:06.59 | bufh | you never read the texts of the BOFH ? |
| 17:07.05 | bufh | how to punish users |
| 17:07.16 | salviadud | bofh, no |
| 17:07.26 | bufh | if you are an admin you should ;) |
| 17:07.31 | jpabuyer | what was camelon's problem?? I didn't get it |
| 17:07.38 | salviadud | where could i find that? |
| 17:07.43 | ManxPower | frk2, your poor english skills makes it difficult to understand what you are saying. Perhaps someone that speaks your native language can help? |
| 17:07.55 | bufh | salviadud < google then BOFH should be a good starting point |
| 17:08.05 | jpabuyer | yo hablo espa#ol |
| 17:08.07 | bufh | going to eat, bbl |
| 17:08.09 | ManxPower | bufh, I refer to BOFH as "The Good Book" |
| 17:08.25 | bufh | manxpower < hell, you are one ! |
| 17:08.56 | camelon | jpabuyer . . . I used to have a frecuent problem with some Zap extensions that suddenly are giving the busy messege! What I can do to overcome this embarrising situation? |
| 17:09.10 | ManxPower | bufh, The "local support person" for one of the offices I work with is unable to do a factory reset on the Polycom phones here. |
| 17:09.42 | ManxPower | jpabuyer, I believe his users are accidenlty creating three-way calls. |
| 17:09.45 | salviadud | bofh is nutz! its friggin' awesome |
| 17:10.12 | jpabuyer | camelon, you mean, the channels are bridged and you are speaking and everything and then all out of the blue comes the busy signal?? |
| 17:10.13 | tzanger | ManxPower: out of curiosity, what does your set-ring agi do? |
| 17:10.16 | Hmmhesays | ARGH |
| 17:10.24 | bufh | manxpower < i worked with a lot of high graduated engeeners like that too |
| 17:11.04 | ManxPower | tzafrir, at the moment? nothing. LOL!. It's supposed to set the Alert Info for internal .vs. external calls. I now handle it in a different way. |
| 17:11.55 | *** join/#asterisk trbldwine (n=trbldwin@vpn166141.vpn.northwestern.edu) |
| 17:12.12 | ManxPower | now I set the Alert Info before exten => _NXXNXXXXXX,1,Goto(extensions,${EXTEN:6},1) |
| 17:12.39 | ManxPower | and since only external calls would be dialing the full 10-digit DID..... |
| 17:13.27 | camelon | ManxPower . . . what happen with the other funcionalities threeway related?? |
| 17:14.37 | salviadud | i love my box too |
| 17:14.46 | salviadud | it's a prankcalling machine from HELL |
| 17:14.56 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:15.09 | tzanger | ManxPower: yeah that makes sense |
| 17:15.11 | salviadud | i just need to work on my .call files |
| 17:15.17 | [TK]D-Fender | ManxPower : later (substantially) I'd appreciate a sample of a generical Alert-info for syntax so I can start having mine do that as well... |
| 17:18.39 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 17:19.15 | DeeJay[2] | erm.... I had to leave.. we were talking about call transfer ... |
| 17:19.19 | DeeJay[2] | ;) |
| 17:19.40 | DeeJay[2] | a1fa: Still there? ;) |
| 17:21.30 | DeeJay[2] | as I said.... is it possible to have a conference, leaving the conference while keeping the 2 peers in communication even if I'm the one who created the conference? |
| 17:21.56 | *** join/#asterisk Thock (n=FreePBX3@216.119.93.253) |
| 17:22.12 | [TK]D-Fender | DeeJay[2] : Depends on the phone. Polycom's do it, can't vouch for others |
| 17:22.18 | DeeJay[2] | polycom yes |
| 17:22.20 | DeeJay[2] | we have polycoms |
| 17:22.31 | DeeJay[2] | [TK]D-Fender: how do we achieve it? |
| 17:22.33 | [TK]D-Fender | DeeJay[2] : May depends on the SIP revision |
| 17:22.38 | DeeJay[2] | 1.6.2 |
| 17:22.38 | [TK]D-Fender | DeeJay[2] : Its automatic |
| 17:22.47 | [TK]D-Fender | you should be good to go already last I checked |
| 17:22.57 | DeeJay[2] | Currently, If I start a conference with 2 other persons... If I hang up, the conference closes... |
| 17:23.06 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 17:23.12 | dlynes_ | If an x100p card isn't detecting hangup, that would be a gain control issue? |
| 17:23.23 | *** join/#asterisk Noky (n=dnardell@200.68.89.23) |
| 17:23.33 | Noky | hi |
| 17:23.35 | [TK]D-Fender | DeeJay[2] : upgrade |
| 17:24.01 | Noky | how can i know in my extension what is the context where i am... |
| 17:24.04 | [TK]D-Fender | dlynes : no its an X100 (and analog period) sucks at disconnect supervision issue |
| 17:24.08 | Noky | the extension i have with ${EXTEN} |
| 17:24.16 | Noky | and context? ${CONTEXT} ??? |
| 17:24.24 | dlynes_ | [TK]D-Fender: ah...just never ran into the problem before |
| 17:24.25 | DeeJay[2] | [TK]D-Fender: Do you know where I could find a newer version? |
| 17:24.40 | [TK]D-Fender | Noky : how do you NOT know where you are? |
| 17:24.42 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 17:25.03 | [TK]D-Fender | DeeJay[2] : Freedom phones usualy is 1 revision behind, or best to check with your reseller |
| 17:25.14 | Noky | [TK]D-Fender: i want know my context, because i want to do a Goto(${CONTEXT),1,1) or something else |
| 17:25.17 | Noky | understand me ? |
| 17:25.23 | Noky | my english suck, sorry |
| 17:25.28 | Noky | :[ |
| 17:25.56 | [TK]D-Fender | Noky : No need. Goto(1,1) does it |
| 17:26.05 | Thock | Quick Q: If i'm running two interface cards, (A200D and A104D), how would the channel setup work? Would i create the two channel areas with different [bracket] names? Like [channela200] and [channel104d] etc? |
| 17:26.10 | Noky | i want to configurate an IVR.. |
| 17:26.14 | Thock | also /wave fender |
| 17:26.19 | [TK]D-Fender | Noky : you don't need to specifiy the context to jump within it |
| 17:26.22 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 17:26.38 | Noky | ok |
| 17:26.39 | Noky | thanks |
| 17:27.04 | [TK]D-Fender | Thock : That doesn't describe where you are at all really... |
| 17:27.19 | [TK]D-Fender | Thock : Which file are you working on? |
| 17:27.23 | Thock | Okay, i have a A200D installed and configured just fine. |
| 17:27.32 | Thock | i'm working on etc/asterisk/zapata.conf |
| 17:27.55 | Thock | just two channels with FXO ports on the card |
| 17:27.59 | [TK]D-Fender | no, everything in zapata.conf is under [channels] |
| 17:28.00 | *** join/#asterisk SparFux (n=player@e182023106.adsl.alicedsl.de) |
| 17:28.09 | Thock | but what if you have two different interface cards? |
| 17:28.23 | Thock | do you separate those or is it just one or the other? |
| 17:28.23 | [TK]D-Fender | Thock : Your channel def in zaptel.conf is what it should follow. |
| 17:28.39 | SparFux | I have two bri cards installed. One AVM Fritz classic and one HFC-PCI A card. I use mISDN. How can I tell which one a program would use? Which one is THE FIRST one? |
| 17:28.41 | [TK]D-Fender | yes, you need to seperat them, but not by contexts. |
| 17:29.04 | [TK]D-Fender | Thock : Go read up on mixing TDM & T1 cards in a server for an idea of how channels are divided. |
| 17:29.08 | dlynes_ | [TK]D-Fender: btw...does the a200 come standard with an on-board ec? |
| 17:29.18 | [TK]D-Fender | dlynes : its optional. |
| 17:29.21 | dlynes_ | ah |
| 17:29.26 | Thock | [TK]D-Fender: Thanks again man, i'll go look it up. It's on voipinfo, yes? |
| 17:29.31 | [TK]D-Fender | dlynes : with adds +/- 300$ |
| 17:29.38 | dlynes_ | Just didn't see the echo canceller product number on their website |
| 17:29.40 | [TK]D-Fender | thock : yup |
| 17:29.57 | [TK]D-Fender | dlynes : may not be listed there seperately, thats all |
| 17:30.06 | dlynes_ | Only seen the one for the quad t1/e1 card |
| 17:30.12 | [TK]D-Fender | dlynes : Sometimes you list the finished product, and not the permutations. |
| 17:30.24 | marcus2 | anyone here using polycom desk phones? |
| 17:30.30 | dlynes_ | ic |
| 17:30.31 | DeeJay[2] | [TK]D-Fender: what version are you using for your polycom phones? |
| 17:30.32 | [TK]D-Fender | dlynes : its got it... (as an option). it should be bundled by the place you buy it all from. |
| 17:30.48 | dlynes_ | [TK]D-Fender: buying directly from sangoma |
| 17:30.59 | [TK]D-Fender | DeeJay[2] : various. for my work IP600's I use 1.5.3, for home and my work 601 I use 1.6.5 (lateest) |
| 17:31.01 | Noky | [TK]D-Fender |
| 17:31.06 | [TK]D-Fender | dlynes : Ask for it :) |
| 17:31.12 | Noky | do u know some example of extension with IVR ? |
| 17:31.18 | Noky | but a strong IVR |
| 17:31.22 | DeeJay[2] | [TK]D-Fender: Would you mind sharing 1.6.5? ;) |
| 17:31.27 | [TK]D-Fender | marcus2 : I own every model. |
| 17:31.31 | asteriskmonkey | [TK]D-Fender: i have the 1.6.6 release |
| 17:31.33 | [TK]D-Fender | marcus2 : what about them? |
| 17:31.40 | [TK]D-Fender | asteriskmonkey : its out? |
| 17:31.40 | DeeJay[2] | I have IP500 and IP600.. |
| 17:31.43 | asteriskmonkey | it supports side cards woo :D |
| 17:31.48 | [TK]D-Fender | asteriskmonkey : Cool, will have to go DL it. |
| 17:31.48 | asteriskmonkey | < beta monkey heheheh |
| 17:31.57 | [TK]D-Fender | asteriskmonkey : side cards? |
| 17:32.03 | marcus2 | is it possible to make it show the the name of the party i am calling? |
| 17:32.15 | [TK]D-Fender | marcus2 : It yes, Asterisk = no |
| 17:32.20 | asteriskmonkey | yes the attachemnts for the 601 the side cars it now adds the asterisk and blf support |
| 17:32.25 | marcus2 | lame. |
| 17:32.36 | [TK]D-Fender | asteriskmonkey : you mean they lifted their ass limitation? :) |
| 17:32.49 | [TK]D-Fender | marcus2 : Don't blame me.. I don't code. |
| 17:32.50 | asteriskmonkey | yes |
| 17:32.56 | marcus2 | i wonder if someone will fix that |
| 17:32.57 | p1p | asteriskmonkey: is this available for partners yet? |
| 17:32.58 | asteriskmonkey | we bugged them daily ehee |
| 17:33.03 | [TK]D-Fender | asteriskmonkey : Is it beta or release? |
| 17:33.17 | asteriskmonkey | think a beta |
| 17:33.22 | asteriskmonkey | dont know might be release now |
| 17:33.27 | p1p | =o |
| 17:33.33 | Noky | i don't found a example of ivr =( |
| 17:33.51 | [TK]D-Fender | Noky : you obviously didn't look very hard |
| 17:34.00 | [TK]D-Fender | Noky : just type in "ivr" on the WIKI. |
| 17:34.02 | dlynes_ | Noky: look up the documentation for the Background() dialplan application |
| 17:34.10 | Noky | thanks |
| 17:35.47 | *** join/#asterisk Arcu (i=Arcu@67.108.111.144.ptr.us.xo.net) |
| 17:36.14 | Noky | the wiki doesn't work |
| 17:36.17 | *** join/#asterisk Rick_Hunter (n=rhunter@04-181.008.popsite.net) |
| 17:36.31 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 17:36.33 | Noky | mmm... nono, it's work :D |
| 17:36.35 | Noky | sorry |
| 17:36.49 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 17:40.22 | [TK]D-Fender | Noky : YOU don't work.... go call for RMA :D http://www.voip-info.org/wiki/view/IVR |
| 17:43.11 | [TK]D-Fender | Noky : Actually that link isn't the best, but its all in thersomewhere to be found quickly. |
| 17:43.18 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:43.44 | paolob | Hi guys! Where could I find spanish prompts for asterisk? thank you! |
| 17:44.15 | nahirean | record them |
| 17:44.32 | [TK]D-Fender | paolob : Go look on the WIKI! |
| 17:44.35 | [TK]D-Fender | :D |
| 17:45.09 | *** join/#asterisk mugawump (n=mugawump@rrcs-24-172-3-11.midsouth.biz.rr.com) |
| 17:47.30 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 17:48.07 | znoG | [TK]D-Fender: unfortunately the dual servers page on the Wiki focuses more on IAX than SIP :( |
| 17:48.31 | key2 | !seen kram |
| 17:48.38 | key2 | ~seen kram |
| 17:48.45 | jbot | kram is currently on #asterisk, last said: 'uhm lol'. |
| 17:49.31 | X-Gen | dont bother the kram god ! |
| 17:53.48 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:54.15 | generalhan | hey all |
| 17:54.29 | websae | hey |
| 17:54.52 | generalhan | Ive been hearing a few people talking about "click to dial" applications ... anyone in here working on that ? or got that working ? |
| 17:57.08 | *** part/#asterisk runa (n=martin@200.123.150.237) |
| 17:58.48 | nahirean | you mean have asterisk log caller's phone numbers and create a little GUI that redials DIDs, etc? |
| 17:58.51 | jake1932 | click to dial is easy if you know manager |
| 17:59.08 | generalhan | hmmm |
| 17:59.16 | generalhan | im not real familiar with manager either ! lol |
| 18:00.07 | jake1932 | do you want to do it yourself - or want someone else to do it for you? |
| 18:00.14 | generalhan | we have some software developers working on some front and back end software for us. and they want to have any "leads" that need calling to be in a list for the reps calling. then they could click the link and their phone would dial |
| 18:00.18 | [TK]D-Fender | znoG : it shows everything you need. BUt I suggest setting it up like you would for an ITSP on one side and a phone on the other |
| 18:00.27 | generalhan | jake1932: i would like to learn it i think. |
| 18:00.54 | jake1932 | check the wiki on the manager API and look at originate |
| 18:00.57 | camelon | how looks this map interrupts CPU0 CPU1 CPU2 CPU3 |
| 18:00.57 | camelon | 0 341158694 341226495 341170954 341166086 IO-apic-edge timer |
| 18:00.58 | camelon | 1 371 358 4582 6053 IO-apic-edge i8042 |
| 18:01.02 | camelon | 2 0 0 0 0 XT pic cascade |
| 18:01.05 | jake1932 | woah |
| 18:01.07 | camelon | 3 121 169249 0 164988 IO-apic-level aic7xxx, a |
| 18:01.12 | camelon | 4 17571028 137 0 0 etho |
| 18:01.12 | jake1932 | can you pastebin it |
| 18:01.16 | camelon | 7 0 0 0 0 |
| 18:01.16 | generalhan | camelon: stop it please |
| 18:01.20 | camelon | 8 0 0 0 0 |
| 18:01.20 | generalhan | ~pb |
| 18:01.22 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:01.26 | camelon | 10 550122 368190123 313714661 681328720 IO-apic-level wct4xxp |
| 18:01.27 | nahirean | dude! |
| 18:01.28 | nahirean | stop |
| 18:01.30 | camelon | 12 544 2204 1669 971 IO-apic-i8042 |
| 18:01.35 | camelon | NMI 0 0 0 0 |
| 18:01.39 | camelon | loc: 1364787655 1364787533 1364787661 1364787660 |
| 18:01.39 | snitt | omfg |
| 18:01.43 | camelon | ERR: 0 |
| 18:01.43 | [TK]D-Fender | camelon : CUT IT THE ^#% OUT |
| 18:01.47 | snitt | PASTEBIN! |
| 18:01.47 | camelon | MIS: 0 |
| 18:01.49 | camelon | sorry!! |
| 18:02.11 | camelon | SORRY!!!!!!!!!! |
| 18:02.26 | *** join/#asterisk paolob_ (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:02.39 | bufh | pompom |
| 18:02.58 | snitt | pompom |
| 18:03.52 | paolob_ | Hi guys! I'm getting various messages about mp3 at CLI: "res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?!" - "res_musiconhold.c:336 spawn_mp3: /var/lib/asterisk/mohmp3 is not a valid directory" - "res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player". Any idea what should I do? |
| 18:04.25 | [TK]D-Fender | paolob_ : pick a valid directory |
| 18:04.26 | jake1932 | paolob_: are you using format_mp3? |
| 18:04.40 | paolob_ | jake1932, where? |
| 18:04.45 | jake1932 | add-ins |
| 18:04.50 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 18:04.52 | x86 | is there a way i can make calls coming from a certain caller ID go to a given macro? |
| 18:04.53 | jake1932 | asterisk-add-ins |
| 18:04.55 | [TK]D-Fender | jake1932 : He's not using native. |
| 18:04.57 | paolob_ | [TK]D-Fender, a valid directory for what? |
| 18:05.04 | [TK]D-Fender | paolob_ : You picked a bad folder. |
| 18:05.05 | jake1932 | yeh - just caught the last line |
| 18:05.19 | [TK]D-Fender | paolob_ : that folder doesn't exist or have MP3's in it from what the error says |
| 18:05.26 | x86 | eh? |
| 18:05.31 | fjean | hi guys |
| 18:05.37 | [TK]D-Fender | x86 : You can make any all do whatever you want |
| 18:06.16 | x86 | I want to place certain extensions that are temporarily disabled into a macro whenever they dial out, so they hear Allison saying the account has been temporarily disconnected |
| 18:06.24 | x86 | and it wont allow them to place outbound calls :P |
| 18:06.52 | x86 | I've got it working for calls inbound to the given extension, but having a bit of trouble with outbound from the given extension |
| 18:07.17 | generalhan | jake1932: do you have any links with sample manager scripts .. i want to figure out wha the heck im doing here. |
| 18:07.18 | fjean | can someone give me a hand on a zaptel install ? |
| 18:07.25 | jake1932 | why would you not do a db lookup and use goto to send them to a message? |
| 18:07.47 | [TK]D-Fender | x86 : I'd suggest changing the context of the phone to one with exten => _X. to do that. |
| 18:08.11 | *** join/#asterisk abcdic (n=naoeda@copernico.inatel.br) |
| 18:08.12 | camelon | jbot . .sorry . . just a newby . . this is the paste: http://pastebin.ca/54567 |
| 18:08.14 | jbot | that's too long, camelon |
| 18:08.14 | [TK]D-Fender | x86 : my way = easy & reversable |
| 18:08.27 | generalhan | hahaha |
| 18:08.30 | x86 | [TK]D-Fender: i tried changing the context, but i used exten => s |
| 18:08.40 | x86 | would it work better with _X. ? |
| 18:08.57 | GerbilWrk | x86, that should work fine for incoming, but outgoing would be best with a _X |
| 18:08.58 | jake1932 | generalhan: i got everything from the wiki - it's pretty straightforward http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 18:09.08 | x86 | GerbilWrk: cool |
| 18:09.13 | [TK]D-Fender | x86 : you are talking about DIALING. exten _X. |
| 18:09.21 | camelon | jbot . . . .is all the mapping interrupts that my system give us . . I only need to know if all in there is OK?? |
| 18:09.23 | jbot | camelon: that's too long |
| 18:09.27 | abcdic | newb question: What does SPAN mean? Is it a protocol? O a shortcut term for some thing? Like, the zt_span structure.. |
| 18:09.32 | [TK]D-Fender | x86 : you don't wantoto disable an analog line, just an EXTENSION. |
| 18:09.42 | fjean | hey guys where would I see why the zap devices (udev) are not created at boot ime ? |
| 18:09.55 | generalhan | jake1932: yea that stuff makes perfect sense ,,, but i need more direction now ... like after i make that how exactly do i call on it, and how to i pass variables in there so that i can have it take the number and insert it into the Exten: perameter ? |
| 18:10.04 | generalhan | jake1932: that kind of stuff |
| 18:10.36 | tzafrir | ~bot abuse |
| 18:10.37 | jbot | ACTION huddles in the corner, whimpering 'please, please stop' |
| 18:10.46 | jake1932 | generalhan: it's just text over TCP/IP, you can use whatever program you like to insert the variables |
| 18:11.05 | jake1932 | heck, you can even test everything with a telnet client |
| 18:11.05 | Nugget | telnet is eeeeeeevil! |
| 18:11.12 | camelon | jbot . . . wich line must be reviwed from your point of view?? |
| 18:11.22 | generalhan | camelon: please stop talking to jbot |
| 18:11.23 | generalhan | lol |
| 18:11.30 | *** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca) |
| 18:11.31 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 18:12.11 | jake1932 | ~spam |
| 18:12.13 | jbot | ACTION sings, Spam, Spam, Spam, Spam, Spam, Wonderfull spam! |
| 18:12.14 | jake1932 | ~span |
| 18:12.16 | jbot | Span data across multiple removable disks. URL: http://users.gtn.net/fraserm/span.html |
| 18:12.16 | camelon | generalhan . . . I've missed something? (hahaha) |
| 18:12.22 | x86 | [TK]D-Fender: still not working... I'm using realtime, and I changed the context in MySQL, but asterisk still shows it as the original context when I do sip show peer 100 |
| 18:12.34 | *** part/#asterisk trbldwine (n=trbldwin@vpn166141.vpn.northwestern.edu) |
| 18:12.55 | jake1932 | span across multiple disks - in a telephony channel??? |
| 18:13.01 | gandhijee | whats better for CDR, postgres or MySQL> |
| 18:13.07 | generalhan | camelon: jbot is a bot .. lol |
| 18:13.24 | generalhan | ~dict PSTN |
| 18:13.27 | Hmmhesays | anyone have one of those customers that just rubs them the wrong way? |
| 18:13.48 | camelon | get it!!! |
| 18:13.49 | generalhan | Hmmhesays: yeah my boss lol |
| 18:13.50 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 18:13.53 | Hmmhesays | yeah |
| 18:13.56 | Hmmhesays | i hate guys like that |
| 18:13.58 | generalhan | only i cant move on to the next one ! lol |
| 18:14.08 | jsharp | All my customers are like that. |
| 18:14.14 | jsharp | They demand that I deliver what they pay me for. |
| 18:14.17 | jsharp | Bastards. |
| 18:14.17 | *** join/#asterisk DrPete (n=Pete@host-84-9-255-194.bulldogdsl.com) |
| 18:15.24 | jsharp | this world me a great job if it weren't for the freakin customers. |
| 18:15.35 | DrPete | llol |
| 18:16.40 | camelon | someone could take a look over this paste: TIA . . . http://pastebin.ca/54567 |
| 18:17.00 | camelon | How the interrupts looks?? |
| 18:17.24 | jsharp | Looks good to me. You're not sharing interrupts and your TDM card is generating interrupts. |
| 18:17.30 | *** join/#asterisk Arpheis_ (n=Arpheis@mna75-4-82-225-77-91.fbx.proxad.net) |
| 18:18.21 | [TK]D-Fender | x86 : then REALTIME is not working.. NMP :) |
| 18:18.51 | x86 | seems to be not a problem with realtime, but how asterisk caches realtime to astdb |
| 18:18.56 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 18:19.13 | camelon | jsharp . . .from those information could I be looking for my * problems in another place??? |
| 18:19.24 | copland | Is there a list of Asterisk Friendly VOIP providers out there ? |
| 18:20.09 | x86 | gah, i'll just write an AGI for it ;) |
| 18:20.36 | camelon | jsharp . . so frecuents channel busy messeges in my * daily operation!! |
| 18:20.42 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 18:20.55 | VoicePulse | copland: http://www.voip-info.org/wiki/view/VOIP+Service+Providers |
| 18:21.37 | asteriskmonkey | dont suppose digium could add fqdn support to there iaxys |
| 18:21.54 | tzafrir | capelon: a simple advide: replace "??" with "?" and remove all "!"-s . You'll actually get more answers |
| 18:22.15 | tzafrir | camelon, that is |
| 18:22.27 | jaybuffet | so i just had a meeting with our current phone providers trying to keep us tied to them... the say a lot of business have issues with pure voip |
| 18:22.30 | camelon | tzafir . . .TIA |
| 18:22.36 | tzafrir | abcdic, that's here |
| 18:23.09 | camelon | sorry world |
| 18:23.26 | jaybuffet | is that true |
| 18:23.50 | websae | jsharp: how are you doing fellow rescuer? |
| 18:24.31 | bufh | hum, i'd like to send a call directly to a voicemail (say the extension 102) whithout ringing the phone(102), is there a "out of the box" macro for that ? |
| 18:24.50 | fjean | Mandrake 10.1 : kernel 2.6.8.1-12mdksmp (i585) on a dual processor i686 ; am I starting in the wrong path ? :-) |
| 18:25.27 | glm2k | fjean: MDK 10.1 is old dude |
| 18:25.52 | bufh | btw, mandrake doesn't exist anymore |
| 18:25.57 | glm2k | that too :) |
| 18:26.11 | fjean | glm2k: you think it could give me issues with current zaptel ? :-) |
| 18:26.29 | Corydon-w | Yeah, but all the guys who used to work for Mandrake are all still sitting at the same desks, getting paid the same |
| 18:26.41 | fjean | so tell me guys which one should I install, frankly |
| 18:26.44 | glm2k | i only compiled on an mdk10.1. no recent operating experience with it |
| 18:27.02 | Corydon-w | It's like saying you've worked at 5 companies in as many years, and you never changed desks |
| 18:27.16 | glm2k | Corydon-w: all except the founder |
| 18:27.51 | bufh | beware the ostroll ;) |
| 18:27.57 | glm2k | lol |
| 18:28.10 | Corydon-w | In any case, the company is now called Mandriva |
| 18:28.20 | fjean | the motherboard is D915GAG ; any one would be better suited then ? |
| 18:28.32 | glm2k | fjean: just use the latest mandriva. whether you compile * or get the cooker packages, is up to you |
| 18:28.43 | fjean | k |
| 18:28.48 | jake1932 | is that the same as D915GAG1? |
| 18:29.17 | bufh | hum, is there a limitation in the numbers of an extension ? may a make extensions with four digits ? |
| 18:29.26 | glm2k | bufh: yep |
| 18:29.32 | glm2k | er, no, and yes |
| 18:29.36 | bufh | tx |
| 18:29.40 | fjean | jake1932: the one I have is GAG-L |
| 18:29.47 | jake1932 | ok |
| 18:30.10 | fjean | jake1932: that's what you play with too ? |
| 18:30.15 | *** part/#asterisk abcdic (n=naoeda@copernico.inatel.br) |
| 18:30.52 | jake1932 | no - i couldn't even tell you - but it would be interesting if someone had that exact motherboard and was able to give you a recommendation |
| 18:31.11 | fjean | yeah |
| 18:31.58 | fjean | I think I\ll go with FC |
| 18:33.10 | jaybuffet | so is what the phone company telling me accurate... is pure voip still too immature for business ? |
| 18:33.16 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:33.22 | Qwell[] | jaybuffet: Is your business immature? |
| 18:33.44 | Qwell[] | Can you afford T1? Do you need the reliability of a T1? |
| 18:33.46 | fjean | jaybuffet, if they don't sell it, for sure they will tell you this... |
| 18:34.07 | jsharp | voip itself rocks. Its voip over the public internet that sucks balls. |
| 18:34.09 | Qwell[] | The trouble isn't "immaturity" |
| 18:34.14 | Qwell[] | ^^^ What he said |
| 18:34.17 | jaybuffet | i'm just wondering from people here that use it / implement it what experience they have had |
| 18:34.18 | gandhijee | i am gonna be using asterisk w/ a fake-me-out VoIP |
| 18:34.33 | Qwell[] | jaybuffet: If you use voip over the LAN, it works incredibly well |
| 18:34.37 | jake1932 | it's based on your net connection (if using it over the net) |
| 18:34.53 | jaybuffet | Qwell[]: so thats basically what they were saying |
| 18:34.55 | Qwell[] | but, unless you've got a private link and an SLA to your ITSP...it's gonna suck |
| 18:35.31 | jsharp | I use voip all the time over our VSAT links and it rocks. Voip from office to office over the public internet, though, sucks sometimes. |
| 18:38.12 | jarrod | :q! |
| 18:38.13 | jarrod | w |
| 18:38.20 | mikefoo | this isn't vi! |
| 18:40.05 | Zodiacal | anyone know how i can get asterisk to let go of the sound card after it uses it? i.e. paging over a loud speaker. it keeps other applications from using the sound card.. |
| 18:41.00 | Hmmhesays | nested while loops make my head hurt |
| 18:41.40 | jsharp | Zodiacal: the instant asterisk loads chan_oss/chan_alsa, it opens /dev/dsp and keeps it open. |
| 18:42.04 | Zodiacal | hmmhesay better than recursive functions.. |
| 18:42.41 | Zodiacal | jsharp, it seems like it doesn't open it until its used.. then keeps it open |
| 18:42.55 | Zodiacal | just wonderin if theres some way to close it, maybe manualy? before i run another app that needs the sound card? |
| 18:43.38 | Qwell[] | Zodiacal: add proper alsa support for chan_alsa |
| 18:43.41 | jake1932 | modprobe -r :) |
| 18:43.54 | dlynes_ | Zodiacal: You could rewrite the channel code |
| 18:44.00 | Zodiacal | :/ |
| 18:44.14 | Qwell[] | shouldn't be hard...there are probably hundreds of apps that had to change |
| 18:44.16 | elvisthedj|work | Qwell : I got my firmware upgraded on the 7940.. i'll be anxious to try skinny when it can do things like .. ring :) |
| 18:44.20 | Zodiacal | when using lsof it only shows dev/dsp in use after i use the paging features |
| 18:44.38 | Zodiacal | qwell i woudn't know where to start :P |
| 18:44.52 | Qwell[] | elvisthedj|work: it rings...kinda :p |
| 18:44.56 | dlynes_ | Zodiacal: I would suspect whoever wrote the chan_alsa and chan_oss didn't expect someone to make the asterisk server double as a desktop machine |
| 18:45.02 | Qwell[] | elvisthedj|work: I actually know what causes it not to ring..just need to work around it |
| 18:45.17 | Zodiacal | dlynes just a simple "play filename.wav" is all i need |
| 18:45.18 | Qwell[] | elvisthedj|work: It'll either not ring, or crash your phone...I chose the former ;) |
| 18:45.24 | Zodiacal | maybe i can have asterisk play my file? |
| 18:45.26 | jake1932 | couldn't you just add another sound card? |
| 18:45.29 | elvisthedj|work | Qwell I choose sip .. for now :) |
| 18:45.34 | Qwell[] | yeah |
| 18:46.07 | elvisthedj|work | Qwell I appreciate your help with that though. I wouldn't have even got interested in fixing it without having seen it actually function in some capacity |
| 18:46.14 | dlynes_ | Zodiacal: What's the purpose of the application? |
| 18:46.19 | p1p | Can anybody point me to some documentation on setting up a paging ext that outputs through the * server? |
| 18:46.26 | Zodiacal | dlynes to play a sound file at specific times of the day, i.e. break times etc.. |
| 18:46.53 | *** join/#asterisk mjackson (n=mjackson@69.85.202.186) |
| 18:46.56 | dlynes_ | Zodiacal: sorta like some muzak interlude during break times to make it a little less boring? |
| 18:47.02 | elvisthedj|work | Qwell I'm pretty happy with it. Now I'm going to search for a way to split a conference call.. i'm having to go to the CLI and do a soft hangup on the channel i want rid of.. that can't be right |
| 18:47.12 | dlynes_ | Zodiacal: i.e. it would play during the entire break? |
| 18:47.16 | jsharp | Call file to call the console, Playback() a file, then Hangup() |
| 18:47.23 | Zodiacal | dlynes no no, not elevator music :), just to signal the start and end times of break |
| 18:47.33 | Zodiacal | like a buzzer |
| 18:47.37 | dlynes_ | Zodiacal: why not just run it as a cron job? |
| 18:47.44 | dlynes_ | Zodiacal: and take asterisk out of the picture, completely? |
| 18:47.53 | Zodiacal | dlynes yeah i am, but asterisk paging takes hold of the sound card |
| 18:48.08 | mjackson | Anybody know why asterisk would just stop playing sounds? In the log it appears to play sound file fine, no errors anywhere, but no playback. |
| 18:48.30 | Zodiacal | jsharp i tried playing with call files, but they seem to need an ext to bridge the call too. am i missing somthing, is that not nessisary? |
| 18:49.19 | jsharp | No, you don't need an extension to bridge to. |
| 18:49.21 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 18:50.33 | jsaunders | iax2 show netstats shows following... |
| 18:50.51 | Zodiacal | jsharp okie i'll give it a try.. |
| 18:50.55 | jsaunders | -------- REMOTE -------------------- |
| 18:51.00 | Zodiacal | dlynes, jsharp Thank You! |
| 18:51.01 | jsaunders | Jit Del Lost % Drop OOO Kpkts |
| 18:51.05 | jsharp | NO PASTE FLOOD. |
| 18:51.07 | jsharp | ~pb |
| 18:51.14 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:51.18 | jsaunders | jsharp... your hilarious. |
| 18:51.23 | jsaunders | I was going to put in 4 lines. |
| 18:51.25 | jsaunders | Ohhh nooooooooo. |
| 18:51.33 | jsaunders | Talk about trigger happy. |
| 18:51.36 | jsharp | Zodiacal: http://pastebin.ca/54571 |
| 18:51.39 | p1p | Can anybody point me to some documentation on setting up a paging ext that outputs through the * server? |
| 18:51.44 | jsharp | Try that call file. That should do what you need it to do. |
| 18:52.13 | Zodiacal | jsharp ooo i can send it directly to oss? |
| 18:52.21 | jsharp | That'll play getcher-butt-back-to-work.[wav|gsm] over the oss channel. |
| 18:52.23 | jsharp | Yes. |
| 18:52.23 | Zodiacal | trying right now |
| 18:53.50 | jsaunders | anyways... 'iax2 show netstats' shows 16-30% loss on REMOTE side. I am sending calls via local network using IAX2, and outbound to provider using SIP. Does this 'REMOTE' measurement denote the outbound SIP leg? |
| 18:54.14 | jsaunders | Or does LOCAL refer to *, and REMOTE refer to host of client? |
| 18:56.03 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:56.25 | Zodiacal | jsharp hrm.. doesn't seem to work... i tried putting the full path to /var/lib/asterisk/sounds/beep.gsm |
| 18:56.34 | Zodiacal | and just beep by itself, etc.. |
| 18:57.35 | *** join/#asterisk backblue (n=moo@87-196-44-8.net.novis.pt) |
| 18:59.22 | [TK]D-Fender | Zodiacal : remove the extension. |
| 18:59.42 | Zodiacal | tkd-fender this is what im using http://pastebin.ca/54571 |
| 18:59.45 | Zodiacal | jsharps example |
| 18:59.53 | Zodiacal | with out an ext |
| 19:00.35 | [TK]D-Fender | verified the full path to that file? |
| 19:00.49 | Zodiacal | i changed it to /var/lib/asterisk/sounds/beep.gsm |
| 19:00.55 | Zodiacal | and it exists |
| 19:01.16 | Zodiacal | my cli doesn't show any activity when i put the .call file in /var/spool/asterisk/outgoing |
| 19:01.21 | Zodiacal | that is where im suposed to move it right? |
| 19:01.32 | [TK]D-Fender | Zodiacal : Couldn't confirm on that, sorry. |
| 19:01.58 | Zodiacal | i think so, when i tested other .call files thats where it put it, but this is using the channel OSS insted of an ext's like when i tested it.. |
| 19:02.09 | mjackson | my asterisks is broken! *cry* format time! |
| 19:02.25 | Qwell[] | elvisthedj|work: donations welcome :P |
| 19:02.49 | [TK]D-Fender | mjackson : clarify "broken" and how bad it could be that you'd need to reformat... |
| 19:03.02 | Qwell[] | donations and bug reports... 6859 I believe |
| 19:04.49 | Zodiacal | i changed it to channel: console/dsp |
| 19:05.02 | Zodiacal | that put this in the cli http://pastebin.ca/54575 |
| 19:05.04 | Zodiacal | but no sound :/ |
| 19:05.34 | Zodiacal | so close :) |
| 19:05.44 | *** part/#asterisk fjean (n=fjean@201.29.130.118) |
| 19:06.25 | Zodiacal | any ideas? |
| 19:08.09 | Zodiacal | WORKING! |
| 19:08.14 | Zodiacal | had to use no path |
| 19:08.16 | Zodiacal | beep |
| 19:08.21 | Zodiacal | thanks again guys! |
| 19:08.24 | [TK]D-Fender | np |
| 19:08.33 | Zodiacal | and console/dsp |
| 19:08.37 | Zodiacal | or OSS/dsp |
| 19:08.38 | [TK]D-Fender | full path works MINUS the extension. |
| 19:08.44 | Zodiacal | yep minus ext |
| 19:08.49 | Zodiacal | oh ic |
| 19:08.50 | Zodiacal | yeah |
| 19:08.54 | [TK]D-Fender | never add it, * will search whats available |
| 19:08.54 | Zodiacal | makes sence |
| 19:09.14 | [TK]D-Fender | I need to start making call files as well... |
| 19:09.32 | Zodiacal | theres another issue too, some times the paging is chopy.. |
| 19:10.04 | Zodiacal | doesn't matter if i use the phones or just playing a sound file |
| 19:10.11 | Zodiacal | anyone one ever seen that? |
| 19:10.12 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:10.25 | Zodiacal | only some times, i would say 1 out of 10 pages is chopy |
| 19:11.03 | Zodiacal | maybe its the first one on boot and its slow to open the sound card... |
| 19:11.20 | [TK]D-Fender | Zodiacal : Queue up a silent track |
| 19:11.46 | Zodiacal | just a silent sound file? |
| 19:12.40 | [TK]D-Fender | Zodiacal : yup |
| 19:12.45 | [TK]D-Fender | to "prime" the system |
| 19:12.56 | mut | O_O |
| 19:13.04 | Zodiacal | oic okie |
| 19:13.10 | Zodiacal | i'll see if thats the cause in a sec |
| 19:14.09 | mjackson | D-Dender: It stopped being able to play any kind of sound file through sip phones or dialed in lines over the T1. It's a new install anwway, and something we did this morning broke it, so I'm going to start it from scratch. Trying to get vicidial working. |
| 19:15.06 | mjackson | Honestly I think what happened was I did a source install from the cvs stable version, and somebody else did a cvs install of asterisk-extras from the cvs head version and it went goofy |
| 19:15.37 | mut | hey [TK]D |
| 19:15.43 | mut | how do ya prnounce nenad? |
| 19:15.59 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 19:16.56 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 19:16.58 | SexyKen | Hello |
| 19:17.16 | SexyKen | How can I make an asterisk compatible recording (.gsm) |
| 19:18.07 | RaYmAn-Bx | Anyone around who happens to know the right zaptel settings for a x100/101p card in Denmark? (no callerid necessary) |
| 19:18.16 | RaYmAn-Bx | google seems to give me nothing. |
| 19:19.56 | [TK]D-Fender | mut : neh- nad |
| 19:20.44 | [TK]D-Fender | mjackson : so just download STABLE release and recompile everything |
| 19:20.56 | [TK]D-Fender | SexyKen : "Record" |
| 19:22.41 | *** join/#asterisk my007ms (n=my007ms@196.202.70.12) |
| 19:23.35 | mut | know an extension |
| 19:23.43 | mut | or should they be able to transfer me |
| 19:23.46 | my007ms | any one know any idea how pay all this asterisk guys or any othere ppl that work in free and open software how they keep a life |
| 19:24.12 | [TK]D-Fender | mut : should be able to transfer. |
| 19:24.30 | mut | last name? |
| 19:24.55 | [TK]D-Fender | my007ms : Talk Yoda does funny hmmMMMMMM??! |
| 19:25.01 | [TK]D-Fender | mut : Korvic |
| 19:25.30 | my007ms | realy i wish find answer for this Q |
| 19:25.40 | [TK]D-Fender | my007ms : Try asking a bit clearer. |
| 19:25.41 | my007ms | how this ppl eat |
| 19:25.48 | mut | we'll see if this works i guess |
| 19:26.25 | mut | no answer at sales |
| 19:26.46 | mut | oo |
| 19:26.47 | mut | holding |
| 19:27.26 | jake1932 | yoda - LOL |
| 19:27.27 | [TK]D-Fender | be ready to take it down though.... |
| 19:27.53 | [TK]D-Fender | jake1932 : I pull no punches when it comes to dark humour :) |
| 19:28.34 | jake1932 | funny thing is that rerading with yoda in mind was pretty darn funny |
| 19:28.57 | my007ms | where all this developers that work in open sorce and free software get the money to keep work in this software |
| 19:29.11 | jake1932 | i'll anser |
| 19:29.15 | jake1932 | answer too |
| 19:30.04 | *** join/#asterisk triple-e (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 19:30.08 | jake1932 | we're multitasking |
| 19:30.18 | [TK]D-Fender | my007ms : We contribute for fun, and Digium contributes so they can keep selling their compatible HARDWARE |
| 19:30.29 | *** join/#asterisk eido (n=eido@m815f36d0.tmodns.net) |
| 19:30.54 | my007ms | so asterisk for fun |
| 19:31.05 | triple-e | for thats the reason i buy Digium hardware |
| 19:31.08 | jake1932 | <PROTECTED> |
| 19:31.13 | [TK]D-Fender | my007ms : For a lot of us, yes. For some its business like it is for Digium |
| 19:31.14 | mut | sweeet |
| 19:31.17 | mut | got him |
| 19:31.39 | eido | hiya folks. i just got my first softphone working from linux (twinkle) connected to an asterisk server hosted at a VOIP company. I'm able to place calls just fine from my desktop (yay opensource!) - but inbound calls are not ringing through. when i ask the asterisk server for a list of registrations, it shows that my desktop client is connected (as is the phone at home)... but it's not rinigng me. is this a problem with twinkle, or with asterisk, |
| 19:31.39 | eido | or i'm just doing something wrong? |
| 19:32.00 | my007ms | this is asterisk what abut othere free software that no one have it for business |
| 19:34.04 | [TK]D-Fender | my007ms : You need to learn about OSS. Go read up elsewhere on how OSS works for the free community and for business |
| 19:34.06 | *** join/#asterisk freakGB (n=mark@host-84-9-66-232.bulldogdsl.com) |
| 19:34.23 | eido | hmm. any ideas on my 'not ringing me' problem? or are we still arguing FOSS vs commercial? :) |
| 19:34.32 | mercestes | eido: Do you have a phone # associated wtih your softphone or are you dialing by IP or......?? |
| 19:35.03 | eido | i don't have a specific phone number for my softphone, i assumed that when i logged into the VOIP accont it would ring both places (the phone at home and my softphone). is that not the case? |
| 19:35.20 | mercestes | Not if you don't port your number to the voip company. |
| 19:35.21 | sevard | [TK]D-Fender: if somebody calls in on zap/1-1 and gets pushed to a queue, and an agent set up as member => ZAP/3/5555555 where 5x == his cellphone, how come it just patches though the caller to zap/3 ? |
| 19:35.22 | sevard | oh oh oh |
| 19:35.25 | mercestes | It will continue to ring your home phone as always. |
| 19:35.29 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 19:35.30 | sevard | because once a zap line picks up it's considered answered |
| 19:35.31 | sevard | damnit! |
| 19:35.37 | sevard | how do i get around that foo |
| 19:35.40 | my007ms | i am ready to read all oss site if it will answer my Q |
| 19:35.53 | eido | hmm. well, there's no real 'port' to happen. theres no old number to add in. i can't just say "I'm also xxx-xxx-xxxx" ? |
| 19:35.54 | [TK]D-Fender | sevard : you DON'T do it. period |
| 19:36.07 | sevard | [TK]D-Fender: but what if an agent is mobile :| |
| 19:36.19 | mercestes | Asterisk will never see the number if you dial from a land line. |
| 19:36.29 | mercestes | There is no "DID broadcast." |
| 19:36.40 | mercestes | otherwise I could steal your number and talk to your mom. |
| 19:36.43 | [TK]D-Fender | sevard : sorry, not like that you don't.... |
| 19:36.47 | mercestes | you don't want that to happen. |
| 19:36.53 | sevard | [TK]D-Fender: then how :| |
| 19:37.31 | [TK]D-Fender | sevard : What part of YOU DON'T is not registering? Maybe on a PRI with call supervision, even then you'd have to about before Cell's VM kicked in |
| 19:37.34 | eido | mercestes: can i tell asterisk i'm 'also' at the primary number? |
| 19:37.43 | eido | so if someon were to call the priary, i'll hear it here too? |
| 19:37.44 | *** join/#asterisk noky (n=dnardell@200.68.89.23) |
| 19:37.45 | noky | hi |
| 19:37.55 | sevard | [TK]D-Fender: :'( |
| 19:37.59 | mercestes | you can program a rule for a DID internally so if you call from your softphone you can have it ring itself via your DID...because the Asterisk won't pass it off and handle it itself... |
| 19:38.13 | noky | at incoming,01150316030,1 failed so falling back to exten 's' :( |
| 19:38.23 | mercestes | but....no one else will be able to call you via that number unless they happen to be calling you via the same VoIP company. |
| 19:38.33 | noky | i'm using realtime's extensions |
| 19:38.41 | noky | the call in for this extension |
| 19:38.41 | noky | | 9998 | incoming | _0115031603[0234] | 1 | Goto | America|${EXTEN:7}|1 | |
| 19:38.44 | eido | hmmmm. |
| 19:38.46 | noky | have an error ? :S |
| 19:38.50 | noky | it's OK |
| 19:42.00 | bufh | so |
| 19:42.08 | bufh | thank you for your support guys |
| 19:42.10 | bufh | see ya |
| 19:43.13 | mercestes | noky, are you trying to send _0115....etc. to context America,6030, priority 1? |
| 19:43.18 | sevard | [TK]D-Fender: alright, one more thing, you still around |
| 19:43.42 | [TK]D-Fender | sevard : shoot |
| 19:43.52 | sevard | I get this a lot lately |
| 19:44.00 | sevard | [May 9 14:42:34] WARNING[11941]: channel.c:2049 ast_indicate: Unable to handle indication 3 for 'SIP/2010-ecca' |
| 19:44.04 | sevard | no idea where it got set at |
| 19:44.30 | noky | mercestes yes |
| 19:45.00 | noky | must match with this |
| 19:45.00 | noky | | 20003 | America | 6030 | 1 | Goto | America_IVR_1||1 | |
| 19:45.07 | noky | i dial 01150316030 |
| 19:45.28 | [TK]D-Fender | sevard : not a clue |
| 19:45.38 | sevard | [TK]D-Fender: HA!!! |
| 19:45.39 | *** join/#asterisk op3r (n=op3r@202.124.131.132) |
| 19:45.40 | sevard | STUMPED YOU! |
| 19:46.15 | [TK]D-Fender | sevard : ... well I HAVE a clue, but don't feel like spending the effort to explore it :) |
| 19:47.03 | sevard | :\ |
| 19:47.11 | op3r | does anyone knows any docs for AGI perl and AGI php howtos? |
| 19:47.23 | sevard | op3r: voip-info.org |
| 19:47.23 | [TK]D-Fender | op3r : All on the WIKI..... |
| 19:47.28 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 19:47.37 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 19:49.48 | jpabuyer | I want to understand the Asterisk source code.. is there an explanation of the structure somewhere or something like that? for example, if I want to know when cdr records are logged in the source code, where should I look... |
| 19:51.41 | mercestes | CDR's are created on hangup events. |
| 19:51.56 | mercestes | not sure where that's located in the source...but that's when the CDR Is created.. |
| 19:52.11 | jpabuyer | yeah, and where are hangup events in the source code?.. |
| 19:52.14 | sevard | [TK]D-Fender: what's your clue |
| 19:52.33 | jpabuyer | I'd like to understand the code |
| 19:52.51 | [TK]D-Fender | sevard : whatever channel you are using has an in-progress indicator that it can't pass on to a SIP channel in a meaningful way. |
| 19:55.14 | op3r | thanks |
| 19:55.57 | sevard | [TK]D-Fender: extension 2010 rings extension 1024 and it spits out that error while no rings are present on the 2010 phone while the 1024 phone rings. the other way around 1024 dials 2010 and boths phones have indications of ringing |
| 19:56.03 | sevard | they are both part of the same dial plan |
| 19:56.08 | eido | now. s ee.. this is odd. the VOIP provider folks say "No, you don't have to do anything special. If you log in with a softphone, we ave multiring set up by default. it should ring all registered clients." |
| 19:56.19 | eido | so i'm back to wondering if this is a problem with twinkle. |
| 20:00.19 | eido | hmm. |
| 20:01.22 | mut | now this dude is cool [TK]D-Fender |
| 20:01.27 | mut | thx |
| 20:01.50 | eido | wow. the ersion of twinkle that's in Debian is -ancient-. (well, kubuntu). 0.4.2. 0.7.1 is out. |
| 20:02.26 | mercestes | Did you give them a number when you registered? |
| 20:02.33 | eido | me? |
| 20:02.41 | mercestes | -> Eido |
| 20:02.44 | eido | ah. nope. |
| 20:03.09 | mercestes | Then how do they know what number to multiring you to? |
| 20:03.45 | eido | well, the phone number is regstered in asterisk. it's what SIP client to ring, yes? so if i say "show me regsitrations" I see the 2 IP addresses i'e connected in from. |
| 20:04.03 | mercestes | so you used your phone number as your sip username? |
| 20:04.12 | eido | i used the account name. |
| 20:04.20 | eido | sorry if this is sounding confusing - i'm very new to VOIp. |
| 20:04.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:04.48 | mercestes | You have a phone number...and a SIP username.... |
| 20:04.54 | eido | correct. |
| 20:05.00 | *** join/#asterisk boch (n=boch@unirc.com.ar) |
| 20:05.09 | boch | sup |
| 20:05.09 | mercestes | lets say you have you rnumber, 1234567, your ext, 1234, and your sip username...FooBar. |
| 20:05.09 | eido | the phone number + username etc is in use by a phone dongle i have at home - patched into my cable mode. |
| 20:05.26 | eido | well, no extension, but okay. |
| 20:05.29 | mercestes | You can only ring Sip/Foobar. |
| 20:05.49 | mercestes | If you want to extenstion dial, you must have 1234,1,Dial(Sip/FooBar). |
| 20:05.58 | mercestes | That means, "When you see 1234, dial FooBar." |
| 20:06.15 | mercestes | same wiht your phone number, 1234567,1,Dial(Foobar) |
| 20:06.23 | eido | right, but what if i log in with the same sip username multiple times? |
| 20:06.37 | eido | the voip guys are saying "a call into 12345 will ring all the sip clients" |
| 20:06.41 | mercestes | now....if your number, 1234567 goes to your home phone....then it will go directly to your home phone...not any other asterisk switches. |
| 20:07.07 | mercestes | So your voip guys will never see 1234567. |
| 20:07.15 | eido | ... |
| 20:07.19 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 20:07.21 | eido | byut the voip provider -is- my home phone. |
| 20:07.36 | mercestes | So the voip provider has your home phone number now? |
| 20:07.37 | eido | or, more to the point, my home phone is servicded by my voip provider. |
| 20:07.48 | mercestes | Ok, your home phone is VoIP? |
| 20:07.51 | eido | YES |
| 20:07.54 | mercestes | Yay.... |
| 20:07.56 | eido | sorry, didn't make that clear apparently. |
| 20:07.57 | mercestes | getting somewhere. |
| 20:07.57 | [TK]D-Fender | mut : Good to hear |
| 20:08.10 | asteriskmonkey | my home phone is string and can technology |
| 20:08.10 | asteriskmonkey | :D |
| 20:08.24 | mercestes | LOL. |
| 20:08.28 | asteriskmonkey | quality far surpasses voip and pstn woo |
| 20:08.40 | mercestes | Ask your Voip Provider ot verify that your softphone is being sent a "ring." and they are getting a "Sip peer is ringing." on your username. |
| 20:08.42 | mut | no fax support tho? |
| 20:08.48 | mut | less you use the airplane protocol |
| 20:09.05 | eido | hmmm. |
| 20:09.44 | *** join/#asterisk IceManRISK (n=kart@201.66.47.9) |
| 20:10.13 | mercestes | If yoru username is "bob" you should see "calling Bob" "Bob is ringing." |
| 20:10.39 | mercestes | if that is the case...,you can look at your softphone...if not...you can look at yoru VoIP provider. |
| 20:12.42 | [TK]D-Fender | mut : so how'd it go? |
| 20:12.55 | mut | least i got some stuff to try tonight |
| 20:13.28 | [TK]D-Fender | mut : If you ask and can schedule it right he may be around to help you. He was instrumental in getting me up and running blind the first time. |
| 20:14.41 | eido | mercestes: the requests are in the queue to look at the logs, thanks :) |
| 20:14.46 | mut | heh |
| 20:15.02 | mut | he said i'de have to give him a pretty nice thank you to get him out at 2am |
| 20:15.03 | mut | :P |
| 20:15.18 | *** join/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 20:15.19 | imperfect- | Howdy |
| 20:16.00 | imperfect- | chan_sip.c is complaing about Initizaling already initilized SIP dialog |
| 20:16.13 | imperfect- | and when I get a call and ring my extensions, when I answer it dies with |
| 20:17.17 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 20:17.19 | imperfect- | han_sip.c:10370 handle_response: Remote host can't match request BYE to call '11c0059-1c@147.135.12.128'. Giving up. |
| 20:17.25 | boch | im using .call files to generate monitoring calls, but how can i know the results? |
| 20:18.16 | *** join/#asterisk terrapen (n=cjs@166.70.183.109) |
| 20:18.44 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
| 20:21.16 | boch | or is there a better way to test the state of a provider? |
| 20:23.23 | ManxPower | boch, you cannot know the results without actually testing the audio path. |
| 20:23.56 | ManxPower | For example, use your .call file to call a PSTN number that answers and sends DTMF, then on the other leg of the call send it to something that waits for DTMF. |
| 20:26.58 | *** join/#asterisk magaf__ (n=Heaven@acxl172.neoplus.adsl.tpnet.pl) |
| 20:27.01 | magaf__ | hello |
| 20:27.16 | magaf__ | does anoyone has an experience with mISDN and avm fritz card? |
| 20:27.32 | magaf__ | i have to set a outbound and inbound context , i dont know how |
| 20:27.54 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:28.29 | generalhan | boch: where are the docs regarding .call files ? |
| 20:33.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:34.46 | *** join/#asterisk |cleric| (n=dacleric@p54822FDB.dip0.t-ipconnect.de) |
| 20:34.57 | *** join/#asterisk PrivalAC (n=someone@64.235.216.178) |
| 20:35.06 | boch | generalhan http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 20:35.22 | PrivalAC | Hi, anyonw has exterience with CT-5 or CallTransferDisconnect with Asterisk over a PRI? |
| 20:35.53 | SplasPood | hey anyone know the legalities involved, within the US, with charging customers to call toll free numbers? |
| 20:36.07 | generalhan | boch: yea i found it ... this stuff confuses the hell outta me |
| 20:37.05 | ckwall | ok, quick question... is there a way to make the dial plan automatically append digits to send with the dialed numbers? ie 1+area code? I have only an LD T1, and want to be able to specify in my dial plan that if only 7 numbers are dialed, append 1+801 in front of it. |
| 20:37.16 | mercestes | If the point of VoIP is to be cheaper than PSTN I fail to see why one would want to attempt to charge users for calling toll free numbers. |
| 20:37.33 | generalhan | ckwall: yes ill pastebin mine to show you cause i do thatsame thing |
| 20:37.33 | ckwall | my current lines look like: |
| 20:37.41 | ckwall | ok, thanks |
| 20:38.20 | PrivalAC | Ok, no ones knows Ct5... Anyone knows how to do a hook flash on a softphone? |
| 20:38.39 | generalhan | http://generalhan.pastebin.ca/54602 |
| 20:39.13 | ckwall | thanks... checking it out. |
| 20:39.20 | generalhan | np |
| 20:39.59 | *** join/#asterisk jahani (n=k@41.250.39.59) |
| 20:40.48 | PrivalAC | Anyone knows how to do a hook flash on a softphone? |
| 20:41.18 | docelm0 | w00t! |
| 20:41.29 | Druken | why would you need a hookflash on a softphone? |
| 20:41.31 | MikeJ[Laptop] | PrivalAC, I beleie you can do flash in 2833 |
| 20:41.32 | docelm0 | yes close the window and reopen it |
| 20:42.30 | PrivalAC | CT5 or CallTransferDisconnect requires to to a hookflash to perform the transfer. |
| 20:42.50 | kruz123 | all: is matt b in here? |
| 20:43.16 | *** join/#asterisk asterboy (n=root@S010600485480f4be.ed.shawcable.net) |
| 20:43.27 | PrivalAC | Basically you are with a party, you hookflash to get a second line and you dial a third party. You hookflash again and you have all 2 parties with you. You hangup and the 2 other parties stay connected but they don't use any lines on your PRI. |
| 20:44.15 | magaf__ | when i have a asterisk box connected to real telephony central, how to assign a number to my asterisk box , |
| 20:44.19 | magaf__ | in telephony central? |
| 20:44.28 | magaf__ | im contected via isdn card |
| 20:44.33 | PrivalAC | http://www.callamericacom.com/pdf/ctd_instructions.pdf |
| 20:44.37 | asterboy | Any suggestions on hardware to help managers listen in next to a Polycom phone? |
| 20:44.56 | *** join/#asterisk Dr-Linux (n=huh@202.59.73.131) |
| 20:45.41 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 20:45.52 | Dr-Linux | Hi all |
| 20:46.54 | asterboy | high |
| 20:47.05 | *** join/#asterisk ToTo (n=ToTo@host253-91.pool8256.interbusiness.it) |
| 20:47.19 | docelm0 | HO! |
| 20:47.43 | Dr-Linux | hey! |
| 20:48.25 | Hmmhesays | apparently ser doesn't like having a few hundred if statements in it |
| 20:48.36 | *** join/#asterisk jahani (n=k@41.250.39.59) |
| 20:49.03 | asterboy | what is the lingo for listening in on a conversation right next to the phone? |
| 20:49.10 | asterboy | tapping is one |
| 20:49.37 | *** join/#asterisk fjean (n=fjean@201.29.130.118) |
| 20:49.38 | docelm0 | listening in |
| 20:49.46 | docelm0 | eves droppin |
| 20:49.48 | docelm0 | g |
| 20:50.03 | docelm0 | monitoring |
| 20:50.04 | asterboy | eves might help me find the equipment I'm looking for |
| 20:50.06 | docelm0 | ??? |
| 20:50.08 | asterboy | or monitoring |
| 20:50.36 | docelm0 | asterboy you looking for a way to tap a POTS line? |
| 20:50.49 | *** join/#asterisk Inkubot (n=inkubot@200.74.182.50) |
| 20:50.57 | asterboy | only for a manager right next to the person on the telephone. |
| 20:51.03 | asterboy | want to listen in but no mic |
| 20:51.11 | docelm0 | ohh call center or something? |
| 20:51.17 | asterboy | crisis center |
| 20:51.24 | docelm0 | ahh.. what kinda phones? |
| 20:51.29 | asterboy | Polycom |
| 20:51.39 | asterboy | I know they have a headset...but |
| 20:51.48 | docelm0 | You could get a splitter for for the headset.. |
| 20:51.50 | asterboy | ...you can only choose one or the other not both |
| 20:51.54 | docelm0 | Most call centers have em |
| 20:52.09 | Mystiq | ZapBarge |
| 20:52.15 | mercestes | Speaker phone. lol |
| 20:52.22 | asterboy | ya but you need another phone for zapbarge |
| 20:52.33 | docelm0 | you have to use TDM |
| 20:52.34 | Dr-Linux | any wireless phone works with asterisk? |
| 20:52.38 | docelm0 | thats an even bigger issue |
| 20:52.42 | docelm0 | all |
| 20:52.52 | docelm0 | Dr-Linux, all that is.. |
| 20:53.03 | docelm0 | Dr-Linux, just have to be sipv2 compatible |
| 20:53.03 | mercestes | Wireless as in WiFi, Wireless as in cordless phones, or wireless as in cellular? |
| 20:53.34 | docelm0 | come on Dr-Linux spit it out.. :) |
| 20:53.34 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-73.apple.com) |
| 20:54.28 | Dr-Linux | well, i know only cisco, spa and polycom ip phone.s |
| 20:54.40 | mercestes | Dr- Linux: None of which are wireless. |
| 20:55.05 | docelm0 | true.. |
| 20:55.09 | docelm0 | newb? |
| 20:55.09 | mercestes | Dr-Linux The wireless headsets for POlycom's seem to be hit and miss at best.. |
| 20:55.10 | docelm0 | :) |
| 20:55.20 | Dr-Linux | mercestes: i mean like maybe some one phas has bluetooth or something, which works. |
| 20:55.21 | docelm0 | ohh those |
| 20:55.28 | mercestes | Dr-Linux The Wifi phones, as far as I can tell, have very short range. |
| 20:55.38 | docelm0 | check out plantronics.. |
| 20:55.42 | docelm0 | there shit is decient |
| 20:55.42 | mercestes | Dr-Linux Best success I've had is cordless phones on an ATA device.... |
| 20:55.48 | Dr-Linux | mercestes: i'm sorry i don't know what's Wifi |
| 20:55.53 | mercestes | Got a bunch of Plantronics here, I have heard good things of the 550. |
| 20:56.14 | Mystiq | or gnnetcom |
| 20:56.24 | mercestes | Dr-Linux Wifi is Wireless Networking just like on a laptop..they make WiFi phones that work off of wireless networking...but..the phones seem to have a very short range. |
| 20:56.31 | Dr-Linux | we have plantronics phones in our Pakistan call center |
| 20:57.02 | docelm0 | Dr-Linux you guys calling or being called? |
| 20:57.12 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 20:57.34 | mjackson | Yay, now i've got a hampsterdance ringtone on our polycom's.... they'll love that upstairs tomorrow |
| 20:57.55 | b00mer_ | I am getting echo and poor quality on calls going out my PRI which is the last place I would have expected. Is there something I can do to debug / diagnose the issues? |
| 20:57.57 | docelm0 | you will be fired by friday |
| 20:58.00 | Dr-Linux | docelm0: have both facilities, but ofcos being called, and provide support to Americans |
| 20:58.15 | docelm0 | ohh lord |
| 20:58.20 | mjackson | ^^ |
| 20:58.20 | docelm0 | For what companies? |
| 20:58.42 | mjackson | Debt consolidation inbound call center |
| 20:59.08 | docelm0 | as if Dell going to India wasnt bad enough |
| 20:59.24 | Dr-Linux | b00mer_: what about your rx/tx ? and echotraining stuff ? |
| 20:59.28 | docelm0 | now we have to tell people half way around the world we dont know how to manage our money? shit.. |
| 20:59.53 | Dr-Linux | docelm0: we hare our own transaction company |
| 21:00.09 | docelm0 | meaning what? |
| 21:00.27 | mercestes | you should hire illegal american immigrants. |
| 21:00.28 | docelm0 | There are 1000's of transaction types.. |
| 21:00.30 | Dr-Linux | docelm0: we have 3 call centers, Pakistan/USA/DR |
| 21:00.40 | docelm0 | DR? |
| 21:01.07 | b00mer_ | Dr-Linux : rxgain = 0.0 txgain = 0.0 echocancel=yes echocancelwhenbridged=yes |
| 21:01.09 | Dr-Linux | Dominicom Republic << that's for spanish, bcoz we can't get spanish guys here in pakistan |
| 21:01.22 | docelm0 | ya think |
| 21:01.49 | Dr-Linux | b00mer_: increase your rxgain to 2.0 |
| 21:02.01 | Dr-Linux | and echotraining=yes |
| 21:02.43 | Dr-Linux | b00mer_: but in priority we are not using asterisk, but TV |
| 21:02.53 | Dr-Linux | we are just planning to move for Asterisk |
| 21:03.12 | b00mer_ | Dr-Linux : huh? TV? |
| 21:03.44 | Dr-Linux | docelm0: actually we provide IVR solutions as well, now building them in AGI asterisk |
| 21:03.58 | Dr-Linux | b00mer_: TV = Televantage |
| 21:04.04 | docelm0 | what language? |
| 21:04.18 | Dr-Linux | docelm0: english and spanish |
| 21:04.51 | docelm0 | good lord.. programming language.. You have established broken english and spanish |
| 21:05.10 | Dr-Linux | lol |
| 21:05.16 | b00mer_ | Dr-Linux : thanks... I'll try and get some feed back with those new settings |
| 21:05.21 | Dr-Linux | docelm0: C for AGI |
| 21:06.16 | docelm0 | when did panasync pull that one-liner out his ass? |
| 21:06.17 | Dr-Linux | b00mer_: you should always play with your rx/tx for low/hight audio |
| 21:06.37 | Dr-Linux | sorry? |
| 21:06.48 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 21:07.18 | jpabuyer | that must have hurt |
| 21:07.31 | *** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net) |
| 21:07.37 | r0d3nt|m | quit |
| 21:07.39 | r0d3nt|m | opps |
| 21:08.12 | freakGB | hi, is there any way i can try to diagnose a problem with an x100p. The problem is i cannot get it to answer an incoming call. I have set up a catch all inbound route in aah 2.8. outbound is working fine. |
| 21:08.41 | docelm0 | freakGB are you useing ANSWER()? |
| 21:08.47 | Dr-Linux | aah 2.8 ? |
| 21:08.59 | docelm0 | A@H and if you READ THE DAMN TOPIC! |
| 21:09.04 | jpabuyer | he means Asterisk@home 2.8 |
| 21:09.17 | freakGB | sorry |
| 21:09.20 | mercestes | lol |
| 21:09.23 | docelm0 | #asterisk are for power users.. not newbies who try to learn |
| 21:09.27 | freakGB | second line missed that |
| 21:09.35 | docelm0 | all good tho |
| 21:09.37 | docelm0 | :) |
| 21:09.37 | Dr-Linux | docelm0: i'm a newbie :S |
| 21:09.45 | mercestes | Dr-Linux: We know. |
| 21:09.47 | docelm0 | Dr-Linux, and it shows.. |
| 21:09.48 | docelm0 | :) |
| 21:09.59 | Dr-Linux | ahh ? nice pbx |
| 21:10.09 | Dr-Linux | aah ohh uff ouch! |
| 21:11.34 | Dr-Linux | mercestes: honestly i never seen any of IP phone in real, and neither it available in my country |
| 21:12.07 | Dr-Linux | but i have configured many 7940/60 |
| 21:12.13 | mercestes | Not bad. |
| 21:12.35 | mercestes | My 7940 has custom logos for ME. |
| 21:12.42 | mercestes | *struts* |
| 21:13.12 | Mystiq | in black & white, w00ptidoe |
| 21:13.42 | Mystiq | cisco needs good backlit phones |
| 21:14.04 | mercestes | I had a linux penguin on our linux guy's phone..I had a celtic knotwork moon. |
| 21:14.12 | mercestes | our sales manager had a middle finger bmp. |
| 21:14.28 | asterboy | Is there a bluetooth headset that can be used with line taps? |
| 21:15.02 | gandhijee | where is a good guide to setting up asterisk w/ CDR? |
| 21:15.19 | Dr-Linux | gandhijee: Indian? |
| 21:15.30 | gandhijee | yeah |
| 21:15.35 | gandhijee | guju |
| 21:16.01 | Dr-Linux | gandhijee: lolzz gandhi jee ka nick kahan say rakh liya? :P |
| 21:16.57 | asterboy | I'm thinking my only choice is for a pickup coil...not very high tech |
| 21:17.03 | brif8 | what is the difference between (a) stop now and then asterisk -vvvvvgc VS (b) from CLI restart ? What occurs differently ore are they basically the same esp memory leakage etc.. |
| 21:17.04 | Dr-Linux | gandhijee: there is third party CDR GUI system on WIKI , look for it .. it's cool |
| 21:17.26 | asterboy | What do the call centers use when a manager is listening in on a call right next to the agent? |
| 21:17.46 | gandhijee | man, i think i know what you said, but i don't really understand hindi, nor can i read Gujarati |
| 21:17.59 | gandhijee | voip-info wiki? |
| 21:18.08 | Dr-Linux | gandhijee: yes |
| 21:18.20 | Dr-Linux | asterboy: QueueMetrics? |
| 21:18.31 | asterboy | looking |
| 21:18.38 | gandhijee | and i think you asked where i live, either that or where i've been roaming around |
| 21:18.49 | *** part/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 21:18.53 | *** join/#asterisk gandhijee (n=gandhije@pool-71-161-34-140.clppva.east.verizon.net) |
| 21:19.38 | Dr-Linux | gandhijee: no, lolzz actually you nick is, i asked how you got her over here :) |
| 21:19.48 | Dr-Linux | s/you/your |
| 21:19.56 | Zodiacal | anyone know if theres a way to disable the phones speaker for a specific call? or just reduce the volume to 0 or somthing? the reason is when i page over the sound card, the handsets speaker volume echos back what the caller says at like 1000% the volume.. its quite anoying |
| 21:20.04 | gandhijee | her? |
| 21:20.48 | asterboy | oh, QueueMetrics is software...I need a hardware solution |
| 21:20.59 | Dr-Linux | gandhijee: who was Gandhi ? |
| 21:21.08 | gandhijee | mohandas |
| 21:21.31 | gandhijee | and the free-er of india |
| 21:21.31 | asterboy | Wasn't Gandhi a weed smoker? |
| 21:21.40 | Dr-Linux | gandhijee: andragandhi or Raju Gandhi ? |
| 21:22.28 | Dr-Linux | <gandhijee> and the free-er of india >> the one was HER |
| 21:22.50 | gandhijee | you mean gandhi's daughter? |
| 21:23.01 | mercestes | gandhi had a daughter? |
| 21:23.11 | mercestes | bet she was hot. |
| 21:23.24 | gandhijee | or one of the women from the gandhi family |
| 21:23.52 | Dr-Linux | gandhi was a Lady and her SON name was Raju Gandhi, both were Killed. |
| 21:24.02 | mercestes | aw....that's sad. |
| 21:24.31 | Dr-Linux | bother were killed by Indian's Sikh peoples |
| 21:24.37 | gandhijee | learn something new every day |
| 21:24.43 | mercestes | Even sadder?? |
| 21:24.54 | gandhijee | are you talking about indira ghandi? |
| 21:25.11 | Dr-Linux | gandhijee: Yes |
| 21:25.22 | gandhijee | i think it was some of her bodygaurds that tried to kill her for messin up the golden temple |
| 21:26.34 | Dr-Linux | gandhijee: well, both were killed on the accasion when they was wearing bullit proof jackets. |
| 21:26.47 | Dr-Linux | Raju gandhi was fired during his speach. |
| 21:27.01 | Dr-Linux | and killer were from Sikh traditions |
| 21:27.14 | gandhijee | yeah |
| 21:27.38 | Dr-Linux | gandhijee: however you maybe know better, as you are an Indian i'm not :) |
| 21:27.45 | brif8 | what is the difference between (a) stop now and then asterisk -vvvvvgc VS (b) from CLI restart ? What occurs differently ore are they basically the same esp memory leakage etc.. |
| 21:28.02 | Dr-Linux | gandhijee: i'm a tribal |
| 21:28.21 | gandhijee | a tribal??? |
| 21:28.58 | Dr-Linux | gandhijee: thre is only tribal exist in all the world, and that is in Pakistan. |
| 21:29.14 | gandhijee | rajiv was killed by some Tamil rebels |
| 21:29.21 | gandhijee | ahh |
| 21:29.44 | Dr-Linux | gandhijee: but sikh were after Him. |
| 21:30.04 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 21:30.06 | gandhijee | its possible. |
| 21:30.43 | Dr-Linux | gandhijee: do you like Wajpayeeee ? |
| 21:31.44 | ckwall | I am looking for information on outbound caller id... but I am not seeing anything helpfull. What is the appropriate term I should be searching for? ANI? DNIS? I am trying to set all my calls to one spcified caller id number. |
| 21:31.49 | mercestes | Welcome to channel Asterisk. The number one choice of call centers outsourced to India. |
| 21:32.24 | mercestes | context,1,Set(CALLERID= "Name" <number>) |
| 21:32.30 | mercestes | example. |
| 21:32.33 | Hmmhesays | exten => s,1,Set(CALLERID(num)="123456"); exten => s,2,Dial(Tech/foo) |
| 21:33.01 | mercestes | 6969,1,Set(CALLERID = "Mercestes" <12223334444>) |
| 21:33.02 | *** join/#asterisk SoMeOnEnUlL (n=morris@p849-adslbkkct1.C.csloxinfo.net) |
| 21:33.45 | ckwall | so that is by extension... right? I want to set it globaly. I am having to go through to each extension and specify it. |
| 21:33.55 | mercestes | could do it in sip.conf. |
| 21:34.03 | *** join/#asterisk pbx321 (n=pbx321@203.177.234.49) |
| 21:34.03 | mercestes | under global. |
| 21:34.21 | mercestes | callerid = "Mercestes" <12223334444> |
| 21:34.23 | trelane_ | you know, I love it when competitors in the area make asterisk look hard |
| 21:34.31 | trelane_ | my $APPARENT_WIZARDRY increases 10 fold |
| 21:34.35 | ckwall | hmm. ok. let me look at that |
| 21:35.14 | mercestes | That's my real number, so don't use that...I don't want people calling me. |
| 21:35.27 | asterboy | too bad you can't have BOTH the headset and handset on the Polycom active at the same time. |
| 21:35.44 | Dr-Linux | mercestes: i never sale out this number? how you got? :S |
| 21:36.37 | gandhijee | Dr-Linux: wtf is Wajpayeeee? |
| 21:36.41 | jpabuyer | What's masquerading channels for un channels.h ? |
| 21:37.04 | ckwall | ok, i dont think i did it correctly. |
| 21:37.17 | Dr-Linux | gandhijee: what was the name of X president of India? :S |
| 21:37.18 | ckwall | i added it under my [general] section |
| 21:37.22 | ckwall | is that not correct? |
| 21:37.43 | mercestes | Should be. |
| 21:37.43 | ckwall | [general] |
| 21:37.43 | ckwall | callerid = "Spherous" <1-888-777-6666> |
| 21:37.48 | docelm0 | Hay anyone familiary w/ early media on Asteris? |
| 21:37.50 | docelm0 | asterisk |
| 21:38.06 | mercestes | Shouldn't have dashes. |
| 21:38.10 | ckwall | ah |
| 21:38.14 | ckwall | lemme retry |
| 21:38.45 | mercestes | exactly as I typed....except...use your own name and number. |
| 21:39.00 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 21:39.02 | ckwall | hmmm. still no |
| 21:39.19 | ckwall | thats what i thought when I got into it, but I cannot get it to work. |
| 21:39.37 | ckwall | the name and number are not associated with anything except for what to display, right? |
| 21:39.50 | mercestes | basically. |
| 21:39.58 | mercestes | If you are calling out via a provider they could be resetting yoru caller ID. |
| 21:40.09 | mercestes | if you are calling within * or * to * it should work tho. |
| 21:40.19 | ckwall | well, it works correctly when i set it by extension in the sip.conf |
| 21:40.28 | mercestes | could just set caller ID in all your sip entries then, but callerID under global *should* work. |
| 21:40.43 | SoMeOnEnUlL | anyone experienced sip - NAT - Asterisk enviroment? |
| 21:40.44 | mercestes | but if everything worked as it should I would be out of a job so, I shall not complain. |
| 21:40.56 | ckwall | when you say under global... what do you mean. I do not have a global section in my sip.conf |
| 21:41.01 | ckwall | is that maybe what the problem is? |
| 21:41.02 | mercestes | [global] |
| 21:41.07 | mercestes | yess..... |
| 21:41.11 | mercestes | should be at the top of your sip.conf. |
| 21:41.18 | mercestes | you didn't happen to make samples did you? |
| 21:41.24 | ckwall | yes |
| 21:41.41 | generalhan | Anyone in here working with .call files ? |
| 21:41.44 | ckwall | my first section is general |
| 21:42.03 | mercestes | oh.. |
| 21:42.06 | mercestes | s/global/general |
| 21:42.11 | mercestes | :S |
| 21:42.17 | mercestes | Yea, it's general. |
| 21:42.19 | mercestes | Should work. |
| 21:42.24 | ckwall | suck! |
| 21:42.26 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 21:42.30 | mercestes | Sorry. |
| 21:42.33 | Dr-Linux | mercestes: that's what i was thinking .. i thought maybe i'm learning a new thing |
| 21:43.12 | *** join/#asterisk Bentley (n=Bentley@S010600301baf55dd.cg.shawcable.net) |
| 21:43.30 | ckwall | does it matter where in the [general] i put it? |
| 21:44.45 | mercestes | before the next context hopefully. |
| 21:45.01 | mercestes | That should set it everywhere and subsequent values should overwrite that... |
| 21:45.12 | mercestes | but it's possible that "Callerid" doesn't have that functionality. |
| 21:45.39 | ckwall | ok... well, onto the next issue |
| 21:46.03 | Dr-Linux | mercestes: what does that mean, as we define in sip.conf for each user "mailbox=2343" |
| 21:46.16 | ckwall | is anyone here using the polycom soundpoint ip 501? |
| 21:46.44 | jpabuyer | yes |
| 21:46.50 | ckwall | do i really have to use an ftp server with it to specify all of the username and secret settings for use with my sip.conf file? |
| 21:47.05 | jpabuyer | no |
| 21:47.07 | mercestes | could turn it off I guess. |
| 21:47.16 | mercestes | set it via........phone interface? web interface? tftp is nice. |
| 21:47.29 | jpabuyer | use the web interface |
| 21:47.33 | ckwall | ok. |
| 21:47.39 | ckwall | i can set the user name and secret there? |
| 21:47.46 | jpabuyer | yes |
| 21:48.22 | jpabuyer | then phone takes about 2 mins to start up with the web admin interface upon reboot |
| 21:48.26 | ckwall | ok, so do i set each line up as a user with secret, and make an entry in sip.conf based off of the lines? |
| 21:48.35 | ckwall | or can i just do it by phone? |
| 21:49.00 | jpabuyer | zZZ |
| 21:49.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:49.13 | mercestes | each line can be a different user/secret...or the same user/secret.....or be blank...up to you. |
| 21:50.50 | mercestes | Hvae you read the Polycom manual and atleast the basics on the Asterisk WIKI? |
| 21:51.00 | jpabuyer | of course not |
| 21:51.10 | ckwall | actually, i have read the asterisk wiki... |
| 21:51.18 | mercestes | Polycom has a very good manual. |
| 21:51.19 | ckwall | using the info there i have not been able to connect this phone. |
| 21:51.47 | ckwall | I will check out the manual. bought phones used... no info. |
| 21:52.04 | ckwall | googleing is getting me confused |
| 21:52.18 | mercestes | google polycom user manual look for a IP501 pdf. |
| 21:54.30 | ckwall | thank you |
| 21:55.00 | mercestes | NP. |
| 21:55.15 | mercestes | <PROTECTED> |
| 21:55.16 | fjean | anyone knows to authenticate SER in sip.conf to allow incoming DIDs ? |
| 21:55.21 | mercestes | .........what...no msg? Damn! |
| 21:56.34 | *** join/#asterisk mspiceland (n=mike@gateway.digium.com) |
| 21:56.38 | *** part/#asterisk mspiceland (n=mike@gateway.digium.com) |
| 21:58.16 | fjean | I am willing to paypal a small amount for succesfull help :-) |
| 21:58.34 | gandhijee | is libpri needed for sangoma hardware? |
| 21:58.36 | mercestes | lol nice answer. |
| 22:00.52 | mercestes | http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 22:02.11 | Dr-Linux | gandhijee: yes, if you are going for T1/E1 |
| 22:02.17 | mercestes | Autocreatepeer = yes|no : If set, anyone will be able to log in as a peer (with no check of credentials; useful for operation with SER). Default no. |
| 22:02.31 | fjean | mercestes: the thing is I cannot use a username and password, and cannot use autocreatepeer as it's not safe |
| 22:02.39 | gandhijee | Dr-Linux: its connecting to a Rhino Bank. i set up the machine about 5 months ago and forgot how i did |
| 22:03.05 | gandhijee | Dr-Linux: has digium hardware too, i remember i had to go through some funky stuff |
| 22:03.05 | Dr-Linux | Rhino bank |
| 22:03.12 | gandhijee | Dr-Linux: yep |
| 22:03.22 | Dr-Linux | gandhijee: sorry i was unawar what's asterisk 5 month ago |
| 22:03.26 | gandhijee | rhino channel bank |
| 22:03.54 | gandhijee | it was prolly more like 3 |
| 22:04.31 | fjean | mercestesL i tried by specifying the host ip and insecure, but did not work |
| 22:04.43 | fjean | got a 403 |
| 22:08.13 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 22:08.28 | copland | Is there any telephone techs in here that I can ask a stupid question |
| 22:09.03 | mercestes | Sure |
| 22:09.04 | mercestes | Ask away |
| 22:09.54 | terrapen | sometimes mediawiki really pisses me off |
| 22:10.55 | copland | I am looking for the name of the wire anchors that teleco line techs use to fashion cabling to the side of buildings |
| 22:11.07 | asterboy | somebody has to make a mono headset without mic for eaves drop? |
| 22:11.18 | Dr-Linux | question, how it possible i want give out my own caller ID, if calling into the Pakistan. My * box is also in Pakistan with FXOs ? |
| 22:11.40 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 22:12.53 | mercestes | Um..that's a good q....actually....I always called them wall anchors. |
| 22:13.38 | copland | mercestes: i am trying to find some to buy and cant find any |
| 22:13.48 | *** join/#asterisk IceManRISK (n=kart@201.66.47.9) |
| 22:13.56 | Dr-Linux | mercestes: i tried differnt things, but caller always get my PSTN line actual number :S |
| 22:14.05 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 22:14.15 | copland | And the verizon tech who was by the house today kinda refused to give me like the 5 i needed some are nice others are assholes |
| 22:14.21 | IceManRISK | Hey, anyone here use IAXCLient ? |
| 22:15.21 | IceManRISK | Hey, anyone here use IAXCLient ? |
| 22:15.45 | Dr-Linux | ~suggestion |
| 22:15.59 | Dr-Linux | ~suggestions |
| 22:16.00 | jbot | it has been said that suggestions is 1) Don't ask to ask. Just say your problem, 2) Don't repeat until 5 mins after, 3) Read and re-read the docs first, then admit it if you REALLY don't understand. You're wasting your time and ours if you haven't at least tried. 4) If your problem ain't solved, come back in 12 hrs or 24 hrs later. We're very international. 5) Be ... |
| 22:19.11 | gandhijee | Dr-Linux: you using POTS lines or an T1/E1 to place the call? |
| 22:19.31 | *** part/#asterisk AsteriskAddict (n=speedy@r172h230.dixie-net.com) |
| 22:19.55 | gandhijee | if u r using POTS lines, you can't set the outbound caller ID, on BRI/PRI's you can |
| 22:20.18 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 22:20.41 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:21.26 | *** join/#asterisk japerry (n=japerry@216.231.51.208) |
| 22:21.31 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:22.50 | japerry | heya, anyone in here dealt with em_w T1 interfaces before? I have a TE100P |
| 22:23.59 | gandhijee | japerry: what you tryin to do? i setup some stuff w/ sangoma hardware |
| 22:24.01 | japerry | basically I get a call on a did number, and asterisk sees it, but the phone doesn't ring and there is no message |
| 22:24.05 | *** join/#asterisk syle2 (n=blag@unaffiliated/syle) |
| 22:24.28 | japerry | basically I have 4 channels that are em_wink, and I think the timing is off |
| 22:24.40 | japerry | in zttool I can't get it to say anything but internally clocked |
| 22:24.41 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 22:24.54 | gandhijee | dunno, my stuff goes to a channel bank, i was a TDM400 for connectivity |
| 22:25.35 | japerry | hmm. yah makes matters worse, this line is from Verizon .. a 'flexgrow' .. its not a PRI line |
| 22:25.50 | gandhijee | isn't that the Voice and data lines? |
| 22:25.57 | japerry | yah but we just have it doing voice |
| 22:26.11 | gandhijee | i am gonna setup a place in MD w/ cavliers phonom service... |
| 22:26.26 | gandhijee | its kinda the same |
| 22:27.06 | japerry | okay |
| 22:27.17 | japerry | yah I was told that the lines are 4 DIOD |
| 22:27.21 | gandhijee | Voice and Data over one line, but they give us a channel bank that is a MGCP endpoint and combines the signal back in to T1 format |
| 22:27.42 | japerry | hmm |
| 22:28.08 | gandhijee | i dunno if u have them in your area, or how far you are into your contract w/ verizon |
| 22:28.17 | gandhijee | but that might be the better option.... |
| 22:28.50 | *** join/#asterisk imperfect- (n=tbw@c-68-58-148-186.hsd1.in.comcast.net) |
| 22:28.50 | japerry | what? cavliers phonom service? |
| 22:28.58 | gandhijee | yeah |
| 22:29.09 | imperfect- | I've got a problem using broadvoice for my home phone |
| 22:29.18 | imperfect- | I'm using asterisk to peer w/ a sipura 2100 |
| 22:29.29 | imperfect- | the calls come in, it rings muh phones, and when I answer, fast busy... |
| 22:29.30 | imperfect- | Any ieas? |
| 22:29.48 | japerry | eehh we have another T1 doing our data, but I'm worried that VoIP could be flaky |
| 22:30.06 | japerry | we just started with verizon so I could get out of it, but I'm sure its just an asterisk configuration program |
| 22:30.53 | gandhijee | the Cav VoIP is over thier private network |
| 22:31.12 | japerry | but you still have to have an internet connection .... |
| 22:31.23 | gandhijee | no you don't |
| 22:31.29 | japerry | ? hmm |
| 22:31.34 | IceManRISK | Anyone here know where i can download IAXCLIENT ? |
| 22:31.38 | dlynes_ | japerry: voip does not require internet |
| 22:31.44 | gandhijee | they roll it out to you off an ethernet hand off |
| 22:31.45 | dlynes_ | japerry: just a network |
| 22:32.01 | gandhijee | its VoIP, but think about it as VoIP over a LAN |
| 22:32.14 | gandhijee | you have a direct connection to there office |
| 22:32.30 | japerry | d1ynes_ right but we have one connection going to the outside world at the moemnt, so gandhijee is saying that they bring in another line, correct? |
| 22:33.07 | gandhijee | yes, its is similar to flexgrow |
| 22:33.31 | gandhijee | but they give you a channel bank that is an MGCP endpoint that can do 12 loopstart lines |
| 22:33.39 | japerry | hehe too bad they don't service Seattle =/ |
| 22:33.45 | gandhijee | O |
| 22:33.52 | gandhijee | that sucks |
| 22:33.56 | japerry | yah but that sounds like what I want |
| 22:33.58 | japerry | geh |
| 22:34.05 | gandhijee | Caviler is pretty rockin |
| 22:35.27 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:36.24 | japerry | well meh that still doesn't answer the question--how do you get the telco to be your clock for the TE100P |
| 22:36.43 | gandhijee | it should be in the manual for it somewhere |
| 22:36.48 | gandhijee | i know it was for my sangoma |
| 22:45.34 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-73.apple.com) |
| 22:45.41 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:58.46 | gandhijee | anyone know what IP phone walmart supposedly offers? |
| 23:02.30 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:06.41 | asterboy | yep, WalMart will offer the cheapest piece of outsourced crap imaginable. |
| 23:07.28 | asterboy | of course it will fall under their "price point" marketing scheam |
| 23:08.05 | gandhijee | so what would it be? a budgetone? |
| 23:08.41 | asterboy | suck the white trailer trash into that flourescent lit square box and sell them on the cheapest price, but then offer those other overpriced goods. |
| 23:09.13 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.33.Dial1.SanJose1.Level3.net) |
| 23:09.20 | asterboy | we really are devolving as a species...fucking each other over for a % at every chance. |
| 23:09.43 | asterboy | canablistic capitalism at its finest |
| 23:10.27 | tainted- | gandhijee didn't know walmart offered ip phones |
| 23:10.57 | asterboy | maybe they will brand it with their new trademark...the smiley face |
| 23:11.36 | asterboy | AOL is offering free incoming phone service too |
| 23:11.36 | gandhijee | i just saw it in the AsteriskTFOT book... |
| 23:11.46 | tainted- | well it must be true then |
| 23:12.44 | gandhijee | i never said it was, i was just asking if it was |
| 23:13.23 | mjackson | Anybody know why the asterisk server would end up playing silence when it was suppoed to be playing sound files? |
| 23:13.54 | tainted- | mjackson is the sound file a recording of silence? |
| 23:14.19 | mjackson | no ^^ |
| 23:14.40 | mjackson | CLI shows sounds playing successfully w/o error, but it never makes it to the phone. Not when dialing in, or internally, wherever |
| 23:14.52 | asterboy | mjackson, check your musiconhold.conf |
| 23:14.55 | mjackson | did a reinstall / reconfigure today, and ended up with the same problem |
| 23:14.56 | tainted- | check client firewall |
| 23:15.21 | mjackson | it's not musiconhold... anything played with Playback or Background |
| 23:15.40 | asterboy | well does musiconhold work? |
| 23:15.43 | mjackson | Tested it between some polycom SIP phones all behind the firewall |
| 23:15.48 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 23:16.01 | mjackson | good question. I'm reinstalling the OS again right now... when I get it back up I'll let ya know |
| 23:16.04 | dlynes_ | mjackson: i ran into the same problem the other day |
| 23:16.21 | asterboy | mpg123 installed and working? |
| 23:16.25 | dlynes_ | mjackson: i don't know if it affected music on hold or not, but it definitely affected autoattendant |
| 23:16.47 | dlynes_ | mjackson: are you using 1.2.7.1 as well? |
| 23:16.55 | asterboy | asterrecipies has a great trick to turn those songs into a native format and save your system the extra process |
| 23:16.58 | mjackson | *nodnod* mpg123, specifically version 0.59r |
| 23:17.10 | dlynes_ | Don't need mpg123 |
| 23:17.13 | dlynes_ | It's a pos |
| 23:17.17 | dlynes_ | Use native mode |
| 23:17.23 | mjackson | but i don't think mpg123 is necessary for playback() and background() with gsm files |
| 23:17.38 | mjackson | dlynes: using the latest stable version |
| 23:17.47 | dlynes_ | mjackson: mpg123 isn't necessary for anything, except placing a huge, buggy load on your system |
| 23:17.55 | mjackson | 1.2.7.1 |
| 23:18.23 | dlynes_ | mjackson: and your voice works in both directions, when you have this issue with autoattendant files, right? |
| 23:18.28 | mjackson | yup |
| 23:18.52 | dlynes_ | mjackson: and it also affects playback of the files for voicemail menus, right? |
| 23:18.56 | mjackson | can also pass the call into contexts on another asterisk box via IAX, do stuffs, and pass back |
| 23:19.02 | mjackson | that's right |
| 23:19.16 | dlynes_ | mjackson: yeah...sounds like you've got the exact same problem as I did |
| 23:19.24 | dlynes_ | I just didn't have the patience to deal with it |
| 23:19.43 | dlynes_ | I just said forget it, bought a new box, formatted it, installed all new hardware, and deployed a new mahcine |
| 23:19.50 | mjackson | lol |
| 23:19.58 | dlynes_ | I've got the old box sitting in the van |
| 23:20.04 | dlynes_ | when I get a chance, I'll take a look at it |
| 23:20.32 | mjackson | the boss loves to buy these overpriced dell servers :P Dual zeon, raid 5.... |
| 23:20.40 | dlynes_ | asterboy: btw, the problem we're having is not a zero volume gsm file, or anything like that |
| 23:20.51 | mjackson | throwing it on another box would be touch |
| 23:21.00 | mjackson | espeically since i now it was working at one point on this hardware |
| 23:21.01 | dlynes_ | asterboy: The file plays, but asterisk loses track of the audio for whatever reason |
| 23:21.35 | dlynes_ | 1.2.7.1 is the only version I've had this happen on, too |
| 23:22.19 | asterboy | is there an mpg123 process running? |
| 23:22.31 | dlynes_ | asterboy: I don't even use mpg123...it's a huge piece of crap |
| 23:22.52 | mjackson | hmm.... think i might try 1.2.6 |
| 23:22.57 | asterboy | ok, well try it then, just to see if it's * having the problem with the gsm file |
| 23:23.07 | asterboy | try an mp3 |
| 23:23.13 | asterboy | a different new one |
| 23:23.20 | dlynes_ | asterboy: in my case, it's a wav file |
| 23:23.23 | asterboy | that you know works in windows or something |
| 23:23.33 | dlynes_ | asterboy: It works in linux, too |
| 23:23.43 | asterboy | use a different format...just change things up, so you can try to find the failure point |
| 23:23.57 | dlynes_ | asterboy: I downloaded an entire snapshot of the whole config system, all the voicemail, and the autoattendant files |
| 23:24.05 | dlynes_ | asterboy: and put them onto the new machine before deploying it |
| 23:24.10 | dlynes_ | asterboy: the new machine has no issues |
| 23:24.39 | asterboy | still doesn't change the need to try different things to find the failure point. |
| 23:24.50 | asterboy | something is not the same |
| 23:24.55 | dlynes_ | asterboy: I changed two things |
| 23:24.58 | *** join/#asterisk bweiss (n=bweiss@72.54.41.2) |
| 23:24.59 | dlynes_ | asterboy: the hardware |
| 23:25.03 | dlynes_ | asterboy: and the kernel |
| 23:25.14 | dlynes_ | asterboy: the new machine isn't using 2.6 at all....only 2.4.29 |
| 23:25.37 | asterboy | I use to crack C64 games by just trying different things...I still do this today in my troubleshooting |
| 23:25.47 | dlynes_ | asterboy: same here |
| 23:26.03 | asterboy | 2.4 is better in that it's so much lighter |
| 23:26.26 | dlynes_ | asterboy: the only thing i did between the day before and that day when it stopped working was one thing: |
| 23:26.28 | asterboy | They are starting to bloat the kernel bad |
| 23:26.37 | dlynes_ | asterboy: I recompiled the kernel to add smbfs support |
| 23:27.01 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 23:27.07 | asterboy | shouldn't change anything, but then again, you never know. |
| 23:27.07 | dlynes_ | smbfs SHOULD NOT cause shit like that to stop working...especially when it's not even being used |
| 23:27.18 | Jaxxan | hey guys |
| 23:27.24 | dlynes_ | heya jaxxan |
| 23:27.25 | asterboy | I'd suspect the hardware change |
| 23:27.34 | dlynes_ | asterboy: there wasn't any hardware change |
| 23:27.38 | Jaxxan | anyone familiar with Glenayre Voicemail ? |
| 23:27.44 | asterboy | thought you said there was |
| 23:27.54 | dlynes_ | asterboy: I changed hardware to fix the problem |
| 23:28.00 | dlynes_ | asterboy: it wasn't the cause of the problem |
| 23:28.05 | asterboy | ah |
| 23:28.14 | dlynes_ | after I added smbfs support to the kernel |
| 23:28.20 | dlynes_ | two things screwed up |
| 23:28.23 | asterboy | anyway, start playing around |
| 23:28.29 | asterboy | I'd try mpg123 first |
| 23:28.29 | dlynes_ | the x100p driver stopped loading |
| 23:28.33 | Jaxxan | I wanna replace our old as hell Glenayre voicemail platform that costs an arm and a leg with a $4000 server and Asterisk handling voicemail |
| 23:28.38 | dlynes_ | wtf does mpg123 have to do with anything? |
| 23:28.39 | asterboy | oh x100p....gag |
| 23:28.45 | dlynes_ | I wasn't even using mp3 files |
| 23:28.48 | asterboy | run ztcfg -vvvvv |
| 23:28.54 | asterboy | and it will load |
| 23:29.03 | dlynes_ | asterboy: really? |
| 23:29.10 | dlynes_ | asterboy: what's the -vvvvv do? |
| 23:29.13 | asterboy | works for me on my 233 |
| 23:29.15 | asterboy | MHz |
| 23:29.31 | asterboy | won't load on boot, so I add ztcfg into rc.local |
| 23:29.36 | asterboy | then it starts |
| 23:29.45 | dlynes_ | asterboy: but the driver is getting loaded |
| 23:29.48 | asterboy | some kind of timing issue I think |
| 23:29.52 | dlynes_ | asterboy: it just fails initialization |
| 23:29.56 | asterboy | same |
| 23:30.02 | dlynes_ | asterboy: i.e. before it even gets to ztcfg |
| 23:30.07 | asterboy | yep |
| 23:30.27 | dlynes_ | ah |
| 23:30.30 | asterboy | your going to get old using the x100p |
| 23:30.37 | asterboy | I have a few grey pubics now. |
| 23:30.38 | dlynes_ | i've got another box that's using that crappy card |
| 23:30.43 | dlynes_ | I'll try that trick on it |
| 23:30.52 | *** join/#asterisk Junior_Payne (n=junior@CPE000f66364a01-CM00407b87bbbd.cpe.net.cable.rogers.com) |
| 23:31.06 | Junior_Payne | Hello everyone. |
| 23:31.10 | asterboy | just try the mpg123 to see if * can play a music file. |
| 23:31.13 | dlynes_ | well, i just ordered an a200d with 4 fxo ports yesterday |
| 23:31.25 | asterboy | yep, mine is on its way |
| 23:31.30 | dlynes_ | i've heard it's a hell of a lot better |
| 23:31.45 | dlynes_ | I've got isp status with them, so i'm getting decent pricing |
| 23:31.47 | asterboy | ya that's what I heard too. |
| 23:31.58 | asterboy | how much? |
| 23:32.01 | gandhijee | you get them w/ the echo canceller module? |
| 23:32.09 | dlynes_ | Not this time |
| 23:32.10 | Junior_Payne | I have a strange problem, when I put in a register => command in the iax.conf I can no longer make any outgoing calls. |
| 23:32.12 | dlynes_ | Probably next time though |
| 23:32.15 | asterboy | mine was $300 |
| 23:32.25 | dlynes_ | gandhijee: I just got mine for our own office |
| 23:32.27 | asterboy | no echo can is too much, and not necessary with 4 ports |
| 23:32.33 | asterboy | the CPU can handle that. |
| 23:32.47 | asterboy | err...well maybe not on my 233 |
| 23:32.59 | asterboy | but I'm upgrading to a P4 1.8 |
| 23:33.16 | asterboy | picked up 2 of them for $220 |
| 23:33.29 | asterboy | how much was your A200d? |
| 23:33.53 | gandhijee | i'm tryin to drop in asterisk for my parents to replace thier old mitel system |
| 23:34.05 | dlynes_ | 2 for 220? |
| 23:34.13 | drray | which mitel? |
| 23:34.19 | gandhijee | sx-20 |
| 23:34.24 | drray | sx-50 is what we are flusing |
| 23:34.28 | dlynes_ | asterboy: which is it? 1 for 300, or two for 200? |
| 23:34.33 | asterboy | yes, P4 1.8, 128Mb Ram no HD |
| 23:34.34 | gandhijee | isn't that the new one from them? |
| 23:34.43 | asterboy | 1 sangoma card for $300 |
| 23:34.48 | drray | no, it's 10 plus years old |
| 23:34.53 | gandhijee | O |
| 23:34.54 | asterboy | the 2 P4s for $220 |
| 23:34.55 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 23:35.02 | dlynes_ | asterboy: 4 port fxo sangoma? |
| 23:35.05 | asterboy | yes |
| 23:35.08 | gandhijee | yeah |
| 23:35.11 | gandhijee | the A200's |
| 23:35.13 | dlynes_ | asterboy: yeah...$290 i think it was for me |
| 23:35.20 | drray | do your parents use the 90v MWI? |
| 23:35.25 | gandhijee | they are supposta be really nice |
| 23:35.26 | asterboy | sounds good...sure be nice to get them lower |
| 23:35.30 | gandhijee | nah |
| 23:35.30 | asterboy | especially the EC |
| 23:35.37 | asterboy | they want $300 just for EC |
| 23:35.39 | dlynes_ | asterboy: you're in canada, right? |
| 23:35.44 | asterboy | eh? |
| 23:35.46 | gandhijee | but supposedly the Rhino Channel banks can support that now |
| 23:35.48 | dlynes_ | guess not :) |
| 23:35.51 | asterboy | yes |
| 23:35.55 | gandhijee | EC is worth the money, IMO |
| 23:35.56 | dlynes_ | smart ass :) |
| 23:36.00 | asterboy | RedNeck Albertan |
| 23:36.03 | drray | gandhijee - yeah, we are going to order one |
| 23:36.05 | drray | as a test |
| 23:36.14 | asterboy | EC is not worth it for 4 ports |
| 23:36.32 | dlynes_ | the ec is worth it if you go 8 ports or more, though |
| 23:36.45 | gandhijee | asterboy: depends on the line quality i think |
| 23:36.48 | asterboy | ya, it starts to get important |
| 23:36.57 | dlynes_ | gandhijee: yeah...if the line quality sucks ass |
| 23:37.02 | asterboy | and on the system your putting it in. |
| 23:37.06 | gandhijee | i still get echo w/ the aggresive quality on my test line |
| 23:37.11 | gandhijee | and my system is a beast |
| 23:37.12 | dlynes_ | gandhijee: but then, if the line quality's that bad, how much are you going to be able to actually improve it? |
| 23:37.44 | asterboy | A P4 3GHz 800FSB, 1G DDR 400 should not need EC for 8 ports |
| 23:37.54 | asterboy | but I'll find out and let you know |
| 23:38.22 | gandhijee | asterboy: mine is a Pentium D 3.0 800FSB and 2 GIG DDR400... |
| 23:38.30 | asterboy | how many ports? |
| 23:38.37 | gandhijee | w/ the digium Wildcard 400 for FXO |
| 23:38.51 | asterboy | and how is the echo? |
| 23:38.54 | gandhijee | and a Sangoma AFT102 that runs to a rhino channel bank |
| 23:39.09 | gandhijee | on the internal motel line is come and go |
| 23:39.34 | gandhijee | i'm hopin to move the the real lines tonight, depends on if i get these apps for the phone coded up |
| 23:39.46 | asterboy | is the motel line on the TDM? |
| 23:40.09 | gandhijee | yeah, it comes of the PBX right now. |
| 23:40.20 | gandhijee | so we can test the basic system to make sure calls and stuff work |
| 23:40.22 | asterboy | then you better read this: |
| 23:40.23 | asterboy | http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 23:40.42 | asterboy | bet if that was a sangoma, you would'nt have that problem |
| 23:40.59 | asterboy | SATA drives btw are bad in combo with TDM |
| 23:41.05 | gandhijee | fuck |
| 23:41.07 | gandhijee | really? |
| 23:41.13 | asterboy | read that link |
| 23:41.31 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-73.apple.com) |
| 23:41.46 | gandhijee | zttest seems to be fine.... gettin like >=99.98% |
| 23:41.55 | Junior_Payne | Any one have any ideas why I'm getting this problem? |
| 23:41.59 | asterboy | and this: http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation |
| 23:42.12 | asterboy | zttest over how many minutes? |
| 23:42.35 | gandhijee | how long should i let it run? |
| 23:43.05 | asterboy | another one: http://www.voip-info.org/wiki/index.php?page=Causes+of+Echo |
| 23:43.17 | asterboy | 5 min at least and with traffic |
| 23:43.23 | drray | gandhijee - until it drops |
| 23:43.40 | asterboy | I get good stats at the beginning always |
| 23:43.40 | gandhijee | what kinda traffic? phone traffic |
| 23:43.46 | asterboy | everything |
| 23:43.56 | gandhijee | or just peg the machine with hd accessing |
| 23:43.57 | asterboy | throw what you can at it |
| 23:43.58 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 23:44.03 | diclophis | howdy all |
| 23:44.16 | diclophis | so what is the best way to detect a fax.. while still accepting DTMF input |
| 23:44.55 | asterboy | oh and ztmonitor is a great help |
| 23:44.56 | asterboy | http://www.voip-info.org/wiki/index.php?page=Asterisk+X100P+Echotraining |
| 23:46.23 | Druken | hehehe |
| 23:46.29 | Druken | kids are funny |
| 23:46.44 | drray | kids are funny until they happen to you |
| 23:46.46 | drray | er |
| 23:46.55 | gandhijee | lol @ drunken |
| 23:47.17 | asterboy | pwd |
| 23:47.19 | Druken | drray: my son just noticed the 18foot pool that i am filling in the backyard |
| 23:47.35 | drray | that he never wanted to swim in before then |
| 23:47.36 | Druken | he came running into my room all excited over it.. hehe |
| 23:47.52 | asterboy | water I don't mind...it's the shit |
| 23:48.07 | Druken | asterboy: that's the toilet, not a pool.... |
| 23:48.21 | asterboy | How is it a dog takes 2 weeks to train how to shit but a human being takes fucking years??? |
| 23:48.38 | drray | you can't rub a kids nose in it without cps getting involved |
| 23:48.45 | asterboy | lol |
| 23:49.12 | Druken | i've yet to see a dog flush the toilet |
| 23:49.28 | drray | we had a cat that learned how to |
| 23:49.34 | drray | not use it, just flush it |
| 23:49.37 | drray | and the doorbell |
| 23:49.38 | asterboy | actually there was a video clip on American's Funniest Home Videos |
| 23:49.50 | Druken | my cats used to use the damn water cooler... |
| 23:49.53 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.124) |
| 23:50.09 | drray | cats are assholes |
| 23:50.13 | Druken | agreed |
| 23:50.28 | Druken | cats are like women... |
| 23:50.52 | Druken | they'll make nice to you, if it's convient to them... |
| 23:51.09 | drray | or inconvienent to you |
| 23:51.24 | asterboy | I wouldn't mind cleaning up shit if it was in one trained spot...but when you spell something ugly on your kids fingers, you left wondering what has been touched. |
| 23:51.39 | *** join/#asterisk Mavvie (n=edwin@203.222.131.252) |
| 23:51.40 | asterboy | /spell/smell |
| 23:51.55 | gandhijee | rofl |
| 23:52.05 | Druken | can't say i've ever had that experince... |
| 23:52.41 | asterboy | how can you tell...I've had enough, I want them trained ASAP |
| 23:54.03 | asterboy | more on EC, make sure you read page 38 of theBible...it has the fix that worked for me. |
| 23:54.17 | gandhijee | theBible?? |
| 23:54.20 | asterboy | moving to MARK2 EC |
| 23:54.30 | asterboy | the gospel according to MARK |
| 23:54.43 | gandhijee | asteriskTFOT book? |
| 23:54.53 | asterboy | yes, that is the Bible |
| 23:55.16 | drray | i turned my sound down on my channel bank |
| 23:55.20 | drray | which helped my echo |
| 23:55.49 | asterboy | ya, use ztmonitor for that too |
| 23:56.11 | asterboy | I could only get to -6.3 before DTMF was ignored |
| 23:56.17 | drray | heh |
| 23:56.18 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 23:58.53 | Druken | can someone like, fastforward the time for me? make it next thursday ? |
