| 00:00.15 | Ahrimanes | xp_prg: $AGI->stream_file('queue-on', 0); <- that's how i use stream_file from perl |
| 00:00.32 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:00.32 | xp_prg | rpm can I paste the output to you? |
| 00:00.46 | justinu|laptop | ~pb |
| 00:00.47 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:02.27 | xp_prg | http://pastebin.com/723714 |
| 00:02.35 | xp_prg | tell me what I am doing wrong please |
| 00:03.12 | Ahrimanes | xp_prg: line 8 is broken |
| 00:03.18 | rpm | it looks like its never running your script. |
| 00:04.11 | Ahrimanes | xp_prg: do perl -c script.agi |
| 00:04.43 | xp_prg | Can't locate Asterisk/AGI.pm |
| 00:04.53 | Ahrimanes | there's ya trouble |
| 00:04.59 | xp_prg | ok awsome thanks! |
| 00:05.21 | Ahrimanes | also my ($text=@ARGV; <- unmatched ( |
| 00:05.46 | xp_prg | anyone know the gentoo package for perl Asterisk::AGI? |
| 00:06.14 | Ahrimanes | no idea if there is one |
| 00:06.23 | rpm | perl -e shell -MCPAN 'install Asterisk::AGI' |
| 00:06.35 | Ahrimanes | uhm |
| 00:06.42 | justinu|laptop | or jut cpan install for the lazy |
| 00:06.47 | Ahrimanes | perl -MCPAN -e 'install Asterisk::AGI' |
| 00:07.05 | Ahrimanes | if it's gotten on to cpan now |
| 00:08.29 | Ahrimanes | nope |
| 00:08.34 | xp_prg | I don't think it is on there, how did you install it Ahrimanes? |
| 00:09.22 | Ahrimanes | xp_prg: wget http://asterisk.gnuinter.net/files/asterisk-perl-0.08.tar.gz && tar xzf asterisk-perl-0.08.tar.gz && cd asterisk-perl && perl Makefile.PL i guess |
| 00:10.05 | Ahrimanes | POE::Component::Client::Asterisk::Manager <- really nice for manager events |
| 00:10.09 | *** join/#asterisk ethzer0 (n=ethzer0@cheetah-tiv-ve3000-124-0179.bmts.com) |
| 00:10.23 | xp_prg | Ahrimanes I got that yesterday but don't understand it AAAAAGGGGGHHHHH |
| 00:10.33 | xp_prg | I can telnet and put in my commands |
| 00:10.36 | Ahrimanes | xp_prg: the .tar.gz file ? |
| 00:10.39 | xp_prg | do you have any exampled? |
| 00:10.45 | xp_prg | no the POE Manager |
| 00:10.47 | xp_prg | you just mentioned |
| 00:10.51 | *** part/#asterisk ethzer0 (n=ethzer0@cheetah-tiv-ve3000-124-0179.bmts.com) |
| 00:11.01 | Ahrimanes | xp_prg: ah, POE Manager is not all that easy to get started with |
| 00:11.19 | xp_prg | Ahrimanes can you tell me what to do after perl Makefile.pl? |
| 00:11.25 | Ahrimanes | xp_prg: make |
| 00:11.28 | Ahrimanes | xp_prg: then make install |
| 00:11.46 | xp_prg | ok cool ran it |
| 00:11.58 | xp_prg | Ahrimanes help me to understand POE Manager and I promise to worship you |
| 00:12.23 | Ahrimanes | xp_prg: after make install your Agi should work.. but run perl -c on it again to see if there are syntax errors |
| 00:12.38 | xp_prg | yes no errors, trying the extension now |
| 00:12.58 | xp_prg | it worked!! |
| 00:13.08 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 00:13.08 | Ahrimanes | there ya go |
| 00:13.10 | justinu|laptop | imagine that |
| 00:13.11 | justinu|laptop | :) |
| 00:13.40 | Ahrimanes | xp_prg: i have a sample program that uses POE Manager to look for sip peer registrations and log them to database and such... |
| 00:13.42 | xp_prg | Ahrimanes can you show me an example with POE Manager where I make an originate command via the manager api? |
| 00:14.05 | justinu|laptop | i did it all in java |
| 00:14.06 | justinu|laptop | :P |
| 00:14.15 | xp_prg | why do you like java? |
| 00:14.27 | Ahrimanes | xp_prg: to originate commands i would recommend Asterisk::Manager from the Asterisk-perl dist you just installed, i can show you a small sample for that |
| 00:14.42 | justinu|laptop | why not? |
| 00:14.51 | xp_prg | did that install Asterisk::Manager when I did the make? |
| 00:15.03 | xp_prg | g-cpan shows Asterisk::Manager, I could install it from there |
| 00:15.10 | xp_prg | want me to do that? |
| 00:15.44 | Ahrimanes | xp_prg: do perldoc Asterisk::Manager to see if it's installed now |
| 00:15.51 | *** join/#asterisk juice (n=juice@mo-67-77-179-198.dyn.sprint-hsd.net) |
| 00:16.00 | xp_prg | it is installed |
| 00:16.09 | Ahrimanes | http://pastebin.com/723731 |
| 00:16.09 | supjigatr | Any ideas: Loading WAN drivers: wanpipe WARNING: /etc/modprobe.conf line 3: ignoring bad line starting with 'post-install' |
| 00:16.18 | FuriousGeorge | han anyone ever interfaced a telephone handset with a sound card? im starting to find adapters that do it, though most of them are designed for recording, not playback |
| 00:16.29 | *** join/#asterisk Vorondil (n=vorondil@209-42-156-161.wispnet.net) |
| 00:16.31 | justinu|laptop | supjigatr: missing comment? |
| 00:16.40 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 00:17.01 | FuriousGeorge | i got a loudspeaker that interfaces via rj11 to a station on an existing pbx, and i wanted to see if i could interface that somehow with chan/dsp |
| 00:17.12 | Ahrimanes | xp_prg: there's a program to use Asterisk::Manager to do sip show peer $peer and fish out useragent and ip address of the peer |
| 00:17.14 | *** join/#asterisk PBXtech (i=nik@179.sub-70-218-100.myvzw.com) |
| 00:17.14 | supjigatr | ??? What is it suppose to look like. |
| 00:17.20 | FuriousGeorge | rather than buy a whole new amp and speaker |
| 00:17.24 | Blackthorn | Hi all, I just moved my asterisk system from a standard case to a rack mount case and now I can't get the pri's to come up. whats the utility for checking the pri's? zztools or somehting like that? |
| 00:17.26 | *** join/#asterisk andymul (n=iCallAnd@cpe-69-203-74-36.nyc.res.rr.com) |
| 00:17.28 | justinu|laptop | not sure, just came off the top of my head |
| 00:17.36 | justinu|laptop | zttool |
| 00:17.46 | andymul | Does anyone know why "show channels" chops off the last few characters of the channel name? |
| 00:17.55 | xp_prg | Can't locate object method "new" via package "Asterisk::Manager" |
| 00:18.04 | Ahrimanes | xp_prg: hm |
| 00:18.32 | supjigatr | ztcfg isn't loading |
| 00:18.33 | Ahrimanes | strange |
| 00:18.37 | xp_prg | oh I didn't have the use command at the top |
| 00:18.44 | Ahrimanes | xp_prg: ah |
| 00:18.52 | Blackthorn | thanks. they show in red alarm. thing it's a configureation problem or perhaps my provider killed the pri's since they were down for 5 minutes? |
| 00:18.56 | Ahrimanes | xp_prg: that was missing from my sample, sorry :D |
| 00:19.02 | PBXtech | i have a stupid 7940 with sip 2.3 it wont flash crap. not even the 3.0 is there magic to this? |
| 00:19.06 | Blackthorn | thing = think |
| 00:19.06 | xp_prg | no output from the script, not even errors |
| 00:19.16 | Ahrimanes | xp_prg: remember to change manager user and secret settings :) |
| 00:19.18 | justinu|laptop | Blackthorn: 5 minutes of downtime usually isn't enough for a telco to turn them off |
| 00:19.21 | xp_prg | Ahrimanes I did |
| 00:19.25 | Ahrimanes | xp_prg: hm.. lemme rewrite real quick |
| 00:19.33 | *** part/#asterisk Vorondil (n=vorondil@209-42-156-161.wispnet.net) |
| 00:20.50 | Ahrimanes | xp_prg: http://pastebin.com/723739 <- change manager user/secret etc.. put in a file and do perl file.pl <peer> |
| 00:21.25 | xp_prg | what is a good value for peer, my extension? |
| 00:21.45 | supjigatr | I think I am having udev problems. |
| 00:21.51 | Ahrimanes | xp_prg: http://pastebin.com/723744 <- smal change |
| 00:21.56 | Ahrimanes | xp_prg: yeah any sip peer |
| 00:22.00 | supjigatr | Anyone have sangoma card a104d? |
| 00:22.10 | orlok | supjigatr: which sort is that? |
| 00:22.42 | xp_prg | Ahrimanes I don't know what a sip peer is though |
| 00:22.56 | Ahrimanes | xp_prg: a sip extension from sip.conf |
| 00:23.48 | xp_prg | oh cool it is outputing something! |
| 00:23.58 | xp_prg | registered;262;;;1147911825 |
| 00:24.16 | Ahrimanes | oh so your extension is 262 ? |
| 00:24.22 | xp_prg | ya |
| 00:24.24 | Ahrimanes | ok |
| 00:24.29 | Ahrimanes | so it's basically working |
| 00:24.33 | supjigatr | orlok: 4 port T1 card. |
| 00:24.35 | *** join/#asterisk miztic (n=gerard@rarcoa.com) |
| 00:24.36 | xp_prg | thanks! |
| 00:24.56 | Ahrimanes | xp_prg: you should be able to figure out the basics from that script? |
| 00:25.09 | xp_prg | now if only I could tell it to dial an extension, then when it connects, tell to connect that call to antoher extension |
| 00:25.14 | xp_prg | can you help me to do that last little part? |
| 00:25.26 | Ahrimanes | ah originate |
| 00:25.35 | xp_prg | right |
| 00:25.47 | xp_prg | the context is workshop and the extension is 1 |
| 00:26.54 | *** join/#asterisk xanderp (n=pphillip@12.222.95.177) |
| 00:27.14 | Ahrimanes | hm i thought i had a script somewhere for tha |
| 00:27.14 | Ahrimanes | t |
| 00:27.38 | xanderp | can a cisco ubr925 or ubr924 cable modem (they have phone ports on the back) be configured to drive ananlog phones for asterisk? |
| 00:30.22 | Ahrimanes | xp_prg: http://www.azxws.com/asterisk/ |
| 00:30.55 | xp_prg | does the use Asterisk::Manager use poe? |
| 00:31.09 | Qwell | power over ethernet? |
| 00:31.16 | Ahrimanes | xp_prg: not poe |
| 00:31.20 | xp_prg | poe is a perl library |
| 00:31.24 | Ahrimanes | xp_prg: well i'll check |
| 00:31.30 | file | Qwelllllllllllllllllllllllllllll |
| 00:31.31 | xp_prg | Ahrimanes I thought you were going to show me the poe stuff |
| 00:31.40 | Qwell | file: you! |
| 00:31.47 | file | hihihi |
| 00:31.59 | Qwell | wtf is Hellmanns Mayo? |
| 00:32.07 | file | some sort of Mayo? |
| 00:32.25 | Qwell | yes, but...why does it have the Best Food slogan/logo? |
| 00:32.30 | Ahrimanes | xp_prg: the poe stuff is mostly for receiving events from the manager api, Asterisk::Manager is for running commands |
| 00:32.38 | file | Qwell: unknown |
| 00:32.40 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 00:32.51 | xanderp | what's the linux version of hyperterminal? I need to console into a router, and don't know how to access the com port. |
| 00:32.54 | xp_prg | oh ok |
| 00:32.59 | Qwell | anybody, feel free to answer the above question |
| 00:33.05 | Qwell | xanderp: minicom |
| 00:33.08 | xanderp | thanks |
| 00:33.20 | xp_prg | so you can't send commands with the poe? |
| 00:33.24 | Ahrimanes | xp_prg: hm taci uses Net::Telnet ... not pretty |
| 00:33.34 | xp_prg | ya I don't want to use that garbage |
| 00:33.40 | justinu|laptop | hellman's is what it was called on the east coast |
| 00:33.56 | Ahrimanes | xp_prg: maybe, i havent gotten it working, but then again, as i have Asterisk::Manager i didnt try too hard |
| 00:34.18 | Ahrimanes | xp_prg: well, all manager interaction is done with Telnet on some level, Asterisk::Manager just hides it :) |
| 00:34.18 | xp_prg | well show me an example with Asterisk::Manager doing an originate pretty please |
| 00:34.23 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
| 00:34.31 | Ahrimanes | xp_prg: hm, guess i could whip that up |
| 00:35.27 | xp_prg | thanks! |
| 00:37.01 | *** join/#asterisk SlackUser_ (n=FullT@201008199065.user.veloxzone.com.br) |
| 00:37.46 | *** join/#asterisk cp5 (n=cp5@rtr3-t1.fonality.com) |
| 00:38.06 | cp5 | hey guys, is this possible -- registering with multiple SIP providers but with different source ports (port=[portno]) |
| 00:38.11 | xanderp | shouldn't the first com port on a machine be ttyS0 ? |
| 00:38.21 | Qwell | should |
| 00:38.26 | xanderp | hmm |
| 00:38.38 | xanderp | minicom is not playing nice. |
| 00:38.51 | Qwell | got the right speed settings? |
| 00:39.03 | Qwell | 9600N1 is common |
| 00:39.10 | xanderp | yep |
| 00:39.14 | *** part/#asterisk andymul (n=iCallAnd@cpe-69-203-74-36.nyc.res.rr.com) |
| 00:39.19 | xanderp | that's the first thing i checked... odd |
| 00:39.22 | Qwell | right kind of cable? |
| 00:39.31 | xanderp | yup, blue cisco console cable |
| 00:39.59 | orlok | xanderp: i find sometimes you need to quit out and restart minicom after setting the serial speed and saving the config |
| 00:40.06 | xanderp | will try that thanks |
| 00:41.08 | Ahrimanes | xp_prg: done |
| 00:41.13 | xp_prg | cool |
| 00:41.54 | xp_prg | can I please see it? |
| 00:42.30 | Ahrimanes | xp_prg: yep.. just need to clean out extensions and passwords |
| 00:42.37 | xp_prg | oh ok |
| 00:43.36 | Ahrimanes | http://pastebin.com/723777 <- REPLACE = enter your own values |
| 00:44.15 | Ahrimanes | it spits out an error, but that can be ignored safely |
| 00:45.22 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 00:45.40 | xp_prg | I just got it to work wahhoooo!!! |
| 00:45.53 | xp_prg | thanks Ahrimanes, can I ask one more question, then my proejct will be finished? |
| 00:45.54 | Ahrimanes | guess there should be a lot of error checking on running the command, but it's 2:45 am |
| 00:46.01 | Ahrimanes | xp_prg: hehe sure |
| 00:46.44 | xp_prg | in my originate command, once it connects to the SIP/262 I want it to then immediately connect to extension 1 so it will play the pre-recorded message, it isn't doing that currently, what must I add to the originate command to do this? |
| 00:46.58 | Ahrimanes | that's actually reall small number of lines to do originate :D |
| 00:47.37 | Ahrimanes | xp_prg: hm, then you need to configure an extension in extensions.conf that plays the recorded message and then call that with originate |
| 00:47.44 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 00:48.02 | yxa | how do i Dial() using iax to ext1234 @ a server say voice.abc.com ? |
| 00:48.42 | xp_prg | Ahrimanes but how do I call it and the other # at the same time so they connect to each other? |
| 00:49.11 | Ahrimanes | xp_prg: you make the extension in extensions.conf first playback the recorded message, then call the other # |
| 00:50.06 | dlynes_home | yxa: Dial(protocol/servername/exten) |
| 00:51.49 | xanderp | odd, must be a problem with this laptop's com port/linux. i plugged the console cable into my one remaining windows machine and hyperterminal comes right up with a router console.... odd. |
| 00:53.01 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) [NETSPLIT VICTIM] |
| 00:53.10 | *** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 00:54.11 | xp_prg | it works man |
| 00:54.29 | Ahrimanes | xp_prg: my paypal account is michael.aronsen@gmail.com ;) |
| 00:54.38 | bzbw | hi, does everyone know that if I can do a "one to many" intercom? |
| 00:54.49 | yxa | dlynes_home under sip.conf, there is context=local for guest incoming calls. is there an equivalent for iax? |
| 00:55.06 | dlynes_home | yxa: yes...same directive |
| 00:55.06 | xp_prg | I will note this |
| 00:55.26 | *** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net) |
| 00:55.28 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:55.39 | xp_prg | I am getting this weird error when I run it though: Use of uninitialized value in string eq at /usr/lib/perl5/site_perl/5.8.7/Asterisk/Manager.pm line 372, <GEN0> line 18. |
| 00:56.08 | *** join/#asterisk s0lid (i=s0lid@gr-153-200.eglobalreach.net) |
| 00:56.17 | op3r | does anyone know how many users can an asterisk install on a dual xeon with 2gig of memory? |
| 00:56.44 | Ahrimanes | xp_prg: yep, what i said earlier.. you can ignore it or remove -w from #!/usr/bin/perl -w line |
| 00:56.50 | justinu|laptop | more than 10, less than 10,000? |
| 00:56.58 | s0lid | yes i know |
| 00:57.17 | s0lid | op3r, give us info about you rboc |
| 00:57.19 | s0lid | box |
| 00:57.44 | op3r | because we kept on 100% cpu usage if we have like 24 users using vicidial and the asterisk box bogs down |
| 00:58.00 | Ahrimanes | hm vicidial uses meetme afaik |
| 00:58.21 | s0lid | does meetme uses lots of processor power? |
| 00:58.39 | Ahrimanes | it can |
| 00:59.04 | Blackthorn | zttool reports my pri's in red alarm think it's pretty good chance the remote end has the lines down after i had my box turned off? |
| 00:59.06 | s0lid | hmmm... how can you lower it? |
| 00:59.09 | xp_prg | Ahrimanes when I took the -w off it still shows |
| 00:59.29 | Ahrimanes | xp_prg: hm ok |
| 01:01.12 | dlynes_home | Blackthorn: if i remember correctly, red alarm means nothing's plugged in; if remote end is having issues, you usually have a yellow alarm |
| 01:03.10 | cp5 | anyone know how to register with multiple SIP providers but with different source ports (port=[portno])? can i somehow assign a register to a peer so that i have a unique socket per register? it seems right now they might all be using the same socket |
| 01:04.24 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 01:04.27 | kernel20 | hi |
| 01:04.42 | kernel20 | aside from asterisk what should I install in order to have MeetMe? |
| 01:04.52 | file | zaptel |
| 01:05.10 | Qwell | file: don't..trust me |
| 01:05.15 | Qwell | rather |
| 01:05.18 | Qwell | don't don't trust me |
| 01:05.25 | Qwell | but just don't.. |
| 01:05.25 | file | ok! |
| 01:05.39 | kernel20 | ? |
| 01:06.04 | Qwell | well |
| 01:06.08 | Qwell | don't trust me either |
| 01:06.25 | dlynes_home | kernel20: a timer, be it some digium hardware with the appropriate zaptel driver, or ztdummy |
| 01:06.27 | kernel20 | i've been here days ago, and someone(i dont remember who) told me that I need to have MeetMe to have voice conferencing |
| 01:06.50 | kernel20 | dlynes_home: i only use x-lite |
| 01:06.54 | dlynes_home | kernel20: meetme is a dialplan application that comes with asterisk |
| 01:06.56 | kernel20 | only softphones |
| 01:07.17 | dlynes_home | kernel20: then you need to load ztdummy (which in turn needs the rtc driver and the ccitt_crc driver |
| 01:07.29 | kernel20 | pbx.c:1697 pbx_extension_helper: No application 'MeetMe' for extension (tutorial, 8000, 1) |
| 01:07.47 | dlynes_home | kernel20: you don't have the module loaded that's needed for it |
| 01:07.58 | kernel20 | how can i load the module? |
| 01:08.12 | kernel20 | like ztdummy |
| 01:08.13 | dlynes_home | kernel20: i don't know which one it is offhand...one second |
| 01:08.13 | kernel20 | ? |
| 01:08.20 | dlynes_home | ztdummy is a kernel driver |
| 01:08.28 | dlynes_home | it comes with zaptel-1.2.5 |
| 01:08.51 | kernel20 | so i need to install zaptel? |
| 01:09.06 | dlynes_home | kernel20: correct |
| 01:09.34 | supjigatr | Anyone using a sangoma a104d? |
| 01:09.37 | kernel20 | after traditional make and make install, my conferencing would work? |
| 01:10.29 | dlynes_home | kernel20: you need to do a load => app_meetme.so in your modules.conf file, too |
| 01:10.47 | dlynes_home | kernel20: and of course do a reboot after your make install on the zaptel stuff |
| 01:11.01 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 01:11.09 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:12.25 | dlynes_home | kernel20: http://www.voip-info.org/wiki/view/Asterisk+timer |
| 01:12.44 | dlynes_home | anyways...gotta run |
| 01:12.52 | kernel20 | thanks dlynes_home |
| 01:13.00 | dlynes_home | tennis court and cute chinese girls in nice short white skirts are calling :) |
| 01:13.21 | kernel20 | :D |
| 01:13.45 | dlynes_home | summer is definitely the best season of the year :) |
| 01:14.31 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 01:16.11 | Cybertoy | hi ... quick q ... probably an easy one ... I use Dial(....,,t) to be able to transfer callers coming in with # ... I can only transfer them to extensions that are in the default context though. Any way to transfer them to extensions in other contexts?? |
| 01:17.46 | Flauto | how can i start festival at boot up? |
| 01:17.57 | Flauto | now, i have to start the server on kernel |
| 01:18.02 | Cybertoy | Flauto, put the command into /etc/rc.local |
| 01:18.04 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:19.50 | Flauto | cybertoy, what should i put in that file? |
| 01:19.56 | Flauto | if you can tell me |
| 01:20.21 | Flauto | and festival sounds pretty bad too |
| 01:21.20 | Flauto | just put festival_server? |
| 01:21.26 | Cybertoy | Flauto, yes |
| 01:21.30 | Cybertoy | if that's the command to start it |
| 01:21.34 | Cybertoy | better would be the full path |
| 01:21.37 | Flauto | okay |
| 01:21.41 | Cybertoy | like /usr/bin/festival_server |
| 01:21.53 | kernel20 | i have install zapata |
| 01:21.54 | Cybertoy | and make sure you add a & if it doesn't fork itself |
| 01:21.59 | kernel20 | but still problems |
| 01:22.04 | kernel20 | i cant do conferencing |
| 01:22.53 | kernel20 | pbx.c:1697 pbx_extension_helper: No application 'MeetMe' for extension (tutorial, 8000, 1) |
| 01:22.56 | kernel20 | any ideas? |
| 01:23.10 | file | did you recompile Asterisk after installing zaptel? |
| 01:23.12 | km- | is the meetme module loaded? |
| 01:23.21 | km- | ah yeah, you need to recompile if you didnt have zaptel previously |
| 01:23.49 | kernel20 | so at asterisk folder i will do make again? |
| 01:24.33 | Flauto | where to add & |
| 01:24.39 | kernel20 | ? |
| 01:25.06 | kernel20 | load => app_meetme.so |
| 01:25.08 | kernel20 | ? |
| 01:26.53 | *** join/#asterisk jaybuffet (n=jperron@43-159.127-70.tampabay.res.rr.com) |
| 01:27.58 | jaybuffet | i need help.. i am using voiceone web front end and i dont think it is working correctly.. i add extensions yet the extensions_table remains empty, any body familiar with voiceone or can recomend a good mature web front end |
| 01:32.15 | kernel20 | <PROTECTED> |
| 01:32.17 | kernel20 | any ideas? |
| 01:32.27 | kernel20 | my conf number is 100 |
| 01:32.40 | kernel20 | when i dial 100 it says invalid conference number please try again |
| 01:32.44 | kernel20 | any ideas? |
| 01:33.01 | enkrypted | From earlier, my * is now working properly. |
| 01:33.12 | kernel20 | ? |
| 01:33.14 | op3r | kernel20: whats does the cli shows u? |
| 01:33.18 | kernel20 | <PROTECTED> |
| 01:34.13 | kernel20 | http://pastebin.com/723846 |
| 01:34.16 | kernel20 | for more details |
| 01:34.50 | file | do you have a zaptel timing device, with the kernel module loaded? |
| 01:34.55 | file | like ztdummy for example |
| 01:35.29 | kernel20 | how file? |
| 01:35.37 | kernel20 | am just a newbie of this |
| 01:35.41 | Qwell | file: I'm telling you... |
| 01:36.16 | file | kernel20: this is well documented online, I suggest you look there or enlist the help of someone here to help you more closely |
| 01:36.41 | kernel20 | dlynes_home told me that i need to install zaptel |
| 01:36.46 | kernel20 | i installed it |
| 01:36.56 | kernel20 | and have modules loaded at modules.conf |
| 01:37.10 | file | yes, but meetme requires a zaptel timing device to exist in order to be used |
| 01:37.16 | file | this comes from hardware, or using the ztdummy kernel module |
| 01:38.13 | kernel20 | so if i dont have the zaptel timing device i cant do voice conferencing? |
| 01:38.36 | Qwell | kernel20: yes |
| 01:38.42 | file | correct, but ztdummy is for those who don't have hardware for it... if you have a 2.6 kernel, chances are ztdummy will work |
| 01:39.00 | kernel20 | i have ztdummy |
| 01:39.02 | kernel20 | no |
| 01:39.08 | kernel20 | i am on kernel2.6 |
| 01:39.13 | file | did you load ztdummy? |
| 01:39.16 | kernel20 | modprobe ztdummy? |
| 01:39.16 | bzbw | hi, anyone knows if I can run an application against a list of predefined extensions from an variable? |
| 01:39.28 | supjigatr | Anyone in chan yet that uses sangoma a104d? |
| 01:39.44 | bzbw | say, I have Ext_List=601&602&603 |
| 01:39.52 | enkrypted | kernel20: lsmod |grep ztdummy |
| 01:40.17 | kernel20 | k wait |
| 01:40.35 | kernel20 | nothing found |
| 01:40.41 | bzbw | then use Page(${Exten}) where ${Exten} is extracted from Ext_List? |
| 01:40.43 | kernel20 | how can i install ztdummy module? |
| 01:40.48 | Blackthorn | I've talked to the telco they sware up and down the "d" channels are down on my * box. but * zztool shows pri is down "red". Have any suggestions or anyting i shoudl check? |
| 01:40.53 | enkrypted | kernel: modprobe ztdummy |
| 01:41.07 | Qwell | file: see what I'm saying, already? |
| 01:41.42 | Lino` | whoah |
| 01:41.44 | Lino` | what the fuck |
| 01:41.49 | kernel20 | Notice: Configuration file is /etc/zaptel.conf |
| 01:41.50 | kernel20 | line 0: Unable to open master device '/dev/zap/ctl' |
| 01:41.50 | kernel20 | 1 error(s) detected |
| 01:41.50 | kernel20 | FATAL: Error running install command for ztdummy |
| 01:41.53 | kernel20 | ? |
| 01:41.54 | bzbw | the reason Iam doing this is that I normally use Variable to define the list of contents so that if there is any change, I just change the variable contents, so need to touch the actual operation |
| 01:42.02 | Qwell | ~pb |
| 01:42.03 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 01:42.22 | kernel20 | sorry |
| 01:42.54 | enkrypted | kernel20: google.com |
| 01:43.50 | Lino` | i'm pretty ashamed. just got a mail... assignment for a piece of service... done with old programmes on my good ol' OS/2 box... some days ago i thought about throwing it away, now it'll earn a five figure sum... |
| 01:43.57 | Lino` | with a piece of junk |
| 01:44.42 | Lino` | i'm feeling so guilty for just thinking about throwing that piece of shit away |
| 01:44.45 | Lino` | ;D |
| 01:45.13 | enkrypted | kernel20: http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 01:45.51 | *** join/#asterisk camelon (n=chiardon@206.106.254.228) |
| 01:46.27 | camelon | Night |
| 01:48.53 | camelon | wich could be the origin of so frecuent extensions down that force to do frecuents ra to put the extensions up? |
| 01:49.56 | Blackthorn | well about the 50'th reset it started working :\ anyway. i'm going home. cya |
| 01:50.29 | camelon | all the extensions are zap |
| 01:50.39 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 01:51.01 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 01:55.36 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:55.37 | *** part/#asterisk gursikh (n=guriskh1@dsl254-123-245.nyc1.dsl.speakeasy.net) |
| 01:56.17 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:00.10 | camelon | somehing bad in this debug:http://pastebin.ca/56754 |
| 02:00.30 | *** join/#asterisk razu (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
| 02:02.09 | jaybuffet | any recommendations on a web interface for administration |
| 02:04.18 | Ahrimanes | jaybuffet: freepbx seems popular |
| 02:05.05 | jaybuffet | Ahrimanes: i am actually looking at that one right now |
| 02:05.21 | jaybuffet | i wonder if it will be good for a business scenario |
| 02:07.13 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 02:07.35 | Ahrimanes | jaybuffet: depends... i guess its ok or a company that have the resources to assign staff to managing the pbx |
| 02:07.53 | *** join/#asterisk mog_home (n=mogorman@64.241.203.173) |
| 02:08.48 | Ahrimanes | ok for a.. |
| 02:11.13 | jaybuffet | Ahrimanes: once configured, asterisk shouldnt require too much maintenance, right? |
| 02:12.48 | Ahrimanes | jaybuffet: well, if you have a static number of extensions and dont use too much fancy stuff it should "just work" [tm] |
| 02:16.16 | yxa | why is chan_sccp2 not in cvs tree? |
| 02:16.17 | SkramX | echo "stream file /tmp/$agi_uniqueid #" |
| 02:16.25 | Faithful | anyone got gnokii working as extensions? |
| 02:16.33 | SkramX | how do i, in an agi script, make something stream like it does when using "background"? |
| 02:18.27 | SwK | exec background |
| 02:19.01 | SwK | you can exec any dialplan application from an AGI and it will return to the AGI with and answer |
| 02:20.12 | *** join/#asterisk juice (n=juice@mo-71-50-26-181.dhcp.sprint-hsd.net) |
| 02:23.56 | supjigatr | Would "Failed to configure EC device" cause no audio on a sangoma card? |
| 02:30.40 | *** join/#asterisk junglicious (n=junglici@user-24-214-39-200.knology.net) |
| 02:31.45 | *** join/#asterisk loud (n=ariel@cypher.punk.net) |
| 02:32.13 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 02:35.31 | kernel20 | thank u file |
| 02:35.38 | kernel20 | i was able to do conferencing now |
| 02:35.45 | kernel20 | using ztdummy |
| 02:35.46 | kernel20 | thanks |
| 02:39.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 02:42.13 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 02:42.36 | *** join/#asterisk PBXtech (i=nik@179.sub-70-218-100.myvzw.com) |
| 02:57.50 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-132-61.dsl.chcgil.sbcglobal.net) |
| 02:58.34 | Flauto | how can i find out forecast mp3 file in my local area |
| 03:03.52 | *** join/#asterisk papa_e (i=papa_e@ip68-4-40-21.pv.oc.cox.net) |
| 03:04.18 | papa_e | sphinx work well for voice recognition? we only need to identify numbers |
| 03:04.30 | papa_e | ie, "press or say 1 for sales" |
| 03:10.26 | supjigatr | Anyone see a faxback for asterisk? |
| 03:11.13 | *** join/#asterisk elvisthedj (n=kris@host-69-145-70-130.bln-mt.client.bresnan.net) |
| 03:11.58 | elvisthedj | anybody here using mythphone? having trouble with garbled audio and google isn't helping me much |
| 03:13.02 | papa_e | supjigatr, could probably hack it together really quick with agi |
| 03:13.17 | papa_e | but i haven't personally seen anything |
| 03:14.04 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 03:16.24 | supjigatr | I finally got everything stable and wanted to move to the next application. |
| 03:18.37 | papa_e | you could probably hack something together with hylafax |
| 03:18.58 | papa_e | shouldn't require that much scripting, just basically passing callerid/phone number information to an external application |
| 03:20.02 | supjigatr | I was looking to use txfax. We have a hylafax solution but no PRI. |
| 03:25.05 | *** join/#asterisk suge (i=none@adsl-065-006-189-182.sip.bct.bellsouth.net) |
| 03:33.38 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 03:37.52 | hansin321 | I am running Astlinux 4.0 as a virtual machine in VM Player (though I have encountered this problem in other Asterisk set-ups). Whenever I call into a "service" that gives me back a voice response from the Asterisk server, the voice sounds choppy and stuttery (for lack of better words). When I get through (to say an inter-Asterisk eXchange with the servers at Digium, or a SIP to SIP call)... |
| 03:37.53 | hansin321 | ...the server, this doesn't occour. Any ideas? |
| 03:38.51 | hansin321 | For example the Voice Mail prompts. I think I would have enough resources available for that. |
| 03:40.33 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-24-49-168-129.albyny.adelphia.net) |
| 03:41.50 | Faithful | general oppinions on Aastra phones? |
| 03:43.26 | znoG | hey anyone have a SPA 841 phone? |
| 03:44.23 | Az_au | anyone have this problem before? Say someone is on a call (SIP (ALAW) to asterisk then IAX2 trunk (G729) to provider) if someone else picks up a phone and dials out it causes a few seconds of silence on the call that was already up |
| 03:44.41 | pjchilds | hansin321, ztdummy probably can't generate a consistent set of timing running under vmware... |
| 03:47.17 | hansin321 | pjchilds: Thanks, that actually makes sense from what I understand (I am fairly new, but have done a little reading). Eventually I will get a native Linux box running and try it there. |
| 03:47.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:49.59 | hansin321 | pjchilds: The time I had it running native on Linux (Fedora 3 or 4) I didn't have broadband, so I ran the box through a windows machine that had dial-connectivity, acting as a gateway. I can't remember, but the prompt from the Asterisk server may have been fine, but the other end of an Inter Asterisk eXchange at Digium sounded choppy. But that would be expected on a dial-up link I... |
| 03:50.00 | hansin321 | ...suppose. Thanks. |
| 03:51.47 | *** join/#asterisk bmg505 (n=leon@c1-81-10.rndf.isadsl.co.za) |
| 03:54.27 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 04:01.26 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 04:05.30 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 04:05.35 | Flauto | for the playback function |
| 04:05.45 | Flauto | is there a way to play multi-fiiles |
| 04:05.47 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:13.39 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:15.12 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 04:24.35 | tclark | hey anyone use the the 4 port sangoma cards ? |
| 04:24.43 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 04:27.23 | supjigatr | I do. |
| 04:27.26 | supjigatr | a104d |
| 04:41.36 | De_Mon | callparkandannounce is another one of those features thats worth backporting and keeping somewhere |
| 04:41.49 | De_Mon | well.. you know what I mean |
| 04:46.15 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 04:46.24 | revslowmo | Anyone know if it is possible to count the rings "90v" singnal on the line without picking it up? |
| 04:46.39 | revslowmo | On a simple fxo card |
| 04:50.16 | *** join/#asterisk dorphalsig (n=kjshdkfh@200.106.223.5) |
| 04:51.18 | dorphalsig | hello ... I'm trying to build an appliance box for a * VoIP Server ... can anybody recommend me any kind of hardware? |
| 04:53.02 | De_Mon | dorphalsig have you tried voip-info.org? |
| 04:54.07 | dorphalsig | Yes, I mean they recommend processor and such |
| 04:54.32 | De_Mon | ? |
| 04:54.33 | dorphalsig | but I'm trying to build something like a firewall box ... you just mount it in a rack and thats all |
| 04:54.46 | dorphalsig | no video card or anything of the likes |
| 04:55.20 | dorphalsig | I dont really know if those things are just a plain PC with no video card and with no peripherials but a network card |
| 04:55.39 | tclark | De_Mon: A CLUSTER of wrt54gs :) |
| 04:56.17 | De_Mon | the dorphalsig you mean like this: http://voip-info.org/wiki/view/Asterisk+dimensioning |
| 04:57.26 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:57.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:57.28 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 04:57.46 | De_Mon | if you're shopping for a rackable server... you woln't find many recomendations. Anything that runs linux and was built within the last 3 years is plenty for most uses |
| 04:58.25 | orlok | dorphalsig: go for a quality motherboard |
| 04:58.28 | *** join/#asterisk angler_ (n=angler@gateway.digium.com) |
| 04:58.37 | orlok | dorphalsig: Tyan, Supermicro, etc |
| 04:58.38 | dorphalsig | but anything will do? |
| 04:58.49 | orlok | depends on the load and reliability you are after |
| 04:59.15 | orlok | if you want a company to run its phone system off it, you dont want anything less than harddware raid and hot swap redundant psu's imo |
| 04:59.31 | dorphalsig | 16 SIP extensions, about 10 calls / hour |
| 04:59.38 | dorphalsig | (each) |
| 04:59.56 | De_Mon | dorphalsig how many at the same time |
| 05:00.11 | De_Mon | go read that page |
| 05:00.57 | dorphalsig | De_Mon ... reading ... |
| 05:01.24 | revslowmo | Anyone know a good howto to setup a box with a fxo card and mycomputer to dialout on it with say twinkle or some other voip software. |
| 05:02.29 | De_Mon | dorphalsig concurrent calls and codec are two of the more important questions. a P266 or better would probably be fine for what you need |
| 05:02.45 | De_Mon | i'm going to bed. Cya! |
| 05:02.48 | *** join/#asterisk BugKham (i=BugKham@202.8.86.162) |
| 05:03.35 | dorphalsig | Now my question is ... DO i need to purchase special things to make the appliance work on its own? (you know ... generally computers wont start if they dont have a video card set) |
| 05:04.41 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 05:04.44 | dlynes_home | dorphalsig: you don't need a video card or a network card to run asterisk |
| 05:04.47 | BugKham | Hi, anyone using 1.2.7.1? |
| 05:05.15 | dlynes_home | dorphalsig: however, you're somewhat limited in your options if you don't have a network card |
| 05:05.15 | dorphalsig | dlynes_home --> I know, but at least my computer will refuse to start if I dont get it a video card ... |
| 05:05.32 | dlynes_home | dorphalsig: it shouldn't...just disable the check for a video card |
| 05:05.35 | orlok | dorphalsig: how do you know? |
| 05:05.44 | L|NUX | i have configured asterisk |
| 05:05.46 | dorphalsig | orlok --> Tried it :$ |
| 05:06.10 | orlok | dorphalsig: it can be dependant on boards. we have a pile of old socket 7 systems that are fine headless |
| 05:06.12 | L|NUX | when i put some one on hold then my moh run but files its play too fast. what to do to fix this issue |
| 05:06.19 | orlok | but of course so many boards come with on-board video |
| 05:06.19 | BugKham | dlynes_home: what version of * r u running? |
| 05:06.25 | dlynes_home | I only use the video card on two of my linux boxes; all the rest are headless |
| 05:06.48 | dlynes_home | BugKham: 1.2.4, 1.2.7.1, trunk, and oej's test-this-branch |
| 05:07.27 | BugKham | dlynes_home: have you encountered any problem with cmd Dial in 1.2.7.1? |
| 05:07.35 | BugKham | dlynes_home: like L(${TIMELEFT}:61000:30000) |
| 05:07.36 | dorphalsig | dlynes_home --> thankx =) |
| 05:08.05 | dlynes_home | BugKham: never but then again, i don't use that option |
| 05:08.28 | BugKham | dlynes_home: it won't disconnect when the variable TIMELEFT is zero |
| 05:08.31 | dlynes_home | BugKham: I don't do any prepaid billing; only postpaid |
| 05:09.10 | dlynes_home | BugKham: Well, that begs the question why are you even letting it get to that point when timeleft is 0? |
| 05:09.11 | *** join/#asterisk xanderp (n=pphillip@12.222.95.177) |
| 05:09.15 | BugKham | dlynes_home: ok |
| 05:09.32 | dlynes_home | BugKham: but yeah, i can see you point |
| 05:09.44 | *** join/#asterisk rushowr (n=team_z@cpe-24-26-133-52.columbus.res.rr.com) |
| 05:09.46 | dlynes_home | BugKham: have you checked bugs.digium.com to see if someone's reported that bug already? |
| 05:09.51 | BugKham | dlynes_home: http://bugs.digium.com/view.php?id=6979 |
| 05:10.00 | rushowr | hey all |
| 05:10.07 | SkramX | none of my agi scripts seem to work |
| 05:10.16 | xanderp | anybody here used a cisco ubr925 with asterisk? it has 2 phone ports in it, and there is an firmware that supports SIP. |
| 05:10.34 | SkramX | well, one does.. but none of the others do |
| 05:10.36 | dlynes_home | BugKham: you see at the bottom there? |
| 05:10.41 | dlynes_home | BugKham: it says it's fixed in trunk |
| 05:10.44 | SkramX | even examples that came with the perl module |
| 05:10.50 | dlynes_home | BugKham: that means when asterisk 1.2.8 comes out, that bug will be fixed |
| 05:10.53 | BugKham | dlynes_home: not the 1.2.7.1? |
| 05:11.04 | rushowr | anyone know of a decent way to create connections and run queries to a PostgreSQL db from within the dialplan? PGSQL() used to do it but it's not in the compile anymore. |
| 05:11.08 | dlynes_home | BugKham: 1.2.8 should be coming out any day...it was marked for release about 3 days ago |
| 05:11.53 | BugKham | dlynes_home: i see |
| 05:12.37 | *** join/#asterisk neilbags (n=neil@149.171.94.137) |
| 05:13.20 | BugKham | dlynes_home: but the one on svn should be fixed, right? |
| 05:13.31 | dlynes_home | BugKham: correct |
| 05:13.34 | dlynes_home | BugKham: but.... |
| 05:13.47 | dlynes_home | BugKham: I wouldn't try trunk in a production environment |
| 05:13.57 | dlynes_home | BugKham: there's been a few nasty bugs in there recently |
| 05:14.13 | dlynes_home | BugKham: so its state might be a little edgy when you go to check it out |
| 05:14.22 | dlynes_home | BugKham: what i would do instead |
| 05:14.29 | dlynes_home | BugKham: is check out 1.2.8 |
| 05:14.33 | *** join/#asterisk dasmoot_ (n=dasmoot@S0106001195567012.no.shawcable.net) |
| 05:14.57 | rushowr | anybody have a clue on using postgresql from the dialplan? |
| 05:14.58 | BugKham | dlynes_home: ok, just have to wait then |
| 05:15.11 | SkramX | <PROTECTED> |
| 05:15.16 | SkramX | <PROTECTED> |
| 05:15.24 | SkramX | it foesnt actually execute or do anything |
| 05:15.26 | SkramX | any idea? |
| 05:15.32 | SkramX | thats direct from the example |
| 05:15.53 | rushowr | <PROTECTED> |
| 05:15.58 | SkramX | yes |
| 05:16.02 | SkramX | it just shows the channel, etc |
| 05:16.06 | SkramX | nothing about the agi script |
| 05:16.16 | SkramX | this one script, swift.agi works, but nothing else |
| 05:16.58 | rushowr | sorry then, don't know. I'd tryto help but I'm fighting a wicked battle with the no missing application PGSQL() |
| 05:17.43 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
| 05:18.11 | *** part/#asterisk rushowr (n=team_z@cpe-24-26-133-52.columbus.res.rr.com) |
| 05:18.13 | websae | hey if anyone need SIP phones or ATAs in here, give me a hollar -- willing to extend a hand to my fellow asterisk users :)! -- I have many in stock, and have some great connections if I don't have it... |
| 05:18.30 | SkramX | websae: hmmm, website? |
| 05:18.36 | BugKham | dlynes_home: again, how do you know when the next version will be released? I can't find the release schedules anywhere |
| 05:18.37 | websae | nah |
| 05:18.47 | websae | i don't do this for a living |
| 05:18.49 | SkramX | i need one to do a presentation on asterisk, and virtual private servers |
| 05:18.51 | SkramX | okay |
| 05:18.56 | SkramX | rather buy from a real business |
| 05:18.56 | websae | just thought I'd mention it though |
| 05:19.01 | SkramX | unless its free |
| 05:19.02 | SkramX | :P |
| 05:19.05 | websae | well I am a real VoIP company |
| 05:19.21 | websae | well understood then |
| 05:19.27 | dlynes_home | BugKham: no release schedules |
| 05:19.42 | dasmoot_ | wondering if someone could give me a hand... DNS on my asterisk machine is fine and I can ping sip.provider.com, but I'm getting the following in my Asterisk logs: |
| 05:19.42 | dasmoot_ | chan_sip.c: No such host: sip.provider.com |
| 05:19.42 | dasmoot_ | chan_sip.c: Probably a DNS error for registration to username@sip.provider.com, trying REGISTER again (after 20 seconds) |
| 05:19.44 | dlynes_home | BugKham: however, 1.4 is slated to be released in another month or so |
| 05:19.53 | dasmoot_ | someone here suggested that I should ditch my debs and compile from source, so I'm using a freshly built 1.2.7.1 with the config files from 'make samples' -- the only change I've made is to setup my VoIP provider in sip.conf and change the TRUNK variable in extensions.conf |
| 05:22.35 | dlynes_home | BugKham: you can also monitor the asterisk-dev mailing list and the #asterisk-dev channel on irc |
| 05:22.43 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:22.58 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 05:23.09 | BugKham | dlynes_home: oh, ok |
| 05:23.12 | dlynes_home | BugKham: Every once in a while, they mention possible release dates, or at least give you some kind of an idea when something will be coming out, and what will be included in the new release |
| 05:23.27 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 05:23.42 | rabelais | where can I read more about the asterisk database functionss? |
| 05:23.49 | BugKham | dlynes_home: for now, which's considered the most stable version? |
| 05:24.12 | dlynes_home | BugKham: but also, I think i remember something on the main website that mentioned something about when they're planning to have 1.4 and 1.6 out |
| 05:24.14 | BugKham | dlynes_home: for me, 1.2.6 was good |
| 05:24.26 | dlynes_home | BugKham: 1.2.7.1 is considered to be the most stable |
| 05:24.48 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 05:27.47 | dlynes_home | wow....down to 242 bugs |
| 05:28.08 | dlynes_home | 28 bugs fixed in less than a week |
| 05:28.24 | Qwell | about 100 fixed in about 2 weeks |
| 05:29.19 | rabelais | if I'm looking to put a database of my calls (callerid, call duration, etc) would that best be done with the MYSQL commands or with dbget dbput, etc? |
| 05:29.35 | Qwell | rabelais: like cdr? |
| 05:29.57 | Ahrimanes | rabelais: cdr_mysql, cdr_odbc or cdr_pgsql |
| 05:29.58 | rabelais | Qwell: I don't know what cdr is |
| 05:30.06 | Qwell | ~cdr |
| 05:30.07 | jbot | rumour has it, cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 05:30.26 | dlynes_home | hrm |
| 05:30.35 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 05:30.37 | dlynes_home | i like the second definition better...less painful |
| 05:30.50 | rabelais | ooh, I see...it's sorta already done for me |
| 05:30.51 | rabelais | hehe |
| 05:31.07 | dlynes_home | not sorta |
| 05:31.12 | dlynes_home | it is done for you already :) |
| 05:31.46 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 05:31.51 | rabelais | so then I'm guessing there must be applications that can parse this? |
| 05:31.55 | Ahrimanes | dlynes_home: who would have thought anyone else would have had that need.. :P |
| 05:32.07 | dlynes_home | Ahrimanes: no kidding :) |
| 05:32.07 | Ahrimanes | rabelais: asterisk-stat is one |
| 05:32.37 | dlynes_home | rabelais: Perl's Text::XSV |
| 05:32.47 | dlynes_home | erm Text::xSV I mean |
| 05:32.56 | rabelais | ah, I see |
| 05:32.58 | Ahrimanes | rabelais: for nice webviewing: http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 |
| 05:33.30 | rabelais | is there a way to rotate the master.csv file? or should I not need to do that? |
| 05:33.32 | dlynes_home | rabelais: also, mysql's mysqlimport(?) command will import it into a table, if you have the table already defined |
| 05:33.41 | dlynes_home | rabelais: add it to your logger.conf file |
| 05:33.46 | rabelais | wonderful! |
| 05:33.51 | rabelais | they've thought of everything, haven't they? |
| 05:34.15 | dlynes_home | rabelais: Unfortunately, I haven't been able to get Asterisk to do my dishes for me yet :((( |
| 05:34.19 | Ahrimanes | not _everything_.. but most of the common stuff :) |
| 05:34.27 | rabelais | now if only it was documented in a way that I could find it without being confused with all these crazy acronyms or having to ask you nice fellows |
| 05:34.29 | rabelais | :T |
| 05:35.02 | rabelais | I've been using asterisk for over a year now and had no idea there was an call detail...I had just been procrastinating making it myself :/ |
| 05:35.10 | Ahrimanes | dlynes_home: hint: record "do my dishes *beeeeep*" and use a predictive dialer to dial mother, sister and any other females you may have a number for ;) |
| 05:35.25 | dlynes_home | Ahrimanes: not gonna happen |
| 05:35.37 | dlynes_home | Ahrimanes: my closest relative is 3500 miles away |
| 05:35.39 | Ahrimanes | hehe |
| 05:37.07 | rabelais | do I need a database for asterisk-stat? or can it just parse the csv file? |
| 05:37.57 | Ahrimanes | rabelais: asterisk-stat need postgresql or mysql database |
| 05:38.07 | rabelais | hmm, I see |
| 05:38.19 | rabelais | so asterisk has to be setup to dump the cdr to a mysql database? |
| 05:38.30 | rabelais | or postgres |
| 05:38.53 | websae | anyone know how to have a SMS gateway with asterisk? |
| 05:39.05 | dlynes_home | websae: www.kannel.org |
| 05:39.36 | websae | dlynes: have you done much with it? |
| 05:39.36 | Ahrimanes | rabelais: i would install postgresql, make sure the -devel version is there too and then recompile asterisk to get the cdr_psql aplication.. then setup cdr_pgsql.conf... all calls are logged to postgresql and asterisk-stat can easily read the data |
| 05:39.42 | dlynes_home | And Kannel's a far more mature project than asterisk |
| 05:39.53 | SkramX | any tools for using asterisk realtime? |
| 05:39.53 | dlynes_home | websae: I've done almost nothing with it |
| 05:40.02 | dlynes_home | websae: But a local company here had used it a lot |
| 05:40.04 | SkramX | like better than phpmyadmin but not as bulky as freepbx? |
| 05:40.07 | pjchilds | websae, how about http://www.talktous.net.au/VoiceBlue.html |
| 05:40.35 | dlynes_home | websae: If you see nxu7 around, he knows a lot about integrating asterisk with kannel...he used to work at the company that implemented that solution here, locally |
| 05:41.24 | dlynes_home | SkramX: isn't phpmyadmin pretty bulky, too? |
| 05:41.28 | *** join/#asterisk califa54 (n=zeke@adsl-69-231-123-169.dsl.irvnca.pacbell.net) |
| 05:42.15 | dlynes_home | SkramX: you could try www.scriptyourownasteriskconfiguration.pl |
| 05:42.50 | SkramX | thanks. |
| 05:43.20 | Ahrimanes | dlynes_home: silly, that domain does not exist.. |
| 05:44.45 | dlynes_home | Ahrimanes: lol |
| 05:44.47 | Ahrimanes | find ~/ -iname "*.pl" | wc -l |
| 05:44.48 | Ahrimanes | 214 |
| 05:44.52 | Ahrimanes | damn |
| 05:44.54 | Ahrimanes | dlynes_home: ;) |
| 05:45.31 | Ahrimanes | i do too many small perl scripts for everything |
| 05:45.54 | dlynes_home | I do too many small bash shell scripts for everything :) |
| 05:46.31 | enkrypted | I have a neural interface to the shell. |
| 05:47.01 | pjchilds | she sells sea shells by the sea shore... |
| 05:47.26 | Ahrimanes | good way to fracture your tongue when drunk? |
| 05:50.26 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 05:50.37 | Ahrimanes | python even |
| 05:50.48 | Ahrimanes | guess i should have gone home last night |
| 05:52.32 | dlynes_home | perl's enough of a nightmare for me |
| 05:52.38 | dlynes_home | I don't need python, too :) |
| 05:53.06 | Ahrimanes | hehe |
| 05:56.52 | FuriousGeorge | anyone use chan_dsp as a pager for a large area |
| 05:56.57 | FuriousGeorge | something LOUD |
| 05:57.20 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 05:57.26 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 05:58.09 | FuriousGeorge | i have an existing loudspeaker installed but it interfaces via rj11 |
| 05:58.44 | dlynes_home | FuriousGeorge: why can't you use it? |
| 05:59.13 | FuriousGeorge | dlynes_home: its basically a handset with an amplifier as i see it. how do you plug a handset into your soundcard |
| 05:59.45 | dlynes_home | FuriousGeorge: does it ring? |
| 05:59.50 | *** part/#asterisk dasmoot_ (n=dasmoot@S0106001195567012.no.shawcable.net) |
| 06:00.11 | FuriousGeorge | dlynes_home: no, its not like a telephone, its like a handset for a telephone |
| 06:00.40 | FuriousGeorge | like if i wanted to put a sip phone there i could put it on autoanswer with a headset and it would work |
| 06:00.50 | dlynes_home | FuriousGeorge: hook it up as a speaker to your amplified out jack on your soundcard |
| 06:00.58 | FuriousGeorge | but the headset would be the loudspeaker |
| 06:01.02 | dlynes_home | FuriousGeorge: see if it does anything |
| 06:01.08 | FuriousGeorge | it wont |
| 06:01.49 | dlynes_home | FuriousGeorge: put it on an analog jack, ring that analog line, and see if does anything? |
| 06:02.00 | FuriousGeorge | there is no simple in-line adapter for this sort of thing |
| 06:02.15 | dlynes_home | FuriousGeorge: i'm thinking plug it into an fxs jack |
| 06:02.27 | dlynes_home | FuriousGeorge: and then ringing that fxs port, and broadcasting on it |
| 06:02.31 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 06:02.47 | FuriousGeorge | i didnt want to use another tdm400p just for this one phone that is only a loudspeaker when that is what chandsp is for |
| 06:02.58 | Ahrimanes | it wouldnt have the intelligence to handle ringtone etc.. |
| 06:03.18 | FuriousGeorge | sure any phone with an rj11 port for a headset and autoanswer would work |
| 06:03.40 | FuriousGeorge | i could plug the loudspeaker into the phone |
| 06:04.14 | Ahrimanes | yeah, but not directly plugging the handset in.. |
| 06:04.30 | FuriousGeorge | exactamundo |
| 06:05.05 | dlynes_home | Why not just hook up a paging amplifier and a horn? |
| 06:05.25 | FuriousGeorge | just cuz there already is an amplifier on the roof |
| 06:05.28 | dlynes_home | Then you can control the volume as well |
| 06:05.59 | FuriousGeorge | i can control the volume of chan_dsp with alsamixer or something im sure |
| 06:06.09 | Lino` | hmmm |
| 06:06.39 | Lino` | an fxs jack just for that speaker would be overkill |
| 06:06.43 | Lino` | thats a fact. |
| 06:07.18 | Lino` | maybe you can solder an adaptor for rj11 -> snd card line out or something |
| 06:07.46 | FuriousGeorge | http://www.jkaudio.com/voice-path.htm |
| 06:07.49 | Ahrimanes | typical rj11 handset seems to have 4 wires.. which should be mapped where? |
| 06:07.53 | FuriousGeorge | thats an example of the types of things im finding |
| 06:08.09 | Lino` | there are plenty of diagrams available |
| 06:08.13 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:08.41 | Ahrimanes | morning oej |
| 06:08.46 | oej | Morning |
| 06:08.50 | oej | Did the patch help you? |
| 06:08.56 | FuriousGeorge | i read that most devices made to record to a soundcard from a phone line will work in the other direction |
| 06:09.15 | FuriousGeorge | and most of these types of devices are made for the purpose of recording |
| 06:10.09 | Ahrimanes | oej: uhm... not sure if its this snom... but when i hit the transfer button (which _should_ do sip refer), it hangs up the call |
| 06:10.17 | Lino` | well |
| 06:10.25 | Ahrimanes | oej: plus.. with my current setup i can crash * at will :) |
| 06:10.34 | Lino` | FuriousGeorge are you from the USA? |
| 06:10.38 | oej | Ahrimanes: That's a feature, not a bug. You don't really want to transfer any calls, if you ask yourself |
| 06:10.40 | FuriousGeorge | Lino`: yeah |
| 06:10.44 | oej | Ahrimanes: That's weird |
| 06:10.47 | Lino` | because there is an adaptor available @ radioshack |
| 06:10.51 | Lino` | which has a switch on it |
| 06:10.54 | Lino` | for recording / playback |
| 06:11.05 | Lino` | so that officially works in both directions |
| 06:11.20 | Lino` | http://cgi.ebay.de/TANDY-Radio-Shack-Telefon-Mitschneideadapter_W0QQitemZ9133075414QQcategoryZ30464QQssPageNameZWDVWQQrdZ1QQcmdZViewItem <= heres one of them @ ebay germany |
| 06:11.38 | oej | Seems like Danish and Norwegian users have really weird bugs. What's up with these fellow scandinavians |
| 06:11.55 | FuriousGeorge | my next question would be: would the loudspeaker know if its on the hook or off the hook? its just a handset with an amplifier after all |
| 06:12.08 | Lino` | hm? |
| 06:12.17 | Lino` | if you connect it to the soundcard |
| 06:12.25 | Lino` | there should be no problem |
| 06:12.36 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 06:12.36 | Lino` | if you connect it to the phone line it'll get complicated |
| 06:12.48 | FuriousGeorge | well, via some 3rd part device as yet to be determined, it needs to connect eventually to the soundcard |
| 06:13.11 | Lino` | well |
| 06:13.16 | Lino` | if you had ISDN |
| 06:13.28 | Lino` | i'd say get a cheap phone which has intercom functionality |
| 06:13.34 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 06:13.38 | Lino` | and solder a bigger speaker to it |
| 06:13.41 | Lino` | quick cheap and dirty |
| 06:13.51 | littleball | hello, how to disable the sip debug info from CLI console? |
| 06:14.07 | FuriousGeorge | hmm, ive never done this. perhaps i should just get an amplifier and a speaker of some sort |
| 06:14.07 | kaldemar | littleball: sip no debug |
| 06:14.13 | littleball | thanks |
| 06:14.17 | Lino` | hm |
| 06:14.17 | FuriousGeorge | ever shopped for loudspeakers Lino`? |
| 06:14.21 | Lino` | :D |
| 06:14.31 | Lino` | yeah i have a huge electronics distributor right next to my house |
| 06:14.54 | Lino` | they always have several 10k items in stock |
| 06:15.19 | Lino` | and they also tell you what you need |
| 06:15.20 | FuriousGeorge | seriously? it should be relatively easy to interface a loudspeaker with a soundcard, no? |
| 06:15.25 | Lino` | well |
| 06:15.29 | Lino` | basically yes |
| 06:15.30 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 06:15.39 | FuriousGeorge | isnt this what chan_dsp is really for, after all? |
| 06:15.47 | Lino` | yes |
| 06:15.59 | Lino` | thats what i said, just connect that speaker to the soundcard |
| 06:15.59 | Ahrimanes | FuriousGeorge: if you want it to be really loud, soundcard amp probably has way too little power tho |
| 06:16.01 | Lino` | thats it |
| 06:16.11 | Lino` | why don't you use an active speaker anyway? |
| 06:16.15 | FuriousGeorge | Ahrimanes: thats why i figured i would need an amplifier anyway |
| 06:16.17 | FuriousGeorge | warehouse |
| 06:16.18 | Lino` | or an amplifier |
| 06:16.30 | Lino` | if its a warehouse |
| 06:16.33 | FuriousGeorge | si |
| 06:16.42 | Lino` | you can use those big ass "horns" |
| 06:16.48 | Lino` | know what i mean? |
| 06:16.50 | Ahrimanes | ouch |
| 06:16.53 | FuriousGeorge | sure thats what they got now |
| 06:16.59 | Lino` | sounds bad but works |
| 06:17.00 | Lino` | :D |
| 06:17.02 | FuriousGeorge | but it interfaces via an rjll jack |
| 06:17.11 | Ahrimanes | Lino`: run app_milliwat on one of those? |
| 06:17.14 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 06:17.14 | Lino` | then use the radio shack converter |
| 06:17.15 | Lino` | :D |
| 06:17.34 | Lino` | although i dont know how much power is consumed by those things |
| 06:17.40 | Lino` | and if a soundcard can handle it |
| 06:17.57 | FuriousGeorge | i can see its plugged into the wall, so to speak |
| 06:18.08 | FuriousGeorge | its got its own amp/power |
| 06:18.11 | Lino` | ok |
| 06:18.14 | Lino` | thats good news |
| 06:18.29 | FuriousGeorge | and that radioshack thing, im guessing, is meant to sit behind a phone. this thing is just a handset basically, im thinking |
| 06:18.37 | Lino` | yes |
| 06:19.01 | Lino` | what the radioshack thing does is simply taking a signal from a 3.5mm or 2.5mm jack |
| 06:19.06 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 06:19.15 | Lino` | and putting it into the rj11 thing for a handset |
| 06:19.50 | Lino` | you may need an rj11 extension thing to connect the two rj11 things |
| 06:19.55 | Lino` | but those are cheap ;) |
| 06:20.01 | FuriousGeorge | what do people normally plug into chan dsp when they want to be loud |
| 06:20.12 | Lino` | an amp? |
| 06:20.13 | Lino` | i guess |
| 06:20.14 | Lino` | :D |
| 06:20.23 | FuriousGeorge | and a stereo speaker? |
| 06:20.26 | Ahrimanes | Lino`: how many "things" are involved by now? heeh |
| 06:22.33 | FuriousGeorge | http://www.omnicronelectronics.com/PC/TSA-3PC.htm |
| 06:22.40 | FuriousGeorge | that could work |
| 06:23.32 | FuriousGeorge | i think |
| 06:24.58 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:25.31 | Lino` | öh |
| 06:25.32 | Lino` | ? |
| 06:25.42 | kmilitzer | Morning everyone ... |
| 06:25.42 | Lino` | dont know the word |
| 06:25.43 | Lino` | like |
| 06:25.46 | Lino` | plug |
| 06:25.59 | Lino` | do you have a rj11 socket in the wall that connects to the speaker? |
| 06:26.03 | Lino` | or to the amp? |
| 06:26.08 | Lino` | then you dont need the extension thing |
| 06:26.36 | Lino` | because i thought that you have two cables like -----> and <----- |
| 06:26.40 | Lino` | and that does not work :D |
| 06:26.56 | Lino` | so you'd need that little device which is basically an extension |
| 06:27.03 | Lino` | ----->][<------ |
| 06:27.04 | satlan32 | good mornin' all |
| 06:27.11 | satlan32 | i need help pleae |
| 06:29.03 | satlan32 | i'm trying to use the sipaddheader() and get this log |
| 06:29.04 | satlan32 | http://pastebin.ca/56798 |
| 06:29.18 | satlan32 | and the header is not added |
| 06:30.54 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 06:30.58 | Tier_1 | hey al |
| 06:31.04 | satlan32 | any idea? |
| 06:31.07 | Tier_1 | http://pastebin.ca/56799 need help with this |
| 06:31.31 | Tier_1 | need to know if it looks right |
| 06:37.08 | kmilitzer | satlan32: In your dialplan you add the headers, but no SIP package is added, because the session is already established. Try to set the extra header before you answer |
| 06:37.26 | satlan32 | oh... ok ... |
| 06:37.39 | satlan32 | and then it will add the header to all the packets of this call? |
| 06:37.49 | kmilitzer | satlan32: I meant: no SIP package is sent ... |
| 06:38.08 | satlan32 | i understood |
| 06:38.24 | FuriousGeorge | lino sorry i zoned out |
| 06:38.33 | kmilitzer | satlan32: Don't know, as I have used it only once for a outgoing call, but the way it is now, it cannot work |
| 06:38.39 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 06:39.05 | satlan32 | ??? |
| 06:40.25 | kmilitzer | satlan32: Just did a help for SipADDHeader: Adds a header to a SIP call placed with DIAL. |
| 06:40.34 | kmilitzer | satlan32: So it won't work the way you want it ... |
| 06:40.42 | satlan32 | ok |
| 06:40.53 | satlan32 | thanks i will think about another way |
| 06:44.12 | satlan32 | and sipgetheader will work for a call in progress |
| 06:44.13 | satlan32 | ? |
| 06:53.50 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 06:54.03 | tparcina | hi everybody! |
| 06:54.43 | tparcina | it's beautifull morning in Split (HR)... sun is shining, birds are singing... :)) |
| 06:55.06 | Ahrimanes | HR ? |
| 06:55.16 | tparcina | hrvatska - croatia |
| 06:55.23 | Ahrimanes | ah o |
| 06:55.24 | Ahrimanes | k |
| 06:55.42 | nextime | tparcina : same here ( north italy ) |
| 06:55.57 | Ahrimanes | not here (denmark) |
| 06:56.14 | tparcina | yes, birds are singing and i'm wondering can i put #6 in my dialplan as dialing number for turning on/off some options |
| 06:56.37 | tparcina | well, you should mouve from denmark to some not so well organized countries on south :)) |
| 06:56.40 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 06:58.07 | tparcina | so, what i would like to do is to make this possible |
| 06:58.08 | tparcina | exten => #6,1,Answer |
| 06:58.08 | tparcina | exten => #6,n,Wait(1) |
| 06:58.16 | tparcina | and so on... |
| 06:58.29 | satlan32 | i'm trying to work out SIP_HEADER |
| 06:58.41 | satlan32 | exten => 11,n,Set(foo=${SIP_HEADER(X-Salamat)}) |
| 06:58.41 | satlan32 | exten => 11,n,NoOp(${foo}) |
| 06:59.10 | tparcina | newer worked with sip header. why do you need them? |
| 06:59.34 | satlan32 | i have sipp to simulate the call with the header "X-Salamat: 123" |
| 06:59.43 | satlan32 | and i want to identify this header |
| 06:59.44 | boddy | hii all I configured bchan=1-15,17-31 for euro E1/pri in zaptel.conf how can I define group and channel defination on zapata.conf |
| 06:59.56 | satlan32 | but when i trace the call i get this log |
| 07:00.14 | boddy | I have to create 2 group in zapata ? |
| 07:00.27 | satlan32 | http://pastebin.ca/56804 |
| 07:01.10 | nextime | boddy : not really |
| 07:01.25 | satlan32 | as you can see there i have this at line 190 |
| 07:01.29 | nextime | you can use only one channel group |
| 07:01.44 | satlan32 | this function can only be used in sip calls |
| 07:01.47 | satlan32 | ... |
| 07:02.18 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:02.54 | nextime | boddy : simply setup all your needed parameters, like switchtype=euroisdn and signalling=pri_cpe, a context and so on, at the end of those, put dchannel => 16 and channel => 1-15,17-31 |
| 07:03.38 | boddy | ok thanks nextime |
| 07:03.44 | boddy | I will try |
| 07:04.40 | boddy | nextime:last question is what is the function of $(EXTEN) |
| 07:05.25 | *** join/#asterisk Tili (n=Tili@cm33.gamma249.maxonline.com.sg) |
| 07:06.00 | nextime | boddy : is ${EXTEN} not $(EXTEN), and cointain the called extension, so you can do many interesting things knowning the called number |
| 07:06.24 | nextime | boddy : maybe you need to make a tour on voip-info.org and read some docs, do you think? |
| 07:06.42 | boddy | yes |
| 07:07.11 | boddy | can I create group=1 in zapata.conf |
| 07:07.12 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:08.05 | satlan32 | anyone can help |
| 07:08.06 | satlan32 | ? |
| 07:08.18 | nextime | boddy : of course yes |
| 07:08.19 | tparcina | exten => #6,1,Answer and exten => _#6,1,Answer none of them work for me. can i use #6 as extension number? |
| 07:09.50 | iDunno | only if your phone is sending the # ;) |
| 07:09.55 | *** join/#asterisk delmar (n=delmar@203-114-178-231.static.inspire.net.nz) |
| 07:10.39 | tparcina | iDunno, it seams that sjphone doesn't send them. do you no do cisco phones send # ? |
| 07:11.02 | iDunno | sorry, never had a cisco phone :/ |
| 07:11.05 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:11.28 | tparcina | do you know for grandstream or any other ardwer phone? |
| 07:12.31 | iDunno | the budgetone don't unless you configure them to. |
| 07:12.34 | Ahrimanes | hardware ? |
| 07:12.40 | iDunno | (which is grandstream) |
| 07:12.47 | iDunno | # is used for "send" ;) |
| 07:13.04 | tparcina | ok, thank you |
| 07:13.13 | iDunno | IIRC the Snom190 does send the # |
| 07:17.23 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 07:21.43 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:34.01 | *** join/#asterisk kernel20 (n=kernel20@203.160.223.26) |
| 07:34.04 | kernel20 | hi |
| 07:34.08 | kernel20 | i am getting this |
| 07:34.17 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:34.46 | kernel20 | rpt.c330 process_rfr3389: Comfort noise support incomplete |
| 07:34.50 | kernel20 | any ideas? |
| 07:35.08 | zoa | turn of voice activity detection ? |
| 07:35.17 | kernel20 | how? |
| 07:35.24 | pif | hi, looking for the latest cisco 7960 firmware... |
| 07:36.57 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 07:40.51 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 07:42.03 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 07:42.12 | rvhi | hi |
| 07:42.29 | rvhi | anyone uses ast_data? |
| 07:42.44 | rvhi | had some issues with pattern matching on dial plan. |
| 07:43.00 | *** join/#asterisk catch23 (n=catch23@c-67-191-252-3.hsd1.ga.comcast.net) |
| 07:44.57 | *** join/#asterisk lorinc (n=ang@caracas-2722.adsl.interware.hu) |
| 07:45.42 | kernel20 | ? |
| 07:45.44 | kernel20 | rpt.c330 process_rfr3389: Comfort noise support incomplete |
| 07:45.46 | kernel20 | ? |
| 07:45.56 | *** join/#asterisk my007ms (n=noor@217.139.224.194) |
| 07:45.59 | my007ms | hello all |
| 07:46.02 | zoa | kernel i just told you |
| 07:46.14 | kernel20 | how to turn it off? |
| 07:46.17 | kernel20 | i dont know of it |
| 07:46.24 | my007ms | did any one know what hardware i need to deal with isdn |
| 07:46.25 | kernel20 | coz its new to me |
| 07:46.33 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 07:46.44 | mitcheloc | good morning everyone =) |
| 07:46.45 | kernel20 | u told nothing |
| 07:46.56 | pif | read the fucking admin manual to your phone |
| 07:46.59 | kernel20 | u only tell disable the damn voice |
| 07:47.06 | kernel20 | i use xlite |
| 07:47.11 | kernel20 | FYI |
| 07:47.16 | mitcheloc | heh, cheerful in here isn't it |
| 07:47.21 | mitcheloc | kernel20: audio problems? |
| 07:47.45 | kernel20 | Transmit Silence = no |
| 07:47.53 | kernel20 | what the hell should i look into |
| 07:48.25 | enkrypted | is kernel20 problem a client or asterisk or kernel issue? |
| 07:48.34 | enkrypted | because, no, you did not make that clear. |
| 07:48.37 | kernel20 | i have that error in the asterisk |
| 07:48.49 | enkrypted | kernel20: i have seen the error as well. |
| 07:48.52 | *** join/#asterisk tparcina_ (n=tparcina@wr-lama.iskon.hr) |
| 07:49.03 | enkrypted | mitcheloc: morning! :) |
| 07:49.05 | my007ms | is there is card that make me use ISDN line |
| 07:49.06 | kernel20 | rpt.c330 process_rfr3389: Comfort noise support incomplete in Asterisk (RFC 3389) |
| 07:49.17 | kernel20 | thats what i see when i dot asterisk -vvvc |
| 07:49.24 | mitcheloc | enkrypted: it's not really morning here ;) |
| 07:49.26 | Qwell | kernel20: Show us the ENTIRE line |
| 07:49.38 | mitcheloc | kernel20: i have that, just ignore it and be done with it |
| 07:49.40 | Qwell | no, fuck it |
| 07:49.43 | Qwell | zoa told you |
| 07:49.49 | enkrypted | zoa did not |
| 07:49.52 | Qwell | I'm not in the mood to deal with you |
| 07:50.00 | enkrypted | i can read as well. |
| 07:50.13 | Qwell | enkrypted: not you |
| 07:50.33 | tparcina_ | i can't make exten => _*6 or exten => _#6 to work with cisco, or sjphone. can anybody sugest anything? |
| 07:50.51 | enkrypted | 03:35 < zoa> turn of voice activity detection ? |
| 07:50.51 | enkrypted | 03:46 < zoa> kernel i just told you |
| 07:51.00 | enkrypted | I'd hardle call that an explanation. |
| 07:51.04 | enkrypted | hardly, even. |
| 07:51.04 | mitcheloc | tparcina_: look up the dialplan.xml file to put on your tftp server |
| 07:51.12 | Qwell | enkrypted: You want to deal with him? Go right ahead. |
| 07:51.15 | Qwell | It's your headache now |
| 07:51.17 | tparcina_ | i can't set some other extension (like 999 od 888) because those extension's are allready taken (or will be) |
| 07:51.20 | enkrypted | haha |
| 07:52.00 | mitcheloc | it sure is cheerful in here ;) |
| 07:52.03 | enkrypted | here goes, kernel20: I'm assuming its a client issue (xlite). There is a setting under audio to turn off voice detection |
| 07:52.15 | kernel20 | rpt.c330 process_rfr3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible |
| 07:52.23 | tparcina_ | mitcheloc, thank you, i'll double check for my dialplan |
| 07:52.24 | enkrypted | kernel20: shutup now |
| 07:52.29 | Qwell | "Please turn off on client if possible" |
| 07:52.32 | Qwell | dude...come on |
| 07:52.36 | Qwell | RTFEM |
| 07:52.39 | kernel20 | yeah i know |
| 07:52.45 | kernel20 | u have used x-lite? |
| 07:52.49 | kernel20 | which option? |
| 07:52.56 | zoa | in xlite it is called: transmit silence |
| 07:52.58 | zoa | set it to no |
| 07:53.03 | kernel20 | yeah i did |
| 07:53.06 | kernel20 | but the same |
| 07:53.08 | pif | dig the menus dude |
| 07:53.11 | zoa | then it should go away |
| 07:53.12 | kernel20 | i know u will tell that |
| 07:53.16 | zoa | its one of those settings |
| 07:53.17 | Qwell | It's asking for CNG, not VAD |
| 07:53.17 | kernel20 | no it wont |
| 07:53.35 | mitcheloc | tparcina_: http://www.loligo.com/asterisk/Cisco/79xx/2003-04-27.examples/dialplan.xml, you also need to enable in SIPDefault.cnf |
| 07:53.48 | kernel20 | rpt.c330 process_rfr3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible |
| 07:53.50 | zoa | look around the transmit silence settings, it will be an option that is probably very close to the other one |
| 07:53.51 | kernel20 | ? |
| 07:54.01 | kernel20 | hmm |
| 07:54.23 | enkrypted | kernel20: please stop pasting your error, you are merely aggravating your situation. |
| 07:54.49 | zoa | http://www.asteriskguru.com/tutorials/comfort_noise_support_incomplete.html |
| 07:54.54 | zoa | first link in google |
| 07:54.58 | tparcina_ | mitcheloc, can you tell me what are those for - http://pastebin.ca/56813 |
| 07:55.16 | tparcina_ | they are cisco defaults, and I belive that i can delete them |
| 07:55.50 | mitcheloc | tparcina_: i wouldn't know, think of them like the sample extensions.conf file, they are just an example, modify at will |
| 07:56.55 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:57.28 | enkrypted | zoa: sorry you were right the first time.. |
| 07:57.45 | enkrypted | i misread the second statement |
| 07:58.07 | zoa | no need to say sorry! |
| 07:58.17 | tparcina_ | mitcheloc: you have mentioned that i need to enable it in sipdefault.cnf, did you think only about dialplan line or something else? |
| 07:58.32 | zoa | just gimme all your money! |
| 07:58.35 | zoa | and your first born! |
| 07:58.37 | mitcheloc | tparcina_: dialplan line should be good |
| 07:58.44 | mitcheloc | *already claimed his first born*' |
| 07:58.58 | tparcina_ | mitcheloc: thank you, i'll try to adapt my dialplan.xml |
| 07:59.03 | enkrypted | haha |
| 08:02.53 | pif | how can I combine auth and extension data on a SIP Dial() string? |
| 08:03.02 | zoa | ok, lets divide and conquer, i will take the left leg and some of his fingers |
| 08:03.12 | mitcheloc | ewww |
| 08:03.27 | pif | I'm using Dial(SIP/${MACRO_EXTEN}@host) |
| 08:03.54 | pif | but I had to deactivate auth in my sip.conf for that to work |
| 08:05.07 | kernel20 | pif: read the fucking manual |
| 08:05.31 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 08:05.43 | mitcheloc | woa there girls |
| 08:07.07 | mitcheloc | pif: probably something like this, but don't shoot me if i'm wrong: Dial(IAX2/user:pass@voip-a.host.com/${EXTEN},,r) |
| 08:07.29 | pif | mitcheloc : this works for iax2, not for sip |
| 08:07.34 | mitcheloc | tparcina_: i didn't mean *you*, your first born would be be your first-born-child |
| 08:08.03 | mitcheloc | pif: i dunno, i just put that in sip.conf and i'm done with it |
| 08:08.23 | pif | what did you put ? |
| 08:08.35 | tparcina_ | mitcheloc: dialplan.xml doesn't help. it seams that cisco sends something, but asterisk doesn't get it (i have exten => ?*6,1,Answer= |
| 08:08.40 | *** part/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:09.41 | mitcheloc | tparcina_: if the cisco isn't beeping at you then it's a dialplan issue, i don't know about the question mark though? |
| 08:09.47 | mitcheloc | *it's probably a dialplan issue |
| 08:11.38 | tparcina_ | mitcheloc: i'm preaty sure that i have make my dialplan well - loke at here http://pastebin.ca/56817 |
| 08:13.55 | mitcheloc | tparcina_: i don't know, but why do you have _ in there? you aren't matching an unknown extension |
| 08:15.02 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 08:15.12 | boddy | I bought te110 p which module I have loaded but I received this error "wcte11xp: module not supported by Novell, setting U taint flag" |
| 08:15.41 | boddy | sorry I have loaded wcte11xp |
| 08:15.47 | boddy | module |
| 08:15.53 | boddy | on suse |
| 08:17.38 | *** join/#asterisk dpryo (i=hn@donatello.nesland.net) |
| 08:18.19 | tparcina_ | mitcheloc: question mark was error |
| 08:18.40 | mitcheloc | tparcina_: ahh, cool =) |
| 08:19.34 | tparcina_ | no, i'm not matching unknown extension, but wen extension starts with * or # it has to have _ in front. or have i got it wrong? |
| 08:20.55 | mitcheloc | my guess is no you don't need it |
| 08:21.44 | tparcina_ | i have remouved _ from exten and I get the same tone in my headset... that's it for me (for now at leasat :)). i'll have to find some free extension to use them instead of * or # |
| 08:22.18 | mitcheloc | tparcina_: oh i thought you meant the ? mark fixed it |
| 08:22.51 | mitcheloc | i don't have a cisco in front of me to test =/ |
| 08:23.27 | tparcina_ | no, it didn't fix it, it was just an error that i have mistyped in irc :) |
| 08:23.39 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 08:24.15 | mitcheloc | tparcina_: i can't help you any further at the moment |
| 08:24.52 | mitcheloc | tparcina_: are you sure the cisco loads up the dialplan.xml file? |
| 08:26.29 | tparcina_ | yes, i have rebooted the phone |
| 08:26.44 | mitcheloc | but are you sure it loaded the file? |
| 08:27.12 | tparcina_ | yes, because when i dial *6 it dials emidietly |
| 08:27.49 | mitcheloc | in your sipdefault.cnf the line is like this: "dial_template: dialplan" |
| 08:28.19 | Alex | Anyone know of any *good* Howto/FAQs for Asterisk + SIP? |
| 08:28.24 | tparcina_ | yes, i have dial_template: dialplan |
| 08:28.55 | tparcina_ | Alex, look at http://www.voip-info.org/wiki/ |
| 08:28.58 | Alex | Thanks tparcina_ |
| 08:29.05 | tparcina_ | you are welcome |
| 08:29.17 | mitcheloc | Alex: did you not know about voip-info? |
| 08:29.38 | Alex | mitcheloc: "It's my first day" ;) |
| 08:30.04 | mitcheloc | Alex: cool! |
| 08:30.28 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:30.28 | Alex | New job, doing sys admin, and they decided to lump a load of Asterisk/VoIP stuff on me as well. |
| 08:30.44 | mitcheloc | cool, where at? |
| 08:31.25 | Alex | http://aql.com/telecoms/ |
| 08:32.13 | mitcheloc | cool, hosted pbx, or just termination? |
| 08:32.56 | Alex | Hosted |
| 08:33.33 | mitcheloc | tparcina_: did you look at the link i sent you? |
| 08:34.09 | mitcheloc | Alex: cool, well the wiki is the best place for documentation, the next is poking around the /etc/asterisk directory |
| 08:34.20 | Alex | Thanks. |
| 08:34.25 | mitcheloc | np |
| 08:36.55 | *** join/#asterisk FRUMUSHELUL (n=nomody@195.138.120.151) |
| 08:36.58 | FRUMUSHELUL | Hello |
| 08:37.01 | FRUMUSHELUL | i have some question |
| 08:37.06 | FRUMUSHELUL | anybody can help me? |
| 08:37.51 | Alex | Ask and you shall receive |
| 08:37.52 | FRUMUSHELUL | I have a network with a lot of users, and i want to make able calls from network via NEtMeeting to POTS |
| 08:38.14 | FRUMUSHELUL | and viceversa, from POTS to NEtmeeting |
| 08:38.24 | mitcheloc | no netmeeting here |
| 08:38.38 | mitcheloc | can you use something else? |
| 08:39.01 | FRUMUSHELUL | can i do this using Asterisk software and Wildcard X100P |
| 08:39.02 | FRUMUSHELUL | <PROTECTED> |
| 08:39.07 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:39.08 | FRUMUSHELUL | sure |
| 08:39.20 | FRUMUSHELUL | what windows client i have to install on user's computers? |
| 08:39.46 | mitcheloc | is cost an issue? |
| 08:39.50 | FRUMUSHELUL | Client->Softare -> Gate -> POTS -> PHONE |
| 08:40.00 | mitcheloc | an x100p means you will only have one outgoing call? |
| 08:40.08 | FRUMUSHELUL | mitcheloc> yes |
| 08:40.30 | FRUMUSHELUL | first i want to check with one line |
| 08:40.43 | mitcheloc | FRUMUSHELUL: what you are looking for is called a soft-phone, now you need to decide what is best for you |
| 08:40.57 | mitcheloc | first decide if you want free or paid, and what protocol, sip or iax (not h323 like netmeeting) |
| 08:41.08 | mitcheloc | then look on www.voip-info.org at the softphones listed there and pick what you want |
| 08:41.13 | FRUMUSHELUL | why i can't use netmeeting? |
| 08:41.57 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 08:42.04 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 08:42.06 | mitcheloc | it's archaic, it uses h323, it's not well supported, you'll get poor voice quality compared to these other phones |
| 08:42.46 | FRUMUSHELUL | what softphone you can recommend me? |
| 08:43.09 | mitcheloc | personally i use eyebeam from counterpath.com |
| 08:43.23 | mitcheloc | but it's paid, they do have a free version, but it doesn't exactly work (at least not for me) |
| 08:44.19 | zoa | mitcheloc: did you ever try idefisk ? |
| 08:44.22 | FRUMUSHELUL | How it is possible to make calls from outside (Phone) to a softphone? |
| 08:44.51 | mitcheloc | zoa: nah |
| 08:45.12 | mitcheloc | FRUMUSHELUL: you have a server with asterisk on it, and a card (like x100p), and the phone makes the call through the asterisk server |
| 08:45.24 | mitcheloc | much like an web server-browser analogy |
| 08:45.37 | *** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it) |
| 08:46.01 | FRUMUSHELUL | i have one analog number, and 100 users. How can the call be redirected to a specific user ? |
| 08:46.10 | FRUMUSHELUL | ex: my number is 323322 |
| 08:46.29 | FRUMUSHELUL | and i want to call to a softuser (ip: 10.1.1.5) from outside phone |
| 08:46.49 | FRUMUSHELUL | sa, i have to call to 323322, and to enter an additional number? or how? |
| 08:49.51 | boddy | I insert te110p and configured zaptel and zapata but there is no led running on card |
| 08:50.00 | boddy | how can I check connectivity ? |
| 08:50.14 | boddy | between meridian and digium |
| 08:50.21 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 08:51.33 | Agrajag- | gday, anyone happen to be around that could help me setup a spa-3000 to get incoming PSTN calls go through to asterisk? i've turned off the "PSTN Ring Thru Line 1" but have "PSTN-To-VoIP Gateway Enable" set to yes |
| 08:51.57 | Agrajag- | and i have the PSTN line registered with asterisk |
| 08:53.53 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:54.10 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 08:59.30 | boddy | is there anyone who running asterisk on suse |
| 09:00.19 | boddy | ? |
| 09:02.24 | *** join/#asterisk h3x0r4t0r (i=hex@ip68-224-57-17.lv.lv.cox.net) |
| 09:02.46 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.39 | *** join/#asterisk motu (n=motu@192.165.166.183) |
| 09:06.51 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 09:07.51 | Assid | err.. what exactly is a 'bridged' call ? |
| 09:09.11 | qdk | Assid: that is when Asterisk connects 2 channels. |
| 09:09.36 | satlan32 | hi |
| 09:09.39 | satlan32 | need help |
| 09:09.47 | Assid | so a native bridge would be one that doesnt need any transcoding |
| 09:09.56 | satlan32 | i tried to run exten => 15,n,System(echo | mutt -s "hello" -a /tmp/wav.wav XX@XX.xom) |
| 09:10.04 | satlan32 | but i get nothing in the mailbox |
| 09:12.04 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:12.41 | Assid | wtf.. something just went haywire |
| 09:12.54 | Assid | my files just started disappearing |
| 09:14.09 | *** join/#asterisk redondos_ (n=redondos@190.48.35.242) |
| 09:21.08 | satlan32 | hi |
| 09:21.10 | satlan32 | need help |
| 09:21.12 | satlan32 | i tried to run exten => 15,n,System(echo | mutt -s "hello" -a /tmp/wav.wav XX@XX.xom) |
| 09:21.14 | satlan32 | but i get nothing in the mailbox |
| 09:21.34 | satlan32 | how i use the system() function? |
| 09:24.50 | kmilitzer | satlan32: Is you mail system configured correctly, what does you mail log say, what does your asterisk verbose log say? |
| 09:27.11 | boddy | anybody help me ? |
| 09:31.23 | boddy | wcte11xp: module not supported by novel |
| 09:31.29 | boddy | I received this error |
| 09:31.33 | boddy | on suse |
| 09:32.39 | boddy | ? |
| 09:33.50 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 09:34.29 | boddy | :( |
| 09:35.35 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:37.16 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 09:37.18 | xbit` | hi |
| 09:37.40 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 09:37.54 | xbit` | what is that mean? anyone knows? Maximum retries exceeded on transmission 00007da6446c3f9d161e87ba43353d7a@xxx.xxx.xxx.xxx for seqno 102 (Critical Response) |
| 09:38.44 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 09:40.05 | RoyK | <PROTECTED> |
| 09:40.51 | RoyK | xbit`: i guess it means asterisk has tried to retransmit a (SIP?) packet the maximum retry count (5?) without getting a reply |
| 09:43.37 | Agrajag- | for my outgoing calls, i want to define a couple of extensions that go via certain routes, but then have a rule for "and anything else, send to this". are extensions read in the order listed or something? i don't want users to have to prefix numbers |
| 09:44.27 | Agrajag- | for example, if i have "exten => _9XXXXXXX,1,Dial(foo)", how do i then do "and anything else, Dial(bar)" |
| 09:50.10 | qdk | Agrajag-: most specific exten "wins". |
| 09:51.33 | Agrajag- | excellent - so if i had exten => _X.,1,Dial(bar) anything that doesn't start with a 9 in that case would then go to bar |
| 09:52.46 | *** join/#asterisk redondos (n=redondos@190.48.36.217) |
| 09:53.20 | qdk | Agrajag-: yes |
| 09:58.03 | motu | when i get a call from an outside caller on my fxo device (zap/3), how do i see the number number of the outside caller? |
| 09:58.56 | X-Rob | CallerID? |
| 10:01.08 | RoyK | qdk: wrong |
| 10:01.45 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 10:02.07 | RoyK | qdk, Agrajag- to make this work, you need to put exten => _9xx.... into one context and exten => _X.,..... into another, and then from your base context, include => the two in the order of processing |
| 10:02.42 | RoyK | qdk: see http://bugs.digium.com/bug_view_page.php?bug_id=4940&history=1 |
| 10:03.19 | *** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 10:07.50 | RoyK | ~seen zoa |
| 10:07.56 | jbot | zoa is currently on #asterisk (34m 6s), last said: 'mitcheloc: did you ever try idefisk ?'. |
| 10:08.11 | mitcheloc | ;) |
| 10:10.04 | qdk | RoyK: oh, a stupid bug.... will i would ha made it _9XXXXXXX vs _XXXXXXXX or _9XXXXXXX. vs _XXXXXXXX. and then im pretty sure that it works. |
| 10:10.20 | RoyK | it doesn't |
| 10:10.47 | RoyK | or, that is, with the same amount of digits, perhaps, but with _X. vs _9XXXXXXX, the _X. wins |
| 10:10.54 | RoyK | which is bloody stupid |
| 10:11.31 | RoyK | but some people seems to think having it another way is just a nice feature |
| 10:11.47 | *** join/#asterisk hgaillac (n=Harry@61.10.119-80.rev.gaoland.net) |
| 10:11.49 | qdk | RoyK: as i said... with the same amount im pretty sure that it works, or i might just have been very lucky/good ordering my extens |
| 10:11.58 | mut | best match rather than first match? |
| 10:12.06 | hgaillac | Hello, |
| 10:12.07 | mut | it's nice, thats about it th |
| 10:12.09 | mut | o |
| 10:12.53 | hgaillac | Is their somebody with a h323 terminal for testing oh323/asterisk channel ? |
| 10:13.14 | zoa | roy |
| 10:13.16 | zoa | im here |
| 10:13.21 | zoa | im about to go out to lunch |
| 10:13.27 | zoa | but i check the irc every few minutes |
| 10:14.01 | RoyK | zoa: can you and slav take a look at my audio problems, please? |
| 10:14.05 | RoyK | with the jitterbuffer |
| 10:14.10 | RoyK | we need this fixed |
| 10:14.37 | zoa | i already asked him |
| 10:14.39 | zoa | he is puzzled |
| 10:14.43 | zoa | he doesnt know where they come from |
| 10:15.01 | RoyK | then we need to dig down |
| 10:15.04 | RoyK | to find them |
| 10:15.06 | zoa | yes |
| 10:15.11 | zoa | he is not here now |
| 10:15.18 | zoa | but i will talk to him again when he gets back |
| 10:19.21 | boddy | Unable to specify channel 1: No such device or address |
| 10:19.24 | boddy | May 18 13:04:56 ERROR[8598]: Unable to open channel 1: No such device or address |
| 10:19.24 | boddy | here = 0, tmp->channel = 1, channel = 1 |
| 10:19.38 | boddy | <PROTECTED> |
| 10:19.56 | boddy | anybody help me |
| 10:23.35 | *** part/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 10:25.59 | motu | when i get a call from an outside caller on my fxo device (zap/3), callerid only shows zap/3-1, how do i see the number number of the outside caller? |
| 10:35.33 | Assid | can i include a file in sip.conf as well? |
| 10:35.59 | *** join/#asterisk Sebastian[1] (n=Sebastia@ip-5-112.sn2.eutelia.it) |
| 10:36.31 | Sebastian[1] | Hello. Sorry for the off topic. Is there any "web based" phone client ? |
| 10:37.04 | Ahrimanes | Sebastian[1]: hm there's at least one java phone |
| 10:37.18 | mitcheloc | you mean a softphone |
| 10:37.40 | Sebastian[1] | i mean a softphone but embedded in a browser, like an applet or activex |
| 10:38.06 | mitcheloc | ah probably, i wouldn't know though, if you can do activex why not a seperate phone? |
| 10:38.55 | Ahrimanes | Sebastian[1]: http://www.sip-communicator.org/ i guess if it's java it could be made an applet |
| 10:38.58 | Sebastian[1] | i want to give my site visitors the possibility to call me |
| 10:39.12 | mitcheloc | you could do a call-back feature ? |
| 10:39.21 | mitcheloc | have them type in their phone number, and your pbx rings them |
| 10:39.34 | Sebastian[1] | how can i do that? |
| 10:40.09 | Ahrimanes | mitcheloc: do you know perl? |
| 10:40.18 | mitcheloc | it'll require some programming |
| 10:40.27 | Ahrimanes | argh |
| 10:40.27 | mitcheloc | Ahrimanes: nope, php works though |
| 10:40.31 | Ahrimanes | Sebastian[1]: do you know perl |
| 10:40.33 | Sebastian[1] | can it be done in php? |
| 10:40.40 | Ahrimanes | mitcheloc: hehe, was meant for Sebastian[1] |
| 10:40.41 | mitcheloc | yes |
| 10:40.50 | Ahrimanes | Sebastian[1]: yeah, it can be done in shell if needed :P |
| 10:40.54 | Sebastian[1] | Ahrimanes: no... |
| 10:40.54 | mitcheloc | use .net though with mono ;) |
| 10:41.07 | mitcheloc | *c# |
| 10:41.09 | Sebastian[1] | thanks guys |
| 10:41.16 | mitcheloc | np |
| 10:41.20 | Ahrimanes | C#.net ? |
| 10:41.28 | mitcheloc | C# with mono yes |
| 10:41.33 | *** part/#asterisk Sebastian[1] (n=Sebastia@ip-5-112.sn2.eutelia.it) |
| 10:43.21 | boddy | Unable to register channel '1-15' |
| 10:43.30 | boddy | I still receive this error |
| 10:44.49 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 10:46.48 | kmilitzer | Does anyone know what the option dispositionstring=yes in cdr_odbc.conf is good for? |
| 10:53.52 | motu | how do i show the callerid from an outside call to zap/3? currently only showing zap/3-1 |
| 10:57.37 | *** join/#asterisk kay2 (n=key2@sd-420.dedibox.fr) |
| 11:01.01 | *** join/#asterisk zotz (n=zotz@24.231.36.155) |
| 11:01.03 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 11:01.07 | satlan32 | hi |
| 11:01.10 | satlan32 | quiestion |
| 11:02.27 | RoyK | question? |
| 11:03.22 | kaldemar | it will be a long one. |
| 11:03.37 | X-Rob | pi is long. |
| 11:04.52 | satlan32 | i'm trying to run this command from the dial plan |
| 11:05.01 | *** join/#asterisk _Paulo_ (n=Paulo@c906d4b0.virtua.com.br) |
| 11:05.05 | satlan32 | exten => 15,n,System(mutt -s "hello" -a /tmp/wav.wav moti@varicom.co.il < /tmp/inst.txt) |
| 11:05.13 | satlan32 | but it's not doing anything |
| 11:05.14 | _Paulo_ | ~seen generalhan |
| 11:05.20 | jbot | generalhan <i=general_@ip67-90-64-2.z64-90-67.customer.algx.net> was last seen on IRC in channel #asterisk, 12h 2m 24s ago, saying: 'so now i just need to figure out how to get it to work after i turn it back on ... causei really dont want to leave it disabled'. |
| 11:07.49 | kaldemar | satlan32: what's the n between 15 and System? |
| 11:08.05 | satlan32 | ??? |
| 11:08.10 | satlan32 | next |
| 11:08.24 | *** join/#asterisk viperdude (n=jon@84.45.193.6) |
| 11:08.30 | mitcheloc | kaldemar: it's new |
| 11:09.04 | kaldemar | lookie, lookie. *looks at wiki* |
| 11:09.44 | kaldemar | who came up with such a great idea? |
| 11:10.10 | mitcheloc | heh, i have no clue, but i do thank them, whenever i go look at my dialplans i might just have to update to use it =) |
| 11:14.14 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) |
| 11:14.53 | *** join/#asterisk raggi (n=ra66i@host-84-9-127-182.bulldogdsl.com) |
| 11:16.00 | raggi | i'm trying to call another asterisk server over iax, from a sip phone, but the context on teh final IAX endpoint seems to be coming through wrong, it's saying "s@default does not exist" |
| 11:16.38 | X-Rob | so don't call s@default 8) |
| 11:16.56 | raggi | yeah |
| 11:17.39 | raggi | exten=>number,1,Answer \n exten=>number,2,Dial(IAX2/user:pass@server/context) that's the right form? |
| 11:18.02 | X-Rob | IAX2/user:pass@server/xtn |
| 11:18.28 | raggi | you mean, no dial? |
| 11:18.30 | X-Rob | or xtn@context if you have 'context=*' in the remote servers type=peer context |
| 11:18.40 | raggi | ah ok |
| 11:20.00 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 11:20.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:20.56 | *** join/#asterisk keyhack (n=keyhack@68.236.93.224) |
| 11:23.47 | *** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com) |
| 11:30.18 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 11:34.23 | hgaillac | Anybody use an H323 terminal ? |
| 11:34.48 | raggi | May 18 11:34:15 ERROR[5260]: chan_sip.c:10831 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 01273808770 in context default |
| 11:34.51 | raggi | <PROTECTED> |
| 11:35.04 | raggi | can you explain what this means, i've tried adding a hint, but it's not helping |
| 11:37.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:37.20 | RoyK | you need a hint :) |
| 11:37.34 | raggi | clearly |
| 11:37.40 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 11:37.41 | raggi | as i said, i've tried adding one |
| 11:38.02 | raggi | but i get May 18 11:37:19 WARNING[5327]: pbx_config.c:1700 pbx_load_module: Invalid priority/label ' hint' at line 31 |
| 11:38.31 | raggi | oh right |
| 11:38.36 | raggi | the parser for hints is broken |
| 11:38.37 | raggi | heh |
| 11:38.45 | raggi | or at least, it can't cope wiht spaces |
| 11:44.47 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 11:48.33 | Sonderblade | how do you setup asterisk so that when you get a special error code back from the sip-provider like 504, asterisk sends a sample back to the phone? |
| 11:52.14 | kmilitzer | Are the uniqueid's in cdr_odbc really unique? So if I have two asterisk machines both logging cdrs into the same table I will never get the same uniqueid for the same call? |
| 11:53.03 | RoyK | kmilitzer: we just use a unique serial/autoincrement instead |
| 11:53.08 | RoyK | i'd recommend thatt |
| 11:55.38 | kmilitzer | RoyK: Thanks ... I guess I will do that ... |
| 11:57.42 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 11:58.24 | *** join/#asterisk slav_jb (n=k@213.16.46.130) |
| 11:58.31 | *** join/#asterisk littlejohn (n=little@host146-255.pool8289.interbusiness.it) |
| 11:59.39 | raggi | i want to crete a dialplan which dials an extension on our trunk server, if it can't reach our other asterisk server over IAX |
| 11:59.47 | raggi | at the momoent, extensions.conf looks like this: |
| 11:59.53 | raggi | exten => 01273808770,1,Answer |
| 11:59.53 | raggi | exten => 01273808770,2,Dial(${ISKI}) |
| 11:59.54 | raggi | exten => 01273808770,3,Dial(${TRUNK}/447887666248) |
| 12:00.07 | raggi | but, that seems to stop at 2, if 2 fails |
| 12:00.12 | raggi | am i missing a keyword? |
| 12:00.47 | *** join/#asterisk tuman77 (n=tuman77@210.21.227.66) |
| 12:01.19 | tuman77 | hello |
| 12:02.41 | raggi | anyone? |
| 12:03.57 | raggi | do i need to make a hunting group etc? or is it possible to keep it simple in extensions.conf |
| 12:04.23 | *** join/#asterisk tuman77 (n=tuman77@210.21.227.66) |
| 12:05.35 | tuman77 | hello |
| 12:10.15 | *** join/#asterisk jpbotelho (n=jpbotelh@201.7.108.130) |
| 12:10.21 | *** join/#asterisk Eight (n=blake@12-227-169-99.client.mchsi.com) |
| 12:13.42 | raggi | ok, is it possible to stop asterisk from giving up after "2" in this configuration: |
| 12:13.45 | raggi | exten => 01273808770,1,Answer |
| 12:13.47 | raggi | exten => 01273808770,2,Dial(${ISKI}) |
| 12:13.50 | raggi | exten => 01273808770,3,Dial(${TRUNK}/447887666248) |
| 12:14.02 | raggi | if ${ISKI} (an iax connection) is not available? |
| 12:16.18 | kaldemar | if it's not available, asterisk will most likely try to move to priority n+101. |
| 12:16.46 | raggi | so ideally i want 3 to read n+101? |
| 12:17.08 | tuman77 | NAT? |
| 12:17.43 | tuman77 | 怎么解决NAT问题？ |
| 12:18.46 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:18.48 | kaldemar | raggi: yes, 103. or you can make a goto(3) in 103. either way. |
| 12:19.05 | raggi | k, so now it looks like this: |
| 12:19.10 | raggi | exten => 01273808770,1,Answer |
| 12:19.11 | raggi | exten => 01273808770,n,Dial(${ISKI}) |
| 12:19.11 | raggi | exten => 01273808770,n+101,Dial(${TRUNK}/447734059091) |
| 12:19.15 | *** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net) |
| 12:19.50 | kaldemar | i don't know if that works, probably not. let the priorities be 1, 2 and 103. |
| 12:20.34 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:21.03 | kaldemar | usually if an application or a function fails to do what it was supposed to, it tries to move to priority n+101, where n is the current priority. |
| 12:21.32 | raggi | cool |
| 12:21.56 | raggi | as long as n increments before the +101 is resolved, that should work then |
| 12:21.57 | raggi | http://www.voip-info.org/wiki/index.php?page=Asterisk+priorities |
| 12:22.19 | viperdude | i think the n+101 is prior * v 1.2 as the wiki says you can go back to the n+101 by passing j in the dial command |
| 12:22.44 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:22.51 | viperdude | http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 12:24.56 | kaldemar | oh yes, or setting priorityjumping=yes in [general]. |
| 12:29.40 | raggi | hrm |
| 12:30.06 | raggi | Spawn extension (default, 01273808770, 2) exited non-zero on 'IAX2/217.14.132.185:4569-1' |
| 12:30.10 | raggi | and no hunt |
| 12:30.13 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 12:30.19 | raggi | with: |
| 12:30.20 | raggi | exten => 01273808770,1,Answer |
| 12:30.20 | raggi | exten => 01273808770,2,Dial(${ISKI}) |
| 12:30.20 | raggi | exten => 01273808770,103,Dial(${TRUNK}/447734059091) |
| 12:30.42 | raggi | and priorityjumping=yes |
| 12:31.30 | *** join/#asterisk FaithX (n=FaithX@202-6-145-116.ip.adam.com.au) |
| 12:32.21 | raggi | o.0 |
| 12:33.51 | *** join/#asterisk coppice (n=chatzill@99.194.17.210.dyn.pacific.net.hk) |
| 12:34.15 | zoa | hey ho coppice |
| 12:34.31 | coppice | hi |
| 12:34.41 | raggi | oops, do i need to set autofallthrough=no |
| 12:34.45 | [TK]D-Fender | Priority jumping = BAD for most things now... |
| 12:34.55 | raggi | ok, what's a better method? |
| 12:35.41 | _Paulo_ | hi coppice. |
| 12:35.47 | [TK]D-Fender | raggi : For your sample 2 will continue on to 3 automatically if the call isn't answered. |
| 12:36.26 | [TK]D-Fender | raggi : unless told otherwise at the end of Dial exectuation ends at the end of the call. |
| 12:36.41 | [TK]D-Fender | raggi : So your next step # is 3 |
| 12:36.45 | raggi | it doesn't seem to have that behaviour |
| 12:36.51 | raggi | 1 sec |
| 12:37.04 | [TK]D-Fender | raggi : Turn or global priority jumping. |
| 12:37.04 | raggi | i'll post whole config |
| 12:37.06 | tzanger | heh |
| 12:37.07 | tzanger | Genesis 1:32 And God typed :wq! |
| 12:37.46 | coppice | a divine typo? |
| 12:37.55 | raggi | [general] |
| 12:37.55 | raggi | static=yes |
| 12:37.55 | raggi | writeprotect=no |
| 12:37.55 | raggi | autofallthrough=no |
| 12:37.55 | raggi | clearglobalvars=no |
| 12:37.57 | raggi | priorityjumping=yes |
| 12:38.00 | raggi | correct? |
| 12:38.20 | MrChimpy | which is the best CDR driver to use for a MSSQL DB? |
| 12:38.56 | [TK]D-Fender | raggi : I just told you to turn priority jumping off. |
| 12:38.57 | MrChimpy | i can used TDS or ODBC |
| 12:39.04 | MrChimpy | -d |
| 12:39.17 | raggi | sorry, you said 'or', thx though :) |
| 12:39.27 | r_evolution | TK : sup sucka |
| 12:39.32 | raggi | autofallthrough=off is correct? |
| 12:39.58 | [TK]D-Fender | raggi : Good idea. Its a useless option IMO and only served to create a dum Waitexten command |
| 12:40.03 | [TK]D-Fender | r_evolution : y0 |
| 12:40.10 | raggi | ah right, ok |
| 12:40.27 | raggi | and does this want to be: |
| 12:40.28 | raggi | exten => 01273808770,2,Dial(${ISKI}) |
| 12:40.29 | raggi | exten => 01273808770,103,Dial(${TRUNK}/447734059091) |
| 12:40.38 | raggi | or xxx,3,Dial... |
| 12:41.16 | raggi | sorry for posting so much, i'm just trying to get it right this time :) |
| 12:41.26 | r_evolution | ~pb |
| 12:41.33 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 12:41.33 | r_evolution | ewps. |
| 12:41.36 | r_evolution | :) |
| 12:41.37 | [TK]D-Fender | raggi : change 103 for 3 |
| 12:41.45 | r_evolution | jbot is slow today :( |
| 12:41.55 | [TK]D-Fender | raggi : I take it you want to dial a second trunk if the first one fails right? |
| 12:42.11 | raggi | yup |
| 12:42.22 | [TK]D-Fender | raggi : just change the 103 for 3 then |
| 12:42.24 | *** join/#asterisk rkr245 (n=root@81.21.33.35) |
| 12:42.58 | raggi | cool, and this error would be normal for my old configuration: Spawn extension (default, 01273808770, 2) exited non-zero on 'IAX2/217.14.132.185:4569-1' |
| 12:43.03 | raggi | testing new config.. |
| 12:43.23 | [TK]D-Fender | raggi : the default behavior for Dial is to continue on after setting the ${DIALSTATUS} variable with the reason for the failure. You don't actually CARE what the reason is, you just know that you want to try "plan B" afterwards... |
| 12:43.40 | [TK]D-Fender | raggi : indeed it would |
| 12:43.50 | *** part/#asterisk gonzo- (n=gonzo@www.pbxpress.com) |
| 12:43.57 | raggi | good good, dialing... :) |
| 12:44.29 | raggi | shit |
| 12:44.44 | raggi | same error... Spawn extension (default, 01273808770, 2) exited non-zero on 'IAX2/217.14.132.185:4569-1' |
| 12:47.03 | [TK]D-Fender | raggi : Where does that dial go to? |
| 12:47.18 | raggi | that dial goes to my second asterisk server |
| 12:47.28 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) |
| 12:47.42 | raggi | we hve one in the DC, which recieves calls from voiptalk - that 217.14.132.185 |
| 12:48.09 | raggi | that routes over IAX to our second Asterisk at our main office primarily - that's 2 |
| 12:48.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:48.28 | raggi | then if that fails, we want to hunt mobiles, which is 3, 4, .. etc priorities |
| 12:48.47 | raggi | the mobile hunting goes over iax back to voiptalk and dials a number |
| 12:49.24 | puzzled | hi |
| 12:50.56 | raggi | so , the globals look like this: TRUNK=IAX2/user:pass@iax.voiptalk.org |
| 12:50.56 | raggi | ISKI=IAX2/user:pass@217.22.90.1 |
| 12:51.29 | raggi | it would appear though, that it's still not getting to priority 3 |
| 12:51.49 | raggi | autofallthrough and priorityjumping are both no |
| 12:54.26 | satlan32 | hi |
| 12:54.27 | [TK]D-Fender | raggi : pastebin the call attempt |
| 12:54.29 | satlan32 | need help pleased |
| 12:54.58 | satlan32 | i'm trying to run exten => 15,n,System(email.sh ${EXTEN} ${file}) |
| 12:55.31 | satlan32 | but i get May 18 15:52:29 VERBOSE[2774] logger.c: -- Executing System("SIP/504-a240", "email.sh 15 /tmp/wav.wav") in new stack |
| 12:55.32 | satlan32 | May 18 15:52:29 WARNING[2774] app_system.c: Unable to execute 'email.sh 15 /tmp/wav.wav' |
| 12:55.45 | raggi | [TK]D-Fender: what verbosity level? 4? |
| 12:55.51 | satlan32 | but when i run this command from the CLI : |
| 12:55.51 | [TK]D-Fender | raggi : 10 |
| 12:55.59 | satlan32 | !email.sh 15 /tmp/wav.wav |
| 12:56.04 | satlan32 | it's ok. |
| 12:56.11 | [TK]D-Fender | satlan32 : use a full path to your script |
| 12:56.18 | raggi | k, just gotta wait for calls to tear down on pbx2 |
| 12:56.30 | satlan32 | the path is configured on staretup |
| 12:56.43 | [TK]D-Fender | satlan32 : No, put the FULL PATH in the System call line |
| 12:56.48 | satlan32 | ok |
| 12:56.49 | raggi | actually, i can just break the ip |
| 12:56.51 | satlan32 | let me try |
| 12:58.02 | *** join/#asterisk docelmo (n=docelmo@66.237.242.41.ptr.us.xo.net) |
| 13:00.48 | satlan32 | mmm maybe asterisk user cannot run this script? |
| 13:01.01 | satlan32 | how do i give permissions to everyone? |
| 13:02.11 | raggi | [TK]D-Fender: http://pastebin.com/724491 |
| 13:02.22 | [TK]D-Fender | satlan32 : chown 777 myfile.whatever |
| 13:02.26 | raggi | just does a hangup after the fial |
| 13:02.27 | satlan32 | ok |
| 13:03.28 | [TK]D-Fender | raggi : pastebin your dial-plan entry for it as well. If the other side ANSWERS and hangup up onits side then it won't come bak, just FYI |
| 13:03.40 | satlan32 | logger.c: -- Executing System("SIP/504-7f81", "/root/scripts/email.sh 15 /tmp/wav.wav") in new stack |
| 13:03.46 | satlan32 | but still no email |
| 13:03.47 | [TK]D-Fender | satlan32 oops, thats CHMOD, not CHOWN |
| 13:03.57 | satlan32 | i know |
| 13:04.01 | satlan32 | :) |
| 13:05.12 | raggi | [TK]D-Fender: well, the other side has closed the socket |
| 13:05.21 | raggi | which is the scenario i'm trying to test |
| 13:05.29 | *** join/#asterisk MattH (n=MattH@63.174.244.195) |
| 13:05.29 | satlan32 | the this is when i run it from the CLI: !email.sh 15 /tmp/wav.wav |
| 13:05.37 | raggi | we want it to hunt if IAX on pbx2 cannot be reached |
| 13:05.45 | MattH | Hi.. is there some command I can send from a dialplan to turn on a softkey light on a sip phone? ie an aastra 9133i? |
| 13:05.55 | Katty | morning |
| 13:06.47 | [TK]D-Fender | raggi : Oh, ok, then that should work :) |
| 13:06.59 | raggi | [TK]D-Fender: well, it's not :( |
| 13:07.03 | raggi | any other ideas? |
| 13:07.30 | [TK]D-Fender | raggi : ou sure the other side isn't receiving the call and treating it as answered? |
| 13:07.37 | [TK]D-Fender | Katty: mew. |
| 13:08.06 | satlan32 | what is the difference betweek system() and running from the CLI with !? |
| 13:08.07 | satlan32 | ? |
| 13:08.09 | raggi | [TK]D-Fender: yup, very |
| 13:08.22 | jake1932 | the path |
| 13:08.22 | raggi | the last test i did, i couldn't shutdown pbx2 cause it was in use |
| 13:08.32 | raggi | so i changed the ip to a null location instead |
| 13:08.39 | raggi | so there was definately no IAX at the other end |
| 13:09.02 | [TK]D-Fender | satlan32 : What user are you running * as? |
| 13:09.21 | satlan32 | the default of A@H 2.8 |
| 13:09.24 | jake1932 | satlan32: looks like you put an absolute path in the System cmd - not sure if that's making a difference |
| 13:09.33 | [TK]D-Fender | satlan32 : That isn't an answer. |
| 13:09.43 | satlan32 | how do i check which user? |
| 13:10.05 | raggi | [TK]D-Fender: could this be todo with hinting? i'm a little unclear as to what hinting is all about? |
| 13:10.21 | [TK]D-Fender | raggi : Nothing to do with hints, thats for presence |
| 13:10.44 | raggi | i should have a hint pointing to pbx2 right? |
| 13:11.03 | [TK]D-Fender | raggi : no importantce |
| 13:11.08 | raggi | ok, good |
| 13:11.11 | satlan32 | [TK]D-Fender |
| 13:11.18 | satlan32 | user asrerisk |
| 13:11.22 | jake1932 | satlan32: also you might want to add sh |
| 13:11.32 | jake1932 | sh cmd.sh |
| 13:11.37 | satlan32 | in the system()? |
| 13:11.38 | raggi | so... any way i can find out why it's hanging up? |
| 13:11.39 | jake1932 | yes |
| 13:11.40 | satlan32 | i can try |
| 13:11.41 | satlan32 | sec |
| 13:11.45 | [TK]D-Fender | satlan32 : And in what world do you think that user "asterisk" has access to the contents of the ROOT folder? |
| 13:11.56 | raggi | or verify the dialplan logic? |
| 13:12.02 | satlan32 | i gave the email.sh permissin 777 |
| 13:12.20 | [TK]D-Fender | satlan32 : the whole FOLDER is inaccessable to "asterisk" |
| 13:12.29 | [TK]D-Fender | satlan32 : the file doesn't even matter! |
| 13:12.37 | jake1932 | (unless "asterisk" has root permissions |
| 13:12.40 | satlan32 | mmm ok... that might explain... i'll try |
| 13:12.42 | satlan32 | :) |
| 13:12.52 | [TK]D-Fender | satlan32 : Do NOT mess with that folder's permissions. |
| 13:13.00 | [TK]D-Fender | satlan32 : Move your script somewhere SANE. |
| 13:13.05 | satlan32 | i know |
| 13:13.17 | satlan32 | i'll put it in the tmp |
| 13:16.07 | satlan32 | mmm still nothing. |
| 13:16.24 | satlan32 | in the log i can see it's running but no email. |
| 13:16.33 | jake1932 | did you add sh? |
| 13:16.36 | satlan32 | May 18 16:15:11 DEBUG[3189] pbx.c: Launching 'System' |
| 13:16.36 | satlan32 | May 18 16:15:11 VERBOSE[3189] logger.c: -- Executing System("SIP/504-468c", "/tmp/email.sh 15 /tmp/wav.wav") in new stack |
| 13:16.54 | jake1932 | (ok - that's a no) :) |
| 13:16.55 | satlan32 | i did but in that case it gave me unable to execute ... |
| 13:17.33 | satlan32 | May 18 16:17:25 VERBOSE[3213] logger.c: -- Executing System("SIP/504-abfe", "sh /tmp/email.sh 15 /tmp/wav.wav") in new stack |
| 13:17.34 | satlan32 | May 18 16:17:25 WARNING[3213] app_system.c: Unable to execute 'sh /tmp/email.sh 15 /tmp/wav.wav' |
| 13:18.05 | jake1932 | <PROTECTED> |
| 13:18.40 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 13:18.50 | satlan32 | the thing that drive me crazy is that "!email.sh 15 /tmp/wav.wav" from asterisk CLI is working |
| 13:19.11 | satlan32 | what is the different between the two methodes? |
| 13:19.31 | raggi | bbs |
| 13:19.52 | jake1932 | i have it working (not @home) - System(sh cmd args) |
| 13:20.02 | [TK]D-Fender | satlan32 : Just because you don't this e-mail you're expecting doesn't imply that its not running. Maybe the script isn't doing its JOB. |
| 13:20.03 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 13:20.18 | *** join/#asterisk kevinfcn_ (n=kevinfcn@c-68-39-64-129.hsd1.nj.comcast.net) |
| 13:20.26 | [TK]D-Fender | satlan32 : What user are you running the script as at CLI? Would "asterisk" have the same priveledges? |
| 13:20.52 | satlan32 | how do i know? |
| 13:21.01 | [TK]D-Fender | satlan32: Its your script, you tell ME. |
| 13:21.21 | satlan32 | i just type !email.sh 15 /tmp/wav.wav |
| 13:21.29 | [TK]D-Fender | satlan32 : As what user? |
| 13:21.37 | satlan32 | i don't know |
| 13:21.58 | [TK]D-Fender | satlan32 : if you don't know what user you are logged in as at a linux console you shouldn't even be here... |
| 13:22.00 | satlan32 | asterisk |
| 13:22.34 | satlan32 | oh logged in as root |
| 13:22.57 | [TK]D-Fender | satlan32 : Well who says that "asterisk would have the priveledges to do what that script does? |
| 13:23.27 | brettnem | #linux |
| 13:23.29 | satlan32 | dunno |
| 13:23.32 | snitt | May 18 15:23:24 NOTICE[8916]: chan_sip.c:5289 sip_reg_timeout: -- Registration for '710579@sip.neophonex.hu' timed out, trying again (Attempt #633) |
| 13:23.33 | FaithX | I having very interesting garbled sound during calls... fine then 2min or so in one direction then fine then 2min or so in the other direction and so on. Jitter is constantly low (about .7ms) |
| 13:23.34 | satlan32 | not me |
| 13:23.36 | snitt | oops |
| 13:23.44 | snitt | not here |
| 13:23.50 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 13:24.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:24.16 | *** join/#asterisk eds0n (n=edson@c-3969e353.03-53-6f736c2.cust.bredband.no) |
| 13:24.31 | [TK]D-Fender | satlan32 : Ok, sounds like you're not equiped to be trying these thigs you are doing then. |
| 13:24.59 | MrChimpy | where can I find a recent schema for CDRs? |
| 13:25.15 | eds0n | hey, is there a set of codecs in asterisk that are "most used" ? I know it supports a bunch, but are they all widely used? |
| 13:25.40 | MrChimpy | trying to set up CDR logging to a MSSQL DB, and the schema I have is lacking... |
| 13:25.42 | [TK]D-Fender | eds0n : Popular ones are ULAW, ALAW, GSM, and G729. |
| 13:25.45 | satlan32 | i'm trying to understand the different between running the system() command, and running the CLI command. as far as i understand it's the same proccess, same user, and should be the same function |
| 13:26.06 | satlan32 | am i right? |
| 13:26.07 | [TK]D-Fender | satlan32 : its NOT the same user. At CLI you are ROOT <- |
| 13:26.20 | satlan32 | at asterisk CLI i'm running as root |
| 13:26.20 | satlan32 | ? |
| 13:26.26 | [TK]D-Fender | satlan32 : and Asterisk is running as "asterisk" |
| 13:26.37 | [TK]D-Fender | satlan32 : So its NOT the same user |
| 13:26.48 | satlan32 | asterisk1*CLI> is running as root? |
| 13:27.01 | mut | anyone heard of tekelec telcom switches? |
| 13:27.13 | mut | http://www.tekelec.com/solutions/s_detail.asp?id=79 |
| 13:27.16 | [TK]D-Fender | satlan32 : You don't even seem to know what user you are. You definately need to learn Linux first. |
| 13:27.42 | raggi | "whoami" |
| 13:27.43 | eds0n | [TK]D-Fender: so, if I'm going to run tests, I should focus most on ulaw, alaw, gsm and g729 ? |
| 13:28.39 | *** join/#asterisk magaf (n=Heaven@acxb137.neoplus.adsl.tpnet.pl) |
| 13:28.41 | magaf | hello |
| 13:28.46 | [TK]D-Fender | eds0n : If your phone is local to your * box you should always be using ULAW in North America, or ALAW just about everywhere else |
| 13:28.49 | magaf | wha ts igoing on with asteriskguru.com ?? |
| 13:28.56 | magaf | domains seems not to work |
| 13:28.59 | magaf | domain |
| 13:29.07 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.220.Dial1.SanJose1.Level3.net) |
| 13:29.51 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 13:30.07 | websae | maybe they need to renew it |
| 13:30.21 | raggi | no, their nameservers are down |
| 13:30.22 | magaf | no , expire date is set to 2008y |
| 13:30.26 | flujan | Hi guys, I'm configuring a TE406P |
| 13:30.39 | raggi | ns1 and ns2.asteriskguru.com are both down |
| 13:30.45 | magaf | i see |
| 13:30.49 | magaf | any mirror? |
| 13:30.58 | raggi | http://www.dnsstuff.com/tools/lookup.ch?name=asteriskguru.com&type=A |
| 13:31.00 | raggi | dunno. |
| 13:31.04 | flujan | I want to now how to configure the ports... for instance, does the por 1 of the card reference to the span=1 definition? |
| 13:31.04 | magaf | ok |
| 13:31.16 | raggi | you could use archive.org or google cache |
| 13:31.40 | [TK]D-Fender | flujan: if its the only card in your system, it should |
| 13:31.59 | magaf | raggi i need it : http://asteriskguru.com/tutorials/chan_misdn.html |
| 13:32.02 | flujan | ok... |
| 13:32.17 | magaf | there is a faw how to recompile kernel to use chan_misdn |
| 13:32.22 | flujan | is this correct span 0 = port 1 of the card, span 2 = port 3 and so on? |
| 13:33.40 | raggi | magaf: archive.org doesn't have it unfortuantely |
| 13:33.58 | magaf | raggi google seems to have it |
| 13:34.04 | magaf | http://66.102.9.104/search?q=cache:PCb5-lzLhpEJ:www.asteriskguru.com/tutorials/chan_misdn.html+&hl=pl&lr=&strip=1 |
| 13:34.04 | raggi | good |
| 13:34.07 | magaf | google kick ass :) |
| 13:34.14 | satlan32 | i'm looking for help on how to run the script as root. |
| 13:34.17 | [TK]D-Fender | flujan : No, I just confirmed your question that port 1 = SPpan 1. |
| 13:34.29 | satlan32 | can you help? or point me to a good tuturial? |
| 13:34.34 | raggi | satlan32: su root -c 'command' |
| 13:34.57 | raggi | man su first, and read |
| 13:36.03 | satlan32 | thanks |
| 13:36.24 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 13:36.26 | raggi | [TK]D-Fender: do you have a working configuration that falls through priorities in a dialplan somewhere, so i can compare, as somethings not right here. |
| 13:37.00 | Sonderblade | is there a way to write dialplans in a proper programming language? using asterisk's builtin dialplan language is annoying |
| 13:37.04 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |
| 13:37.35 | raggi | Sonderblade: define proper programming language |
| 13:37.52 | Makenshi | Sonderblade, yes, you can use agi and just specify _. in your dp |
| 13:37.52 | *** join/#asterisk shiznatix (n=shiznati@213-35-236-104-dsl.end.estpak.ee) |
| 13:37.59 | flujan | [TK]D-Fender, OK... Thanks for the reply... And another question, I search the meaning of span for asterisk . Can it be considered as a network tap? Which is the best meaning of the word span for asterisk? |
| 13:38.04 | r_evolution | you could always just use AEL Sonder :) |
| 13:38.15 | Sonderblade | r_evolution: whats that? |
| 13:38.28 | satlan32 | raggi: the script is running... |
| 13:38.38 | r_evolution | http://www.voip-info.org/wiki/view/Asterisk+AEL |
| 13:38.40 | satlan32 | but it seems that it's not finishing.. |
| 13:38.43 | r_evolution | ^ |
| 13:38.43 | Sonderblade | Makenshi: ah, so from agi you can call all asterisks builtin functions? |
| 13:38.49 | satlan32 | and the call is still connected |
| 13:38.57 | Makenshi | Sonderblade, yip |
| 13:39.03 | r_evolution | proper enough for you? |
| 13:39.06 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 13:39.07 | raggi | satlan32: if it seems to not finish, debug the script outside of asterisk / a call firts |
| 13:39.36 | magaf | does anyone have an experience with asterisk + ISDN cards? |
| 13:39.42 | satlan32 | raggi: i did it outside of asterisk and it works just fine |
| 13:39.49 | Sonderblade | Makenshi: cause it seems like for large dialplans, writing it all in extensions.conf isn't very nice, right? |
| 13:39.52 | raggi | [TK]D-Fender: know you're getting busy mate, but .. do you have a working configuration that falls through priorities in a dialplan somewhere, so i can compare, as somethings not right here. |
| 13:40.19 | raggi | satlan32: is it recieving the same context, arguments etc? |
| 13:40.33 | satlan32 | yes |
| 13:40.46 | Makenshi | Sonderblade, right |
| 13:40.57 | Makenshi | Sonderblade, another alternative is to use the realtime extension |
| 13:41.11 | raggi | satlan32: clearly that's not true though, otherwise it would be doing the same thing! ;) |
| 13:42.05 | satlan32 | i'm getting an email when the script completes. |
| 13:42.30 | Sonderblade | Makenshi: so there are atleast four alternatives... extensions.conf, AEL, agi and the realtime extension... |
| 13:42.46 | satlan32 | and when i run it like this [root@asterisk1 tmp]# email.sh 15 /tmp/wav.wav it's working |
| 13:43.15 | r_evolution | personally i get along best with the realtime extensions sonder... |
| 13:43.16 | satlan32 | asterisk dose this: Executing System("SIP/504-ce3d", "su root -c email.sh 15 /tmp/wav.wav") in new stack |
| 13:43.36 | r_evolution | it makes my life a little easier when users want forwarding... or call-hunting... or voicemail turned off/on... or little things like that |
| 13:44.00 | r_evolution | no need to go edit the ext.conf and reload it... just edit the db. ahh :) |
| 13:44.27 | many | uh |
| 13:44.30 | many | ohm. |
| 13:44.39 | many | someone beat some clue into satlan. |
| 13:45.07 | satlan32 | ????? |
| 13:45.36 | satlan32 | ???? |
| 13:45.53 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:45.57 | many | it DOES NOT make any sense to send tehe mail as root. |
| 13:46.14 | many | it DOES NOT make sense to use su root in scripting |
| 13:46.25 | zoa | guys |
| 13:46.29 | zoa | i need some help |
| 13:46.36 | zoa | i need another name for our softphone |
| 13:46.42 | zoa | help me help me |
| 13:47.00 | jake1932 | mr softy |
| 13:47.01 | *** join/#asterisk Ironhand (i=arjen@meek.xs4all.nl) |
| 13:47.08 | websae | lol |
| 13:47.09 | many | zoa: softice |
| 13:47.11 | satlan32 | it really matter how i send the mail? is it really matter hiw i run the script? i'm using it in a testing system not commercial and not on the internet, so i don't really care about root |
| 13:47.26 | r_evolution | name it |
| 13:47.27 | r_evolution | ZOA! |
| 13:47.31 | Makenshi | satlan32, it's not really a realistic test though, is it? |
| 13:47.36 | satlan32 | as far as i care, let everything run as root |
| 13:47.45 | C4T3l | wha? |
| 13:48.00 | r_evolution | easy killa. |
| 13:48.03 | sevard | lemme attem LEMME ATTEM |
| 13:48.16 | brettnem | how about zoaphone |
| 13:48.31 | Makenshi | Super Happy Fun Phone |
| 13:48.38 | r_evolution | eh. poor security practices bleed over sometimes :( |
| 13:48.44 | sevard | Makenshi: Super Happy Fun Phone, GO! |
| 13:48.46 | satlan32 | makenshi, this system will finaly be a part of an internal system. no internet connection will be whatso ever |
| 13:48.48 | puzzled | anyone know if it's normal for a 7960 with 7.4 sip firmware to show a big Cisco logo in the screen when booted? |
| 13:49.14 | jake1932 | mine still has the logo when finished booting |
| 13:49.19 | many | satlan: i dont care wether you care about root. |
| 13:49.28 | many | satlan32: i say that what you do does not make sense at all. |
| 13:49.30 | sevard | satlan32: what happens when one of your scripts running as root accidentally 'rm -rf /' |
| 13:49.33 | sevard | can you say |
| 13:49.34 | sevard | "oops" |
| 13:49.47 | puzzled | jake1932: ok thanks |
| 13:49.49 | flujan | how many channels can I configure in a E1 span? 32 ? |
| 13:50.00 | satlan32 | i have only this script for now, |
| 13:50.04 | jake1932 | with no d chan |
| 13:50.07 | r_evolution | hey sevard... when they happens... |
| 13:50.08 | many | 30, other two should be D-Chan |
| 13:50.14 | flujan | the asterisk book references a E1 as a DS-1 with 32 channels... |
| 13:50.14 | *** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl) |
| 13:50.16 | satlan32 | as i go on i will pay more attension |
| 13:50.18 | r_evolution | satlan will be away for a while :) |
| 13:50.23 | sevard | :) |
| 13:50.29 | satlan32 | :) |
| 13:50.29 | many | satlan: okay. let me put it straight. |
| 13:50.34 | Assid | chan_zap.so]May 18 19:18:44 WARNING[13631]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: pri_restart |
| 13:50.41 | brettnem | flujan: that's messed up |
| 13:50.58 | many | satlan32: you have no clue how unix works. none. zip. zero. youre trying to work on a car you dont know how to use. |
| 13:51.00 | websae | anyone need sip phones/atas? |
| 13:51.06 | *** join/#asterisk Matt (i=matt@freenode/staff/matt) |
| 13:51.08 | r_evolution | i swear... it's easy to make a nemesis in this channel |
| 13:51.12 | Matt | morning |
| 13:51.20 | Makenshi | websae, i use sip phones. |
| 13:51.24 | r_evolution | the VoicePulse people made me want to choke on my own vomit the other day :) |
| 13:51.26 | [TK]D-Fender | many : He's a complete linux newb running A@H. |
| 13:51.33 | r_evolution | now sevard is trying to strangle satlan |
| 13:51.39 | many | satlan32: and to put it even more straight, its frustrating for people to help other people who have no clue. You shall learn the basics before doing such stuff. really. |
| 13:51.39 | satlan32 | listen guys, i'm new to linux, and learning along. ask me about windows, i'll know, ask me PRI SS7 etc.. i know |
| 13:51.42 | many | fender :) |
| 13:51.49 | satlan32 | linux is new to me, and i'm trying my best |
| 13:51.50 | flujan | brettnem, what is messed up? The number of channels? The description? |
| 13:51.52 | flujan | brettnem, :) |
| 13:51.57 | satlan32 | so cut the ... |
| 13:51.57 | many | satlan: good. |
| 13:51.58 | r_evolution | it's ok satlan :) i still have to use wind0ze at work |
| 13:52.00 | brettnem | flujan the description |
| 13:52.15 | Makenshi | websae, please don't spam |
| 13:52.15 | r_evolution | office policies etc. |
| 13:52.19 | many | satlan32: now that weve cleared that, you dont care either which user it mails as, as long as it works? |
| 13:52.19 | sevard | satlan32: don't be retarded. learn the right way now so you aren't retarded with a live system |
| 13:52.46 | flujan | brettnem, how many channels then ? I have in a E1 link and I can configure up to 96 channels whithout ztcfg -v warnings or errors... |
| 13:52.47 | r_evolution | rule number 1 - *nix is not windows. It's not socially acceptable to have 100 gaping security flaws :) |
| 13:52.54 | many | satlan32: or atleast, when you want to "su" in a script, use sudo instead. |
| 13:52.56 | r_evolution | and not care bout them :) |
| 13:52.56 | sevard | haha |
| 13:53.04 | Makenshi | r_evolution, though windows is now posix compliant :o) |
| 13:53.07 | satlan32 | ok. |
| 13:53.09 | brettnem | flujan: I'm pretty sure it's 30 and no more |
| 13:53.12 | many | flujan: builtin channelcompression |
| 13:53.13 | r_evolution | O_o |
| 13:53.16 | sevard | Makenshi: say what? |
| 13:53.19 | r_evolution | :-D sevard |
| 13:53.22 | satlan32 | so how should i run this script, not using root, from the system()? |
| 13:53.32 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:53.38 | many | but e1 are 30 voice channels, 2 data channels |
| 13:53.38 | r_evolution | windows is thrown out my window flying till hitting the ground compliant too! |
| 13:53.45 | sevard | hahaha |
| 13:53.50 | many | satlan: test one simple thing. |
| 13:53.58 | Makenshi | we're getting rid of a lot of our linux infrastructure and replacing it with windows over the summer |
| 13:54.00 | many | satlan32: do what you want to system() from a console. |
| 13:54.05 | many | and check what happens. |
| 13:54.07 | r_evolution | I'm sorry Makenshi :( |
| 13:54.10 | many | look at your logs |
| 13:54.13 | satlan32 | ok |
| 13:54.14 | brettnem | many: that sounds like euroISDN not E1 |
| 13:54.21 | satlan32 | debug is 10 verbose is 10 |
| 13:54.21 | r_evolution | we have to use winbl0ws here... people have enough trouble moving a mouse |
| 13:54.27 | Makenshi | r_evolution, it's ok, there are a lot of things that are a heck of a lot easier to do in windows |
| 13:54.34 | r_evolution | like get a virus! |
| 13:54.37 | r_evolution | ;) |
| 13:54.40 | flujan | brettnem, But asterisk must emerge a error when i try to configure more, mustn't it? |
| 13:54.42 | Makenshi | the main thing is certificate autoenrollment |
| 13:54.46 | many | brettnem: mh. mh. gotme |
| 13:54.47 | Makenshi | try doing that on anything but AD |
| 13:54.49 | sevard | Makenshi: edit 8,000 text files for me in windows. |
| 13:54.58 | brettnem | flujan: asterisk is full of problems.. add that to the list |
| 13:54.59 | shiznatix | if i have registered a external sip account in my [general] thing how do I dial with that? like I want to use that to dial a number but i cant figure out how |
| 13:55.03 | Makenshi | sevard, can do it in perl or monad |
| 13:55.08 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 13:55.15 | Makenshi | monad is a lot like perl anyway |
| 13:55.31 | r_evolution | you know... i caught the second half of that at first makenshi |
| 13:55.32 | flujan | brettnem, thanks for the reply... |
| 13:55.34 | flujan | i found this: |
| 13:55.36 | r_evolution | and i thought you said perl or gonad |
| 13:55.37 | many | brettnem: youre right. damn. :-) |
| 13:55.42 | flujan | E1 has 32 channels. Channel 0 is used for framing. Channel 16 is often used for signaling. Channels 1-15 and 17-31 are used to carry voice. |
| 13:55.45 | r_evolution | i was like who names a language after balls? |
| 13:55.45 | *** join/#asterisk vooduhal (n=cmcbee@64.18.101.53) |
| 13:55.47 | Makenshi | sevard, we're not completely ditching linux, still going to use it for dhcp and dns |
| 13:55.59 | brettnem | channel 0 is used for framing?! |
| 13:56.02 | brettnem | wtf |
| 13:56.09 | websae | time to feed the cows here in Wisconsin -- have a great day all |
| 13:56.19 | brettnem | what the hell is channel 0? |
| 13:56.20 | satlan32 | Many: i don't see anything in the logs |
| 13:56.29 | many | satlan: your MTA should log something |
| 13:56.34 | r_evolution | yer mom is a channel 0 :-\ |
| 13:56.37 | Makenshi | it's just not feasible to manage 3000 clients with the requirements placed upon us (legal and organizational) with linux |
| 13:56.41 | iDunno | assume 0 based numbering :p |
| 13:56.45 | littleball | hello, my xlite can connect to sip server (i have one account in the sip provider). Now, i run my own asterisk server. I want to connect my own asterisk box to SIP provider. How to config? |
| 13:56.46 | many | and i bet its either an unresolving hostname or relaying denied from the next mta. |
| 13:56.49 | satlan32 | Many: about the !email.sh 15 /tmp/wav.wav |
| 13:56.54 | brettnem | I believe the DS-1 equivalant in euro standards is called a "VC-12" |
| 13:56.56 | iDunno | so channel 0 is the first channel. |
| 13:57.14 | sevard | Makenshi: I'm not angry I'm just curious as to why, could you break it down furthure? |
| 13:57.17 | brettnem | r_evolution: your mamma's a channel 0! |
| 13:57.19 | vooduhal | Hello all. Question about app_queue and persistence. I have persistentagents=yes for a start. When ever I asterisk -rx 'reload', all agents who were logged in but paused get put back into a ready state. Is there anyway to resolve this? |
| 13:57.19 | r_evolution | What are you guys doing Makenshi? |
| 13:57.22 | jake1932 | littleball: configure a peer entry in sip.conf, point your sip phone, and reload |
| 13:57.23 | r_evolution | haha exactly! |
| 13:57.30 | r_evolution | im in teh same boat as sevard |
| 13:57.30 | brettnem | channel 0 is a bunch of baloney |
| 13:57.38 | sevard | teh lulz |
| 13:57.39 | many | i give up, i once saw a stm1 set up |
| 13:57.40 | r_evolution | im just giving you hell man :) don't take it personally/seriously |
| 13:57.52 | flujan | so, this configuration is correct? |
| 13:57.53 | littleball | jake1932, "point your sip phone"? |
| 13:58.01 | jake1932 | littleball: to asterisk |
| 13:58.09 | flujan | <PROTECTED> |
| 13:58.09 | flujan | <PROTECTED> |
| 13:58.09 | flujan | <PROTECTED> |
| 13:58.12 | Assid | errr.. anyone know whats up witht his: May 18 19:27:12 WARNING[13685]: loader.c:554 load_modules: Loading module chan_zap.so failed! |
| 13:58.12 | flujan | to a E1 link? |
| 13:58.12 | Assid | Ouch ... error while writing audio data: : Broken pipe |
| 13:58.16 | r_evolution | if i meant for it to be personal -- I'd probably start my sentences with "now THIS idiot" |
| 13:58.20 | r_evolution | Hey assid... check your PMs |
| 13:58.23 | Makenshi | my employer is an education provider (gotta love all the pc crap they spew [pc as in politically correct]) |
| 13:58.38 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 13:58.39 | vooduhal | Anyone have any ideas? |
| 13:58.41 | many | flujan: i bet that nobody here can say for sure :) |
| 13:58.42 | brettnem | flujan: well you are missing a channel, I think.. |
| 13:58.47 | littleball | jake1932, then how my asterisk connect to sip provider? i mean how my asterisk authenticate itself to the sip provider? |
| 13:58.51 | r_evolution | <---politically very INcorrect |
| 13:58.53 | brettnem | flujan: I'm not as fimilar with the euro standards |
| 13:59.02 | many | brettnem: mh? as you said, 0 is framing and 16 is signalling :) |
| 13:59.07 | many | 0-31 = 32 channels |
| 13:59.16 | brettnem | many: no, 0 is baloney |
| 13:59.17 | jake1932 | littleball: by using " register " in sip.conf |
| 13:59.28 | brettnem | many: channel 0 doesn't exist.. |
| 13:59.46 | many | i didnt see any other working config til yet. |
| 13:59.53 | raggi | Can someone help me wiht my dialplan, i'm trying to setup a failver configuration, but my Dial to an IAX server seems to generate a Hangup when the server cannot be reached, detail is here: http://pastebin.com/724574 |
| 13:59.53 | many | but, well, as i said, i only once.. and so on. |
| 13:59.54 | Assid | err.. anyone know whats wrong iwith that |
| 14:00.15 | brettnem | visdn? |
| 14:00.39 | Assid | do i need the chan_zap ? |
| 14:01.38 | shiznatix | if i have registered a external sip account in my [general] thing how do I dial with that? like I want to use that to dial a number but i cant figure out how |
| 14:01.40 | brettnem | hmmm. maybe channel 0 does exist. |
| 14:02.16 | many | Yes. visdn standard 2B:1D euro ISDN. |
| 14:02.27 | jake1932 | shiznatix: you have an account with VOIP termination? |
| 14:02.28 | many | or rather, any HFC Chipset. |
| 14:02.46 | shiznatix | jake1932, yes |
| 14:02.58 | jake1932 | shiznatix: they don't have docs to tell you how? |
| 14:03.04 | brettnem | many: heh.. euro bri |
| 14:03.09 | shiznatix | jake1932, no, I can't find anything |
| 14:03.10 | jake1932 | (i always got an e-mail) |
| 14:03.17 | jake1932 | who are you with? |
| 14:03.26 | many | brettnem: BRI/PRI. |
| 14:03.36 | brettnem | flujan: ok, I might be wrong.. I definately am the wrong one to ask euro standards questions to http://en.wikipedia.org/wiki/E-carrier |
| 14:03.45 | brettnem | many: in the US 2B+D is ISDN BRI |
| 14:03.53 | many | actually |
| 14:03.56 | brettnem | ISDN PRI is 23B+D |
| 14:03.59 | many | i have 6B+3D |
| 14:04.04 | many | on the same number. |
| 14:04.32 | *** join/#asterisk parag7732 (n=root@83.110.244.106) |
| 14:04.40 | shiznatix | jake1932, how do I setup my conf files to dial using that registered SIP account? |
| 14:04.54 | parag7732 | <PROTECTED> |
| 14:04.58 | brettnem | 6B+3D? interesting.. well the ISDN standards take into account all sorts of configurations.. most not actually utilized.. like "H" channels |
| 14:05.30 | jake1932 | in general terms, sue exten=> _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@provider) - but not sure about your provider |
| 14:05.41 | jake1932 | s/sue/use |
| 14:06.03 | jake1932 | (that's why i asked who you're with) |
| 14:06.19 | MrChimpy | hmm. I'm trying to use cdr_odbc - i've installed freetds and latest stable unixodbc. freetds worked (sort of) previously. with unixodbc all I get is : |
| 14:06.31 | MrChimpy | May 18 14:57:09 NOTICE[13778] res_odbc.c: registered database handle 'asterisk' dsn->[asterisk] |
| 14:06.31 | MrChimpy | May 18 14:57:09 NOTICE[13778] res_odbc.c: Connecting asterisk |
| 14:06.31 | MrChimpy | May 18 14:57:09 WARNING[13778] res_odbc.c: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/libtdsodbc.so' : /usr/lib/libtdsodbc.so: cannot |
| 14:06.31 | MrChimpy | May 18 14:57:09 NOTICE[13778] res_odbc.c: res_odbc loaded. |
| 14:06.55 | flujan | brettnem, thanks... It's sad thad digium reformulate its web page. |
| 14:07.04 | raggi | Can someone help me wiht my dialplan, i'm trying to setup a failver configuration, but my Dial to an IAX server seems to generate a Hangup when the server cannot be reached, detail is here: http://pastebin.com/724574 |
| 14:07.13 | MrChimpy | /usr/lib/libtdsodbc.so isn't there, yet i've built and installed unixodbc. no sign of such a thing in the build dirs. |
| 14:07.17 | brettnem | flujan: I'm not sure what you are talking about |
| 14:07.26 | flujan | brettnem, now almost all links in the voip-info.org are broken... :( |
| 14:07.44 | Assid | err.. anyone up ? |
| 14:07.53 | brettnem | flujan: the digium website dosen't contain useful information.. |
| 14:08.11 | flujan | http://www.digium.com/index.php?menu=faq#Configuration_6 |
| 14:08.30 | *** join/#asterisk fugitivo (n=ajf@201.255.176.56) |
| 14:08.30 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 14:08.36 | fugitivo | hello |
| 14:08.47 | fugitivo | I'm getting this warnings pri_fixup_principle: Can't fix up channel from 2 to 1 because 1 is already in use |
| 14:09.20 | fugitivo | zaptel 1.2.5 |
| 14:10.07 | fugitivo | any idea? also it seems i can't transfer calls, i don't know if that warning is part of the problem |
| 14:10.45 | parag7732 | I want to use inbound routing according to zaptel channel....how i have to specify "Zaptel Channel:"...Is it Zaptel Channel: Zap/3-1..???I want that if the ring is comming on Zap/3-1 my perticurar extention should ring ??? |
| 14:11.22 | *** join/#asterisk kay2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 14:11.49 | raggi | grr and arg |
| 14:12.10 | bkw_ | fugitivo, thats not zaptel... you have libpri skew I feel |
| 14:12.57 | fugitivo | i'm using libpri 1.2.2, didn't get that warning with zaptel 1.2.4 and libpri 1.2.2 |
| 14:13.23 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 14:13.24 | fugitivo | mmm, it seems to be solved on svn |
| 14:13.32 | fugitivo | http://bugs.digium.com/view.php?id=6963 |
| 14:14.36 | fugitivo | i have to upgrade zaptel and libpri from svn... i think i'm going to downgrade |
| 14:16.47 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 14:17.35 | shiznatix | if i have registered a external sip account in my [general] thing how do I dial with that? like I want to use that to dial a number but i cant figure out how |
| 14:18.04 | jake1932 | shiznatix: what's the deal - i already told you how to do it |
| 14:18.14 | shiznatix | jake1932, you just said read the docs, there are no docs |
| 14:18.26 | jake1932 | shiznatix: who is the provider? |
| 14:18.33 | satlan32 | i'm back |
| 14:18.41 | satlan32 | please no jokes, i want to learn |
| 14:18.54 | shiznatix | jake1932, oh wait, you had said stuff after that, i did not see as it was after the other stuff |
| 14:18.56 | shiznatix | let me try that |
| 14:19.31 | satlan32 | exten => 15,n,System(/tmp/email.sh ${EXTEN} ${file}) is not working |
| 14:20.21 | satlan32 | the script has 755 perm. |
| 14:20.39 | satlan32 | but the log show |
| 14:20.40 | satlan32 | app_system.c: Unable to execute '/tmp/email.sh 15 /tmp/wav.wav' |
| 14:21.59 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.220.Dial1.SanJose1.Level3.net) |
| 14:22.44 | sevard | yay, i <3 bash scripts |
| 14:22.49 | *** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 14:22.57 | Tier_1 | need input on a issue |
| 14:23.02 | Tier_1 | http://pastebin.ca/56883 |
| 14:23.13 | Tier_1 | it loops and I pointed out where |
| 14:23.25 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 14:24.39 | MrChimpy | yipes. astricon is pricey |
| 14:24.46 | MrChimpy | $420 |
| 14:25.30 | *** join/#asterisk DirtyD (n=DirtyD@ool-44c24fb7.dyn.optonline.net) |
| 14:25.33 | DirtyD | hohoho |
| 14:26.11 | shiznatix | jake1932, this is still not working. I have this line: exten => 00,1,Dial(SIP/4020@sip.squillo.it) |
| 14:26.22 | DirtyD | How do i get ATT Natural voices working with asterisk? |
| 14:26.38 | shiznatix | jake1932, and it gives me this error: Failed to authenticate on INVITE to '"test" <sip:test@192.168.1.208>;tag=as25b7b070' -- SIP/sip.squillo.it-0d22 is circuit-busy |
| 14:26.46 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
| 14:27.50 | *** join/#asterisk S4w (n=saw@adsl-3-161-117.mia.bellsouth.net) |
| 14:27.55 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 14:28.02 | dlynes_home | shiznatix: username, password, or realm is incorrect |
| 14:28.47 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 14:29.31 | S4w | hello guys, I caould not find this info in the internet hope you could help me. I have an IVR, after I run the Background() app and the extension is invalid it jumps to 1,1. Is there any way to know what number the user entered so that I can SayDigits(NUMBER) to the user and then Playback(INVALID)? |
| 14:30.22 | fugitivo | quit |
| 14:30.27 | sevard | fugitivo: / |
| 14:30.33 | Tier_1 | sox file.wav -r8000 file.gsm |
| 14:31.33 | flujan | defining groups in the zapata.conf file. I already configure the zaptel.conf to work with a E1 card, now I want to configure groups in the zapata.conf |
| 14:31.45 | flujan | signalling = fxs_ks |
| 14:31.45 | flujan | context = incoming |
| 14:31.45 | flujan | channel => 3 |
| 14:31.58 | flujan | it was working with my Dvkit |
| 14:32.12 | flujan | now, what I put in the channel variable? A group? |
| 14:32.47 | raggi | ah ha1 |
| 14:32.48 | raggi | <PROTECTED> |
| 14:32.48 | raggi | <PROTECTED> |
| 14:32.48 | raggi | May 18 14:32:03 WARNING[5764]: pbx.c:2412 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 14:33.15 | raggi | the Dial application does not generate this output unless you have 'rj' options |
| 14:33.22 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 14:34.12 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-188.z143-154-67.customer.algx.net) |
| 14:34.41 | raggi | tried adding option 't' aswell, but still get May 18 14:34:15 WARNING[5773]: pbx.c:2412 __ast_pbx_run: Timeout, but no rule 't' in context 'default' |
| 14:34.46 | raggi | how do i solve that? |
| 14:37.18 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 14:37.58 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
| 14:38.06 | mr_horsepower | hi, ppl |
| 14:38.40 | *** join/#asterisk chapeaurouge (n=chapeaur@vilhost1.vision.lu) |
| 14:39.02 | mr_horsepower | when i do disa(no-password|context) it will jump to ${Exten} colected by disa in context, when finishing colecting? |
| 14:39.35 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:40.50 | S4w | anyone can help me with an IVR problem? |
| 14:41.02 | jake1932 | press 0 or say agent :) |
| 14:42.04 | r_evolution | whoa. |
| 14:42.04 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) [NETSPLIT VICTIM] |
| 14:42.04 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) [NETSPLIT VICTIM] |
| 14:42.05 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) [NETSPLIT VICTIM] |
| 14:42.05 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk coppice (n=chatzill@99.194.17.210.dyn.pacific.net.hk) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk michael-i (n=michael-@141.41.38.58) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk Assid (n=assid@203.115.83.214) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk syle (n=blah@unaffiliated/syle) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk miztic (n=gerard@rarcoa.com) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk mut (n=animenod@65.111.222.120) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk Falle (n=falle@falle.se) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk pjz (n=pj@zachs.place.org) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk rabelais (n=blank@santamonica-cuda4-24-55-43-249.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk duckz (n=duckz@193.192.47.26) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-252-202.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk reth (n=reth@hjaerterdam.csbnet.se) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 14:42.06 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) [NETSPLIT VICTIM] |
| 14:42.06 | *** join/#asterisk flynux (n=prout@2a01:38:0:0:0:0:0:1) [NETSPLIT VICTIM] |
| 14:42.07 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 14:42.07 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk RevK-Busy (n=RevK@excalibur.bracknell.aaisp.net.uk) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk many (i=many@krikkit.ukeer.de) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk jpabuyer (i=macpato@blackbird.tecnoera.com) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff/madkiss) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk fontp (i=cecil@unon.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk pythos (i=pythos@unaffiliated/pythos) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk markus99 (n=markus@165.154.121.219) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk monthos (n=monthos@pewp.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk enots (i=dimka@freelsd.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk mds2 (n=mds@thewife.inspirednetworks.co.nz) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk vgster (n=vgster@84.18.199.68) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk citats (n=james@69.54.200.117) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) [NETSPLIT VICTIM] |
| 14:42.08 | *** join/#asterisk jesster (i=jesster@jesster.org) [NETSPLIT VICTIM] |
| 14:42.08 | r_evolution | whoawhoa. |
| 14:42.11 | jake1932 | oh my |
| 14:42.11 | S4w | I have an IVR, after I run the Background() app and the extension is invalid it jumps to 1,1. Is there any way to know what number the user entered so that I can SayDigits(NUMBER) to the user and then Playback(INVALID)? |
| 14:42.50 | S4w | any ideas? |
| 14:42.55 | S4w | the agent wont repond when I dial o :-D |
| 14:43.17 | jake1932 | ${INVALID_EXTEN} |
| 14:43.22 | S4w | :-O |
| 14:43.23 | jake1932 | that should do it |
| 14:43.25 | S4w | sweet |
| 14:43.29 | S4w | let me try |
| 14:43.31 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 14:45.18 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 14:45.36 | ravenpi | I want a button on my Polycom 501 to be able to transfer the current call automatically to a given extension. Any way to do this? |
| 14:45.46 | ravenpi | (Sorry -- it's actually a Polycom 601.) |
| 14:47.14 | *** join/#asterisk ajx (n=brian@242680hfc45.tampabay.res.rr.com) |
| 14:48.39 | S4w | jake1932: hmm if the extension is not defined in the context it will juts jump to i without letting the user finish dialing.. does background needs parameters? |
| 14:49.03 | *** join/#asterisk protocoldoug (n=doug@69-160-165-210.sbtnvt.adelphia.net) |
| 14:49.56 | jake1932 | S4w: as soon as it finds no valid possibilities, it will go to i |
| 14:50.17 | S4w | ohh :s |
| 14:50.45 | jake1932 | if you do X. it will wait until timeout |
| 14:51.04 | jake1932 | (unless your phone dialplan won't allow for that) |
| 14:51.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:51.24 | S4w | you mean X instead of i? |
| 14:51.35 | *** join/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com) |
| 14:52.08 | *** join/#asterisk ^unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 14:52.37 | jake1932 | X. collect digits till timeout |
| 14:52.38 | *** join/#asterisk Teeli (n=Tili@cm33.gamma249.maxonline.com.sg) |
| 14:53.15 | *** join/#asterisk djacob-wcgs (n=djac@c-68-32-12-249.hsd1.pa.comcast.net) |
| 14:53.20 | jake1932 | actually _X. |
| 14:53.26 | DagMoller | <PROTECTED> |
| 14:54.11 | S4w | jake1932: ok, now it works sweet, thanx man |
| 14:54.11 | jake1932 | should look like exten => _X.,1,Application(blah) |
| 14:54.18 | jake1932 | n[ |
| 14:54.19 | jake1932 | p |
| 14:55.29 | brettnem | hey all, I have a really weird problem maybe someone can help me out with? |
| 14:55.34 | brettnem | it's a good one |
| 14:55.59 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 14:56.12 | sevard | brettnem: sounds like a personal problem you should see your doctor about. |
| 14:56.13 | brettnem | customer is complaining of a call that is up NOW where they have an established 2 way talk path AND they hear the auto attendant at the same time |
| 14:56.32 | brettnem | I'm not sure where to even start to debug this problem |
| 14:57.07 | jake1932 | digital crosstalk? |
| 14:57.14 | brettnem | yeah, what the heck? |
| 14:57.23 | jake1932 | can you see if the call is bridged? |
| 14:57.38 | brettnem | it is |
| 14:57.39 | file | brettnem: that's mildly amusing |
| 14:57.56 | brettnem | file: I get shit like this all the time.. weird eh? |
| 14:58.07 | file | is it a zap channel? |
| 14:58.14 | brettnem | No, SIP->SIP |
| 14:58.22 | file | how very very odd |
| 14:58.23 | jake1932 | it's gotta be bridged |
| 14:58.26 | *** join/#asterisk ajx (n=brian@242680hfc45.tampabay.res.rr.com) |
| 14:58.44 | brettnem | bridged? well the two way talk path is... |
| 14:58.55 | file | you can't do two things at once on a SIP channel, which is why it's odd... on a zap channel you can though |
| 14:59.00 | jake1932 | the two rtp streams just can't mix without special processing |
| 14:59.08 | jake1932 | FWIK |
| 14:59.10 | brettnem | jake1932: I'm sayin! |
| 14:59.13 | file | that is really really weird |
| 14:59.48 | brettnem | it's gotta be asterisk |
| 15:00.04 | jake1932 | did you do an rtp debug yet? |
| 15:00.16 | brettnem | not sure how to do one where the data will be usable |
| 15:00.40 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:01.36 | jake1932 | debug on the ip that's getting both streams |
| 15:01.40 | brettnem | stupid asterisk state table |
| 15:01.52 | Damin | As much as I think VoIP over the Public Internet is a bad idea, there still needs to be a public Internet, so you all need to go to http://www.savetheinternet.com and sign the Net Neutrality petition.. |
| 15:02.16 | brettnem | oh I don't know.. lets get rid of the public internet |
| 15:02.29 | Makenshi | who needs the internets |
| 15:02.50 | jake1932 | is there a public internet anyways? Last I checked everything was owned by private companies |
| 15:05.16 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 15:05.32 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 15:06.20 | *** part/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com) |
| 15:06.35 | brettnem | file: any ideas of where to start looking for problems? |
| 15:07.50 | raggi | how do i get rid of the pbx_ael contexts? |
| 15:07.55 | boddy | hi all I am trying to call phone that on meridian My asterisk connected to meridian via E1 |
| 15:08.03 | boddy | bu failed |
| 15:08.26 | boddy | asterisk looking like phonenumber@sip ip no |
| 15:09.02 | file | brettnem: no, plus I'm making lunch so I'll bbs |
| 15:09.03 | protocoldoug | very new to asterisk here, i'm wondering why this: http://pastebin.com/724712 won't play an mp3 just rings my line |
| 15:10.06 | pif | how to Set() a variable only if empty? |
| 15:10.23 | Assid | oh cmon.. there has to be a way to use instant messaging fromt he phone |
| 15:10.24 | pif | should I test it first or is there an expression? |
| 15:10.28 | Assid | its not letting me send messges |
| 15:10.41 | sevard | grr |
| 15:10.49 | sevard | seems we have fat electrons on the pipe |
| 15:11.06 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 15:11.41 | brettnem | grr |
| 15:12.35 | sevard | probably a spaghetti cable cause packet failure though |
| 15:12.45 | boddy | ?? |
| 15:13.06 | *** part/#asterisk S4w (n=saw@adsl-3-161-117.mia.bellsouth.net) |
| 15:13.27 | sevard | boddy: haven't you ever had a complete transient lockout? |
| 15:14.01 | *** join/#asterisk elvisthedj (n=elvisthe@th20.montanavision.com) |
| 15:14.11 | *** join/#asterisk yain (n=yain@86-102-1-17.xdsl.primorye.ru) |
| 15:14.11 | yain | hi. |
| 15:14.11 | raggi | how do i get rid of the pbx_ael contexts? |
| 15:14.14 | DirtyD | anyone here use AT&T natural voices with Asterisk? If so, how did you do it without spending lots of $$$? |
| 15:14.19 | boddy | sevard:how can I do this ? |
| 15:14.36 | Hmmhesays | I could tell you, but then i'd have to kill you |
| 15:14.36 | nextime | any hint or little thing to say before to install * on a em64t processor? ( with a 64bit kernel ) |
| 15:14.43 | Juggie | no |
| 15:14.45 | Juggie | it works |
| 15:14.49 | yain | when making a SIP call.. client must send supported codecs in INVITE body? |
| 15:14.51 | DirtyD | Hmmhesays: that's ok you can kill me, just tell me :) |
| 15:14.58 | Hmmhesays | no |
| 15:15.03 | nextime | Juggie : it was for me? |
| 15:15.03 | sevard | boddy: just make sure you don't have any bogon emissions |
| 15:15.11 | Assid | i wonder when asterisk will support text |
| 15:15.16 | Assid | sip messaging |
| 15:15.35 | brettnem | Assid: don't hold your breath |
| 15:15.43 | DirtyD | Hmmhesays, is there a way to get the ATT TTS server engine fairly cheap? |
| 15:15.50 | boddy | sevard:no |
| 15:16.00 | DirtyD | I'm broke. |
| 15:16.14 | Assid | okay bbl |
| 15:16.33 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:16.40 | Hmmhesays | getting your dirty little paws on at&t's natural voice stuff is tricky |
| 15:16.43 | sevard | boddy: basically it's a high pressure system failure caused by transient emissions on the pipe, use googleflex's properly designed transmission system to foil the baddies |
| 15:17.04 | sevard | Hmmhesays: I remember their online demo, that stuff rocks |
| 15:17.35 | coppice | why natural voices? its far from the best available |
| 15:18.01 | yain | anyone? |
| 15:18.01 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 15:18.01 | sevard | boddy: basically to fix your problem you have to recalibrate your CPU |
| 15:18.24 | sevard | boddy: because you're experiencing bit bucket overflows |
| 15:18.41 | DirtyD | coppice: the voices sound awesome? WHat's better? |
| 15:18.56 | Hmmhesays | purchase your ivr files from someone |
| 15:19.10 | Hmmhesays | if you want some ivr's from a sleep deprived smoker I can help you out |
| 15:19.21 | coppice | the voices sound so so. the language processing is weak |
| 15:19.30 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 15:20.13 | raggi | sevard: what are you talking about? |
| 15:20.25 | vooduhal | Does anyone know why an agent who is in queue and paused gets put back into the queue but unpaused when you do an asterisk -rx 'reload' and if not, is there away to get agents.conf reread without a reload? |
| 15:20.30 | coppice | whatever sepakworks/softscan/nuance/whatever they are this week call their stuff this week is a lot better than natural voices |
| 15:21.20 | coppice | its derived from natural voices, which is derived from festival. most TTS is derived from festival in one way or another |
| 15:22.02 | *** part/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net) |
| 15:22.45 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 15:24.27 | boddy | sevard I think I have routing problem between sip and E1 |
| 15:27.00 | boddy | because I am debuggin sip call |
| 15:27.27 | boddy | then I see call going to phonenumber@sipserveripadres |
| 15:28.45 | protocoldoug | can Background(filename) play an Mp3? or... wav, or what type does it play? |
| 15:29.54 | *** join/#asterisk rumba (n=ropawa@cpe-68-201-149-21.sw.res.rr.com) |
| 15:29.55 | sevard | boddy: that stems from an unoptimzied hard drive |
| 15:30.48 | tzanger | ManxPower: hahahahahaha |
| 15:30.51 | tzanger | (-users post) |
| 15:30.59 | tzanger | "We originally had dialplans run in random order, but people found it too |
| 15:31.00 | tzanger | confusing." |
| 15:31.11 | ManxPower | tzafrir, 8-) |
| 15:31.21 | ManxPower | ..er. tzanger |
| 15:31.46 | sevard | raggi: electromagnetic energy loss in transient subsystems |
| 15:31.49 | MrChimpy | guys, how do I alter userfield in CDR from an AGI? is it an application I need to EXEC? |
| 15:32.10 | raggi | sevard: i see, clearly on topic then o.0 |
| 15:32.13 | sevard | MrChimpy: 'pico' |
| 15:32.27 | MrChimpy | ?! |
| 15:32.33 | sevard | raggi: actually nobody has caught on but i'm just streaming bofh fortune shit from my head |
| 15:32.44 | sevard | MrChimpy: 'vi' 'ed' |
| 15:32.48 | *** part/#asterisk yain (n=yain@86-102-1-17.xdsl.primorye.ru) |
| 15:32.54 | tzanger | 'grep,' sed awk. |
| 15:33.01 | MrChimpy | sevard: are you on crack? |
| 15:33.01 | protocoldoug | nano, emacs |
| 15:33.11 | sevard | all hail sed. |
| 15:33.22 | sevard | MrChimpy: it's a csv file, use a text editor. |
| 15:33.33 | tzafrir | vim . Though sed (and perl) are capable of inline editing as well |
| 15:33.40 | MrChimpy | FROM AN AGI |
| 15:33.41 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 15:33.41 | elvisthedj | anybody using mythphone? |
| 15:33.59 | sevard | I really don't like vim, i hate it but if i had to use it i'd use vi |
| 15:34.03 | MrChimpy | also, it's a csv file - but then it's also an SQL DB elsewhere |
| 15:34.05 | sevard | MrChimpy: perl? |
| 15:34.29 | ManxPower | MrChimpy, you should be able to do it via exec. "show applications" |
| 15:34.49 | MrChimpy | thanks manx |
| 15:35.42 | sevard | that's a lot of apps. |
| 15:36.04 | ManxPower | for more detailes info "show application X" |
| 15:36.11 | ManxPower | or "show applications like CDR" |
| 15:36.13 | MrChimpy | ah. setcdruserfield |
| 15:37.36 | sevard | MrChimpy: yeah, that's what I said |
| 15:37.57 | Flauto | anyone is using weather on asterisk by using test2wave? |
| 15:39.11 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:40.00 | blebleble | question: has anyone setup something to where (sort of like how cell phones work) when someone goes over minutes or is past due via billing to setup a custom trunk or something so that when they dial out they get an automated system that points them back to you saying pay up? |
| 15:40.58 | myiagy | blebleble a2billing does that |
| 15:41.39 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 15:44.03 | stack_ | I've been getting this (http://pastebin.ca/56898) sometimes when people dial out... it basically hangs up before it gets out of the PRI... anyone know why? |
| 15:45.34 | Sonderblade | does agi scripts have to have the .agi extension? |
| 15:46.40 | myiagy | Sonderblade no |
| 15:46.44 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 15:47.02 | blebleble | myiagy: really? is there anyway to implement such a feature without a2billing? |
| 15:47.19 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 15:47.20 | myiagy | blebleble only way i know is a2billing |
| 15:48.13 | brettnem | blebleble: you can write that in asterisk.. it's no big deal |
| 15:50.16 | blebleble | brettnem: 'generally' what would be your approach? |
| 15:50.45 | brettnem | use the sql application.. and your imaginatin |
| 15:50.58 | brettnem | see? piece of cake |
| 15:51.59 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 15:52.38 | Hmmhesays | imagination is key |
| 15:52.50 | Hmmhesays | you can unlock a wonderful world you would not normally be able to access |
| 15:53.38 | Hmmhesays | ask sevard: he spends most of his day locked in his mind |
| 15:54.15 | mut | wish i had an imagination |
| 15:55.05 | coppice | well, if you don't have one you can always imagine that you have |
| 15:55.31 | sevard | my fingers have stayed locked in your sister for a quite a while this morning |
| 15:56.12 | Hmmhesays | welcome to jail my friend |
| 15:56.23 | Hmmhesays | she's 15 |
| 15:56.39 | sevard | so am I |
| 15:56.53 | raggi | necrophiliac |
| 15:56.58 | blebleble | brettnem: its sql based? what db/table is it in? I thought it would just be a something i would need to set in my sip confs |
| 15:57.35 | brettnem | blebleble: if you want a canned solution you'll need to go with a commerical product.. if you want to develop yourself you'll need to think outside the box and develop your own database |
| 15:57.59 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 15:58.08 | *** part/#asterisk pythos (i=pythos@unaffiliated/pythos) |
| 15:58.10 | brettnem | necrophiliac?! |
| 15:58.25 | brettnem | umm.. I think you got the wrong word... haha |
| 15:58.27 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 15:58.58 | sevard | Hmmhesays: when are you going to get me trashed? this weekend? bring plenty, I'm irish. |
| 15:59.13 | *** part/#asterisk parag7732 (n=root@83.110.244.106) |
| 15:59.19 | _Paulo_ | in Brazil presumed rape is only if the girl is bellow 14. |
| 16:00.12 | stack_ | right now, if someone is on the phone and an incoming queue calls in, that person gets beeped like call waiting... is there a way to disable that? |
| 16:00.43 | brettnem | stack_: that's a phone thing.. not an asterisk thing.. although you might be able to do it with call-limit |
| 16:00.48 | brettnem | but do it at the phone |
| 16:00.58 | stack_ | brettnew: ok cool, thanks |
| 16:01.13 | myiagy | you can disable call waiting on linksys/sipura noramally |
| 16:01.26 | brettnem | yes you can |
| 16:02.42 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 16:07.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
| 16:07.19 | jaybuffet | hello... so i got the sample to work... but now i cant ring other sip phones |
| 16:08.21 | jaybuffet | any ideas? i can get to voicemail, dial out to the asterisk demo, but another interanl extension doesnt ring the phone.. it waits 20s then goes to voicemail |
| 16:08.44 | jaybuffet | firewalls are off and all internal network |
| 16:09.09 | Sonderblade | is there a reference for all commands that you can send through agi? |
| 16:09.14 | brettnem | ok.. it happened again.. "PSTN call to company A and establishes two way talkpath.. but then hears COMPANY B's auto attendant!" |
| 16:09.25 | file | that makes NO sense |
| 16:09.29 | sevard | _Paulo_: so you stuff your head with useful information that you will apply to your own life in the near future |
| 16:09.30 | sevard | like |
| 16:09.34 | sevard | the brazil fact |
| 16:09.35 | sevard | eh? |
| 16:09.45 | file | get a show channels, rtp debug, then a show channel on each respective channel... |
| 16:09.49 | jaybuffet | exten => 1250,1,Macro(stdexten,1250,SIP/1250) is in my extensions.conf |
| 16:11.13 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 16:11.27 | Meaty | j interplex |
| 16:12.16 | *** join/#asterisk salviadud (n=ralfalfa@201.138.132.204) |
| 16:12.35 | brettnem | file: the call is over.. I get the complaints at the end of the call |
| 16:12.37 | mut | heh |
| 16:12.43 | mut | i've had cellphones switch calls before |
| 16:12.47 | brettnem | file: pretty messed up, eh? |
| 16:12.49 | mut | weird when that happen |
| 16:12.50 | mut | s |
| 16:12.55 | file | brettnem: yes |
| 16:13.06 | mut | talkin to my brother and all the sudden i'm talking to some other random person and so is he |
| 16:13.16 | brettnem | file: asterisk 1.2.0 |
| 16:13.22 | stack_ | I've been getting this (http://pastebin.ca/56898) sometimes when people dial out... it basically hangs up before it gets out of the PRI... anyone know why? |
| 16:15.07 | jaybuffet | when i do "show channels" it shows channel SIP/1249 (where i am calling from) in State: Ring and SIP/1250 (where i am dialing to) in state: Down even though he is connected |
| 16:15.32 | *** part/#asterisk pattieja (n=pcxuser@c-67-163-29-136.hsd1.il.comcast.net) |
| 16:15.56 | *** join/#asterisk burizaa (n=freeee@cm87.omega96.maxonline.com.sg) |
| 16:16.22 | burizaa | hello... is there anyone who live in singapore? |
| 16:16.38 | burizaa | any idea where i can get the FXO devices ? |
| 16:17.09 | puzzled | anyone know if a TDM31B will properly work without the additional power plugged into it? |
| 16:17.27 | mut | doubt it |
| 16:17.31 | vooduhal | Ok, I have MONITOR_FILENAME set, monitor-type, monitor-format, and monitor-join set but I'm still not getting asterisk to record calls. There perms on the directory are correct as well. Any other ideas? |
| 16:17.34 | lunk | electronics run on power, i think |
| 16:17.55 | mut | uh no |
| 16:17.59 | mut | they run on lemon juice |
| 16:17.59 | puzzled | mut: yeah I didn't have high hopes it would |
| 16:18.05 | RoyK | lunk: electricity iirc |
| 16:18.14 | *** join/#asterisk parag7732 (n=root@83.110.244.106) |
| 16:18.36 | lunk | they also run on smoke, when you let the smoke out, they stop functioning |
| 16:18.46 | parag7732 | I m using callback with DISA. When i get a call back from pstn line to my mobile phone and threw DISA when i get a dial-tone and when i press nos quickly then suddenly phone gets disconnects...why ?? Do i need to change anything in zapata.conf.... |
| 16:18.58 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:19.27 | mut | man |
| 16:19.33 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 16:19.34 | Assid | heya |
| 16:19.44 | mut | i took 2 non drowsey benedryl before work today, i can't stay awake at all, it frickin sucks |
| 16:19.51 | Assid | err.. whats the cmd for checking if variable exists? |
| 16:19.52 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:20.00 | parag7732 | I m using callback with DISA. When i get a call back from pstn line to my mobile phone and threw DISA when i get a dial-tone and when i press nos quickly then suddenly phone gets disconnects...why ?? Do i need to change anything in zapata.conf.... |
| 16:20.03 | Assid | for extensions.conf |
| 16:20.05 | burizaa | which device can give me 4 FXO line ? |
| 16:20.14 | parag7732 | TDM400P |
| 16:20.32 | burizaa | parag7732, any idea how much it will cost ? |
| 16:20.52 | parag7732 | its around 600$ |
| 16:21.17 | burizaa | ouch ! |
| 16:22.32 | jaybuffet | if i am able to call and get someones voicemail, but their phone doesnt atually ring (all SIP) where should i start looking ? |
| 16:23.38 | mut | nat/firewall |
| 16:23.59 | mut | if the "are you there still" packet can't get thru to it it'll timeout |
| 16:24.04 | mut | sending ya to voicemail when dialed |
| 16:24.14 | mut | ugh my brain isn't working |
| 16:24.18 | jaybuffet | mut: firewalls off, and on same network |
| 16:24.38 | mut | does the device still think it's logged in? |
| 16:24.38 | jaybuffet | hm.. actually he is connected through wireless |
| 16:24.46 | mut | .. |
| 16:24.48 | mut | nat'd |
| 16:24.57 | jaybuffet | not nat'd though |
| 16:25.04 | *** part/#asterisk parag7732 (n=root@83.110.244.106) |
| 16:25.15 | jaybuffet | pass through dhcp from the rest of the network |
| 16:25.28 | mut | well if it lags out that'de cause it to not ring in |
| 16:25.32 | Damin | As much as I think VoIP over the Public Internet is a bad idea, there still needs to be a public Internet, so you all need to go to http://www.savetheinternet.com and sign the Net Neutrality petition.. |
| 16:25.35 | mut | the device would still think it's logged in |
| 16:25.45 | mut | so it'de just wait it's specified time to relogin so the server recognizes it |
| 16:25.53 | mut | check your logs? |
| 16:26.02 | mut | see if you see where it lags it out.. |
| 16:26.04 | *** join/#asterisk triple-e (n=piaergnj@adsl-70-128-78-22.dsl.stlsmo.swbell.net) |
| 16:26.17 | Sonderblade | how do you do WaitExten in AGI? |
| 16:26.19 | jaybuffet | what would i look for |
| 16:26.38 | jaybuffet | and do the logs give different info than running asterisk -cvvvv |
| 16:26.45 | triple-e | polycom 430 IP rocks --- im in the channel partner conference call right now |
| 16:26.49 | mut | May 18 12:18:07 NOTICE[15673] chan_sip.c: Peer '7281660' is now TOO LAGGED! (3111ms / 3000ms) |
| 16:26.49 | mut | May 18 12:18:07 NOTICE[15673] chan_sip.c: Peer '3332492' is now UNREACHABLE! Last qualify: 145 |
| 16:26.52 | mut | might look like one of those |
| 16:27.04 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 16:27.09 | triple-e | the 430IP is going to be the way to go |
| 16:29.13 | triple-e | 430 = 300 + fullduplex speakerphone + PoE |
| 16:31.49 | *** join/#asterisk fugitivo (n=ajf@201.255.176.56) |
| 16:31.51 | fugitivo | hi |
| 16:31.53 | jaybuffet | i dont see any errors like that in the messages log |
| 16:32.04 | fugitivo | is anyone having problems transfering calls comming from zap? |
| 16:32.15 | fugitivo | i get no audio when transfering a call with asterisk 1.2.7 |
| 16:32.22 | fugitivo | it was working fine with 1.2.4 |
| 16:35.33 | jaybuffet | could it be a config setting within X Lite ? |
| 16:36.20 | ManxPower | what does sip show peers show for that device? |
| 16:36.41 | [TK]D-Fender | triple-e : IP 430? |
| 16:37.11 | MrChimpy | hmm! |
| 16:37.27 | [TK]D-Fender | triple-e : OMG |
| 16:37.29 | ManxPower | burizaa, not "ouch". A 4-port card for a normal PBX would run about 3x that |
| 16:37.34 | MrChimpy | here's a tricky one. how do i get system hostname in dialplan? |
| 16:37.55 | triple-e | yeah ! -- and SIP 2.0 |
| 16:38.02 | [TK]D-Fender | triple-e : Thats distruptive, got photo's, or any other material youcan pass on? |
| 16:38.03 | MrChimpy | I need to pass it to SetAccountCode() |
| 16:38.11 | triple-e | whoah -- 2.0 is gonna fix some shit |
| 16:38.24 | triple-e | it looks exactly like a 301 |
| 16:38.36 | [TK]D-Fender | triple-e : Woken up and put a backlight yet? |
| 16:38.43 | jaybuffet | 'sip show peers' does show the phone |
| 16:38.51 | triple-e | it says something about lit lcd |
| 16:38.54 | triple-e | i can't tell |
| 16:39.15 | ManxPower | MrChimpy, you didn't read README.variables did you? |
| 16:39.18 | triple-e | i can't copy the presentation -- im in a web-ex |
| 16:39.29 | MrChimpy | oooh ${ENV(hostname)} |
| 16:39.33 | [TK]D-Fender | triple-e : Screen cap! |
| 16:39.38 | triple-e | lol |
| 16:39.41 | ManxPower | And you specifically didn't read the ${ENV} section, did you? |
| 16:39.55 | MrChimpy | :) |
| 16:40.03 | triple-e | polycom is cert'ing digium asterisk |
| 16:41.07 | [TK]D-Fender | triple-e : Whats the impact? |
| 16:41.14 | *** join/#asterisk Lino` (n=Lino@i577BCE6C.versanet.de) |
| 16:41.19 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:41.37 | [TK]D-Fender | triple-e : real support? |
| 16:41.41 | triple-e | all good , they used to not support astersik |
| 16:41.42 | triple-e | yeah |
| 16:42.19 | triple-e | i can officially call polycom with some crazy problem like --- if you use an IP instead of a FQDN the phone cant find the voip server |
| 16:42.27 | triple-e | at least with sip 1.6.6 |
| 16:42.54 | triple-e | whoa sip 2.0 is supporting MS LCS2005 |
| 16:43.05 | [TK]D-Fender | triple-e : FUN... never touched "direct dial" on them, and just installed a 301 & 501 for home, and run mostly 600's here at work |
| 16:43.16 | burizaa | well this is sounds like stupid question, but is there anyone here success to integrate Quintum with Asterisk ? |
| 16:43.25 | [TK]D-Fender | triple-e : Is SIP 2.X release or still beta? |
| 16:43.41 | Qwell[] | my guess is prealpha |
| 16:43.43 | triple-e | dunno -- they are talking about releaseing it |
| 16:44.40 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 16:45.02 | flujan | hi all ... I'm trying to put asterisk working with another PBX. |
| 16:45.07 | [TK]D-Fender | triple-e : excellent. I recently upgraded my X01's to 1.6.6 for BLF support and general fixes |
| 16:45.15 | vooduhal | Ok this is frustrating. Asterisk is ignoring Set(MONITOR_FILENAME=/var/spool/asterisk/recordings/QSAMPLE-${UNIQUEID},MONITOR=TRUE) and recording all calls to /var/spool/asterisk/monitor and it's also not merging as specified in the queues.conf Any IDs why? I'm running 1.2.7.1 |
| 16:45.22 | flujan | Both have a e1 interface. I'm trying to use a rj-45 cable |
| 16:45.29 | vooduhal | ideas |
| 16:45.41 | vooduhal | The line is being executed though. |
| 16:45.45 | flujan | Should this cable be a crossover? |
| 16:45.52 | triple-e | but LCS is huge |
| 16:45.56 | Qwell[] | flujan: a t1 cable |
| 16:45.57 | vooduhal | I such also specify that I'm set that before the Queue(myqueue) call. |
| 16:46.16 | [TK]D-Fender | triple-e : LCS? And has 2.0 reduced the image size? |
| 16:46.26 | triple-e | dunno - image size |
| 16:46.28 | flujan | Qwell, what do you mean by a t1 cable? |
| 16:46.37 | Qwell[] | flujan: a t1 cable...not rj45 |
| 16:46.52 | vooduhal | s/such/should/ |
| 16:46.58 | triple-e | LCS is the microsoft live communication server |
| 16:47.04 | [TK]D-Fender | triple-e : Its just psycho to see the firmware 10x larger than Cisco's... |
| 16:47.05 | flujan | http://www.ahk.com/t1_cable.html |
| 16:47.12 | *** join/#asterisk JonR800 (i=jon@p1mp.org) |
| 16:47.18 | [TK]D-Fender | triple-e : LCS *shudder* |
| 16:47.19 | Qwell[] | flujan: You may even need a crossover T1 |
| 16:47.30 | triple-e | LCS is what customers want though |
| 16:47.35 | vooduhal | Does anyone here have experience with call recording and Queue? |
| 16:47.37 | flujan | Qwell, do you have such specification? |
| 16:47.42 | [TK]D-Fender | triple-e : Proof that people are stupid.... |
| 16:47.43 | flujan | i found this website: |
| 16:47.52 | triple-e | lol -- never needed proof |
| 16:47.53 | flujan | http://www.ahk.com/t1_cable.html |
| 16:47.59 | Qwell[] | flujan: google for "t1 cable" "jared smith" |
| 16:48.02 | *** join/#asterisk rob314 (n=rob314@207.58.194.2) |
| 16:48.23 | Qwell[] | maybe not |
| 16:48.38 | Qwell[] | http://www.jaredsmith.net/misc/cables/ |
| 16:48.58 | [TK]D-Fender | triple-e : with the IP 430 when you said PoE do you mean Brick & direct PoE supported, or PoE (/ injector) required? |
| 16:49.22 | triple-e | direct like the 601 |
| 16:49.30 | [TK]D-Fender | triple-e : And brinck included? |
| 16:49.31 | flujan | Qwell, just to make sure: |
| 16:49.34 | flujan | this : http://www.jaredsmith.net/misc/cables/ |
| 16:50.01 | triple-e | TKD: brick included ? |
| 16:50.08 | *** part/#asterisk rob314 (n=rob314@207.58.194.2) |
| 16:50.11 | Qwell[] | flujan: yes, swap 1 with 4 and 2 with 5 |
| 16:50.11 | [TK]D-Fender | triple-e : Power transformer. |
| 16:50.45 | [TK]D-Fender | triple-e : Basically the 501 & 301 each have a STUPID set of cables. 601 includes the brick AND internal support. |
| 16:50.55 | triple-e | i think so, as the 600 includes the brick in the event that you are not going to use the internal support |
| 16:51.32 | triple-e | msrp for 430 is 239$ -- cant wait to see what street price is |
| 16:52.11 | triple-e | 2.0 generaly availabe mid to late june 2006 |
| 16:52.15 | flujan | Qwell, one head with Orange/White |
| 16:52.15 | flujan | 2Orange |
| 16:52.15 | flujan | 3Green/White |
| 16:52.15 | flujan | 4Blue |
| 16:52.15 | flujan | 5Blue/White |
| 16:52.15 | flujan | 6Green |
| 16:52.17 | flujan | 7Brown/White |
| 16:52.19 | flujan | 8Brown |
| 16:52.26 | flujan | and other with 1Blue |
| 16:52.26 | flujan | 2Blue/White |
| 16:52.26 | flujan | 3Green/White |
| 16:52.27 | flujan | 4Orange/White |
| 16:52.29 | [TK]D-Fender | triple-e : Yeah, there isn't a lot of room between the IP301 & 501 at street right now so things could get interesting.... |
| 16:52.29 | flujan | 5Orange |
| 16:52.31 | flujan | 6Green |
| 16:52.35 | flujan | 7Brown/White |
| 16:52.37 | flujan | 8Brown |
| 16:52.38 | [TK]D-Fender | flujan : PASTEBIN! |
| 16:52.39 | [TK]D-Fender | ~pb |
| 16:52.41 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 16:53.33 | triple-e | IAX2 support for polycom is being evaluated !!!!!! |
| 16:53.42 | Qwell[] | won't ever happen |
| 16:53.52 | Qwell[] | would be pretty stupid for them to implement it |
| 16:54.13 | *** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net) |
| 16:54.20 | triple-e | dunno why |
| 16:54.26 | flujan | [TK]D-Fender, sorry. |
| 16:54.32 | [TK]D-Fender | triple-e : A nifty though, but IAX2 endpoints aren't really worth the effort for the most-part. |
| 16:55.04 | Qwell[] | yeah, iax is intended as a server to server protocol |
| 16:55.13 | Nivex | I'd love to see an alternate firmware with SIP/IAX2 instead of SIP/MGCP |
| 16:55.42 | triple-e | for nat transversal and qos i would like to see IAX2 endpoint |
| 16:55.44 | Nivex | Qwell[]: that may be true, but any protocol that can pierce firewalls so easily would be insanely awesome on remote endpoints |
| 16:56.06 | Nivex | ok pierce may not be the right word. traverse would be better. |
| 16:56.19 | [TK]D-Fender | triple-e : The need to clean up their act on firmware policy, PoE onboard across the board (YES, charge me a few more bucks PLEASE), Backlight, MicroBrowser enabled on 501, etc. They are already the best thing I can suggest for a business phone and these additions would squelch the competition. |
| 16:58.58 | *** join/#asterisk xermesx (n=ermsewrk@217.220.121.62) |
| 17:01.18 | mitcheloc | [TK]D-Fender: who actually uses a microbrowser??? |
| 17:01.31 | tzanger | mitcheloc: midgets? |
| 17:01.49 | mitcheloc | tzanger: that was my guess too |
| 17:02.00 | [TK]D-Fender | mitcheloc : On my IP601's at work, hell yeah. Live Queue stats on idle, mass-presence lists, company directory, etc |
| 17:02.23 | Assid | damn there has to be another provider man.. there just has to be |
| 17:02.42 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 17:02.45 | [TK]D-Fender | mitcheloc : had I moved to a large enough place I'd have set up my X-10 gear and run them off it as well. |
| 17:02.46 | mitcheloc | [TK]D-Fender: wouldn't a nice desktop application be better? |
| 17:02.49 | *** join/#asterisk Skarmeth (n=Skarmeth@200164212156.user.veloxzone.com.br) |
| 17:03.19 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 17:03.31 | [TK]D-Fender | mitcheloc : I tend to use it for PBX info. yes a PC is better for several kinds of things, but live queue stats, VM status, etc is nice to have "at a glance" on your desk for CSR's |
| 17:03.33 | *** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net) |
| 17:03.35 | Zodiacal | anyone know how i can get all 6 of my out going numbers to display only one callerid #? verizon provides a ring down for my 6 lines when you call one number, thats the number i want to have show up on peoples callerid. is this something verizon would do or would asterisk do it? |
| 17:03.36 | Assid | need to find an incoming provider .. for cheaper than $5 |
| 17:03.47 | Assid | unltd incoming minutes |
| 17:03.58 | viperdude | i have written a little windows applet that sits in your taskbar and monitors the number of waiting calls in your queues..it pops up a warning if they reach a defined limit |
| 17:03.58 | op3r | how many users can a dual xeon with 2 gig of memory handle? |
| 17:04.05 | [TK]D-Fender | Zodiacal : the CID your lines show is up to the telco to set and you should be able to get them identical. |
| 17:04.26 | viperdude | i could expand it to do other stuff i guess to |
| 17:04.31 | [TK]D-Fender | op3r : Not a good way to phrase that question. How many lines / calls / and on what kind of tech is better. |
| 17:04.36 | triple-e | viperdude: i would have a use for that |
| 17:04.49 | [TK]D-Fender | viperdude : Cool.... I'm a lazy coder myself. |
| 17:04.59 | triple-e | :-) |
| 17:05.02 | [TK]D-Fender | viperdude : is your program public? |
| 17:05.14 | Zodiacal | tkd-fender i emailed verizon, but they said i need to subscribe to a pbx or keysystem.. does that mean i would have to use their pbx? |
| 17:05.18 | viperdude | triple-e: ok I have not looked at it for a while but I will try and put an installer on a website sometime |
| 17:05.27 | Zodiacal | tkd-fender it didn't really sound like they knew what i was talking about |
| 17:05.33 | viperdude | yeah I can give it aaway as beerware |
| 17:06.05 | Assid | [TK]D-Fender: know any decent provider with unlimited incoming minutes.and atleast 2 channels.. for <=$5 ? |
| 17:06.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 17:07.00 | viperdude | actually i have a copy on my work pc here , hang on a few and i will upload it somewhere |
| 17:07.04 | [TK]D-Fender | Zodiacal : It means they're full of shit and want to find every way on this earth to bill you for more "services" |
| 17:07.26 | Zodiacal | tkd-fender haha.. do you know off hand what i should specificly request from them? |
| 17:07.32 | [TK]D-Fender | Assid : No, <5$ is stupid..... |
| 17:08.04 | *** join/#asterisk ToTo (n=ToTo@host87-91.pool8260.interbusiness.it) |
| 17:08.07 | [TK]D-Fender | Zodiacal : Tell them "Set all of these lines I have with you to the PRIMAY number or I'll take my business ELSEWHERE" |
| 17:08.09 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 17:09.22 | Zodiacal | tkd-fender Thank You! |
| 17:10.31 | [TK]D-Fender | Zodiacal : You have to strong are these morons sometimes and you need to let them know that they stand to lose business. Its the only thing they respect (their bottom line) |
| 17:10.54 | *** join/#asterisk niter3 (n=klutch@d57-102-239.home.cgocable.net) |
| 17:11.10 | [TK]D-Fender | I have become quite skilled in recent years realizing the monkeys tell what they are told and it works most of the time. Don't let them win. |
| 17:11.31 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 17:12.01 | niter3 | Hey guys, i got a teleworker phone connect to my asterisk box via SIP. Is it possible to have the teleworker phone ring from two diferent PBX's, one being my asterisk and another with somebody else both using SIP? Would this be dependent on the teleworker phone, or something I have to do with the server? |
| 17:12.35 | jake1932 | yes |
| 17:12.47 | _Paulo_ | niter3, you can always make one pbx call the other. |
| 17:12.49 | jake1932 | have asterisk dial the two SIP extensions |
| 17:13.16 | [TK]D-Fender | niter3 : Depends, if you want to ting it THROUGH the other server direct and can do it in passthrough. Dependsing on the phone you may also be able to have it register to 2 seperate servers at the same time. |
| 17:13.16 | jake1932 | Dial(extn1&extn2) |
| 17:13.21 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 17:13.51 | jake1932 | [TK]D-Fender: you gotta complicate things :) |
| 17:13.57 | niter3 | basically if somebody at this business on their phone system dials extension 22 it will call the teleworker phone, and if somebody on this other PBX calls extension 201 it will call the teleworker hone also. |
| 17:14.16 | *** join/#asterisk dasmoot_ (n=dasmoot@S0106001195567012.no.shawcable.net) |
| 17:14.44 | [TK]D-Fender | jake1932 : Nah! My ways work and are easy... best way, have the phone reg to 2 servers |
| 17:15.05 | niter3 | [TK]D-Fender: that's what i'm thinking, but is there a different way of this? |
| 17:15.20 | _Paulo_ | make the other pbx call * teleworker phone extension |
| 17:15.24 | niter3 | what jake1932 says doesn't make much sense to me. |
| 17:15.47 | tzanger | anyone here ever used/played with PBX Manager? |
| 17:16.00 | niter3 | _Paulo_: so i would have to give it the 7 10 digit code plus the extension? |
| 17:16.20 | [TK]D-Fender | niter3 : pass the calls through 1 server |
| 17:16.24 | jake1932 | niter3: you can have the other pbx interface with asterisk (rather than interfacing w the phone directly) |
| 17:16.30 | viperdude | ok you can download my applet AsteriskSysTray from www.bctech.co.uk... email me if you have comments/problems |
| 17:16.36 | _Paulo_ | niter3, no, make the other pbx call some * sip extension. |
| 17:16.53 | niter3 | ok ok.. i got three people here talking. |
| 17:16.53 | niter3 | :) |
| 17:16.57 | niter3 | Me briain |
| 17:17.47 | niter3 | So I tell one of the PBX machines if they call extension 22 it will pass to the phone, and if they call extension 201 it will passed to another PBX (Asterisk with the extension to him??) |
| 17:17.52 | viperdude | ok i am off to get some food ... cya |
| 17:17.59 | *** part/#asterisk viperdude (n=jon@84.45.193.6) |
| 17:18.07 | _Paulo_ | niter3, yepz |
| 17:18.14 | niter3 | i would have to do this for both systems |
| 17:18.43 | niter3 | now is there a way to do this with no trunk on one of the phone systems. By perhaps pointing to IP addresses or something? No right? |
| 17:19.05 | _Paulo_ | * 22 -> teleworker phone |
| 17:19.12 | niter3 | what is * |
| 17:19.14 | niter3 | the ip address? |
| 17:19.22 | _Paulo_ | other 201 -> * 22 -> teleworker phone |
| 17:19.46 | niter3 | what does * stand for? |
| 17:19.47 | niter3 | ip? |
| 17:19.50 | niter3 | of the pbx server |
| 17:19.50 | _Paulo_ | asterisk |
| 17:20.06 | niter3 | so the pbx server.. ok.. hrm, so that's all i'd have to do.. |
| 17:20.49 | *** join/#asterisk QbY_ (n=Kelvin@66.236.241.68.ptr.us.xo.net) |
| 17:21.33 | niter3 | ok this doesn't make sense though |
| 17:21.33 | niter3 | hold on |
| 17:21.37 | *** join/#asterisk PBXtech (i=nik@49.sub-70-213-221.myvzw.com) |
| 17:21.39 | faberk64 | hi folks |
| 17:21.50 | jake1932 | right - asterisk can register to your other PBX as a client |
| 17:22.03 | faberk64 | I'm having problems, installing zaptel-1.2.5 |
| 17:22.04 | niter3 | if the teleworker phone is logged into the one phone system, and I forward the calls out to the phone system on an extension. His phone won't ring because it's not logged into that phone systme. |
| 17:22.04 | faberk64 | http://pastebin.com/724941 |
| 17:22.25 | faberk64 | is a CentOS-4.3 Server |
| 17:22.35 | PBXtech | i got a 7960 that has a laptop on the back of the phone and i loose pings to the gateway. its like the switch is going in and out. but its on all the phones. |
| 17:22.40 | faberk64 | any ideas? |
| 17:23.24 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:23.25 | russellb | ~centosbug |
| 17:23.26 | jbot | centosbug is, like, a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 17:24.03 | Sonderblade | i want to implement my dialplan as an AGI script written in python, would that scale well? |
| 17:24.34 | *** part/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 17:24.35 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:24.44 | jake1932 | niter3: if asterisk registers to your other pbx system - when the extension on the other pbx rings, it will "ring" asterisk. you have to point it to the context which should have Dial(SIP/you_ phone) |
| 17:25.03 | jake1932 | SIP/your_phone |
| 17:25.36 | niter3 | jake1932: So uhh. is the one system just coming back to the other one? |
| 17:25.45 | jake1932 | yes |
| 17:26.19 | jake1932 | IOW - sip phone reg's to asterisk. asterisk reg's to other pbx |
| 17:26.25 | [TK]D-Fender | russellb: ! |
| 17:26.47 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:26.50 | niter3 | duh i'm stupid |
| 17:26.50 | niter3 | ;0 |
| 17:27.04 | QbY_ | anyone have a good howto for cisco 7940s converting to sip firmware and configuring to asterisk.. the doc on cisco's site is missing something--as i have been unable to make it work.. |
| 17:27.09 | niter3 | it's just like forwarding to a cell phone, but to another extension on a server |
| 17:27.10 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) |
| 17:27.13 | niter3 | server=pbx |
| 17:27.15 | niter3 | simple |
| 17:27.16 | niter3 | ok, |
| 17:27.16 | niter3 | duh |
| 17:27.18 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:27.22 | PBXtech | do 7940/60 not like a cheep ass switch? ive got my laptop plugged into the back of the phone and its up and down |
| 17:27.42 | flujan | Qwell, there are three diagrams in the page. |
| 17:28.05 | flujan | Wich one I use? I clipped a calbe using the T1 crossover and a ethernet crossover without sucess... |
| 17:28.20 | Assid | [TK]D-Fender: i know they are stupid.. but whats the cheapest otherwise |
| 17:29.48 | niter3 | ok but wait a sec |
| 17:29.49 | niter3 | uhhh.. |
| 17:29.51 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-221.cascadeaccess.com) |
| 17:29.57 | *** join/#asterisk aeroillini (n=kevin@c-71-197-210-101.hsd1.or.comcast.net) |
| 17:30.04 | batphone | so what do i do about this stupid situation where users forward the phone to itself and bring my pentium D linux server to its knees? |
| 17:30.22 | niter3 | For instance, I want all people on one PBX to be able to call this extension, and everyone else on the other call that exention too |
| 17:30.40 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:30.40 | faberk64 | THANKS! |
| 17:30.46 | mr_horsepower | disa() sometimes does not work, there is any bug about this function arround? |
| 17:30.46 | dasmoot_ | has anyone run into 482 loop detected error with asterisk? my pbx is behind a nat router, and when I try to to place calls thru my voip provider I get: |
| 17:30.46 | dasmoot_ | <PROTECTED> |
| 17:30.46 | dasmoot_ | <PROTECTED> |
| 17:30.46 | dasmoot_ | <PROTECTED> |
| 17:30.46 | dasmoot_ | <PROTECTED> |
| 17:30.48 | dasmoot_ | chan_local.c:494 local_alloc: No such extension/context 12225551234@default creating local channel |
| 17:31.24 | batphone | dasmoot_: could mean that the number being dialed is being forwarded back to itself? |
| 17:31.25 | flujan | guys, which is the specification for a E1 cable? |
| 17:31.26 | aeroillini | anyone know anything about alsa errors? I can't get asterisk to start up |
| 17:31.30 | flujan | I'm using the link: |
| 17:31.41 | flujan | http://www.jaredsmith.net/misc/cables/ |
| 17:31.53 | jake1932 | niter3: if you link the 2 pbxs through registering asterisk as a SIP client, you should be fine |
| 17:32.03 | flujan | which I use? T1 crossover and straight? |
| 17:32.09 | aeroillini | the error I get is "broken pipe" when asterisk segfaults |
| 17:32.16 | dasmoot_ | batphone: it shouldn't be -- if I take Asterisk out of the picture and connect directly to my voip provider using a sip phone, calls work fine |
| 17:32.48 | aeroillini | and then earlier in the console output, it says it cannot perform sdn_pcm_open: device or resource busy |
| 17:33.30 | Assid | docelm0: yo there? |
| 17:34.11 | *** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be) |
| 17:34.18 | aeroillini | alsa? anyone? |
| 17:34.29 | Assid | err.. the DID's you buy from DIDX .. do they come with minutes? |
| 17:34.49 | Assid | or do you have to pay for minutes seperately |
| 17:34.56 | [TK]D-Fender | Assid : Stop looking for a non-existant free-ride! |
| 17:35.09 | _Paulo_ | aeroillini, do you have alsa working? |
| 17:35.22 | aeroillini | Paulo, yes ... alsa works flawlessly elsewhere |
| 17:35.43 | aeroillini | all my programs use the alsa default device, and they all mix together |
| 17:35.47 | Assid | there just has to be a cheaper option than $11 of voicepulse |
| 17:35.58 | [TK]D-Fender | Assid : Says who, YOU? |
| 17:36.07 | _Paulo_ | put the * user in the alsa group |
| 17:36.25 | aeroillini | Paulo, I will try that ... just a second |
| 17:36.26 | Assid | well.. its the net.. theres always someone cheaper |
| 17:36.28 | [TK]D-Fender | Assid : If you have tried so hard for so long without success maybe you should reevaluate your conclusion. |
| 17:37.03 | [TK]D-Fender | Assid : And I already have a personal flock of wild geese.... |
| 17:37.06 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:38.06 | _Paulo_ | aeroillini, in Debian its the "audio"group |
| 17:38.27 | aeroillini | Paulo, I am running Gentoo ... I have added the asterisk user to the "audio" group, but am still getting the same error |
| 17:38.44 | aeroillini | "Ouch ... error while writing audio data: : Broken pipe" |
| 17:38.59 | generalhan | _Paulo_: whats up man |
| 17:39.36 | generalhan | ~seen sleepy_one |
| 17:39.40 | jbot | sleepy_one <n=chatzill@cpe-24-166-34-22.neo.res.rr.com> was last seen on IRC in channel #asterisk, 5d 23h 55m 45s ago, saying: 'cya all l8r :-D'. |
| 17:39.46 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 17:41.21 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 17:41.35 | jsaunders | Anyone know how to execute a .war? |
| 17:41.46 | jsaunders | on xp |
| 17:42.30 | jsaunders | I think it has somethin' to do w/ Java, not sure. Google is vague thus far. |
| 17:43.07 | dasmoot | anyone have experience with Asterisk behind nat? with sip debugging on it looks like * is connecting to my WAN IP instead of my VoIP provider's SIP server for some reason, and causing a "482 Loop Detected" error since it's connecting to itself |
| 17:44.59 | Alex | This is probably a bad place to ask, but I'm not sure where else to ask... Does anyone know if you can flash older Cisco VoIP phones that don't support SIP to support SIP? |
| 17:45.18 | aeroillini | is it possible to run Asterisk with no sound backend at all? |
| 17:45.34 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 17:45.40 | aeroillini | I assume that only the softphone/hardware phone needs ALSA/OSS/etc. |
| 17:47.04 | vooduhal | Is it possible to tell if a feature patch is supposed to be applied to stable or CVS? I'm wanting to apply http://bugs.digium.com/view.php?id=7084 to 1.2.7.1 but can't get it to patch properly. |
| 17:47.31 | *** part/#asterisk slav_jb (n=k@213.16.46.130) |
| 17:47.41 | *** join/#asterisk amorith (n=nahirean@unaffiliated/nahirean) |
| 17:48.11 | *** join/#asterisk markus99 (n=markus@165.154.121.219) |
| 17:48.35 | flujan | gruys, my board just flash a red light. How can I test to see if it's working. I trying to put one pbx server to communicate with asterisk throught a E1 crossover cable |
| 17:48.48 | flujan | http://www.jaredsmith.net/misc/cables/ |
| 17:49.08 | flujan | a straight head and a T1 crossover head. |
| 17:51.01 | *** join/#asterisk boddy (n=popeye@85.108.34.37) |
| 17:51.25 | boddy | hii all I have a problem |
| 17:51.27 | boddy | I am trying to call phone that on meridian My asterisk |
| 17:51.27 | boddy | connected to meridian via E1.but failed |
| 17:51.27 | boddy | I think I have routing problem between sip and E1 |
| 17:51.27 | boddy | because I am debuging sip call then I see call going |
| 17:51.27 | boddy | to phonenumber@sipserveripadres |
| 17:51.28 | boddy | I just want sip client able to call phones that on |
| 17:51.30 | boddy | meridian and phones that on meridian able to call sip |
| 17:51.32 | boddy | clients |
| 17:51.54 | *** join/#asterisk VxJasonxV (n=jason@unaffiliated/VxJasonxV) |
| 17:52.30 | *** join/#asterisk shanky (i=jramirez@217.11.114.145) |
| 17:52.37 | shanky | hi, a quick question |
| 17:52.42 | boddy | is there anyone |
| 17:52.48 | boddy | help me ? |
| 17:52.49 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 17:52.51 | shanky | I'm trying to add call forward service |
| 17:53.08 | shanky | so I have these lines in the extension.conf |
| 17:53.08 | shanky | exten=>_*72,3,Read(num_desvio,pls-ent-num-transfer,9) |
| 17:53.08 | shanky | exten=>_*72,4,GotoIf($[${LEN(${num_desvio})} < 9]?*72|104) |
| 17:53.37 | shanky | the problem I have is that LEN$foo always return "1" |
| 17:54.26 | shanky | I was using that in an old installation with asterisk 1.0.X, and this one is 1.2.X and I think that's the problem |
| 17:55.11 | shanky | but I've been looking in voip-info and google to see if is there any change in LEN or Read commands, but I can't find anything |
| 17:55.34 | shanky | any idea? |
| 17:56.22 | aeroillini | _Paulo_, I found the problem ... adding the asterisk user to the alsa group got rid of my ALSA error, but ALSA was not causing the segfault |
| 17:56.44 | aeroillini | the segfault was caused by the speex codec ... as soon as I told it not to load it, asterisk started right up |
| 17:57.02 | aeroillini | thanks for your help |
| 17:57.07 | *** join/#asterisk AndyCap_ (n=aoy@pdpc/supporter/sustaining/AndyCap) |
| 17:57.41 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 17:57.45 | generalhan | ~seen _Paulo_ |
| 17:57.48 | jbot | _paulo_ is currently on #asterisk (6h 52m 47s). Has said a total of 13 messages. Is idling for 19m 42s, last said: 'aeroillini, in Debian its the "audio"group'. |
| 17:57.53 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
| 17:57.53 | generalhan | i love jbot ! hahaha ! |
| 17:58.29 | jsaunders | If I have a zap trunk group w/ say 5 channels, and the first 3 lines are in use, will asterisk automatically know to attempt an outbound on channel 4 (next available) or will I have to set something, like busydetect= ? |
| 17:58.44 | mercestes | ~seen jbot |
| 17:58.46 | jbot | jbot <i=ibot@pdpc/supporter/active/TimRiker/bot/apt> was last seen on IRC in channel #debian, 210d 2h 42m 39s ago, saying: 'rumour has it, sarge is Ten-HUT! Fall in! Sarge is the code name for the current stable Debian release, version 3.1, released on June 6th, 2005. Ask me about <install debian>, or i guess sarge is the biggest lump of free as in ... |
| 17:59.16 | mercestes | cool. |
| 17:59.21 | myiagy | how can i tell a2billing an account has unlimited credits? credit = -1? |
| 18:00.19 | vooduhal | Ok, question. I'm needing some patches to be applied to CVS app_queue. We're running 1.2.7.1. Would it be possible to build CVS head and use the app_queue module with 1.2.7.1? |
| 18:00.24 | _Paulo_ | hi generalhan |
| 18:03.08 | *** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net) |
| 18:04.04 | _Paulo_ | generalhan, have you solved the problem with the callfiles? |
| 18:04.29 | generalhan | _Paulo_: one sec i have to put out a fire |
| 18:04.58 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 18:05.43 | generalhan | _Paulo_: yes kind of ... but i had to turn of selinux to do it ... so now i need to figure out a real way to do it ! lol |
| 18:07.05 | Sonderblade | how do you define call-groups in AEL? |
| 18:08.16 | _Paulo_ | I didnt know you were using selinux. |
| 18:16.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:17.41 | _Paulo_ | generalhan, you should have some ": denied" errors in /var/log due to selinux |
| 18:18.38 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 18:18.49 | _Paulo_ | generalhan, you can adjust your selinux polices to get rid of these. then your scripts will work. |
| 18:18.51 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 18:19.12 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 18:19.25 | ManxPower | I just don't understand why carriers have SUCH a fetish for handing off IP to the customer as Ethernet instead of DSX-1 |
| 18:19.36 | boddy | anybody help me ? |
| 18:20.32 | dlynes_office | boddy: Just state what your problem is |
| 18:20.41 | boddy | dlynes_office: |
| 18:20.50 | boddy | I am trying to call phone that on meridian My asterisk |
| 18:20.57 | boddy | connected to meridian via E1.but failed |
| 18:21.07 | boddy | I think I have routing problem between sip and E1 |
| 18:21.13 | boddy | because I am debuging sip call then I see call going |
| 18:21.15 | _Paulo_ | (what is meridian?) |
| 18:21.24 | dlynes_office | _Paulo_: option 11...nortel pbx |
| 18:21.28 | boddy | I just want sip client able to call phones that on |
| 18:21.34 | boddy | meridian and phones that on meridian able to call sip |
| 18:22.03 | CunningPike | boddy: We have a Meridian Option 11C talking to asterisk over a T1 PRI |
| 18:22.13 | boddy | yes |
| 18:22.23 | dlynes_office | CunningPike: you guys want to buy another option 11c, or maybe an option 11e? |
| 18:22.23 | boddy | 11c e1 /pri |
| 18:22.31 | dlynes_office | CunningPike: We've got two we want to get rid of ;) |
| 18:22.33 | CunningPike | dlynes_office: Sure |
| 18:22.35 | _Paulo_ | can you place a call to meridian from cli? |
| 18:22.46 | CunningPike | dlynes_office: And monkeys might fly out of my butt ;) |
| 18:22.51 | dlynes_office | lol |
| 18:23.22 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 18:23.24 | boddy | in my extensin conf exten => 1344,1,Dial,Zap/1 |
| 18:23.37 | boddy | and I am calling 1344 from the sip client |
| 18:23.50 | boddy | in si |
| 18:23.50 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 18:23.53 | boddy | p debug |
| 18:24.05 | boddy | calling going to 1344@sipserverip adres |
| 18:24.10 | CunningPike | boddy: You need to qualify that more - Dial(Zap/1/<insert Nortel extension here>) |
| 18:24.15 | generalhan | _Paulo_: i have no errorls in my error_log about selinux |
| 18:24.28 | ManxPower | boddy, put the CLI output of a call on pastebin.ca |
| 18:24.39 | _Paulo_ | generalhan, nor in dmesg? |
| 18:24.44 | boddy | ManxPower in sip debug |
| 18:24.45 | ManxPower | boddy, we don't care about the SIP part, we care about the Zap part right now. |
| 18:24.52 | generalhan | wheres dmesg ? |
| 18:25.01 | CunningPike | boddy: Dial(Zap/g1/<extension>) would be better |
| 18:25.05 | *** join/#asterisk lars-ut (n=lars-ut@70.103.228.158) |
| 18:25.15 | ManxPower | CunningPike, that's not going to work for most Meridian port types |
| 18:25.19 | _Paulo_ | type dmesg | grep denied |
| 18:25.38 | generalhan | _Paulo_: nadda |
| 18:25.48 | _Paulo_ | oh... |
| 18:26.26 | dlynes_office | Is jbot drunk? |
| 18:26.29 | _Paulo_ | if selinux is not mandatory in your setup, just go on without it. |
| 18:26.33 | generalhan | _Paulo_: i have a TON of them about permissions denied, before i turned selinux off. but now everything is ok |
| 18:26.44 | CunningPike | ManxPower: Really? How do you call a Nortel extension then? |
| 18:26.48 | generalhan | _Paulo_: but much more insecure is it now ? |
| 18:27.27 | _Paulo_ | generalhan, Not much. |
| 18:27.32 | ManxPower | CunningPike, depends on the interface type. For a T-1/E-1 trunk port you don't call extensions directly. For T-1/E-1 DID port, then that's handled by the signaling. for a T-1/E-1 Tie line port, same as for DID port. |
| 18:27.39 | jaybuffet | whats a good cheap hard phone to do my asterisk testing |
| 18:27.59 | dlynes_office | There are no good cheap phones |
| 18:28.11 | dlynes_office | There's cheap phones, and there's good phones |
| 18:28.11 | jaybuffet | is that an oxymoron |
| 18:28.25 | CunningPike | ManxPower: Okay - I guess we're using a tie |
| 18:28.36 | ManxPower | CunningPike, since he has told us NOTHING about the actual nortel interface, I'll assume the cheaper methof |
| 18:28.41 | dlynes_office | jaybuffet: well, how cheap is cheap for you? |
| 18:28.45 | *** join/#asterisk eipi (n=FreePBX6@139-213-126-200.fibertel.com.ar) |
| 18:28.51 | jaybuffet | but there are some phones that are cheaper than other and might be better |
| 18:29.12 | jaybuffet | $50 |
| 18:29.14 | jsaunders | Anyone know of a good, solid IAX2 based ATA? |
| 18:29.16 | ManxPower | jaybuffet, Linksys. Stay away from grandstream |
| 18:29.22 | QbY_ | what is the reset key sequence on a 7940 |
| 18:29.28 | ManxPower | jaybuffet, you will find nothing decent for $50 |
| 18:29.28 | jaybuffet | ok... i was just looking at grandstream |
| 18:29.41 | generalhan | QbY_: if its the same as 7960 its *+6+Settings |
| 18:29.51 | boddy | ManxPower: |
| 18:29.54 | boddy | http://pastebin.com/725042 |
| 18:30.06 | djacob-wcgs | ManxPower: what problems ahve you had with grandstream that you say that? im wondering cause we have 10 of them and want to know if i should expect issues |
| 18:30.10 | dlynes_office | Aastra 9133i's are around $140Cdn, Polycom 501 around $180USD, Grandstream Budgetones around $70USD, ... |
| 18:30.15 | QbY_ | generalhan.. that just takes me to the settings |
| 18:30.20 | jaybuffet | ManxPowere: i just want to test setups and such, i'll get real phones later.. which will be my next question.. whats a good business phone for asterisk |
| 18:30.27 | ManxPower | djacob-wcgs, see the mailing list archices |
| 18:30.32 | generalhan | I have 30 Aastra 9112i's and i love them |
| 18:30.34 | _Paulo_ | dlynes_office, cheap, easy, good: you can have only 2 at the same time. Its hardcoded into the universe source. |
| 18:30.38 | generalhan | they are only like $120 |
| 18:30.39 | ManxPower | jaybuffet, good business phone: Polycom or Cisco |
| 18:31.00 | jaybuffet | ManxPower: ok, so polycom's are good.. i was looking at them also |
| 18:31.14 | stack_ | The Polycom 301's work great here... for just phone usage they are great and give the execs 501's to make them feel special ;p |
| 18:31.25 | websae | most of the Sip phones out there work good...it's all about the configurations and features of each |
| 18:31.25 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:31.44 | [TK]D-Fender | jaybuffet : Polycom is an excellent choice, and only getting better. |
| 18:31.47 | ManxPower | boddy, perhaps you didn't hear me. I said that at this point I don't care about SIP. I care about the ZAP stuff. Turn off SIP debug. Repaste the CLI showing the Dial command being run. |
| 18:31.59 | [TK]D-Fender | stack_ : I gave EVERYONE here IP 600's :) |
| 18:32.27 | stack_ | [TK]D-Fender: lol, my boss is a tight wad, so he wanted as cheap as possible |
| 18:32.41 | ManxPower | stack_, then get out of the project ASAP. |
| 18:32.49 | generalhan | stack_: my boss is the same way .. thats why i went with the Aastra phones ... and they turned out ok |
| 18:32.51 | jaybuffet | im guessing none of these phones i can pick up at bestbuy |
| 18:32.54 | ManxPower | In telecom "cheap = doesn't work well". |
| 18:33.04 | generalhan | but you need the 9113is if you want more than one line on the phone ... |
| 18:33.19 | ManxPower | jaybuffet, any IP phones at BestBuy will be locked to a specific provider. |
| 18:33.38 | QbY_ | My Cisco 7940 is requesting file P0S3-07-5-00.sbn .. The only file I have is P0S3-07-5-00.sb2 -- are they the same file? Simple rename |
| 18:33.45 | ManxPower | boddy, oh, and put your extensions.conf on pastebin.ca too |
| 18:33.47 | jaybuffet | so where can i get a good deal on polycom's, anyone know a good place? |
| 18:33.56 | stack_ | ManxPower: When I showed him how much a proprietary system cost, I had free reign on the rest of the project |
| 18:34.09 | generalhan | QbY_: no they are not the same file you need them all ... there should be 4 i believe |
| 18:34.25 | ManxPower | jaybuffet, voxzilla.com atacomm.com Avenue Computer Supply |
| 18:35.01 | salviadud | stack_. what are 501's? |
| 18:35.09 | QbY_ | generalhan-- i have a bin, loads, and sb2 file |
| 18:36.02 | stack_ | salviadud: Polycom 501's... nice SIP phones |
| 18:36.07 | [TK]D-Fender | stack_ : Even with 600's my project was HALF the cost of a Nortel BCM or Avay IP Office solution. |
| 18:36.10 | salviadud | ohh, that |
| 18:36.30 | [TK]D-Fender | jaybuffet : depends where you are. |
| 18:37.03 | stack_ | [TK]D-Fender: I would give pricing and then my boss would check every price that night and try to find better deals... "Well on eBay I can get them for..." "NO!" |
| 18:37.07 | websae | websae has great deals on phones hehehe :) |
| 18:37.26 | salviadud | stack_. you should be the boos |
| 18:37.30 | salviadud | i mean, boss |
| 18:37.35 | generalhan | QbY_: you should have: POS3-07-5-00, POS3-07-5-00.sb2, POS3-07-5-00.loads, POO3-07-5-00.sbn |
| 18:37.37 | jaybuffet | i'm in tampa, fl |
| 18:37.50 | [TK]D-Fender | jaybuffet : www.antonline.com |
| 18:37.58 | stack_ | salviadud: I'm the only tech guy, so technically I am the boss of my department ;p |
| 18:38.11 | [TK]D-Fender | jaybuffet : atacomm is probably better on support though. |
| 18:38.19 | QbY_ | generalhan.. yes.. i do.. hwoever, the phone is requesting P0S3-07-5-00.sbn |
| 18:38.40 | generalhan | what do you have in your OS79XX file ? |
| 18:38.49 | QbY_ | P0S3-07-5-00 |
| 18:38.52 | ManxPower | Avenue computer is a little higher, but if you buy from them say "Ask ERic to get me the firmware", he'll call me and I'll get you the firmware. |
| 18:39.02 | generalhan | it should say P003-07-5-00 |
| 18:39.05 | generalhan | then it will load |
| 18:39.16 | generalhan | is this the first load of this firmware version ? |
| 18:39.24 | ManxPower | for the most part the firmwares on freedomphones site is good enough for people, however. |
| 18:39.42 | QbY_ | generalhan.. i have version 5 of sip on there now,, i'm trying to get it up to 7 |
| 18:39.56 | boddy | ManxPower:there is nothing on cli when turn off sip debug |
| 18:40.35 | generalhan | QbY_: yes ... IIRC in the instructions for upgrading it even says to start out the first load with "P003" NOT "P0S3" |
| 18:40.48 | QbY_ | k |
| 18:40.52 | generalhan | QbY_: then you can change it to "P0S3" after the first load and it will work ok |
| 18:40.56 | QbY_ | well i'm rebooting it now |
| 18:41.19 | ManxPower | boddy, I'm still waiting for your extensions.conf |
| 18:41.32 | [hC] | If i have a line with smart ring on it (ie one DID tied to a standard ringing tone when someone calls it, another DID with a ring-ring tone) can asterisk differentiate this so that i can treat each number accordingly? |
| 18:41.40 | ManxPower | boddy, what IP phone are you using? |
| 18:41.58 | ManxPower | [hC], in theory. see zapata.conf.sample |
| 18:42.09 | Juggie | distinctive ring, yeah it shuold be able to. |
| 18:42.16 | Juggie | dayton you newb :) |
| 18:42.27 | [hC] | :) dude i am so fuckin swamped i just had a question thrown at me |
| 18:42.28 | boddy | express talk |
| 18:42.36 | [hC] | I wasnt sure if asterisk was smart enough to figure out distinctive ring |
| 18:42.39 | [hC] | You neeeeeeeeeever know. |
| 18:42.40 | [hC] | :P |
| 18:43.20 | ManxPower | [hC], It's one of those features only cheap ass home users care about, so it's not well supported nor well documented. |
| 18:43.24 | boddy | ManxPower:http://pastebin.com/725058 |
| 18:44.59 | ManxPower | boddy, what context= does that specific phone account have in sip.conf? |
| 18:45.19 | tamp4x | why would the MWI light aon a snom stay on when there are no messages |
| 18:45.29 | Zodiacal | anyone know what the AUX rj45 port on my cisco 7960 is for? |
| 18:45.33 | Zodiacal | is it another port on its switch? |
| 18:45.40 | ManxPower | tamp4x, the only time that happens is when I had a typoe in the mailbox- line |
| 18:45.42 | tamp4x | this happened after i upgraded asterisk to latest stable |
| 18:45.54 | ManxPower | Zodiacal, aux is normally a serial port |
| 18:46.05 | tamp4x | why would i typo do it |
| 18:46.07 | Zodiacal | acctualy it looks more like rj11 |
| 18:46.21 | Zodiacal | manxpower okie maybe for programing them all in batch or somthin? |
| 18:46.23 | Zodiacal | hrmm |
| 18:46.26 | Zodiacal | owell |
| 18:46.32 | QbY_ | generalhan.. your rock man.. thanks.. |
| 18:46.33 | Zodiacal | not a rj45 anywayz :) |
| 18:46.36 | Zodiacal | thanks |
| 18:46.47 | ManxPower | tamp4x, mailbox=1234@default was what it was supposed to be, I had mailbox=12346@default which had voicemail messages |
| 18:46.51 | boddy | ManxPower:http://pastebin.com/725065 |
| 18:46.59 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 18:47.53 | generalhan | QbY_: no worries man |
| 18:48.07 | generalhan | i struggled with that for a while myself on my first cisco software upgrade |
| 18:48.37 | *** join/#asterisk H323_Help_Wanted (n=FreePBX3@64.229.199.21) |
| 18:48.50 | H323_Help_Wanted | Hello everyone ;;; I have a big problem with H323 suppport |
| 18:48.57 | H323_Help_Wanted | I have installed it |
| 18:49.09 | H323_Help_Wanted | but I don't know where to check if it is installed ;;; Installation was successfull |
| 18:49.36 | H323_Help_Wanted | Has anyone installed H323 support so far? plz plz plz plz plz help as it has frustrated me a lot |
| 18:49.41 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 18:49.44 | *** join/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 18:49.53 | *** join/#asterisk xp_prg (n=wgheath@67-102-228-17.adsl.lbdsl.net) |
| 18:50.59 | brif8 | jaybuffet: Hi |
| 18:51.18 | distortion | h323: show channeltypes at * cli> |
| 18:51.33 | boddy | ManxPower:did you see me ? |
| 18:51.39 | distortion | if you see h323 in there, its installed. |
| 18:51.43 | _Paulo_ | H323_Help_Wanted, do you can load your H323 module? |
| 18:52.02 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:52.20 | _Paulo_ | H323_Help_Wanted, type load chan_h323.so in the cli |
| 18:52.47 | H323_Help_Wanted | okey guys ;; i will try that ;;; one sec guys |
| 18:53.04 | _Paulo_ | (or whatever the module you have istalled is called) |
| 18:53.08 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:53.34 | generalhan | hey guys ... if i delete my entire voicemails dir... will that just set it all back to nothing ... so everyone will have to record new messages and so on ? |
| 18:53.38 | [TK]D-Fender | twisted[asteria] : ! |
| 18:53.39 | generalhan | or will i damage something ? lol |
| 18:54.06 | ManxPower | boddy, the context for that device is emretest bust you have no Dial line for the nortel in that conext in extensions.conf |
| 18:54.21 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:54.22 | H323_Help_Wanted | distoration = show channel types command doesnt' work |
| 18:54.34 | [TK]D-Fender | twisted[asteria] : Care to lend a hand in adding back in the AMP/FreePBX/A@H referral notice into the channel topic? :} |
| 18:54.35 | sevard | generalhan: i think you just need to leave the voicemails directory else it'll complain, i think you can delete everything inside of it |
| 18:54.38 | distortion | its show channeltypes |
| 18:54.41 | H323_Help_Wanted | _Paulo_ = load chan_h323.so gives unable to load |
| 18:54.47 | sevard | generalhan: but when you do that you'll get rid of the custom prompts that users create, pretty sure |
| 18:54.51 | distortion | then its not installed |
| 18:54.54 | sevard | generalhan: do a backup first then :) |
| 18:55.08 | ManxPower | either put the correct Dial line in [emretest] or change the line in sip.conf to be context=nortel or nortel2 |
| 18:55.09 | *** join/#asterisk littlejohn (n=little@host242-76.pool876.interbusiness.it) |
| 18:55.10 | generalhan | sevard: so i can delete all the folders inside but just leave the voicemails folder itself ? |
| 18:55.22 | _Paulo_ | H323_Help_Wanted, may be its not this name |
| 18:55.31 | H323_Help_Wanted | there is H323 |
| 18:55.35 | H323_Help_Wanted | in show channeltypes |
| 18:55.37 | sevard | generalhan: I believe so, because asterisk creates folders based on voicemail context inside of the voiemail folder |
| 18:55.45 | generalhan | we just got a a couple new offices in the building so i want to completely reformat all my extension numbers so i dont want lingering VM messages |
| 18:55.47 | _Paulo_ | H323_Help_Wanted, type just "load chan_h" and hit tab |
| 18:56.00 | *** join/#asterisk ToTo (n=ToTo@host87-91.pool8260.interbusiness.it) |
| 18:56.12 | distortion | or you may have installed ooh323 which is "load chan_ooh323.so" |
| 18:56.13 | H323_Help_Wanted | There is NO h323 = sorry for mistype |
| 18:56.42 | H323_Help_Wanted | it's not that either |
| 18:57.00 | H323_Help_Wanted | let me get the name |
| 18:57.04 | _Paulo_ | H323_Help_Wanted, type just "load chan_" and hit tab 2 times |
| 18:57.52 | _Paulo_ | H323_Help_Wanted, this should give you a list of channel driver available. |
| 18:57.52 | H323_Help_Wanted | _Paulo_ there is no H323 or so at all |
| 18:58.13 | H323_Help_Wanted | u r right |
| 18:58.20 | _Paulo_ | H323_Help_Wanted, so i think its not correctly installed. |
| 18:58.36 | H323_Help_Wanted | It's an add-on package for Asterisk@home |
| 18:58.41 | H323_Help_Wanted | and all you have to do is unzip it |
| 18:58.48 | H323_Help_Wanted | and type ./install.sh |
| 18:58.50 | H323_Help_Wanted | which installs |
| 18:58.57 | H323_Help_Wanted | and gives succesfuuly installed msg |
| 18:59.04 | H323_Help_Wanted | but I don't know what goes wrong there |
| 18:59.13 | H323_Help_Wanted | I installed in /tmp file |
| 18:59.17 | H323_Help_Wanted | do you think that would be a problem? |
| 18:59.36 | _Paulo_ | sorry, I dont know *@home. |
| 18:59.39 | distortion | there's not much support for *@home here |
| 19:00.03 | sevard | ~aam |
| 19:00.07 | sevard | ~amp |
| 19:00.08 | jbot | i heard amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 19:00.36 | _Paulo_ | H323_Help_Wanted, type "find / -name chan_*.so" in the shell |
| 19:00.47 | H323_Help_Wanted | so where do you guys get the H323 from? |
| 19:00.51 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:01.18 | _Paulo_ | then move your h323 module to the right place. |
| 19:01.32 | [TK]D-Fender | twisted[asteria] : Pretty please? :) |
| 19:02.01 | boddy | ManxPower: Thanks alot |
| 19:02.11 | *** topic/#asterisk by twisted[asteria] -> Asterisk: The Open Source PBX -=- http://www.asterisk.org/ || FreePBX/AMP/Asterisk@Home Users should join #FreePBX for assitance |
| 19:02.12 | ManxPower | H323_Help_Wanted, there are at least FOUR H323 channel drivers for Asterisk. |
| 19:02.32 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 19:02.50 | ManxPower | The one in asterisk-addons was paid for by Digium, so that's the one I'd use if I was in hell and was forced to use H323 as punishment for my sins. |
| 19:03.02 | twisted[asteria] | hahaha |
| 19:03.03 | sevard | HAhahaha |
| 19:03.08 | *** part/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 19:03.24 | H323_Help_Wanted | _Paulo_ the find command should be run in asterisk -r or just in terminal? |
| 19:03.24 | boddy | why I didnt use diffrent context isnt there transition between contexts |
| 19:03.51 | H323_Help_Wanted | ManxPower = It is hell ;;; I realize it now |
| 19:03.54 | ManxPower | boddy, there is no transition between contexts unless you do it via goto or similar |
| 19:03.57 | _Paulo_ | no, you should run outside asterisk, in the linux shell prompt |
| 19:04.00 | [TK]D-Fender | twisted[asteria] : THANKS |
| 19:04.14 | boddy | ok thanks ManxPower |
| 19:04.17 | sevard | I thought H323_Help_Wanted meant Will Work for Food |
| 19:04.26 | ManxPower | boddy, you need to read The Book |
| 19:04.27 | ManxPower | ~thebook |
| 19:04.29 | jbot | thebook is probably somebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Meet Jim Van Meggelen at Cluecon http://www.cluecon.com |
| 19:04.54 | H323_Help_Wanted | _paulo_ = everything is in in module (right place) but there is no H323 ;;; lol |
| 19:05.11 | twisted[asteria] | oh, the cluecon spam has started already? |
| 19:05.42 | ManxPower | H323_Help_Wanted, what H323 module did you install? |
| 19:05.46 | Katty | twisted[asteria]: mew. |
| 19:05.57 | ManxPower | twisted[asteria], I think I still have them sent to /dev/null from the last cluecon |
| 19:06.13 | twisted[asteria] | Katty, hi |
| 19:06.19 | Katty | twisted[asteria]: i didn't 'hi' i mewed. |
| 19:06.26 | dlynes_office | H323_Help_Wanted: cd /usr/local/src/asterisk-addons-1.2.2/asterisk-ooh323c ; ./configure --prefix=usr ; make ; make install |
| 19:06.27 | Katty | twisted[asteria]: let's try again. |
| 19:06.29 | Katty | twisted[asteria]: mew. |
| 19:06.31 | H323_Help_Wanted | so ManxPower can you give me the URL for digium H323 support? |
| 19:06.31 | H323_Help_Wanted | I think it is oh323 |
| 19:06.47 | ManxPower | H323_Help_Wanted, Digium does not support H323 |
| 19:06.48 | dlynes_office | mew mew mew mew, mew mew mew mew, mew mew mew mew mew mew mew |
| 19:06.49 | twisted[asteria] | Katty, mew? |
| 19:06.58 | ManxPower | asterisk.org should have links to asterisk-addons |
| 19:06.59 | Katty | twisted[asteria]: :> |
| 19:07.10 | twisted[asteria] | it does, doesn't it? |
| 19:07.16 | _Paulo_ | H323_Help_Wanted, sorry, if the module was not there, I dont know how i can help. |
| 19:07.31 | Katty | dlynes_office: :< |
| 19:07.36 | dlynes_office | ? |
| 19:07.40 | _Paulo_ | H323_Help_Wanted, try running unzip -l to see if the module is inside the zip file |
| 19:07.48 | dlynes_office | It was the whiskers' theme song :((( |
| 19:07.55 | dlynes_office | You don't speak kat? |
| 19:08.31 | dlynes_office | erm whiskas, i mean |
| 19:08.56 | klasstek | Why does ilbc bring a dual Xeon 2.8 to its knees with only 10 concurrent calls? |
| 19:09.56 | ManxPower | klasstek, did you look at "show translations" to see the relative CPU cost for translating between codecs? |
| 19:10.16 | ManxPower | sorry "show translation" |
| 19:10.24 | klasstek | yes |
| 19:10.42 | H323_Help_Wanted | _Paulo_ the main file contains install.sh, readme.txt, and asterisk-oh323 (zipped) |
| 19:10.48 | klasstek | ilbc was less than g729 |
| 19:10.53 | Katty | dlynes_office: that's not kat. weirdo. |
| 19:11.04 | H323_Help_Wanted | then inside asterisk-oh323 is gatekeeper, oh323 and so on |
| 19:11.09 | H323_Help_Wanted | do I have to unzip that one too? |
| 19:11.13 | ManxPower | or better yet "show translation recalc 15" |
| 19:11.19 | H323_Help_Wanted | I just unzip main file and then go ./install.sh |
| 19:11.26 | ManxPower | Asterisk is not a gatekeeper |
| 19:11.45 | ManxPower | no version of Asterisk H323 drivers support gatekeeper, only gateway |
| 19:12.16 | H323_Help_Wanted | apparently that add-on package is supposed to install that support |
| 19:12.19 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 19:12.25 | *** join/#asterisk darekj (n=darek@chello084010229235.chello.pl) |
| 19:12.31 | H323_Help_Wanted | _Paulo_, distortion, ManxPower, guys can you plz have a look at this URL ;; There is a readme file in the zip (it's small zip file) and I think I followed everything right: http://sourceforge.net/project/showfiles.php?group_id=123387&package_id=151231 |
| 19:12.49 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 19:12.49 | ManxPower | H323_Help_Wanted, no. |
| 19:13.03 | lzhang | anybody use or is familiar with app_amd.c? |
| 19:14.04 | _Paulo_ | H323_Help_Wanted, that link is for festvel |
| 19:14.12 | lzhang | I'm having a problem where it hangs when it tries to detect answering machines |
| 19:14.13 | _Paulo_ | festivel is not h323 |
| 19:14.21 | jpbotelho | i have a error 481: "Call Leg/Transaction Does Not Exist".. how i fix it?? |
| 19:14.24 | H323_Help_Wanted | _Paulo_ just below that is H323 |
| 19:15.02 | H323_Help_Wanted | and to the right of where it says H323 Support package it says (notes) which has the installation instruction |
| 19:15.12 | _Paulo_ | k |
| 19:15.50 | Assid | err.. if i have 192.168.0.14 spherelinx 3760bf96-79 00101/00005 ulaw Yes Rx: ACK .. does this mean any calls from this phone is gonna be on ulaw ? |
| 19:15.53 | jpbotelho | any1 can help me with error 481: "Call Leg/Transaction Does Not Exist" |
| 19:15.54 | jpbotelho | ?? |
| 19:17.01 | ManxPower | jpabuyer, Your extensive search of the mailing list archives was not helpful? |
| 19:17.42 | ManxPower | Assid, no, it means THAT call is ulaw |
| 19:17.57 | ManxPower | jpbotelho, Your extensive search of the mailing list archives was not helpful? |
| 19:19.45 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) |
| 19:21.09 | _Paulo_ | H323_Help_Wanted, I sow no * module inside that package. |
| 19:21.39 | _Paulo_ | H323_Help_Wanted, may be you have to install something else. |
| 19:22.36 | _Paulo_ | H323_Help_Wanted, I see just asterisk-oh323-0.6.5/asterisk-driver/chan_oh323.c |
| 19:22.52 | _Paulo_ | H323_Help_Wanted, this one need to be compiled. |
| 19:23.09 | _Paulo_ | H323_Help_Wanted, and installed under the correct path |
| 19:23.27 | _Paulo_ | H323_Help_Wanted, so you will have chan_0h323.so |
| 19:24.00 | dlynes_office | jpbotelho: it usually means that leg is currently on a call |
| 19:24.14 | dlynes_office | jpbotelho: but, it can mean other things, too |
| 19:24.14 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 19:24.34 | fourcheeze | anyone got a nokia phone to register with asterisk using SIP? |
| 19:25.12 | blitzrage | nope -- check the SIP messages with SIP debug in getting it to work |
| 19:27.20 | twisted[asteria] | nokia phone? |
| 19:27.23 | twisted[asteria] | meh? |
| 19:28.43 | fourcheeze | n80 for instance |
| 19:28.49 | fourcheeze | has a place to enter a sip registrar |
| 19:29.07 | blitzrage | GSM + WiFi? |
| 19:29.12 | fourcheeze | yeah |
| 19:29.13 | twisted[asteria] | cool |
| 19:29.20 | twisted[asteria] | price range? |
| 19:29.38 | blitzrage | available in North America? |
| 19:29.38 | fourcheeze | in the UK they are £45 on a tariff |
| 19:29.47 | twisted[asteria] | 45 pounds? |
| 19:29.50 | fourcheeze | yep |
| 19:29.52 | blitzrage | thats cheap |
| 19:29.53 | twisted[asteria] | that's cheap |
| 19:29.55 | blitzrage | got a link somewhere? |
| 19:29.58 | blitzrage | I want one |
| 19:29.59 | blitzrage | :) |
| 19:30.01 | fourcheeze | hold on |
| 19:30.23 | twisted[asteria] | wtf |
| 19:30.32 | twisted[asteria] | i'm finding them for >$650 US |
| 19:31.19 | fourcheeze | http://shop.orange.co.uk/shop/show/handset/nokia_n80/detail/pay_monthly |
| 19:31.28 | fourcheeze | £50 there |
| 19:31.35 | fourcheeze | only on a tariff though |
| 19:31.39 | Matt | the operative phrase there is "on a tariff" |
| 19:31.45 | fourcheeze | so you have to pay £30 a month |
| 19:31.49 | twisted[asteria] | ooooh |
| 19:31.57 | fourcheeze | but you do get quite a lot of minutes for that |
| 19:32.14 | fourcheeze | don't get exited yet because I've yet to see one register :-) |
| 19:32.17 | Matt | shows how long it's been since I had a uk mobile |
| 19:32.26 | fourcheeze | got a friend trying to get his n80 to register with my * |
| 19:32.27 | blitzrage | I've never had a UK mobile :) |
| 19:32.27 | Matt | "used" |
| 19:32.40 | klasstek | ManxPower, no change in number w/ show translation recalc 15 |
| 19:33.06 | fourcheeze | anyway, the n80 just says "registration failed" |
| 19:33.11 | fourcheeze | asterisk is debugging the IP number |
| 19:33.17 | fourcheeze | nothing appears |
| 19:33.21 | blitzrage | nothing eh? |
| 19:33.22 | fourcheeze | looks like it's not even trying |
| 19:33.26 | blitzrage | hrmmm... |
| 19:33.34 | blitzrage | maybe it can't find the server? no DNS resolv? |
| 19:33.48 | fourcheeze | apparently it can browse web sites etc |
| 19:33.55 | fourcheeze | however IP number might be a way forward |
| 19:35.59 | fourcheeze | the n80 allows you to add a sip url to a contact |
| 19:35.59 | fourcheeze | so in theory it's a cool phone |
| 19:35.59 | blitzrage | yah |
| 19:35.59 | blitzrage | I want one :) |
| 19:35.59 | Matt | theory :&) |
| 19:36.12 | blitzrage | I just want a cell + wifi + sip client + calendar that'll sync with something in Windows :) |
| 19:36.43 | twisted[asteria] | ew windows |
| 19:36.45 | twisted[asteria] | ew ew ew |
| 19:36.56 | blitzrage | what, do you use the Linux desktop? now that's gross |
| 19:37.00 | twisted[asteria] | no |
| 19:37.00 | fourcheeze | yeah, I want one that also does IrMC |
| 19:37.11 | twisted[asteria] | OS X :) |
| 19:37.13 | fourcheeze | blitzrage: which "The" linux desktop do you mean? |
| 19:37.19 | blitzrage | fourcheeze: exactly :) |
| 19:37.29 | blitzrage | twisted[asteria]: thats even worse |
| 19:37.30 | _Paulo_ | motorola A1200 runs linux. |
| 19:37.37 | blitzrage | I hate OSX |
| 19:37.40 | twisted[asteria] | blitzrage, i've had 0 crashes in osx |
| 19:37.40 | blitzrage | like... with a passion |
| 19:37.52 | fourcheeze | well there's no accounting for taste |
| 19:37.57 | fourcheeze | my linux desktop is wonderful |
| 19:38.10 | fourcheeze | pretty soon everyone in the office will have one :-) |
| 19:38.10 | twisted[asteria] | my os x desktop is wonderful |
| 19:38.17 | twisted[asteria] | i can run os x apps, x11 apps, etc. |
| 19:38.23 | fourcheeze | yeah OSX is nice |
| 19:38.34 | _Paulo_ | we are a linux only shop here. |
| 19:38.34 | lunk | fourcheeze: you using LTSP or something similar? |
| 19:38.50 | dlynes_office | We're almost a linux only shop |
| 19:38.54 | fourcheeze | lunk: quite possibly, I haven't worked that bit out yet |
| 19:39.06 | dlynes_office | Just need to get a decent vm that quickbooks'll run under, and then we will be |
| 19:39.17 | lunk | freenx is nice too, smooth remotely too |
| 19:39.18 | fourcheeze | got enough on right now ditching exchange and clients |
| 19:39.43 | fourcheeze | we're going to have to have a few windows servers with RDP for a while |
| 19:39.46 | dlynes_office | or maybe convince intuit to port quickbooks to linux :) |
| 19:39.56 | fourcheeze | but workstations will more or less go straight to linux |
| 19:40.04 | fourcheeze | and thin client stuff looks favourite ATM |
| 19:40.42 | fourcheeze | lunk: do you know if sound can go down LTSP ? |
| 19:41.04 | lunk | pretty sure it can |
| 19:41.11 | lunk | never done it personally though |
| 19:41.28 | fourcheeze | I love the idea of just doing everything off the network |
| 19:42.18 | fourcheeze | actually I think with kde and artsd if you start run an app on another box the sound is transparently sent over |
| 19:43.19 | darekj | does anybody compile t38passthrough and it is working? |
| 19:43.20 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:43.33 | *** part/#asterisk shanky (i=jramirez@217.11.114.145) |
| 19:45.14 | *** join/#asterisk saftsack (n=saftsack@p54A7C8EB.dip.t-dialin.net) |
| 19:46.09 | revslowmo | Anyone know of a good way to detect rings on a incoming line with a x101p card, or if even possible. |
| 19:48.25 | dlynes_office | fourcheeze: you have to enable network sound or whatever it's called, and the application on the other needs to be able to support it |
| 19:49.18 | markus99 | can someone direct me to info on how to setup a dialing context that would route the call according to what was dialed |
| 19:49.36 | fourcheeze | markus99: that sounds like a phone system |
| 19:50.36 | markus99 | fourcheeze: what I mean is if I can route the call to different viop providers or zap if local |
| 19:51.25 | fourcheeze | markus99: dial 9 for voip? |
| 19:52.00 | jsaunders | Is remote % in this bad? http://pastebin.ca/56925 |
| 19:53.02 | _Paulo_ | markus99, look for recipes in vopi-info.org |
| 19:53.25 | *** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net) |
| 19:53.40 | fourcheeze | markus99: things like |
| 19:53.40 | fourcheeze | exten => _6xx,1,Dial(Zap/${EXTEN},30) |
| 19:53.51 | _Paulo_ | s/vopi/voip/ |
| 19:55.12 | _Paulo_ | markus99, http://voip-info.org/wiki/view/Asterisk+Configuration+Examples |
| 19:56.35 | *** join/#asterisk cian (n=cian@g5.cian.ws) |
| 19:58.01 | QbY_ | Anyone have an idea as to why I am able to dial out through my server, but no calls come into my phone--they go straight to voicemail.. |
| 19:58.56 | _Paulo_ | QbY_, may be your incoming calls are landing on the wrong context... |
| 19:59.04 | dlynes_office | QbY_: check your logs |
| 19:59.08 | dlynes_office | QbY_: your logs should tell you |
| 19:59.16 | QbY_ | which log? |
| 19:59.21 | dlynes_office | exactly |
| 19:59.26 | dlynes_office | ~docs |
| 19:59.28 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 19:59.28 | mercestes | God has come down...and held at his hand at your Sip packets and said, "Thou shalt not pass!" and blocked your call.... |
| 19:59.34 | mercestes | and the key to your salvation lies in your voicemail. |
| 19:59.47 | dlynes_office | QbY_: read the wiki that you see jbot telling you about |
| 20:00.08 | QbY_ | <PROTECTED> |
| 20:00.08 | QbY_ | <PROTECTED> |
| 20:00.27 | dlynes_office | QbY_: I'm guessing you're using FreePBX? |
| 20:00.38 | QbY_ | no.. used to be an AMP install, but not anymore |
| 20:00.43 | ManxPower | QbY_, the phone is prolly not registered |
| 20:01.04 | QbY_ | under sip show peers, it shows unspecified--as though it isn't registered.. |
| 20:01.14 | QbY_ | yet, i'm baffled as to how i can make calls |
| 20:01.33 | ManxPower | QbY_, um, an unregistered phone can make calls. |
| 20:01.40 | QbY_ | eh!? |
| 20:01.43 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 20:01.43 | QbY_ | that frightens me |
| 20:01.47 | ManxPower | ALL regustration does is tell the server what IP address to send calls for that user/pass |
| 20:01.48 | Assid | err.. my phones gone crazy |
| 20:02.01 | Assid | the start up text 'polycom soundpoint ip' goes on and off |
| 20:02.03 | Assid | in circle |
| 20:02.05 | ManxPower | "sip show peers" |
| 20:02.07 | blitzrage | I'm looking for an application that'll do more than 2 camera video conferencing -- any suggestions? |
| 20:02.25 | blitzrage | i.e. 3 camera vid conf. |
| 20:02.27 | ManxPower | blitzrage, expanding your videosex company again? |
| 20:02.35 | r_evolution | weird. |
| 20:02.36 | blitzrage | ManxPower: yeppers! |
| 20:02.49 | r_evolution | It seems that some of my users are suddenly unable to qualify themselves to * |
| 20:03.03 | r_evolution | but then in a while... they can qualify again... |
| 20:03.21 | blitzrage | r_evolution: are they behind NAT? I get that too |
| 20:03.30 | r_evolution | yeah... some of them are... |
| 20:03.36 | r_evolution | which leads me to my next statement |
| 20:03.46 | r_evolution | the test-box i have here right? |
| 20:03.46 | blitzrage | yah... I get it all the time -- ever since I started using asterisk... and have no idea why it fails :) |
| 20:03.57 | ManxPower | damn it's windy here today |
| 20:04.00 | r_evolution | was behind a symantec firewall... |
| 20:04.22 | Assid | nevermind.. was my power suply |
| 20:04.25 | r_evolution | asterisk was sending udp rapidly enough to the point where the firewall actually detected it as an attack |
| 20:04.29 | r_evolution | and shutdown for a bit |
| 20:04.34 | r_evolution | i wonder if the two are related? O_o |
| 20:04.43 | r_evolution | annnnnnd if they are... wtf can be done about this? |
| 20:05.05 | mercestes | That's ok... |
| 20:05.09 | r_evolution | it irks me... |
| 20:05.23 | r_evolution | i know what we did on the firewall here... basically told it to chill the fuck out... |
| 20:05.27 | blitzrage | probably have to change the code in chan_sip to not try to often... |
| 20:05.41 | mercestes | I've found that Cisco phones running through a Cisco Switch across a Cisco Router cascades Nat ports into the router until the Nat Tables are trashed and the router can no longer pass traffic. |
| 20:05.45 | r_evolution | but -- how is that supposed to work on a remote customer? >_< |
| 20:05.46 | blitzrage | I had to lower the MAX_RETRANS value because it was taking a full minute for things to failover... |
| 20:06.05 | blitzrage | r_evolution: ummm... it doesnt'? |
| 20:06.16 | r_evolution | that's what seems to be happening mercestes |
| 20:06.36 | r_evolution | :-\ |
| 20:06.39 | r_evolution | solution solution... |
| 20:11.26 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 20:12.04 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 20:13.02 | *** join/#asterisk sung (i=sung@216.106.191.180) |
| 20:13.20 | sung | hey there |
| 20:13.33 | sung | in periods of silence, it seems like asterisk isn't sending rtp |
| 20:13.43 | sung | is it possibly to change this behavior? (if that's what is really happening...) |
| 20:14.11 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:14.50 | mercestes | VAD and Silence Supression. |
| 20:15.46 | Assid | i still cant believe there is no way to send a txt message to a phone |
| 20:16.30 | Assid | VoicePulse: you there |
| 20:17.17 | darekj | <PROTECTED> |
| 20:17.42 | protocoldoug | Where do I change the verbosity level so I can see the output from a call to NoOp(${thing})? |
| 20:18.15 | sevard | Assid: SMS? it's email. |
| 20:18.23 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 20:18.23 | _Paulo_ | help set verbose |
| 20:18.28 | Assid | nah... i meant text message to voip phones |
| 20:18.32 | protocoldoug | _Paulo_, many thanks :) |
| 20:18.32 | ManxPower | protocoldoug, "asterisk -rvvv" or "set verbose X" |
| 20:18.43 | Assid | just cant geyt opver asterisk having all that.. but leaving out sip messaging |
| 20:18.45 | sevard | Assid: NOTIFY? |
| 20:18.45 | protocoldoug | ManxPower, also thanks :) |
| 20:18.49 | ManxPower | Assid, of course there is. Use sipsak or something like that |
| 20:18.52 | sevard | ah |
| 20:19.10 | ManxPower | Why do you expect SIP phones to support IAX2 text messages? |
| 20:19.42 | sevard | my only complaint is that presence support isn't in |
| 20:19.48 | sevard | after so many people complaining about it |
| 20:19.49 | Assid | err.. cause of sip phones connecting to sip server.. |
| 20:19.53 | sevard | i don't need it personally |
| 20:20.01 | ManxPower | Assid, SIP messaging did not exist when chan_sip was written. |
| 20:20.12 | ManxPower | so either stop complainging or write some code. |
| 20:20.16 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) |
| 20:20.29 | Assid | if i knew . i realll would |
| 20:20.50 | _Paulo_ | Assid, I can code for a very fair rate... :-) |
| 20:20.51 | sevard | so learn |
| 20:20.53 | oej | Assid: We do support SIP messaging, really |
| 20:20.56 | ManxPower | then post a bounty |
| 20:20.57 | sevard | or pay paulo |
| 20:20.58 | oej | Assid: But only during a call |
| 20:21.13 | ManxPower | oej, He's trying to use sendtext to a polycom |
| 20:21.18 | oej | Check the sendtext() application |
| 20:21.29 | Assid | oej: i tried that too.. |
| 20:21.31 | oej | Aha, well, we support more than many phones :-) |
| 20:21.37 | ManxPower | oej, when I tried sendtext it disconnected the call |
| 20:21.39 | oej | It works with eyebeam and snoms |
| 20:22.48 | Hmmhesays | send me one |
| 20:23.02 | ManxPower | Hmmhesays, you don't buy me booze |
| 20:23.04 | blitzrage | is there anyway of doing multi-point video conferencing? i.e. gnugk, openh323 MCU... ? |
| 20:23.11 | Hmmhesays | heh |
| 20:23.16 | blitzrage | no one seems to say they either support, or don't, support it :) |
| 20:23.23 | blitzrage | keep checking websites... seems to be silently ignored :) |
| 20:25.34 | _Paulo_ | blitzrage, are you running chan_oh323 and gnugk? |
| 20:25.42 | *** join/#asterisk sunnew (n=nurbina2@200.27.54.226) |
| 20:25.43 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 20:25.43 | *** mode/#asterisk [+o denon] by ChanServ |
| 20:25.55 | blitzrage | _Paulo_: I'm not running anything... just trying to figure out what I need -- doesn't need to be Asterisk :) |
| 20:28.13 | sunnew | hello folks. |
| 20:28.53 | sunnew | how do I compile the cdr_odbc.c source code on asterisk@home? |
| 20:29.43 | ManxPower | ~amp |
| 20:29.45 | jbot | it has been said that amp is NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
| 20:35.12 | *** join/#asterisk p1p (i=tjcomp91@mail.comp911.com) |
| 20:35.34 | p1p | Can anyone shed some light on g729 licensing for me? The more I read the less I understand about it. |
| 20:35.46 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-185-148.lsanca.fios.verizon.net) |
| 20:36.07 | blitzrage | p1p: 1 concurrent call == 1 license |
| 20:36.35 | p1p | blitz: how would it be possible to figure out how many concurrent channels might be in use before an installation?? |
| 20:36.58 | blitzrage | p1p: ummm... research... |
| 20:37.08 | blitzrage | like.. you need to know the call habits of the system you're replacing |
| 20:37.22 | blitzrage | how many phones? are they all going to be on a call at once? or only 25% of them? |
| 20:37.35 | blitzrage | you're getting into stats now -- there is no magic answer |
| 20:39.01 | r_evolution | not statistics :( |
| 20:39.01 | p1p | OK well the super simplified version you offered was actually a big help believe it or not :) |
| 20:40.09 | blitzrage | heh :) |
| 20:40.56 | *** join/#asterisk salviadud (n=dude@dsl-201-129-86-188.prod-infinitum.com.mx) |
| 20:42.59 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) |
| 20:43.54 | r_evolution | im seriously confused about this one blitz... |
| 20:43.56 | r_evolution | it's like |
| 20:43.57 | r_evolution | up down |
| 20:43.58 | r_evolution | up |
| 20:43.58 | r_evolution | down |
| 20:43.59 | r_evolution | up |
| 20:44.00 | r_evolution | down |
| 20:44.01 | r_evolution | up |
| 20:44.06 | blitzrage | I hear you... :( |
| 20:44.15 | blitzrage | I get it all the time, if you figure it out, please tell me :) |
| 20:44.28 | r_evolution | i think it has something to do with how rapidly the packets are sent |
| 20:44.31 | blitzrage | I think it'll require some SIP debugging to track the packets and see why it thinks packets aren't coming back... |
| 20:44.41 | blitzrage | r_evolution: I bet thats a good guess |
| 20:44.42 | r_evolution | that was the problem with the firewall here... |
| 20:44.55 | r_evolution | i dont think it doesnt think they're coming back |
| 20:45.00 | r_evolution | i think the router is just shutting down |
| 20:45.08 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) |
| 20:45.10 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-246.apple.com) |
| 20:45.11 | blitzrage | hrmmm |
| 20:45.16 | blitzrage | now you got me curious :) |
| 20:45.23 | r_evolution | but the question being -- why do some do it... but not others? |
| 20:45.28 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
| 20:45.33 | blitzrage | r_evolution: could be the device... |
| 20:45.39 | r_evolution | is it something in the Broadband Provider? |
| 20:45.45 | r_evolution | i mean they ARE the ones assigning the private IP |
| 20:46.03 | Assid | okay when it says > requested format = ulaw, > actual format = gsm, > host prefs = (gsm), > priority = mine |
| 20:46.03 | r_evolution | oh but get this... |
| 20:46.10 | Assid | what does it use gsm /ulaw? |
| 20:46.12 | r_evolution | in the past... we've had ATAs just... DIE. |
| 20:46.19 | r_evolution | like... the internet falls off |
| 20:46.21 | r_evolution | and never known why |
| 20:46.28 | r_evolution | now i find myself wondering if it's related... |
| 20:46.31 | blitzrage | I doubt it --I get it with devices in all sorts of situations, and none of them have private IPs assigned by an ISP |
| 20:46.31 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 20:46.54 | r_evolution | so you're definitely going with my first theory -- that it's something in the equipment itself |
| 20:47.12 | r_evolution | b/c you know... it actually IS sending the udp packets at a decent clip... |
| 20:47.19 | r_evolution | not like one a millisec |
| 20:47.36 | blitzrage | agreed |
| 20:47.57 | blitzrage | let me look at chan_sip and see if I can find where and how it determines an end point is unreachable |
| 20:48.22 | blitzrage | asterisk 1.2.x? |
| 20:48.27 | r_evolution | yeah |
| 20:48.28 | r_evolution | .7 |
| 20:48.56 | r_evolution | so here's the thing though... i dont think it's *just* that it's unreachable... i think it's either A - closing down the router |
| 20:49.03 | r_evolution | or ignoring all requests from the IP |
| 20:49.20 | blitzrage | well, we need to get a packet trace from both ends to see where the packets are being dropped |
| 20:49.37 | blitzrage | if Asterisk isn't seeing packets come back, then that totally screws with how Asterisk is going to work :) |
| 20:49.53 | blitzrage | if its sending too many in too short a time, then it'd be nice to control the flow |
| 20:50.27 | r_evolution | that's what im sayin... |
| 20:50.31 | r_evolution | i think that's what the problem is... |
| 20:50.34 | r_evolution | too many too quick. |
| 20:50.51 | r_evolution | fux i wish i still had the log from the router here... |
| 20:50.52 | sevard | Does anyone have a problem with ZAP lines not hanging up? |
| 20:50.53 | blitzrage | lets see if I can't figure this out (I can't read code that well :)) |
| 20:52.32 | r_evolution | see here's the second part... |
| 20:52.43 | r_evolution | if the router is intentionally dropping those packets from asterisk |
| 20:52.49 | r_evolution | thus leading asterisk to believe the phone is un-reachable |
| 20:53.03 | r_evolution | why is it that it suddenly becomes qualified? |
| 20:53.35 | r_evolution | Thought - SIP Reg every 3600 secs... so perhaps when it registers again... asterisk realizes it's there... until the router is over-loaded again? |
| 20:53.35 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:54.25 | *** part/#asterisk sunnew (n=nurbina2@200.27.54.226) |
| 20:55.21 | Assid | err. how do you get a number that a sip user has called? |
| 20:55.33 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-246.apple.com) |
| 20:55.52 | blitzrage | r_evolution: just means a packet got through |
| 20:56.17 | ManxPower | Assid, ${EXTEN} same as for any other |
| 20:56.19 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 20:56.23 | blitzrage | r_evolution: no, you wouldn't think so |
| 20:56.46 | r_evolution | so i wonder if it'll do it if qualify is set to like 20 seconds |
| 20:57.01 | ManxPower | r_evolution, so that would be 20000000 |
| 20:57.01 | blitzrage | I had mine set to over a minute, and it still does it |
| 20:57.04 | *** join/#asterisk briiiiian (i=briiiiia@custserv2.netaccnt.net) |
| 20:57.12 | ManxPower | you need QUALIFY SMOOTHING |
| 20:57.18 | blitzrage | oh yah! |
| 20:57.23 | blitzrage | I totally forgot about that shiznit |
| 20:57.25 | ManxPower | I don't know if it's in 1.2, I thought it was, and if it is, it might not be documented. |
| 20:57.36 | ManxPower | <-- smarter than he looks |
| 20:57.38 | Assid | ManxPower: no no i meant from console.. i wanna know who the person called |
| 20:57.57 | ManxPower | Assid, you can't unless you put a NOOP in your dialplan |
| 20:58.08 | blitzrage | 20,000 ms == 20 s |
| 20:58.11 | r_evolution | thats what im saying blitz... if it means a packet got through... why did that packet get through... |
| 20:58.13 | r_evolution | yessir |
| 20:58.25 | Assid | can the manager do it ? |
| 20:58.26 | r_evolution | packet got through because it was initiated on the client side |
| 20:58.34 | r_evolution | or so it would potentially seem |
| 20:58.35 | blitzrage | ManxPower: you need to bone up on your metric :) |
| 20:58.44 | blitzrage | could be the re-register |
| 20:58.48 | r_evolution | i.e. a phone call was attempted... |
| 20:58.49 | ManxPower | blitzrage, I need to get over my cold and off the cold meds |
| 20:59.19 | r_evolution | chya. |
| 21:00.36 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:03.03 | ManxPower | A single lost UDP packet could be caused by any number of things and without qualify smoothing all it takes is 1 lost packet |
| 21:03.26 | blitzrage | ManxPower: exactly |
| 21:03.30 | blitzrage | that makes more sense |
| 21:04.19 | ManxPower | I don't really don't need it because I don't run SIP over the internet. Asterisk's lack of SIP jitterbuffer and all that |
| 21:04.22 | blitzrage | ok ... I need f00d! |
| 21:04.29 | ManxPower | well lack of RTP jitterbuffer, at least |
| 21:04.32 | blitzrage | ManxPower: hear that :) |
| 21:04.35 | r_evolution | define qualify smoothing? |
| 21:04.58 | ManxPower | r_evolution, google "site:lists.digium.com qualify smoothing" |
| 21:05.01 | blitzrage | r_evolution: I think it takes an average over a few packets to determine if it really is a low qualify value, or if we're still in a range close to the qualify value |
| 21:05.57 | r_evolution | yeah but that's only pertinent to iax, is it not? |
| 21:06.04 | blitzrage | SIP |
| 21:06.07 | ManxPower | r_evolution, I hope not. |
| 21:06.30 | r_evolution | http://www.google.com/search?hl=en&q=site%3Alists.digium.com+qualify+smoothing&btnG=Google+Search |
| 21:07.37 | r_evolution | no mention of sip |
| 21:09.44 | ManxPower | just confirmed chan_sop does not have qualify smoothing. I guess it sucks to be you. |
| 21:09.59 | r_evolution | some days ;) |
| 21:10.05 | r_evolution | some days it's pretty fun to be me |
| 21:10.12 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
| 21:10.26 | blitzrage | sucks to be me too |
| 21:10.35 | blitzrage | wonder how easier it would be to port to chan_sip :) |
| 21:10.36 | ManxPower | another reason not to deploy asterisk SIP/RTP in production for Internet |
| 21:11.20 | r_evolution | O_o |
| 21:11.33 | r_evolution | and what would you propose deployment of? iax alone? |
| 21:11.50 | ManxPower | r_evolution, Me? I would not deploy Asterisk for VoIPoInternet at all. |
| 21:12.21 | ManxPower | r_evolution, turn off qualify in sip.conf, make your devices register every 60 seconds. |
| 21:12.55 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 21:13.07 | *** join/#asterisk mgob (n=goldenol@65.171.196.23) |
| 21:13.07 | mgob | hi |
| 21:13.20 | mgob | anyone have any recommendations for 5 lines from a SIP provider? |
| 21:13.53 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 21:13.58 | r_evolution | wb manx. |
| 21:14.07 | Ariel_ | mgob, 5 numbers? or up to 5 inbound calls or outbound |
| 21:14.49 | Dr-Linux | need to discuss something about loadzone and defaultzone |
| 21:14.51 | blitzrage | difference between multiple numbers, and the number of simultaneous trunks is a big difference |
| 21:15.00 | *** join/#asterisk sunnew (n=nurbina2@200.27.54.226) |
| 21:15.15 | blitzrage | r_evolution: I would agree with ManxPower |
| 21:15.38 | blitzrage | re: qualify=no and register... I basically do that now :) |
| 21:15.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:15.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:15.57 | r_evolution | i would be amicable to any suggestions made by people who are more knowledgeable than me :) |
| 21:16.11 | Dr-Linux | In pakistan i'm using loadzone=us and defaultzone=us , everything works fine, but i'm not getting caller ID, what could be the reason? |
| 21:16.21 | r_evolution | ye olde... can't learn if you don't ask theory. |
| 21:16.42 | *** join/#asterisk h323_help (n=FreePBX0@HSE-MTL-ppp71958.qc.sympatico.ca) |
| 21:16.51 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:17.29 | h323_help | I am looking for _Paulo_ is he here? |
| 21:17.44 | ManxPower | !seen _Paulo_ |
| 21:17.51 | ManxPower | ~seen _Paulo_ |
| 21:17.54 | jbot | _paulo_ is currently on #asterisk (10h 12m 53s). Has said a total of 58 messages. Is idling for 52m 20s, last said: 'blitzrage, are you running chan_oh323 and gnugk?'. |
| 21:18.13 | h323_help | thanks manxpower |
| 21:18.26 | _Paulo_ | I in a business meeting right now |
| 21:18.47 | h323_help | Paulo sup ;;; It's me again ;;; take your time ;;; Sorry I got disconnected before |
| 21:19.10 | h323_help | I am the person who asked for H323 and asterisk@home add-on package before |
| 21:19.29 | _Paulo_ | got it working? |
| 21:19.36 | h323_help | nope |
| 21:19.39 | h323_help | I installed it again |
| 21:19.44 | h323_help | and there is no error |
| 21:19.58 | h323_help | It's really confusing when there is no error |
| 21:20.01 | h323_help | and it doesn't install |
| 21:20.17 | h323_help | the funny thing is that absolutly NO ONE else tried it |
| 21:20.19 | h323_help | lol |
| 21:21.44 | h323_help | I prolly go for digium one ;; but I don't know if I succed ;;; caz I am using Asterisk@home |
| 21:22.09 | jaybuffet | my asterisk just died for some reason |
| 21:22.23 | _Paulo_ | now you found chan_oh323.so ? |
| 21:22.28 | jaybuffet | everything looks fine on the server when it starts |
| 21:22.51 | h323_help | nope it's still not in the system |
| 21:22.52 | jaybuffet | but it doesnt show any of the phones registering .. and i can't even get the 1000 demo exten to work |
| 21:23.01 | h323_help | It doesn't make sense at all |
| 21:23.42 | h323_help | I am downloading h323 from digium now ;;; how should I install that? |
| 21:23.45 | h323_help | use makefile?> |
| 21:24.08 | _Paulo_ | unpack the tarball |
| 21:24.51 | _Paulo_ | then run .configure --prefix=/usr inside the source dir |
| 21:25.03 | _Paulo_ | and then make install |
| 21:25.18 | _Paulo_ | but i dont know about *@home. |
| 21:25.21 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:25.28 | _Paulo_ | this is how i make it here. |
| 21:25.32 | h323_help | I will try it ;;;; thanks for the help buddy |
| 21:25.39 | h323_help | where is the source dir? |
| 21:25.53 | h323_help | u mean the place i put the files? |
| 21:25.54 | r_evolution | sup justin... where've you been? |
| 21:26.03 | justinu | workin |
| 21:26.18 | r_evolution | fun. :-\ |
| 21:26.21 | justinu | no kidding |
| 21:26.31 | _Paulo_ | when you run tar -zxvf tarbal.tar.gz |
| 21:26.36 | r_evolution | better than being homeless |
| 21:27.02 | Dr-Linux | anybody have any clue about my question? |
| 21:27.11 | Dr-Linux | In pakistan i'm using loadzone=us and defaultzone=us , everything works fine, but i'm not getting caller ID, what could be the reason? |
| 21:28.04 | *** join/#asterisk jaybuffet (n=jperron@rrcs-24-227-53-138.se.biz.rr.com) |
| 21:28.13 | jaybuffet | nm... im dumb |
| 21:28.44 | *** part/#asterisk QbY_ (n=Kelvin@66.236.241.68.ptr.us.xo.net) |
| 21:28.45 | jaybuffet | my wireless had moved to another router |
| 21:29.35 | *** join/#asterisk ghost99 (n=neville@222-153-92-225.jetstream.xtra.co.nz) |
| 21:29.42 | _Paulo_ | I got to go. |
| 21:29.58 | _Paulo_ | h323_help, I can help you tomorrow |
| 21:30.14 | h323_help | okey _Paulo_ thanks a lot for all the help today |
| 21:30.39 | h323_help | !seen _Paulo_ |
| 21:31.03 | h323_help | ~seen _Paulo_ |
| 21:31.07 | jbot | _paulo_ <n=Paulo@c906d4b0.virtua.com.br> was last seen on IRC in channel #asterisk, 1m 9s ago, saying: 'h323_help, I can help you tomorrow'. |
| 21:31.33 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:31.45 | batphone | ~seen bugz |
| 21:31.46 | jbot | bugz <n=will@69.15.174.114> was last seen on IRC in channel #asterisk, 22h 54m 27s ago, saying: 'lol..'. |
| 21:31.52 | batphone | haha |
| 21:32.25 | generalhan | ~seen generalhan |
| 21:32.26 | jbot | generalhan is currently on #asterisk (3h 55m 20s). Has said a total of 31 messages. Is idling for 1s, last said: '~seen generalhan'. |
| 21:32.49 | generalhan | ~jbot rocks !! |
| 21:33.05 | mercestes | ~seen Janet Jackson's boob? |
| 21:33.07 | jbot | mercestes: i haven't seen 'janet jackson's boob' |
| 21:33.10 | r_evolution | O_O |
| 21:33.16 | mercestes | Aww...poor Jbot. |
| 21:33.22 | mercestes | I thought everyone had seen her boob. |
| 21:33.22 | Dr-Linux | who will answer my question :S jbot? |
| 21:33.33 | Dr-Linux | ~Pakistan caller id |
| 21:38.13 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:38.49 | justinu | r_evolution: so who's homeless? |
| 21:39.07 | r_evolution | lots of people. |
| 21:39.14 | r_evolution | myself not among them... yourself neither |
| 21:39.34 | r_evolution | but i've been there before... in my 'hoodlum' days :-\ |
| 21:39.42 | justinu | yikes |
| 21:40.19 | r_evolution | it happens :) |
| 21:40.47 | r_evolution | worse has happened... so yeah... work sucks some days -- but at least you're not homeless |
| 21:41.03 | justinu | stupid work vpn won't work either |
| 21:42.12 | r_evolution | :( |
| 21:42.33 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 21:42.39 | *** part/#asterisk brif8 (n=Administ@lazyjtrainingcenter.com) |
| 21:42.42 | Assid | hr,mmm anyone here hosted with meganetserve? |
| 21:45.07 | unmanaged | ok. Again with the very little docs on a few things in asterisk. In zapata.conf there is the 'prilocaldialplan=' option.... I need to know the correct way to use this setting but I am not shure what it does .... within respect to asterisk/zaptel hardware ..... |
| 21:45.47 | justinu | it changes a field in the called party number IE in the Q931 setup message |
| 21:46.17 | justinu | if you don't know what it is, mostly likely it's not relavent to you |
| 21:48.55 | unmanaged | what field ? |
| 21:49.39 | justinu | the number type field, i think |
| 21:49.46 | generalhan | ~slap |
| 21:50.00 | generalhan | Damnit Jbot slap someone |
| 21:50.07 | *** join/#asterisk epablo (n=epablo@200.84.4.217) |
| 21:50.19 | epablo | Hi people.. how's it going? |
| 21:50.35 | jsaunders | Anyone had quality/packet loss issues w/ win32 idefisk soft client? Latest build is *ROUGH* w/ asterisk, any codec, for me atleast. Switched to Firefly and it cleared right up. testing was confirmed w/ 'iax2 show netstats' output, and a distinct difference in audio quality. |
| 21:51.17 | generalhan | jbot, no |
| 21:51.18 | jbot | YES |
| 21:51.32 | generalhan | jbot, yes |
| 21:51.33 | jbot | You don't say! |
| 21:51.37 | mercestes | jbot, no |
| 21:51.38 | jbot | YES |
| 21:51.41 | mercestes | supersweet |
| 21:51.44 | generalhan | jbot, oh i DO say |
| 21:51.45 | jbot | You DO say? |
| 21:51.51 | mercestes | ~slap Mercestes |
| 21:51.53 | jbot | ACTION slaps Mercestes, keep your grubby fingers to yourself! |
| 21:51.53 | generalhan | jbot, you know it |
| 21:51.57 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 21:51.57 | mercestes | Awesome. |
| 21:52.09 | mercestes | ~slap generalhan |
| 21:52.10 | jbot | ACTION slaps generalhan, keep your grubby fingers to yourself! |
| 21:52.12 | epablo | Has anyone used MFC/R2 with Digium cards |
| 21:52.22 | generalhan | yea I Love Jbot |
| 21:52.43 | generalhan | ~slap jbot |
| 21:52.44 | jbot | ACTION slaps jbot, keep your grubby fingers to yourself! |
| 21:52.53 | *** join/#asterisk Eggplant (n=none@dsl-216-155-212-221.cascadeaccess.com) |
| 21:54.52 | justinu | ~abuse |
| 21:54.54 | jbot | abuse is, like, http://www.rfc-ignorant.org/policy-abuse.php or RFC 2142 or echo Shut your festering gob, you tit! | mail abuse |
| 21:55.29 | mercestes | ~abuse jbot |
| 21:55.30 | jbot | ACTION smacks jbot across the face. "Take that, sucker!" |
| 21:55.37 | mercestes | LOL |
| 21:55.52 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 21:55.57 | Nivex | ~abuse me |
| 21:55.58 | jbot | ACTION smacks nivex across the face. "Take that, sucker!" |
| 21:56.06 | Nivex | lame... it needs random actions |
| 21:56.24 | mercestes | ~flogger Nivex |
| 21:56.30 | mercestes | ............ |
| 21:56.32 | mercestes | *sighs* |
| 21:56.38 | *** part/#asterisk epablo (n=epablo@200.84.4.217) |
| 21:58.35 | justinu | anyone here from miami? |
| 21:58.45 | justinu | ~miami |
| 21:58.47 | jbot | miami is probably the northern part of cuba |
| 21:59.22 | CunningPike | lol |
| 22:00.30 | mercestes | ~texas |
| 22:00.32 | jbot | texas is, like, a fine place for y'all to have a big ol' time |
| 22:00.38 | mercestes | LMAO |
| 22:00.48 | stack_ | ~ohio |
| 22:00.52 | mercestes | ~Amsterdam |
| 22:00.54 | jbot | methinks amsterdam is cool especially if you like smoking pot |
| 22:01.05 | mercestes | Bwhahaha....that's awesome. |
| 22:01.15 | stack_ | oh i see, no love for ohio... whatever! |
| 22:01.21 | justinu | put it in, stack |
| 22:01.28 | stack_ | ? |
| 22:01.55 | justinu | jbot, ohio is a state no one in #asterisk loves |
| 22:01.56 | jbot | justinu: okay |
| 22:02.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) [NETSPLIT VICTIM] |
| 22:02.18 | mercestes | woah. |
| 22:02.18 | stack_ | ah i see |
| 22:02.18 | stack_ | ~ohio |
| 22:02.19 | jbot | i guess ohio is a state no one in #asterisk loves |
| 22:02.20 | *** part/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 22:02.54 | stack_ | there, that's better |
| 22:02.54 | justinu | jbot, no ohio is cleveland rocks! |
| 22:02.56 | jbot | okay, justinu |
| 22:02.56 | *** join/#asterisk pjz (n=pj@zachs.place.org) [NETSPLIT VICTIM] |
| 22:02.56 | mercestes | ~ohio |
| 22:02.57 | jbot | from memory, ohio is cleveland rocks! |
| 22:02.57 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) [NETSPLIT VICTIM] |
| 22:02.59 | justinu | jbot, forget ohio |
| 22:03.16 | stack_ | night everyone |
| 22:03.27 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) [NETSPLIT VICTIM] |
| 22:04.21 | mercestes | jbot, mercestes is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. |
| 22:04.22 | jbot | okay, mercestes |
| 22:04.26 | mercestes | ~mercestes |
| 22:04.28 | jbot | hmm... mercestes is the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. |
| 22:04.32 | mercestes | YES! |
| 22:04.52 | mercestes | I love, Jbot. |
| 22:04.55 | generalhan | me too |
| 22:04.56 | generalhan | lol |
| 22:05.06 | justinu | ~asterboy |
| 22:05.08 | jbot | extra, extra, read all about it, asterboy is a weed smoker |
| 22:05.28 | generalhan | ~justinu |
| 22:05.29 | jbot | justinu is, like, some other d00d |
| 22:05.34 | generalhan | haha |
| 22:06.31 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 22:06.45 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) [NETSPLIT VICTIM] |
| 22:06.48 | unmanaged | blah |
| 22:07.22 | *** join/#asterisk h323_help (n=FreePBX0@HSE-MTL-ppp71958.qc.sympatico.ca) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk amorith (n=nahirean@unaffiliated/nahirean) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) [NETSPLIT VICTIM] |
| 22:07.22 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 22:07.23 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk miztic (n=gerard@rarcoa.com) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk mut (n=animenod@65.111.222.120) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk Falle (n=falle@falle.se) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk rabelais (n=blank@santamonica-cuda4-24-55-43-249.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk duckz (n=duckz@193.192.47.26) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-252-202.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 22:07.23 | *** join/#asterisk reth (n=reth@hjaerterdam.csbnet.se) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk flynux (n=prout@2a01:38:0:0:0:0:0:1) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 22:07.23 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk RevK-Busy (n=RevK@excalibur.bracknell.aaisp.net.uk) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk many (i=many@krikkit.ukeer.de) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk jpabuyer (i=macpato@blackbird.tecnoera.com) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk fontp (i=cecil@unon.net) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk monthos (n=monthos@pewp.net) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk enots (i=dimka@freelsd.net) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk vgster[away] (n=vgster@84.18.199.68) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk citats (n=james@69.54.200.117) [NETSPLIT VICTIM] |
| 22:07.24 | *** join/#asterisk jesster (i=jesster@jesster.org) [NETSPLIT VICTIM] |
| 22:07.38 | *** join/#asterisk Ahrimanes (n=michael@62.61.133.90.generic-hostname.arrownet.dk) |
| 22:07.38 | unmanaged | netsplit wt f |
| 22:08.00 | Nivex | links go bouncy bouncy |
| 22:08.21 | brodiem | Is there a way of forcing additional cmds in a dial plan after executing Dial()? Basically, I need an ext to ring all phones, but it will enable call waiting before doing the Dial(SIP/001&SIP/002&etc), and I need it to turn off the call waiting once the Dial() has been completede |
| 22:09.43 | unmanaged | justinu.... hmm I need to know what changes when you change prilocaldialplan... guess it is time to go grep chanzap |
| 22:10.21 | justinu | unmanaged: yeah, the source is the only true authoritative source |
| 22:10.32 | justinu | s/source/source code/ |
| 22:10.37 | sevard | does anyone have any 1800 lines that have do continious audio |
| 22:10.45 | sevard | i have to stress test 8 lines at once |
| 22:11.07 | mercestes | 1-800-asshole |
| 22:11.20 | sevard | you're full of it |
| 22:11.25 | mercestes | it's pretty continuous...about 5 minutes. |
| 22:11.28 | mercestes | no, try it. |
| 22:11.33 | mercestes | it's funny |
| 22:11.36 | sevard | i have 1-800-555-tell |
| 22:11.41 | sevard | which plays for about 3 minutes |
| 22:11.44 | sevard | i need longer |
| 22:11.46 | mercestes | Just...don't do it on speakerphone. |
| 22:12.06 | unmanaged | yep |
| 22:12.09 | unmanaged | it workes |
| 22:12.10 | sevard | great, have any other suggestions? :) |
| 22:12.11 | unmanaged | hehehe |
| 22:12.16 | unmanaged | hmm |
| 22:12.16 | mercestes | :) |
| 22:12.24 | mercestes | told ya. |
| 22:12.27 | unmanaged | let me see |
| 22:12.41 | unmanaged | I will try and find a number |
| 22:12.42 | unmanaged | brb |
| 22:12.47 | sevard | w00t |
| 22:13.32 | mercestes | I was using it to test 800 dialing....dialed it at random on speaker phone in a crowded office.....not a good idea. |
| 22:13.39 | sevard | heh |
| 22:13.44 | sevard | all of these phones are going to be on speakerphone |
| 22:13.52 | mercestes | That would be an interesting effect. |
| 22:13.56 | mercestes | stereo even |
| 22:14.08 | sevard | heh |
| 22:15.14 | Dr-Linux | any expert active? :) |
| 22:15.26 | unmanaged | give me a sec |
| 22:15.30 | unmanaged | gota find it |
| 22:15.44 | sevard | No rush man |
| 22:15.59 | sevard | I'm just configuring one of my servers to take a 800, only thing i can only do two concurrent tests |
| 22:20.01 | unmanaged | does it have to be a 800 #? |
| 22:20.43 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 22:20.48 | sevard | it does |
| 22:20.52 | sevard | well |
| 22:20.55 | sevard | it has to be toll free |
| 22:22.16 | Dr-Linux | In pakistan i'm using loadzone=us and defaultzone=us , everything works fine, but i'm not getting caller ID, what could be the reason? |
| 22:23.13 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 22:28.19 | sevard | unmanaged: anything? |
| 22:28.22 | unmanaged | I have a bunch but I can't find one jsut yet that keeps a call up for that long |
| 22:28.31 | unmanaged | if ya wanan wait I can keep going |
| 22:29.07 | mercestes | Try Microsoft Tech support. |
| 22:29.19 | justinu | heh |
| 22:29.22 | mercestes | It's toll free and provides continuous on hold music for atleast six hours. |
| 22:29.30 | mercestes | Or the IRS. |
| 22:30.50 | sevard | mercestes: hahaha |
| 22:32.25 | mercestes | festival(SayText("Dear Mr. Gates. We are calling you from an open source PBX on a Linux operating system, and speaking to you through an open source voice synthisys program. Total software cost, $0.00. This call has cost you 32 cents.") |
| 22:32.36 | sevard | Ruffles. |
| 22:32.40 | sevard | flite :) |
| 22:34.22 | *** part/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 22:35.11 | mercestes | Is there a way to break back out of Comedian Mail and enter back into the extensions.conf dialplan? |
| 22:35.45 | bkw_ | being tied to linux is a pain in the ass |
| 22:36.33 | mercestes | People are complaining that once they hit voicemail they have ot hang up and call back to try another extension. |
| 22:36.35 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 22:37.08 | unmanaged | whoa that is a WTF # 800:213.8378 |
| 22:37.23 | ManxPower | mercestes, perhaps you have not read "show application voicemail" and/or "show application voicemailmain" |
| 22:37.28 | CunningPike | mercestes: Put an 'o' extension in the context from which VoiceMail was called - then they can press '0' to escape |
| 22:38.03 | CunningPike | mercestes: That will transfer them to the 'o' extension and your dialplan can handle it in any way you choose |
| 22:38.36 | mercestes | Sweet! Thanks..you both rock. |
| 22:39.19 | *** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com) |
| 22:40.08 | Malthus | hi all |
| 22:40.21 | Malthus | how do I get call waiting to work with SIP channels? |
| 22:40.42 | Juggie | it just works so long as the phone supports it |
| 22:40.48 | Malthus | when I try to call a SIP channel that is in use it just gives unavailable |
| 22:41.09 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 22:41.20 | unmanaged | sorry no luck |
| 22:41.46 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 22:43.44 | mercestes | Oh well, *waves* bye guys. |
| 22:43.46 | mercestes | ~mercestes |
| 22:43.48 | jbot | [mercestes] the almighty dark overlord. Worship him! Worship or lament and suffer! All hail Mercestes! Dark lord of existance. |
| 22:43.50 | mercestes | muhhahaha |
| 22:47.19 | Malthus | I have several phones, and SIP gateways and it call waiting doesn't work on any of them |
| 22:48.07 | generalhan | jbot, mercestes is The almighty white overlord. Point at him! Point and Laugh or suffer! All laugh at Mercentes! |
| 22:48.08 | jbot | ...but mercestes is already something else... |
| 22:48.16 | generalhan | jbot, change it then |
| 22:48.25 | generalhan | jbot, so ? |
| 22:48.27 | jbot | so what? |
| 22:48.34 | generalhan | jbot, sew buttons |
| 22:48.43 | CunningPike | Malthus: Your user agents are likely set up to only accept one call per line key |
| 22:53.34 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 22:55.08 | Malthus | CunningPike, is that an asterisk setting or on the user agent itself? |
| 22:55.42 | CunningPike | Malthus: On the UA - for example, Polycoms have a setting for this in their config |
| 22:56.08 | CunningPike | Asterisk will present as many calls as are destined for a given UA - it's the UA that returns 'Busy' |
| 22:56.48 | CunningPike | Speaking of Polycoms, is there a 'good' setting for qualify= for Polycom phones on a LAN? |
| 22:58.32 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-58-202.cybersurf.com) |
| 23:06.28 | websae | does anyone here work for a major VoIP company? either manufacturer or service provider... |
| 23:06.37 | sevard | why? |
| 23:06.43 | websae | just curious |
| 23:07.14 | *** join/#asterisk Falle (n=falle@falle.se) [NETSPLIT VICTIM] |
| 23:09.38 | Dr-Linux | websae: was looking for you :) |
| 23:10.14 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 23:10.58 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-116-97.buckeyecom.net) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk mtaht3 (n=m@reserve-64-79-114-30.wiline.com) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk SwK_ (n=Silik0nJ@12-219-147-107.client.mchsi.com) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk bkw_ (n=brian@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk miztic (n=gerard@rarcoa.com) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk rabelais (n=blank@santamonica-cuda4-24-55-43-249.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk duckz (n=duckz@193.192.47.26) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-252-202.telkomadsl.co.za) [NETSPLIT VICTIM] |
| 23:10.58 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 23:10.59 | *** join/#asterisk faberk (n=faberk@host54-228.pool80181.interbusiness.it) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk flynux (n=prout@2a01:38:0:0:0:0:0:1) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-39-36.dsl.tul2ok.sbcglobal.net) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk CMike (i=daemon@c-874071d5.116-1-64736c10.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk RevK-Busy (n=RevK@excalibur.bracknell.aaisp.net.uk) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk jpabuyer (i=macpato@blackbird.tecnoera.com) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk fontp (i=cecil@unon.net) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk monthos (n=monthos@pewp.net) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk enots (i=dimka@freelsd.net) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk citats (n=james@69.54.200.117) [NETSPLIT VICTIM] |
| 23:10.59 | *** join/#asterisk jesster (i=jesster@jesster.org) [NETSPLIT VICTIM] |
| 23:12.15 | *** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com) |
| 23:14.59 | *** join/#asterisk ivanfm (n=ivanfm@201-42-10-225.dsl.telesp.net.br) |
| 23:18.36 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 23:18.51 | SuperLag | "Fewest Completed Calls".... how is this calculated? does it reset every 24 hours? |
| 23:19.54 | *** join/#asterisk willcampos123 (n=willcamp@198.87.100.5) |
| 23:21.03 | willcampos123 | hello |
| 23:21.16 | willcampos123 | I am trying to compile h323 with pwlib and openh323 |
| 23:21.23 | willcampos123 | but what i got is: |
| 23:21.23 | websae | h323 yuck |
| 23:21.34 | willcampos123 | make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.2.7.1/channels/h323' |
| 23:21.34 | willcampos123 | g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_USE_PRAGMA -D_REENTRANT -Wall -fPIC -DPIC -I/usr/local/share/pwlib//include -DPTRACING -I/usr/src/openh323_v1_18_0/include -Os -pipe -felide-constructors -c ast_h323.cxx -o ast_h323.o |
| 23:21.34 | willcampos123 | ast_h323.cxx:1:1: warning: "_GNU_SOURCE" redefined |
| 23:21.34 | willcampos123 | <command line>:4:1: warning: this is the location of the previous definition |
| 23:22.01 | willcampos123 | <websae> I know!! but I have to live with it! |
| 23:22.45 | websae | get a switch and move it to sip |
| 23:23.25 | Malthus | anyone know if the pa168 phones support call waiting? |
| 23:23.43 | willcampos123 | Hope so some day! but now in a near future |
| 23:29.00 | CunningPike | Do zapata.conf settings take effect on a reload? |
| 23:32.01 | Malthus | ok, pa168 supports call waiting |
| 23:32.09 | Malthus | and so does asterisk |
| 23:32.13 | Malthus | and I am just dumb |
| 23:33.05 | justinu | CunningPike: only some. |
| 23:33.33 | CunningPike | justinu: I hope context= is one of them........ |
| 23:33.49 | justinu | i think so... however signalling, and gains I know are not. |
| 23:33.49 | SuperLag | For call queues... I'm reading the docs on the voip-info.org site and I've got a question. "Fewest Completed Calls".... how is this calculated? does it reset every 24 hours? |
| 23:34.45 | CunningPike | justinu: Good news - I'll try it and see :) |
| 23:34.57 | CunningPike | Later, gents..... |
| 23:35.02 | justinu | hasta |
| 23:35.14 | SuperLag | It says, "fewestcalls: ring the one with the fewest completed calls from this queue", but it doesn't say when/how it calculates that value |
| 23:41.18 | Malthus | the count probably restarts when asterisk is restarted |
| 23:50.59 | *** join/#asterisk tuxd00d (n=tuxinato@netblock-68-183-112-22.dslextreme.com) |
| 23:54.35 | *** join/#asterisk ivanfm (n=ivanfm@201-26-6-254.dsl.telesp.net.br) |
| 23:56.09 | *** join/#asterisk frk2 (n=faraz@202.141.251.102) |
| 23:56.23 | frk2 | whatsup people |
| 23:56.47 | frk2 | my TDM cards randomly hangup calls (in one location) |
| 23:56.55 | frk2 | anybody had experience with that? |
| 23:57.04 | frk2 | tried tinkering with busydetect, everything |
| 23:57.11 | frk2 | the calls randomly disconnect |
| 23:57.19 | frk2 | even inbound calls disconnect- which is REALLY weird |
| 23:57.49 | frk2 | any ideas? |
| 23:59.30 | *** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds) |