| 00:02.40 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:06.08 | smackus | looking for a little help with sangoma. each time i reboot the machine my T1s show back up as unconfigured. I have to run wanrouter start and ztcfg each time. what do I need to change this? |
| 00:09.24 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 00:14.26 | *** join/#asterisk tuck3r_ (n=tucker@unaffiliated/tuck3r) |
| 00:14.32 | XARiUS | after reading the wiki.. regarding Queue().. then reading show application queue.. and then reading queues.sample.conf, it's enough to throughly confuse anyone. |
| 00:14.32 | XARiUS | sigh |
| 00:21.39 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 00:24.59 | *** join/#asterisk saftsack (n=saftsack@p54A7E46D.dip.t-dialin.net) |
| 00:25.56 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 00:26.57 | De_mon | XARiUS stick to show application queue |
| 00:27.41 | smackus | anyone here using sangoma t1 cards? |
| 00:29.08 | dlynes_laptop | smackus: yep |
| 00:29.42 | smackus | did you see my earlier question regarding rebooting? |
| 00:29.45 | dlynes_laptop | nope |
| 00:29.55 | dlynes_laptop | if you look back four minutes ago |
| 00:29.59 | dlynes_laptop | you'll see i just signed in |
| 00:30.10 | smackus | sorry, was stepped away for a sec. |
| 00:30.12 | smackus | anyhow... |
| 00:30.30 | smackus | whenever i reboot, i have to restart wanrouter and then re-run ztcfg. |
| 00:30.39 | dlynes_laptop | and? |
| 00:30.40 | smackus | is this normal? or did i possibly miss a step? |
| 00:30.53 | dlynes_laptop | Yeah...you didn't write a startup script |
| 00:31.26 | smackus | ok... can do that. I just expected it to work like with the digium card. |
| 00:31.32 | smackus | so that is normal to require that? |
| 00:31.35 | dlynes_laptop | I guess sangoma assumes everyone installing their hardware is a competent sys admin that knows how to write startup scripts |
| 00:32.00 | dlynes_laptop | Umm....digium cards require ztcfg to be run, too |
| 00:32.29 | dlynes_laptop | Digium chose to do the startup for that inside of modprobe.conf |
| 00:32.40 | dlynes_laptop | However, that doesn't work on all linux distributions, either |
| 00:33.45 | smackus | ok, that is the difference then... digium has it in the modprobe.conf. sangoma leaves it to be handled by the user |
| 00:34.00 | charles___ | 8.3 suckz |
| 00:34.09 | dlynes_laptop | smackus: yeah...I just throw it into the rc.local file, myself |
| 00:36.06 | icyfire0573 | FuriousGeorge, Yo, I checked it out at the NID. Definitely something inside the house. Looks like its time to get some STP wire. |
| 00:36.20 | charles___ | XARiUS: carefull with darwinism, you gonna need some alcohol to fix that |
| 00:37.06 | charles___ | why cisco had to screw the # key on 8.3 ? |
| 00:37.39 | *** part/#asterisk kumamoto (n=eryco@24-178-3-69.dhcp.ftwo.tx.charter.com) |
| 00:38.04 | smackus | dlynes_laptop: yep... just did it, rebooting and testing. thanks... just did not seem right to me. thanks for explaining. |
| 00:38.29 | JT | icyfire0573: ah, not suprising |
| 00:38.48 | icyfire0573 | Yea, I wasn't surprised either. |
| 00:38.58 | icyfire0573 | I don |
| 00:39.05 | icyfire0573 | don't know if I want to run cable though. |
| 00:39.19 | icyfire0573 | Its depressing me just thinking about it. And I don't know the proper method right now. |
| 00:39.30 | JT | if the cable in the walls/ceiling runs parallel to any power that's bad |
| 00:39.39 | JT | but check any leads from the walls |
| 00:40.41 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:41.09 | icyfire0573 | There are like... 5 wires running out of the NID into the house. And we only use 3 of them. Determining which is which would be a great place to start and then deactivating them. Followed by tracing the ones that we do know about and watching their proximity to the powerlines. |
| 00:42.33 | XARiUS | De_Mon: yeah, I've been using queue since.. pre 1.0.. and after today, I realize I must not really understand how ANY of it works. |
| 00:42.34 | XARiUS | lol |
| 00:43.14 | charles___ | XARiUS: darwinism watch out for it |
| 00:43.19 | XARiUS | *nods* |
| 00:43.42 | XARiUS | taking my dolly and going home! well... to the bar actually. |
| 00:44.04 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 00:45.18 | yxa | i have one * box that I can dial overseas thru my ISP. How do I users of another * box dial through that? |
| 00:47.16 | smackus | ok... I have commented out ztdummy from my modules.conf but it is still being loaded. is zaptel loading this regarless? |
| 00:48.08 | tuck3r_ | you have to noload it |
| 00:48.26 | smackus | where do i do that? |
| 00:48.29 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 00:48.54 | ariel_ | ztdummy loads as a driver prior to asterisk starting |
| 00:49.25 | smackus | ok, so what is the way around this. |
| 00:49.40 | dlynes_laptop | smackus: still having troubles with the sangoma drivers? |
| 00:51.08 | smackus | kinda... ztdummy tries to take over span one. so i have to modprobe -r ztdummy, ztcfg, then everything is happy |
| 00:51.36 | dlynes_laptop | smackus: why are you even loading ztdummy? |
| 00:52.10 | dlynes_laptop | smackus: ztdummy doesn't take over any spans |
| 00:52.45 | smackus | I dont want to load ztdummy. i have commented it out of the modules.conf, but it still keeps haunting me :-( |
| 00:53.30 | dlynes_laptop | smackus: it's gotta be getting loaded somewhere...it doesn't get loaded automatically by anything else |
| 00:53.37 | dlynes_laptop | smackus: oh...nvm |
| 00:53.41 | dlynes_laptop | smackus: i'm half awake |
| 00:53.43 | tuck3r_ | uh it's a kernel module not a asterisk module |
| 00:53.50 | dlynes_laptop | smackus: you're on a sangoma system, not a digium system :) |
| 00:53.54 | smackus | right |
| 00:54.01 | smackus | it used to be digium. |
| 00:54.08 | smackus | so this popped up after switching |
| 00:54.39 | dlynes_laptop | yeah...sangoma loads ztdummy instead of wct1xxp |
| 00:54.55 | dlynes_laptop | no |
| 00:54.56 | dlynes_laptop | nvm |
| 00:54.59 | smackus | lol |
| 00:55.00 | dlynes_laptop | sangoma just loads zaptel |
| 00:55.03 | dlynes_laptop | no ztdummy |
| 00:55.39 | dlynes_laptop | smackus: msg me, and i'll send you a list of all the drivers that should be loaded |
| 00:57.11 | *** join/#asterisk skac (n=skac@kakistocracy.co.uk) |
| 00:59.10 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-218.cybersurf.com) |
| 01:06.05 | *** join/#asterisk Rowter (n=Silver@201.135.9.97) |
| 01:09.02 | *** join/#asterisk morex (i=morex@host86-142-188-27.range86-142.btcentralplus.com) |
| 01:09.06 | morex | Eeek |
| 01:09.39 | morex | Zaptel's broke. Error is ZT_CHANCONFIG failed on channel 97: No such device or address (6) |
| 01:09.59 | morex | Anybody seen these before? Our call centre's due to open in 6 hours and I don't know what to do... |
| 01:12.08 | skac | my IP range is getting gangbanged by some E-Gold.com host. |
| 01:12.12 | smackus | ok, so if completely removing the module (all of the digium modules have been removed) from modprobe.conf and it still loads... how do i keep them from still loading? |
| 01:12.15 | skac | sigh. |
| 01:12.26 | smackus | are these installed to the kernel when zaptel is installed? |
| 01:12.53 | smackus | i can modprobe -r them, but I dont want to have to do that every time i boot |
| 01:13.10 | skac | put it in to your rc files then? |
| 01:13.30 | smackus | morex: inbound or outbound call? |
| 01:13.55 | smackus | morex: was it working before? did you make a change? |
| 01:13.58 | morex | Smackus: It's not even getting that far |
| 01:14.09 | smackus | what are you doing to produce the error? |
| 01:14.13 | morex | Yes I upgraded to kernel 2.6 on the advice of Digium support |
| 01:14.17 | morex | Now nothing works |
| 01:14.25 | morex | Problem occurs with modprobe wct4xxp |
| 01:14.26 | smackus | you have to rebuild zaptel for 2.6 |
| 01:14.31 | morex | Yeah I did that |
| 01:14.48 | smackus | have you reinstalled libpri and asterisk as well? |
| 01:14.52 | morex | Yup |
| 01:15.00 | smackus | and that is the only change you made was the kernel? |
| 01:15.01 | morex | make clean |
| 01:15.03 | morex | make install |
| 01:15.04 | morex | Yup |
| 01:15.09 | Bullseye_Network | morex: make linux26 |
| 01:15.13 | Bullseye_Network | then make install |
| 01:15.14 | morex | Tried it |
| 01:15.26 | smackus | ok... hang on. |
| 01:15.28 | morex | THANK YOU |
| 01:15.45 | smackus | so you can start asterisk? |
| 01:15.52 | morex | It's not even getting that far |
| 01:16.08 | morex | (and no, it doesn't like the misloaded wct4xxp module) |
| 01:16.52 | Rowter | anyone here knows where I could find nvlinedetect to detect answer on analog lines? |
| 01:17.18 | *** part/#asterisk dbradley (n=dbradley@tangram.gotadsl.co.uk) |
| 01:17.28 | smackus | if you are not able to modprobe it, I would think it would be an issue with the install of the zaptel driver |
| 01:17.38 | morex | Yes that's what I think too. |
| 01:17.43 | morex | How do I fix it? |
| 01:17.55 | smackus | what distro are you running? |
| 01:18.32 | morex | zaptel 1.2.7 trying to upgrade from 1.2.5 |
| 01:18.44 | morex | Both failing at modprobe wct4xxp t1e1override=15 |
| 01:18.45 | morex | with error |
| 01:18.48 | smackus | distro... flavor of linux |
| 01:18.50 | morex | ZT_CHANCONFIG failed on channel 97: No such device or address (6) |
| 01:18.50 | morex | FATAL: Error running install command for wct4xxp |
| 01:18.55 | morex | Oh sorry Debian |
| 01:19.11 | morex | Kernel 2.6.8-3-686 |
| 01:19.23 | smackus | yeah, i dont know. |
| 01:19.41 | hads | Sounds kinda like a zaptel.conf problem |
| 01:19.42 | Bullseye_Network | thats what I was thinking |
| 01:19.44 | smackus | modprobe is just the kernel isnt it? |
| 01:19.53 | smackus | not even getting to the zaptel.conf? |
| 01:20.18 | hads | Yeah, but I imagine that /etc/modprobe.d/zaptel is trying to run ztcfg |
| 01:20.28 | smackus | hmmm. |
| 01:20.33 | smackus | what t1s are in it? |
| 01:20.36 | smackus | in the card |
| 01:20.45 | morex | It's 4 E1s |
| 01:20.50 | morex | Hence the t1e1override |
| 01:21.06 | smackus | ...didnt know the number of lines.... |
| 01:21.12 | morex | Yes there is a ztcfg line in /etc/modprobe.d/zaptel |
| 01:21.35 | morex | I've even tried compiling with t1e1override set in the source code |
| 01:23.36 | morex | Doesn't seem to be having any effect |
| 01:25.56 | smackus | what are the jumpers on your card set to? |
| 01:26.00 | morex | t1 |
| 01:26.01 | hads | So, I know nothing about PRI... but it sounds like it is being setup as a T1 instead of E1, are the jumpers on the card set? Or can you only use override |
| 01:26.14 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 01:26.14 | morex | Unfortunately the machine is several hundred miles away |
| 01:26.19 | hads | Ah |
| 01:26.28 | morex | So I've been setting using the t1e1override module parameter |
| 01:26.44 | morex | Which now doesn't seem to work :-( |
| 01:26.50 | smackus | i have read a few instances on the net where that does not work for people... and they have to use the jumper. I do not know if this is actually the case |
| 01:27.19 | hads | So I imagine that ls -l /dev/zap would only show up to 96 ? |
| 01:27.33 | tengulre11 | HI,all |
| 01:27.39 | morex | That seems to go up to 249 |
| 01:28.02 | tengulre11 | which audio codec are you selected? |
| 01:28.07 | hads | And the permissions are readable by Asterisk? |
| 01:28.26 | hads | Sorry, and writeable. |
| 01:28.27 | morex | I'm not even trying to start asterisk |
| 01:28.38 | morex | Not until I can sort out zaptel |
| 01:28.44 | *** join/#asterisk brockj49464_home (n=chatzill@63.87.56.153) |
| 01:28.58 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 01:29.12 | tengulre11 | anybody selecte which audio codec? |
| 01:30.14 | morex | OK I've got a theory |
| 01:30.40 | morex | I think that when wct4xxp is started for the first time when the machine boots, it must be started with t1e1override in order for this to take effect |
| 01:30.45 | morex | What loads this module? |
| 01:30.56 | morex | I've tried disabling the /etc/init.d/zaptel script |
| 01:30.59 | *** join/#asterisk nachoguy (n=boster@67.95.66.69) |
| 01:31.13 | hads | put the option in /etc/modules.d/zaptel |
| 01:31.43 | hads | sorry, modprobe.d |
| 01:31.52 | morex | OK was beginning to wonder :-) |
| 01:32.23 | morex | In there I have a line like this: |
| 01:32.27 | morex | install wct4xxp /sbin/modprobe --ignore-install wct4xxp $CMDLINE_OPTS && /sbin/ztcfg |
| 01:32.35 | morex | I wonder where CMDLINE_OPTS comes from? |
| 01:32.46 | hads | grep for it |
| 01:33.34 | morex | It's not in /etc |
| 01:33.37 | hads | It comes from nowhere on my home Debian system |
| 01:34.05 | nachoguy | can you use a wildcard T1 card to terminate a voice T1 and use getty to pick up all incoming calls? |
| 01:34.08 | hads | Just put your own line in a file named 00-zaptel or something, it should take precidence |
| 01:34.35 | morex | I've added the parameter to the line in there and am rebooting |
| 01:34.38 | morex | Wish me luck... |
| 01:35.08 | hads | luck... just be warned that if you reinstall zaptel then that file will probably get overwritten. |
| 01:35.58 | tengulre11 | I building a asterisk server on public network, which audio codec suit to me? |
| 01:36.41 | morex | Nope same problem |
| 01:36.43 | JT | what does it have to connect to on the other side, what sort of "public network", tengulre11 ? |
| 01:37.37 | hads | I don't think I know enough about PRI to help sorry morex |
| 01:37.47 | morex | Thanks for trying - much appreciated |
| 01:37.48 | *** join/#asterisk trbldwine (i=troubled@c-71-194-161-170.hsd1.il.comcast.net) |
| 01:38.01 | tengulre11 | JT: It have a public IP and in city a, we are in city b |
| 01:38.04 | morex | Does anybody know how to downgrade debian safely to an earlier kernel? |
| 01:38.08 | tengulre11 | I think this is my mean. |
| 01:38.11 | morex | At least it worked with 2.4 |
| 01:38.14 | Bullseye_Network | morex if you try ztcfg -vvvvvv does it give the same error? |
| 01:38.21 | morex | Yup |
| 01:38.30 | JT | tengulre11: Internet? |
| 01:38.36 | morex | Yup |
| 01:38.44 | JT | what is the speed of both connections, tengulre11? |
| 01:39.02 | Bullseye_Network | morex: To ME it looks like a problem with zaptel.conf |
| 01:39.16 | morex | It hasn't changed since it worked with 2.4... |
| 01:39.39 | tengulre11 | JT: yes |
| 01:39.40 | hads | morex: You should be able to have 2.4 and 2.6 concurrently, and just boot into whichever you please. |
| 01:39.57 | hads | But it seems odd that you are having trouble with 2.6 |
| 01:40.07 | JT | tengulre11: you really need to provide as much information about the setup as possible so we can help you |
| 01:40.48 | tengulre11 | JT: about 56Kbps between two location. |
| 01:41.03 | JT | dialup or isdn or what? |
| 01:41.15 | *** part/#asterisk nachoguy (n=boster@67.95.66.69) |
| 01:41.38 | morex | OK wct4xxp also mentioned in /etc/default/zaptel with no options. Have commented out. |
| 01:42.19 | hads | morex: Do you have debs installed? |
| 01:42.31 | morex | Debian? Yes |
| 01:42.50 | JT | he said debs not debian |
| 01:42.59 | morex | Oh, I guess not then :-S |
| 01:43.00 | tengulre11 | JT: two defferent telecom provider. ISDN i think! |
| 01:43.06 | morex | What is it? |
| 01:43.18 | hads | `dpkg -l | grep zap` |
| 01:43.31 | morex | No packages matching |
| 01:43.36 | morex | I compile from source anyway |
| 01:44.26 | JT | tengulre11: well you will have fairly poor results if there's any dialup involved |
| 01:44.43 | hads | How did dpkg output 'No packages matching' if you used that command? |
| 01:44.50 | JT | isdn you should be able to run G.729, GSM or G.726 |
| 01:44.56 | JT | 729 is pretty popular |
| 01:45.04 | morex | It didn't output anything |
| 01:45.09 | morex | Usually it gives you a list |
| 01:46.11 | tengulre11 | JT: does the asterisk support g729? sip.conf disallow=all allow=g729, but when I dial out the asterisk got me some error message about audio codecs? why? |
| 01:46.32 | JT | tengulre11: you need to pay a licence fee for every G.729 transcoding session |
| 01:46.39 | JT | digium charges USD$11 or so |
| 01:46.40 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 01:46.40 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 01:46.43 | tengulre11 | when I using gsm codec, the voice is not nice? |
| 01:46.49 | JT | that's gsm for you |
| 01:47.09 | Bullseye_Network | gsm works fine here. |
| 01:47.11 | JT | 729, gsm, they're all highly compressed codecs |
| 01:47.19 | Bullseye_Network | I have 60+ people using gsm. |
| 01:47.20 | JT | they won't sound near as good as pcm |
| 01:48.07 | morex | OK I've commented out every mention of wct4xxp in /etc |
| 01:48.09 | Bullseye_Network | ulaw is better but.. If you have bandwidth issus gsm works good |
| 01:48.15 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 01:50.06 | morex | Bloody hell it's still loading on boot |
| 01:50.09 | morex | How do you stop it? |
| 01:50.25 | JT | Bullseye_Network: 729 sounds better to me |
| 01:50.52 | hads | morex: /etc/modprobe.d/blacklist |
| 01:51.05 | Bullseye_Network | JT:I've never tried 729...What bandwidth does it use? |
| 01:51.08 | morex | New file... |
| 01:51.23 | JT | Bullseye_Network: 13kbit/s for the audio stream |
| 01:51.29 | Bullseye_Network | wow.. |
| 01:51.31 | JT | +overhead |
| 01:51.35 | Bullseye_Network | I might have to try it out. |
| 01:51.43 | hads | morex: I'm using testing, not stable. |
| 01:51.45 | JT | what does your gsm stream use? |
| 01:51.59 | Bullseye_Network | about 13k |
| 01:52.03 | Qwell | g.729 is 8k... |
| 01:52.07 | JT | hmm |
| 01:52.12 | JT | Qwell: oh yeah, oops |
| 01:52.32 | JT | you can run gsm at higher rates can't you? |
| 01:52.36 | tengulre11 | but when I using gsm codec the voice translation slowly,and sometime it auto disconect my client! |
| 01:52.49 | Bullseye_Network | Im gonna have to try 729. Is there a demo 729 to use for a short time? |
| 01:53.32 | JT | apparently you can google for "free g.729 codec" |
| 01:53.39 | JT | so i've seen people say in here |
| 01:53.44 | morex | It's still there :-( |
| 01:53.49 | Bullseye_Network | Gonna have to try that... Thanks. |
| 01:53.52 | Qwell | There is no such thing as a gree g729 codec |
| 01:53.55 | Qwell | free too |
| 01:54.04 | JT | well i dunno if it's legal |
| 01:54.05 | JT | heh |
| 01:54.24 | Bullseye_Network | I have to run... morex I hope you get your problem figured out. |
| 01:54.32 | Bullseye_Network | sorry I couldnt help out more. |
| 01:54.48 | morex | Thankx Bullseye! |
| 01:54.52 | tengulre11 | 61.152.252.29 is my IP, sip user: 0212005 sercet:0212005 you can tested it! |
| 01:55.04 | tengulre11 | I using gsm codec |
| 01:55.45 | Qwell | tengulre11: gsm sounds "bad", it's as simple as that |
| 01:55.59 | morex | OK I don't know grub |
| 01:56.05 | morex | And my machine is 200 miles away |
| 01:56.16 | morex | Can anybody tell me how to safely reboot debian back into 2.4? |
| 01:56.39 | hads | I don't know about safely but Look for the default= line |
| 01:56.58 | morex | in menu.1st? |
| 01:57.09 | hads | Sorry, yes. |
| 01:57.48 | morex | Ok got it. Do I have to run anything to upload the changes? |
| 01:57.52 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 01:58.01 | hads | morex: No, it reads the file on boot. |
| 01:58.10 | morex | So I can just type reboot... |
| 01:58.20 | hads | It's a 0 based index of the entries in there. |
| 01:58.24 | morex | reboot |
| 01:58.32 | morex | In the right window ;-) |
| 01:58.37 | hads | heh |
| 01:58.51 | morex | It's been a loooong night... |
| 01:59.42 | tengulre11 | Qwell: which codec I should chioce? |
| 02:00.02 | Qwell | ~docs |
| 02:00.04 | jbot | hmm... docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 02:00.04 | Qwell | ~wikis |
| 02:00.06 | jbot | hmm... wikis is http://www.voip-info.org |
| 02:00.06 | Qwell | tengulre11: read |
| 02:01.59 | morex | Thanks hads |
| 02:02.46 | hads | np, I haven't really been much help. I wonder if something happened to t1e1override between zaptel 1.2.5 and 1.2.7 |
| 02:03.03 | morex | Well let's see if it reverts correctly now... |
| 02:03.29 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 02:09.52 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 02:12.42 | tengulre11 | Qwell: I readed it before. which codecs I should choice,pls tell me,(Asterisk running city a. we are city b) |
| 02:14.04 | morex | Eeek I've still got red alarms on all my cards :-( |
| 02:14.10 | tengulre11 | <PROTECTED> |
| 02:14.59 | hads | Same thing when you run ztcfg -vv ? |
| 02:15.20 | morex | No ztcfg looks OK |
| 02:15.37 | hads | Interesting. |
| 02:18.32 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 02:19.30 | Faithful | Hmmm... wondering if a Merlin U620 etc can make voice calls... if so a PCMCIA card might turn one into a 3G Zap channel ... any thoughts? |
| 02:24.09 | *** join/#asterisk op3r (n=op3r@202.71.179.140) |
| 02:28.17 | tengulre11 | Qwell: help me! |
| 02:28.21 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 02:28.51 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
| 02:30.11 | Faithful | tengulre11: what's your problem? |
| 02:31.25 | op3r | any ideas on how to install mysql-4.0.27 on centos using apt-get? |
| 02:31.47 | [TK]D-Fender | op3r : why no just use yum like the rest? |
| 02:32.08 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 02:32.17 | op3r | because I used apt-get and removed yum |
| 02:32.30 | [TK]D-Fender | op3r : wierd... how well dos it work? |
| 02:32.48 | op3r | its stable |
| 02:32.59 | morex | Interesting in a bad way |
| 02:33.05 | morex | Man I hope I get some sleep tonight |
| 02:33.06 | op3r | and there are a lot more .deb packages per arch than .rpms |
| 02:33.07 | [TK]D-Fender | op3r : I meant more in terms of package stability due to file location differences. |
| 02:33.16 | tengulre11 | Eaithful: thanks, I have a asterisk server in city a, our customers in ctiy b ,c,..., I don't know how to select audio codecs, I using gsm before but too bad. |
| 02:34.52 | op3r | [TK]D-Fender: I mean I didint encounter any problem since i start using apt-get instead of yum on centos |
| 02:38.11 | *** join/#asterisk asterboy (n=kevin@S01060204ee2b6007.ed.shawcable.net) |
| 02:39.57 | tengulre11 | Faithful:thanks, I have a asterisk server in city a, our customers in ctiy b ,c,..., I don't know how to select audio codecs, I using gsm before but too bad. |
| 02:40.10 | Faithful | tengulre11: bad in which way? |
| 02:40.15 | Qwell | tengulre11: for christs sakes, read |
| 02:40.20 | Qwell | sake.. |
| 02:40.31 | Qwell | This is answered on like page 1 of TFM |
| 02:41.31 | tengulre11 | Faithful: gsm |
| 02:41.36 | Faithful | Qwell: I don't think Christ died for tengulre11's codec ignorance. |
| 02:41.42 | Qwell | Faithful: hey, it's possible |
| 02:41.56 | tengulre11 | :( |
| 02:42.07 | Qwell | BUT, I just read your nick, AND I see that you capitalized the C...so, I'm gonna STFU now |
| 02:42.16 | Faithful | Qwell: well if his ignorance is a sin... then... |
| 02:42.26 | Qwell | ahh, yes, it is |
| 02:42.30 | tengulre11 | friendly is first!! |
| 02:43.11 | tengulre11 | my times is very pressure, so.. |
| 02:43.24 | tengulre11 | thanks everyone! |
| 02:44.32 | smackus | in queues.conf, i see the line Set(MONITOR_FILENAME. can i do something like this: Set(MONITOR_FILENAME=/recordings/${UNIQUEID}) |
| 02:44.38 | smackus | that did not work |
| 02:44.41 | Qwell | smackus: sure |
| 02:45.26 | smackus | here is what I get Unknown keyword in queue 'evolution': Set(MONITOR_FILENAME at line 212 of queues.conf |
| 02:45.41 | Faithful | Qwell: you know what the deal is with "using His Name in vain"? Well, He is also called Faithful & True... if you call His Name He turns up ... everytime ... and finds you in unbelief, provoking Him to do something... not a good deal. |
| 02:45.43 | Qwell | That goes in extensions.conf... |
| 02:46.39 | Qwell | Faithful: See the part about me STFUing? |
| 02:46.56 | Qwell | That implied "let's not go there" |
| 02:47.00 | smackus | i get that info from the queues.conf |
| 02:47.06 | smackus | like I should be adding it in there. |
| 02:47.12 | tengulre11 | SOS!!! |
| 02:47.15 | Qwell | smackus: I'm pretty sure it says it to add it to extensions.conf |
| 02:47.23 | smackus | lemme re-read. |
| 02:47.47 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 02:48.02 | Faithful | Qwell: Yeah... sorry... can't help myself... I will SU too (TF ommited for religious reasons ;-p ) |
| 02:48.28 | Faithful | tengulre11: you didn't explain what is wrong with gsm... |
| 02:48.58 | smackus | http://pastebin.ca/114532 |
| 02:49.08 | Faithful | how can you choose a codec if you don't know what's wrong with the one you got? So you need to read about the differeces and make a choise. |
| 02:50.35 | Faithful | tengulre11: try ilbc or G729 (if you wish to purchase licenses) I use ilbc generally and get a good run out of it. |
| 02:51.00 | hads | There's not a whole lot you can do with 56k |
| 02:51.27 | Faithful | hads: you can skype quite well |
| 02:51.38 | hads | eww |
| 02:52.09 | tengulre11 | Faithful: THANK YOU VERY MUCH! I saw! |
| 02:52.28 | tengulre11 | where can buy the g729 ? |
| 02:52.36 | Faithful | digium... |
| 02:53.03 | Faithful | but ilbc answer problems that g729 does not do well |
| 02:53.39 | JT | such as? |
| 02:54.25 | tengulre11 | Faithful: but my hard VoIP phone can not supported ilbc. only sopport g711,g726,gsm and g729 |
| 02:54.27 | Faithful | JT: do I look like a wiki... do I have to justify my statement? |
| 02:54.30 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 02:54.30 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:55.36 | Faithful | tengulre11: is this not the * channel? Run uLaw internal and ilbc over the internet. |
| 02:55.47 | JT | Faithful: no need to get snappy, i just thought you were going to back up your statements that you volunteer, not use them as a throwaway line |
| 02:55.48 | Faithful | let * transcode |
| 02:56.10 | Faithful | JT: not snappy at all :) |
| 02:56.38 | morex | It was down so long it triggered an alarm at the carrier! |
| 02:56.58 | Faithful | JT FYI ilbc handles jitter better than g729 and it breaks up more gracefully |
| 02:57.28 | Faithful | ilbc better for poor internet |
| 02:57.43 | Faithful | g729 better voice quality |
| 02:57.52 | JT | ah ok |
| 02:57.53 | JT | cool |
| 02:57.59 | JT | that's what i wanted to know |
| 02:58.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:58.28 | Faithful | but then ilbc capable of hearing 'S' on a good link because of higher bandwidth |
| 02:58.50 | Faithful | that's why skype sounds so good... ilbc |
| 02:59.10 | Qwell | skype uses wideband though, doesn't it? |
| 02:59.16 | JT | S? |
| 02:59.23 | Qwell | JT: letter |
| 02:59.26 | JT | skype uses a wider band than telephony frequencies |
| 02:59.30 | morex | Hooray we're back |
| 02:59.41 | JT | < Faithful> but then ilbc capable of hearing 'S' on a good link because |
| 02:59.44 | hads | Yay for morex |
| 03:00.01 | morex | We'd had so many alarms the carrier had disconnected us |
| 03:00.08 | morex | I had to call and get them to turn it back on |
| 03:00.12 | JT | nice |
| 03:00.27 | morex | LOL |
| 03:00.49 | Faithful | yes skype uses a proprietary version of ilbc as I understand it. |
| 03:01.37 | Faithful | What skype does over 56K modem is amazing. We just can't compete... but we are not trying to either... |
| 03:01.42 | morex | Whew! |
| 03:03.04 | JT | hrm |
| 03:03.30 | sevard | dude, it's like 190 F in this room |
| 03:03.39 | sevard | holy friggen' gawd |
| 03:03.52 | JT | uhh |
| 03:03.54 | JT | that's hot |
| 03:04.07 | Qwell | Faithful: ^^ feel free |
| 03:04.12 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 03:04.17 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 03:04.56 | Faithful | Qwell: ?? |
| 03:05.08 | Qwell | nothing.. |
| 03:05.33 | *** join/#asterisk tempest1 (n=asf@adsl-150-199-39.chs.bellsouth.net) |
| 03:06.14 | Faithful | Qwell: Oh, now I get it... |
| 03:06.17 | sevard | the skin just feel off my left arm |
| 03:06.23 | sevard | all of the skin. |
| 03:06.30 | sevard | boiled from the hot air in this tiny office. |
| 03:06.38 | sevard | it's hard to type without skin. |
| 03:08.45 | Faithful | I walked in a server room than noone goes in a while back and the floor was an inch under water... the aircon was on about 16degC and had backed up and was leaking water out the front of it down the wall. Probably been going like that for a week |
| 03:09.02 | JT | nice |
| 03:09.04 | sevard | fun stuff. |
| 03:09.34 | Faithful | yeah all the power cables lying in water... ?:-| |
| 03:10.20 | Faithful | Nice chatting but I better do some work today. |
| 03:10.27 | sevard | work is for fools. |
| 03:10.33 | sevard | #asterisk isn't for work. |
| 03:10.47 | sevard | we play on the internet and with phones all day. |
| 03:10.59 | Faithful | right... |
| 03:11.11 | sevard | right. |
| 03:11.32 | Faithful | anyone got MIAX working? |
| 03:16.39 | JT | sevard: so why is it so hot? |
| 03:20.14 | *** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 03:20.19 | *** part/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 03:21.14 | *** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
| 03:24.04 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 03:35.34 | ManxPower | Life sucks less after a few beers. |
| 03:36.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:36.08 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:40.26 | *** join/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
| 03:40.35 | smackus | anyone here doing queue recordings? |
| 03:40.44 | Scrye | can someone help me, i cant receive calls but not place them |
| 03:40.56 | smackus | Scrye: tell us more |
| 03:41.04 | smackus | are you getting errors? |
| 03:41.44 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:41.48 | shmaltz | hi everyone |
| 03:41.51 | Scrye | i have zingotel as a provider.. |
| 03:41.51 | Scrye | im readying a pastebin |
| 03:41.51 | Scrye | i get a fast busy when i dial |
| 03:42.28 | shmaltz | Scrye, has it ever worked? |
| 03:42.32 | Scrye | no |
| 03:42.57 | smackus | also paste your extensions.conf and tell us what extension you are dialing from |
| 03:43.53 | Scrye | i am |
| 03:44.04 | shmaltz | Scrye, have you ever gotten a dial statement in extensions.conf to work? |
| 03:44.09 | Scrye | i took out 4 extensions that are just redundant |
| 03:44.18 | Scrye | http://pastebin.ca/114621 |
| 03:44.31 | smackus | what are you dialing from? |
| 03:44.50 | Scrye | [ryan] = F1000g (utstarcomm wifi sip phone |
| 03:44.57 | Scrye | [ryan2] = Grandstream 101 |
| 03:45.04 | Scrye | [ryan3] = Xlite soft phone |
| 03:45.23 | Scrye | when i uncomment the 10001->10004 extensions i can dial them direct |
| 03:45.50 | Scrye | i really dont know what im doing so heh |
| 03:45.59 | Primer | polycom ACD...anyone using is successfully? |
| 03:46.02 | Scrye | i had someone help me get the dial-in working, then he got really busy and couldnt help me anymore |
| 03:46.22 | Scrye | (side note .. polycom has an office in my building and they are the most rude people ever...) |
| 03:47.05 | shmaltz | Primer, I don't think so |
| 03:47.18 | Primer | some dude has a branch of asterisk where Polycom ACD supposedly works...emphasis on "supposedly" |
| 03:47.29 | Primer | we've not been able to make it work |
| 03:47.44 | shmaltz | Scrye, it could be the office they have in your building handles their AR :P |
| 03:47.50 | Scrye | ok wtf |
| 03:48.02 | Scrye | i just tried it, and i had to put a 1 first even though the number i was dialing is not long distance |
| 03:48.20 | shmaltz | Scrye, not LD? from where? |
| 03:48.29 | Scrye | vancouver 778 to vancouver 604 |
| 03:48.48 | Scrye | without the 1 it gave me a fast busy |
| 03:48.51 | shmaltz | Scrye, you are using VoIP? right? |
| 03:48.55 | Scrye | yesh |
| 03:49.04 | shmaltz | Scrye, So how is it from vacouver? |
| 03:49.11 | Scrye | zingotel? |
| 03:49.13 | shmaltz | just because *you* are there? |
| 03:49.14 | Scrye | fine so far |
| 03:49.33 | Scrye | huh? |
| 03:49.57 | shmaltz | you said it's not LD because *you* are in vancouver, sorry but if it's voip that doesn't make that not LD |
| 03:50.12 | Scrye | thats whacked |
| 03:50.20 | Scrye | i dont have to dial 1 with my other voip provider |
| 03:50.43 | shmaltz | Scrye, in genral I agree that using VoIP you shouldn't have to dial a 1 ever |
| 03:50.49 | shmaltz | just like with cell phones |
| 03:51.04 | shmaltz | However some providers don't agree |
| 03:51.22 | Scrye | ok so a bit of help then |
| 03:51.22 | shmaltz | in any case you can fix your Asterisk DP that it should work without a 1 |
| 03:51.26 | Scrye | how do i add a 1 if it isnt there |
| 03:51.38 | *** join/#asterisk bmg505 (n=leon@c1-46-8.rndf.isadsl.co.za) |
| 03:51.42 | Scrye | because the wife isnt going to want to dial 1 |
| 03:51.45 | shmaltz | Dial(sip/whatever/1${EXTEN}) |
| 03:51.46 | Scrye | i have to make this idiot proof heh |
| 03:51.57 | Scrye | but what if it has a 1 already? |
| 03:52.05 | shmaltz | then you do like thsi: |
| 03:52.45 | shmaltz | exten => _1X.,1,Dial(sip/whatever/${EXTEN}) |
| 03:52.46 | shmaltz | exten => _X.1,Dial(Sip/whatever/1${EXTEN}) |
| 03:53.02 | Scrye | ok i think i understand |
| 03:53.14 | shmaltz | sorry I made a mistake |
| 03:53.18 | shmaltz | like this: |
| 03:53.28 | [TK]D-Fender | shmaltz : N :) |
| 03:53.38 | shmaltz | exten => _1N.,1,Dial(sip/whatever/${EXTEN}) |
| 03:53.40 | shmaltz | exten => _N.1,Dial(Sip/whatever/1${EXTEN}) |
| 03:53.46 | shmaltz | exactly, thanks D-Fender |
| 03:53.48 | shmaltz | :P |
| 03:53.52 | Scrye | exten => _X.,1,Dial(SIP/1${EXTEN}@zingotel-out) |
| 03:54.04 | Primer | bmg505: sup |
| 03:54.09 | shmaltz | Scrye, thats no good, look at my second example |
| 03:54.11 | ManxPower | How about exten => _NXXNXXXXXX,1,Dial(SIP/sipconfentry/1${EXTEN}) |
| 03:54.16 | Scrye | thats what i have now |
| 03:54.29 | shmaltz | ManxPower, of coure yours is better :P |
| 03:54.54 | ManxPower | . on an exten line delays dialing. |
| 03:55.03 | Scrye | so change that to |
| 03:55.03 | Scrye | exten => _1N.,1,Dial(SIP/${EXTEN}@zingotel-out) |
| 03:55.03 | Scrye | exten => _N.,1,Dial(SIP/1${EXTEN}@zingotel-out) |
| 03:55.03 | Scrye | <PROTECTED> |
| 03:55.21 | Scrye | so get rid of the . ? |
| 03:55.23 | shmaltz | ManxPower, not if it comes from SIP |
| 03:55.36 | shmaltz | Scrye, or to this: |
| 03:55.40 | ManxPower | shmaltz, correct, not if it comes from an ENBLOC device. |
| 03:56.06 | ManxPower | Still, it's a bad habit to get into. |
| 03:56.32 | shmaltz | exten =>_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@zingotel-out) |
| 03:56.33 | shmaltz | exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@zingotel-out) |
| 03:56.42 | shmaltz | ~enbloc |
| 03:56.57 | Scrye | what is the diff between N and X |
| 03:57.03 | shmaltz | N=2-9 |
| 03:57.08 | shmaltz | X=0-9 |
| 03:57.51 | shmaltz | ManxPower whats ENBLOC? |
| 03:58.10 | ManxPower | shmaltz, device that sends the entire dialed string at once. |
| 03:58.27 | shmaltz | ManxPower, that I understood it is, but what does it stand for? |
| 03:58.29 | Scrye | cool thanks |
| 03:58.37 | Scrye | all working now |
| 03:58.39 | ManxPower | i.e. SIP devices and PRI, where MGCP and analog are not. |
| 03:58.51 | shmaltz | or cell phones |
| 03:59.22 | Scrye | does asterisk support ipv6 yet? |
| 03:59.22 | Scrye | (i know my stupid phones dont) |
| 03:59.29 | shmaltz | ManxPower, that is also a good reason for SIP providers to drop the 1 before area code requirement |
| 03:59.35 | shmaltz | Scrye, no |
| 04:00.34 | shmaltz | ~wiki ENBLOC |
| 04:01.11 | Scrye | oh ffs i broke incoming calls |
| 04:02.12 | shmaltz | Scrye, welcome to sip register problems, what do you get in CLI on incoming? |
| 04:02.40 | Scrye | this was working |
| 04:03.04 | Scrye | what the fsck |
| 04:03.32 | shmaltz | Scrye, pb your sip.conf, and extensions.conf and CLI output when incoming call |
| 04:03.36 | Scrye | i did |
| 04:03.43 | Scrye | see previous pb url |
| 04:03.49 | Scrye | http://pastebin.ca/114621 |
| 04:03.59 | Scrye | frank*CLI> sip show peers |
| 04:03.59 | Scrye | Name/username Host Dyn Nat ACL Port Status |
| 04:03.59 | Scrye | zingotel-out/7787863704 69.25.48.85 5060 Unmonitored |
| 04:04.06 | Scrye | that used to be monitored |
| 04:04.40 | Scrye | and im not getting anything when i call it |
| 04:04.44 | ManxPower | shmaltz, I only envinted the ALL CAPS spelling of it. |
| 04:04.55 | JT | Scrye: en bloc is a word/phrase |
| 04:04.58 | JT | means all together |
| 04:06.06 | JT | i meant shmaltz |
| 04:06.09 | shmaltz | Scrye, you don't have a register statement for zingotel |
| 04:06.32 | Scrye | line 12? |
| 04:07.09 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 04:07.56 | shmaltz | Scrye, oh sorry now ic |
| 04:08.07 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 04:08.14 | shmaltz | Scrye, change zingotel in to: |
| 04:08.15 | shmaltz | type=peer |
| 04:08.20 | shmaltz | takeout the host line |
| 04:08.32 | shmaltz | and leave secret blank |
| 04:09.47 | shmaltz | and put in username=username |
| 04:09.50 | Scrye | shmaltz : isnt that what is on line 12? |
| 04:09.50 | Scrye | what should i type to get that back to status monitored |
| 04:09.50 | Scrye | but i need the secret? |
| 04:10.30 | Scrye | wtf |
| 04:10.30 | Scrye | im getting no route to host now |
| 04:10.32 | shmaltz | Scrye, yes it should, but no it's not, line 12 only tells zingotel where asterisk is lcoated, and zingotelin tells asteriks to let zingotel in |
| 04:11.00 | shmaltz | change zingotel out to type=friend |
| 04:12.30 | *** join/#asterisk jgoo (n=foo@ppp117-135.adsl.forthnet.gr) |
| 04:12.58 | jgoo | hey guys, is there a nice list of voip providers like broadvoice? by country (as each seems to have connect centers in various locations) |
| 04:13.17 | *** join/#asterisk CrazyYoss (n=luther@c-24-5-122-56.hsd1.ca.comcast.net) |
| 04:13.48 | Scrye | still busteded |
| 04:15.16 | shmaltz | Scrye, wait |
| 04:15.49 | shmaltz | Scrye, follow this: |
| 04:15.50 | shmaltz | http://pastebin.ca/114648 |
| 04:15.52 | shmaltz | change username to username |
| 04:17.48 | Scrye | ok i got it |
| 04:18.02 | Scrye | i took the sip.conf from the first instance where in was working |
| 04:19.09 | Scrye | and the extensions.conf from when the out only was working |
| 04:19.09 | Scrye | together = both ways |
| 04:19.47 | JT | so was the pastebin from when out was working or in? |
| 04:19.55 | Scrye | in only |
| 04:20.02 | Scrye | out was to alter the dial plan |
| 04:20.08 | Scrye | or extensions.conf |
| 04:22.25 | Rowter | anyone here knows where I could find nvlinedetect to detect answer on analog lines? |
| 04:23.10 | *** join/#asterisk jsharp (n=jsharp@65.88.255.130) |
| 04:23.20 | tuck3r | Ok, I can't figure this out... I have my settings in iax.conf and extensions.conf but no matter what i do the only way i can get it to work is by putting all the settings in the dial string. http://pastebin.ca/114657 |
| 04:23.27 | Scrye | now the real fun begins |
| 04:23.32 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 04:23.49 | jsharp | is t.38 passthru working yet? Don't need termination, just passthru. |
| 04:24.14 | hads | Rowter: http://www.voip-info.org/wiki/view/NVLineDetect |
| 04:26.09 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 04:26.17 | Rowter | hads, ooh, you know where to find the source? |
| 04:26.43 | hads | Rowter: Read the page, it tells you. |
| 04:27.08 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 04:27.13 | Rowter | hads, ok tnx.. they had not answer me mails, and I wondered if someone had it.. |
| 04:28.46 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 04:28.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:29.46 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 04:34.19 | smackus | i found something frightening i need help with. |
| 04:34.39 | tuck3r | zomg! |
| 04:34.52 | smackus | i have my dial plan working, but when someone calls into an auto attendant, they can dial telephone numbers so long as they match my dial plan. |
| 04:35.05 | smackus | so someone can dial LD numbers by calling my local auto attendant |
| 04:35.17 | hads | smackus: contexts |
| 04:36.57 | smackus | wait... ok, it came to me. I need my auto attendant in its own context, right? with no outbound access |
| 04:37.21 | *** join/#asterisk jmang (i=jordan@wnpgmb09dc1-88-150.dynamic.mts.net) |
| 04:37.51 | hads | smackus: Yes. |
| 04:39.01 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:42.13 | *** join/#asterisk lorinc (n=ang@caracas-3040.adsl.interware.hu) |
| 04:42.26 | Scrye | call agent ip address: blah:2427 |
| 04:42.29 | Scrye | is that skinny? |
| 04:42.45 | Scrye | i have another voip service (primus) and i want to integrate that into asterisk instead of using thier lame flakey gateway |
| 04:43.37 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 04:44.25 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 04:46.06 | Scrye | no its mgcp |
| 04:47.12 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 04:51.15 | *** join/#asterisk Splat (n=Splat@220-253-96-73.TAS.netspace.net.au) |
| 04:51.57 | jgoo | so a while back I looked at FOP and thought, eeew flash, and then thought, ah, I could download the flash source and place a hidden flash file as a scripting hook into an ajax/moo interface with ultrasex, and hey, they already did that now, awesome |
| 04:52.07 | jgoo | But I gotta get that version running on my trixbox |
| 04:52.31 | JT | you just mentioned an "evil" word |
| 04:52.42 | jgoo | JT: ultrasex? |
| 04:52.47 | jgoo | heh, trixbox isnät evil |
| 04:52.58 | wunderkin | moo? ultrasex? .... kinky.. |
| 04:52.59 | Scrye | ok i think primus is using MAC authentication with MGCP |
| 04:52.59 | Scrye | me? |
| 04:53.05 | jgoo | my friend just had a mad idea, 'booble' google for boobs, and lo, it was already done, lol |
| 04:53.06 | JT | yeah but you will be told to use a different channel :P |
| 04:53.17 | JT | yeah booble is old as |
| 04:53.33 | jgoo | JT: are you sure you wanted to admit that, you might be asked to use a diff chan ;) |
| 04:53.50 | JT | what |
| 04:53.57 | JT | <PROTECTED> |
| 04:53.58 | Scrye | i think tomorrow im gonna hack at my asterisk conf to get my primus account hooked into it with mgcp |
| 04:54.26 | adorah | A question: DO you think putting an extension both in a ring group and as a queue member can cause troubles in incoming calls? |
| 04:54.31 | Scrye | holy god am i lagged |
| 04:55.43 | jgoo | Scrye, use the lag-shoot technique, aim in front of the people you want to talk to. |
| 04:55.48 | jgoo | sorry, quake 2 flashback. it happens |
| 04:56.10 | Scrye | freenodes ipv6 servers are really bad |
| 04:56.12 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 04:56.18 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 04:57.23 | jgoo | Watching 'crazy german kid' reminds me why I stopped playing games. o.0 that is funny #%& |
| 04:57.40 | ManxPower | I still can't see how a company uses MAC authentication. MACs are not sent outside the local network. |
| 04:58.32 | Scrye | well i guess its how they can do it with dhcp |
| 04:58.40 | Scrye | when you dont know where the device is going |
| 04:58.46 | ManxPower | Uh, DHCP is not sent across the local network either. |
| 04:58.59 | Scrye | the protocol itself sends in the mac address (mgcp) |
| 04:59.09 | ManxPower | Well, not normally, and not without help from every router in the path. |
| 04:59.22 | ManxPower | Doesn't ANYONE here understanding IP networking? |
| 04:59.27 | Scrye | ... |
| 05:00.08 | ManxPower | I guess the box could figure out it's own MAC, then send it as a password, but it doesn't magical networking fairies don't make it work. |
| 05:00.13 | Scrye | Cisco owns the patent on dhcp relaying |
| 05:00.17 | file | ManxPower: nope |
| 05:00.20 | file | IP networking is silly |
| 05:00.31 | Qwell | file: Go to bed |
| 05:00.43 | Scrye | i said it 'sends in' its mac address, not it discovers it |
| 05:00.51 | ManxPower | file, IPX networking forever!!!!!!!!!!!!!!!!! |
| 05:00.55 | Scrye | last time i checked the internet wasnt entirely layer2 |
| 05:00.58 | file | token ring!!! |
| 05:01.39 | Scrye | token ring is a layer 2 technology |
| 05:01.40 | Scrye | im sure there are still token ring routers out there |
| 05:02.40 | Scrye | and before you spout stuff off like that - i happen to design firmware for routers for a living |
| 05:02.46 | Scrye | asterisk is a hobby |
| 05:03.09 | ManxPower | "I know saying something good about Trusted Computing around here is largely akin to taping meat to myself and dancing with lions, so here is an AC post." --seen on Slashdot. |
| 05:03.35 | adorah | Falks: How do u type G.723.1 in .conf file? g.723? g723? or g723.1? |
| 05:03.46 | ManxPower | Scrye, Please don't tell me you work for Linksys, d-Link or others of similar ilk. |
| 05:03.47 | adorah | =folks.. |
| 05:03.54 | ManxPower | adorah, you don't. They are patented. |
| 05:03.57 | Scrye | id love to work for linksys because thats cisco |
| 05:04.08 | denon | kind of |
| 05:04.16 | ManxPower | Scrye, I've just had so many odd problems with Linksys. |
| 05:04.23 | Scrye | but no |
| 05:04.28 | Scrye | i work for a canadian router company |
| 05:04.44 | ManxPower | I liked Linksys until I started running SIP and every time my SIP device registered to a remote server my Linksys box would reboot. |
| 05:04.50 | ManxPower | That's when I stopped liking Linksys. |
| 05:04.52 | adorah | <ManxPowe: so if a service runs only g723 or g729 I have to buy license? |
| 05:05.15 | ManxPower | adorah, The patent holders for G723 refuse to sell licenses for less than USD$30,000 |
| 05:05.25 | JT | heh, to all the fellow people who don't want to use linksys/dlink for business, can anyone recommend a decent ADSL2+ modem? |
| 05:05.46 | ManxPower | JT, Not really. Westel has a reputation for good stuff. |
| 05:05.49 | adorah | Mishuginers |
| 05:06.20 | EyeCue | JT, the millions arent too shabs at all |
| 05:06.23 | EyeCue | billion's |
| 05:06.35 | adorah | <ManxPowe: so how come a service offer termination only with that codec? |
| 05:06.38 | ManxPower | Before Katrina: Storm of Doom I wanted to play around with ADSL cards in my Cisco 1700. |
| 05:06.39 | Scrye | I have my own VoIP equipment. If I tell you what the make and model is, can you help me set it up to work with TalkBroadband? |
| 05:06.39 | Scrye | <PROTECTED> |
| 05:07.02 | ManxPower | adorah, I can imagine that they use only HARDWARE that supports that codec. |
| 05:07.53 | adorah | <ManxPowe: well in fact theu offered me byod they just provision ip or dns.. |
| 05:08.21 | ManxPower | JT, back in the day Paradyne had REALLY good stuff. Check with RAD Datacom, they bought PairGain and PairGain also had good stuff. |
| 05:08.39 | ManxPower | adorah, Tell them your equipment does not support G723 |
| 05:08.47 | adorah | <ManxPowe Do u have an idea how many licenses do I need to contact their trunk with g729? |
| 05:09.11 | adorah | <ManxPowethey support also g729 |
| 05:09.12 | ManxPower | adorah, G729? You can buy a G729 license for 1 call from Digium for $10. |
| 05:09.56 | adorah | <ManxPowe: So I have to buy only 1 license for the trunk? not for each extension to use it? |
| 05:10.02 | JT | ooh, i forgot to say, DSL modem that wasn't uber expensive |
| 05:10.16 | JT | so not complete,ly enterprise level |
| 05:10.18 | ManxPower | adorah, 1 license per simultaneous call |
| 05:10.22 | JT | but at least better than consumer |
| 05:10.43 | JT | EyeCue: that is a crappy consumer brand, i really don't want one of those |
| 05:10.48 | ManxPower | JT, As I said, WesTel has a good reputation for stuff and ISPs use them so they have to be pretty cheap. |
| 05:10.50 | adorah | <ManxPowe: thx a lot..that is diffinitely cheaper than 30,000$..hehe |
| 05:11.24 | adorah | =deffinitely |
| 05:11.30 | ManxPower | hellop, look in pretty much any telecom closet and you'll see Westel and Pairgain stuff. |
| 05:11.37 | ManxPower | ..er.. |
| 05:11.39 | ManxPower | <PROTECTED> |
| 05:11.44 | JT | ManxPower: supports ADSL2+? |
| 05:11.48 | JT | i'm in .au btw |
| 05:11.50 | ManxPower | JT, No idea. |
| 05:11.55 | nestar | wheeeee |
| 05:11.56 | JT | so haven't actually heard of westel |
| 05:12.03 | nestar | pppoe under linux works |
| 05:12.12 | ManxPower | We don't get much of that new fangled telecom stuff here in the USA |
| 05:12.48 | JT | hmm |
| 05:13.20 | nestar | have some mtu issues, it looks like. |
| 05:13.47 | JT | it allows for up to 24000kbit/s up, 1000kbit/s down or so iirc |
| 05:13.53 | JT | ADSL2+ |
| 05:14.01 | JT | swap that |
| 05:14.09 | JT | 24000kbit/s down, 1000kbit/s up |
| 05:14.16 | Scrye | nestar: yeah it does.. |
| 05:14.28 | Scrye | rp-pppoe |
| 05:14.28 | Scrye | its what we based our code off for our firmware |
| 05:15.20 | ManxPower | JT, I hate you. |
| 05:15.34 | JT | heh |
| 05:15.43 | ManxPower | Where I normally live I have 900ms - 3000ms latency |
| 05:15.46 | JT | nowhere in the US has this? |
| 05:15.47 | JT | ouch |
| 05:15.52 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 05:15.52 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 05:15.59 | ManxPower | I live on a mesa in Alabama |
| 05:16.33 | JT | looking at getting an ADSL2+ based SDSL service for voip |
| 05:16.35 | JT | 1024/1024 |
| 05:16.39 | denon | [there is nothing to do here but drink and roast marshmellows] |
| 05:16.42 | ManxPower | The only "broadband" is Satellite |
| 05:16.48 | JT | ah |
| 05:16.50 | JT | yeah |
| 05:16.52 | adorah | <ManxPowe> with that latency u better move to Africa.. |
| 05:17.27 | JT | adorah: you know you can use <tab> to complete nicknames? |
| 05:17.29 | ManxPower | I HOPE to have enough people willing to pay for high speed in the area to get a T-1 installed over the winter. |
| 05:18.00 | adorah | <JT>:) |
| 05:18.03 | denon | [which I'm going to buy from denon] |
| 05:18.10 | JT | adorah: arrgh :P |
| 05:18.29 | JT | it looks like you're quoting people instead of addressing them, adorah |
| 05:18.45 | Scrye | and believe me, you dont want PPPOE |
| 05:18.45 | Scrye | that generally means on the other end is just a reseller which has a network behind it that is over saturated |
| 05:18.45 | Scrye | (see bell canada) |
| 05:19.00 | Scrye | a T1 is not high speed |
| 05:19.20 | adorah | <JT>It is too early in the morning..I'm still half asleep.. |
| 05:19.53 | ManxPower | Scrye, T-1 is high speed where I come from -- corporate world. |
| 05:20.14 | ManxPower | I deal mostly with 384k frame relay links |
| 05:20.55 | Scrye | again, a T1 is not high speed |
| 05:21.00 | Scrye | a T1 is just 23 modems |
| 05:21.14 | ManxPower | A T-1 is 1.5Mbps. |
| 05:21.39 | Scrye | and I have 10 meg service to my house |
| 05:21.44 | ManxPower | Most of the locations I deal with you can't even get anything higher. |
| 05:21.48 | Scrye | that is only starting on high speed |
| 05:21.59 | Scrye | im sorry that the hick south hasnt caught up with the rest of the world |
| 05:22.06 | ManxPower | Scrye, I tend to be pretty skeptical about consumer connections. |
| 05:22.32 | EyeCue | JT, then get a cisco with a dsl wic :) |
| 05:22.37 | Scrye | that is because you deal with monopolistic american corporations |
| 05:22.40 | ManxPower | Every time we try to put bunches of people on "high speed cable modem service" the provider tends to get really bitchy. |
| 05:23.17 | adorah | So try to keep a connection with 150Kbits upload..only GSM can pass thru.. |
| 05:23.18 | ManxPower | Oh, so you sold me 10 Mbps service, but I can't actually use 10Mbps of bandwidth 8 hrs /day 7 days/week |
| 05:23.52 | denon | the nice thing about a T1 from a real carrier .. is that at any given time, you get everything you were promised, and if you dont, you can call a noc 24/7, and they will CARE that you dont |
| 05:24.00 | denon | and have someone on your doorstep if need be |
| 05:24.04 | ManxPower | denon, EXACTLY! |
| 05:24.14 | ManxPower | hellop, our carriers usually call us to tell us there is a problem. |
| 05:24.19 | *** join/#asterisk Splat (n=Splat@220-253-107-146.TAS.netspace.net.au) |
| 05:24.20 | ManxPower | <PROTECTED> |
| 05:24.26 | denon | often you can even talk to someone who can log directly into the switch, test the line cards, etc |
| 05:24.40 | denon | not "We'll open a ticke for you .. I think there's trouble in your area" |
| 05:24.45 | denon | [ticket] |
| 05:24.48 | ManxPower | Even thought AT&T sells everyone's calling records to the NSA, they do have a decent network. |
| 05:24.52 | Scrye | im sorry, anything that you could get in 1980 is not high speed |
| 05:25.10 | denon | Scrye: you could get OC3s in 1980 |
| 05:25.29 | denon | and DS3s were abundant |
| 05:25.36 | denon | you're telling me 45Mbps isnt fast? |
| 05:25.39 | ManxPower | Scrye, I like my weekends to be mine. I do not like my weekends to be on the phone with some cable company that can't spell TCP/IP. |
| 05:26.00 | denon | ManxPower: or guys that send you to a web page to test your speed |
| 05:26.02 | ManxPower | I want it to Just Work and my users expect it to Just Work. |
| 05:26.06 | EyeCue | yeh the t3's and fractionals back in the day were speedy |
| 05:26.24 | EyeCue | i remember in .au entire ISPS's were running off e1's (your t1's) |
| 05:26.27 | denon | we've replaced a LOT of dsl and cable with T1s |
| 05:26.35 | ManxPower | denon, Glad to know SOMEONE from the corporate world still hangs out around here. |
| 05:26.35 | denon | E1 != T1 |
| 05:26.39 | JT | EyeCue: there must be more than cisco available |
| 05:26.50 | denon | E1 has a few extra channels :) |
| 05:26.56 | EyeCue | JT, i wouldnt go anything else if you want to stay away from the soho |
| 05:27.01 | denon | ManxPower: I wouldnt leave you alone in this endluser soup! |
| 05:27.05 | Scrye | 45mbps is not that fast |
| 05:27.06 | JT | like 6 or 7 extra channels :P |
| 05:27.06 | EyeCue | you cant say that billion is consumer grade when you talk about dlink in the same sentence |
| 05:27.06 | ManxPower | denon, My largest client used to use VPN with consumer internet service. Never again. |
| 05:27.23 | JT | EyeCue: sure can |
| 05:27.27 | EyeCue | sif :) |
| 05:27.28 | JT | billion is a real cheap brand |
| 05:27.33 | EyeCue | get a cisco if you dont want to dick around |
| 05:27.37 | denon | Scrye: a constant 45Mbps bi-directional is much faster than whatever you have to your house |
| 05:27.47 | JT | cisco's adsl2+ modems are like $700 |
| 05:27.52 | EyeCue | amen to that |
| 05:27.54 | EyeCue | and worth it |
| 05:27.55 | adorah | <JT>30+2 to be exact.. |
| 05:27.58 | ManxPower | denon, I discovered that Charter ISP (consumer at least) has problems talking to some DHCP clients. |
| 05:28.05 | EyeCue | try getting a 2600 series and a dsl2 wic :D~ |
| 05:28.08 | EyeCue | now that 'is' fun. |
| 05:28.11 | JT | adorah: i know, you don't need to educate me on PRI :P |
| 05:28.11 | Scrye | OC3 isnt 45mbps |
| 05:28.15 | Scrye | oc3 is 155 |
| 05:28.16 | denon | ManxPower: charter also has problems talking to any kind of client :) |
| 05:28.22 | Scrye | DS3 is 45mbs |
| 05:28.22 | JT | EyeCue: are the wics even made? |
| 05:28.23 | denon | Scrye: duh. |
| 05:28.28 | ManxPower | denon, my SIPura box works just fine. |
| 05:28.29 | EyeCue | yeh |
| 05:28.31 | denon | [00:25] <denon> and DS3s were abundant |
| 05:28.41 | JT | how much are they, and any model numbers? |
| 05:28.48 | EyeCue | theres both 1 and 2's around, the 2's are more expensive as you can imagine |
| 05:28.54 | adorah | <JT>well, most americans don't know much about E1..:) |
| 05:28.56 | EyeCue | depends if youre getting them new or not |
| 05:29.01 | EyeCue | JT, whats the router for ? |
| 05:29.03 | ManxPower | egads! It's 12:30pm local time and I have to call a switch tech about a PRI at 9am |
| 05:29.03 | EyeCue | ' |
| 05:29.07 | JT | adorah: this channel doesn't have most americans |
| 05:29.15 | JT | EyeCue: i really only need a modem, not a router |
| 05:29.17 | adorah | <JT>LOL |
| 05:29.26 | denon | actually, most americans who are switch monkeys have deployed T1s and E1s |
| 05:29.27 | Scrye | and no you couldnt get OC3 in 1980 |
| 05:29.28 | JT | for voip over adsl2+ based sdsl |
| 05:29.33 | EyeCue | hmm |
| 05:29.43 | ManxPower | denon, ever heard of a call showing as answered (both legs PRI) on an Asterisk box, but no audio? |
| 05:29.46 | EyeCue | adls2 sdsl? |
| 05:29.50 | EyeCue | whos providing that here? |
| 05:29.57 | JT | people telecom |
| 05:29.59 | JT | 1024/1024 |
| 05:30.03 | EyeCue | pt blow for the record. |
| 05:30.13 | JT | some carriers offer 1500/1500 |
| 05:30.20 | JT | what makes you say that? |
| 05:30.28 | EyeCue | 1mbps sdsl has been around for a while, youre saying theire advertising dsl2 @ 1/1 ? |
| 05:30.34 | Scrye | it wasnt even submitted to the CCIT till 1987 |
| 05:30.34 | Scrye | CCITT |
| 05:30.36 | JT | no |
| 05:30.46 | Scrye | sdsl has been around since 1998 |
| 05:30.51 | JT | they are running an SDSL service over their ADSL2+ DSLAMS |
| 05:30.56 | JT | on 2 wire |
| 05:30.56 | EyeCue | ah |
| 05:30.59 | Scrye | thats when i was working at an ISP in maine, and we were one of the first to offer it |
| 05:31.22 | ManxPower | Uh, May of the T-1s in the USA are SDSL and have been for a while. |
| 05:31.26 | EyeCue | jees, to buy a 2 grand dsl2 wic for 1/1, seems like overkill |
| 05:31.31 | denon | ManxPower: dont think I have |
| 05:31.49 | JT | EyeCue: is that how much it costs? i definately don't want to go that way then :P |
| 05:31.54 | ManxPower | denon, I'm pretty sure it's a problem in the carrier's config of the PRI, but thought I'd ask |
| 05:31.58 | EyeCue | god this tea cant brew any quicker :| |
| 05:32.06 | ManxPower | Off to bed. |
| 05:32.25 | EyeCue | prolly not :) |
| 05:32.50 | EyeCue | linksys perhaps? ;) |
| 05:33.11 | JT | hmm i dunno |
| 05:33.38 | EyeCue | determin eyour budget first, then get what you can |
| 05:34.02 | JT | i want to know what lies between the $150 modems and the $700 ones |
| 05:34.04 | JT | anything? |
| 05:34.45 | denon | JT: you mean analog modems? |
| 05:34.47 | EyeCue | brand and feature sets |
| 05:34.55 | denon | AT&T used to have some spendy modems that had proprietary extensions |
| 05:35.00 | denon | that, say, a courier didnt have |
| 05:35.12 | JT | no, ADSL2+ modems |
| 05:35.23 | denon | oic |
| 05:35.31 | Scrye | Before the jump in Internet traffic in the mid 1990s, DS1s were found almost exclusively in telephone company central offices as a means to transport voice traffic between locations. |
| 05:35.35 | JT | fair enough :) |
| 05:35.54 | JT | let's not forget why they invented DS1s |
| 05:36.17 | JT | to transport NTSC video between tv studios |
| 05:36.43 | JT | because ntsc is hopeless and analogue amplifiers introduce phase changes, and phase changes in NTSC introduces colour changes |
| 05:36.53 | JT | people that were meant to look like people, could look green |
| 05:36.55 | EyeCue | JT, youre gonna find that the cheaper ones are always gonna be focused on providing alot of 'feature's into modem&router type packages |
| 05:36.56 | JT | or worse |
| 05:37.08 | EyeCue | heh |
| 05:37.11 | JT | EyeCue: yeha |
| 05:37.14 | EyeCue | pal for the win. |
| 05:39.31 | *** part/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
| 05:42.04 | JT | there's a reason people call it Never The Same Colour twice |
| 05:42.06 | JT | ;) |
| 05:46.01 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 05:46.48 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 05:48.44 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 05:51.18 | *** join/#asterisk SantaRosaMark (n=mark@c-24-4-207-104.hsd1.ca.comcast.net) |
| 06:02.08 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 06:06.13 | sevard | Question, on the back of my TDM2400P I am only seeing 41 volts for ringing line voltage, can I turn this up to a standard 90? |
| 06:07.32 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 06:07.33 | *** join/#asterisk sandra78 (n=adfvae@200.106.8.106) |
| 06:08.07 | sandra78 | Does anyone knows how to bloking the Caller ID from the FXO PSTN port? |
| 06:08.12 | hads|home | sevard: look at the boostringer option |
| 06:08.33 | hads|home | module option that is. |
| 06:08.36 | sevard | boostringer option? |
| 06:08.37 | *** part/#asterisk stoffell (n=stoffell@pot.catsanddogs.com) |
| 06:08.59 | sevard | sandra78: check out hidecallerid=yes |
| 06:11.04 | *** join/#asterisk kans_1 (n=kans@61.14.86.23) |
| 06:11.24 | kans_1 | hi anyone please refer me good voice to text program ? |
| 06:11.30 | kans_1 | to integrate with asterisk |
| 06:11.38 | tuck3r | what could cause app_dial.c:1049 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)? I'm totally stumped here |
| 06:11.38 | kans_1 | Sorry, text to voice |
| 06:12.06 | kans_1 | tuck3r: it could not find the route to the destination ! |
| 06:12.51 | tuck3r | here are the setting for that http://pastebin.ca/114657 |
| 06:12.53 | sevard | hads|home: that should boost up the voltage from 41 volts to 90? |
| 06:13.09 | tuck3r | i can use the settings in the dial string but not in iax.conf |
| 06:13.29 | kans_1 | what is the best text to voice program for asterisk ? |
| 06:13.37 | tuck3r | festival? |
| 06:14.11 | kans_1 | good one, festival is sleeping one |
| 06:14.18 | kans_1 | i can pay for it, not a problem ! |
| 06:14.29 | tuck3r | free... |
| 06:14.45 | tuck3r | thats about all there is that i know of |
| 06:17.13 | kans_1 | yes |
| 06:17.21 | kans_1 | festival is not good for commercial use |
| 06:17.30 | kans_1 | i'm looking for a solution for commercial use |
| 06:18.04 | sandra78 | <sevard> sandra78: check out hidecallerid=yes<----Where do i put this? in zapata.conf? |
| 06:18.29 | tuck3r | you could google for where it goes... |
| 06:19.13 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 06:19.13 | *** mode/#asterisk [+o russellb] by ChanServ |
| 06:19.36 | sevard | sandra78: yes, if you google site:voip-info.org zapata.conf hidecallerid=yes you'll find lots of other good options |
| 06:20.19 | sevard | kans_1: there's another one that's really good, it's called like smos, or some bullcrap, check voip-info |
| 06:21.49 | kans_1 | sevard: few of them referred me to cepstral |
| 06:22.11 | kans_1 | sevard, but i could not make it to work |
| 06:22.36 | mitcheloc | somebody rebooted my server! it had 450+ days of uptime on it! *sigh* |
| 06:23.14 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 06:23.34 | tuck3r | mitcheloc: lol! |
| 06:23.38 | tuck3r | windows? |
| 06:23.52 | mitcheloc | tuck3r, dude, i'm heart broken, it's not funny |
| 06:23.54 | mitcheloc | (FC3) |
| 06:24.12 | tuck3r | whats the record? |
| 06:24.17 | [hC] | I found a server on my network today |
| 06:24.29 | [hC] | I have -no- idea where it is, or what it does (it doesnt do much, seemingly |
| 06:24.32 | mitcheloc | that was my record... |
| 06:24.32 | [hC] | with a 587 day uptime |
| 06:24.40 | [hC] | I didnt even know it was there. |
| 06:24.45 | tuck3r | hah |
| 06:24.46 | [hC] | it must have been installed ages ago and left and forgotten. |
| 06:24.46 | mitcheloc | [hC], thanks for rubbing it in... |
| 06:25.02 | [hC] | well at least you know now that you didnt have a world record, so you dont have to feel so bad. |
| 06:25.02 | [hC] | :P |
| 06:25.23 | mitcheloc | i heard a story about a university that had the longest running server ever, supposedly one of the services failed on the machine |
| 06:25.32 | mitcheloc | and the IT staff couldn't find it anywhere, so they started tracing cables |
| 06:25.48 | mitcheloc | and then they found one that went into a wall, with no outlet or antyhing, just directly into the wall.... |
| 06:25.59 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:26.07 | mitcheloc | so they had remodeled and the workers had built a wall in front of it without moving it... |
| 06:26.18 | mitcheloc | it was something like 5 years running or something |
| 06:26.28 | [hC] | i heard that story too actually |
| 06:26.28 | [hC] | haha |
| 06:26.35 | tuck3r | so did i... |
| 06:26.37 | JT | it was a netware server |
| 06:26.54 | JT | and they got netware engineers to come in and they worked out the length of cable using ping tools |
| 06:27.06 | JT | so they could work out the area it was in |
| 06:27.38 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 06:27.53 | mitcheloc | lol...http://www.unixsucks.com/unixsucks.aspx |
| 06:27.54 | [hC] | heh |
| 06:29.32 | adorah | Hi how do I limit access to outbound calls? set different contexts or have to set accounts to allow outgoing calls? |
| 06:34.11 | *** join/#asterisk nevyn (n=nevyn@c220-237-154-157.brodm1.vic.optusnet.com.au) |
| 06:34.15 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 06:34.40 | nevyn | dumb question.. what's required for dtmf generation to work from handsets? |
| 06:34.56 | nevyn | I can make calls.. but can't navigate remote IVR's |
| 06:35.38 | *** join/#asterisk rikstah (n=rick@c-24-17-81-231.hsd1.wa.comcast.net) |
| 06:35.45 | adorah | <nevyn>try differnt dtmf mode until u get the right one |
| 06:36.16 | EyeCue | ben? |
| 06:36.43 | nevyn | I can navigate local IVR's but I don't think that's using dtmf |
| 06:37.21 | JT | are you sending it out of band? |
| 06:37.24 | JT | what protocol? |
| 06:37.31 | JT | to a voip provider? |
| 06:38.59 | nevyn | rfc2833 seems to be what I'm using |
| 06:39.22 | nevyn | both with the voip provider (mynetfone) and with the handsets |
| 06:39.34 | *** join/#asterisk norse (n=norse@rootcubed.com) |
| 06:39.43 | nevyn | the extensions are configured as rfc2833 as are the phones |
| 06:40.13 | jgoo | while on a call, can I script the receiving phone? like, I receive a call, can I then listen to a code from the receiver. 33, hold, then wait for extension. so you can transfer internal external using just a phone keypad? |
| 06:40.24 | jgoo | I pick up phone, and press 33, 200 |
| 06:40.49 | jgoo | or 22 XXX-XXX-XXXX for external warm transfer (that is the right terminology?) |
| 06:40.53 | *** join/#asterisk _omer (n=_omer@202.38.51.2) |
| 06:41.03 | jgoo | or is it only incoming DTFM that is scripted? |
| 06:41.36 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:42.16 | *** join/#asterisk ^cLuBk|d^ (n=clubkid@ACCD079E.ipt.aol.com) |
| 06:42.32 | _omer | http://pastebin.ca/114867 anyhelp |
| 06:43.14 | sevard | hads|home: The boostringer option stepped up the voltage from 41 to 52, it seems just enough to ring my phones. thank you very much. |
| 06:44.07 | norse | do astersik run on freebsd |
| 06:44.07 | norse | ? |
| 06:44.10 | jgoo | _omer I think I had that same error! |
| 06:44.13 | tuck3r | _omer: you already have asterisk running |
| 06:44.20 | jgoo | you have to comment out a line in the modules.conf (I think) |
| 06:44.27 | jgoo | where it tries to load that module, and it all worked |
| 06:44.27 | tuck3r | no... |
| 06:44.30 | tuck3r | no... |
| 06:44.43 | _omer | tuck3r : yes.....I tried "service asterisk stop [OK]" |
| 06:44.50 | tuck3r | <PROTECTED> |
| 06:44.57 | tuck3r | you should know to do that |
| 06:45.25 | jgoo | hey tuck3r, this is a fairly abstract question, so I couldn't find anything in voip-info |
| 06:45.36 | jgoo | scripting the key presses on the phone that receives the call |
| 06:45.51 | _omer | tucker : how to kill ? |
| 06:46.00 | tuck3r | _omer: kill. |
| 06:46.29 | jgoo | ps aux | grep asterisk | sed pid or something | kill %i or something I suck at scripting |
| 06:46.35 | tuck3r | jgoo: uh, i forgot :/ it a dial plan commend to get dtmf |
| 06:46.37 | _omer | okey :) |
| 06:47.20 | jgoo | tuck3r: sounds like a useful thing though, to give xfer and hold options directly on a keypad on any phone, even a mobile if you get the call there. |
| 06:47.29 | jgoo | push button record calls on your mobile etc |
| 06:47.50 | jgoo | rexfer call from pbx to your home, pick itup as you walk in and hang up your mobile, and keep talking |
| 06:48.22 | tuck3r | it's really freaking easy i just forgot :/ |
| 06:48.33 | norse | hi im new to asterisk..ive checked the site and see all the hardware requirements my question is how can i connect all phones on the pc |
| 06:48.44 | tuck3r | sigh |
| 06:48.45 | jgoo | aaah ok cool, ill look Ö_o just curious of possible |
| 06:49.23 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 06:49.49 | JT | jgoo: checked out TFOT? |
| 06:49.51 | JT | the book |
| 06:50.03 | linlin | any way to get more detailed information on why an asterisk server is refusing an IAX@ connection? |
| 06:50.07 | jgoo | yes, until page 64 in pdf I believe |
| 06:50.11 | jgoo | I gotta pick it back up |
| 06:50.23 | linlin | iax2 debug just returns that it is refusing the connection, iax2 softphone says it is timing out |
| 06:50.41 | norse | anyone? |
| 06:51.00 | jgoo | norse: you can connect phones depending on the cardyou buy |
| 06:51.11 | JT | nevyn: well i'm not familiar with what that rfc entails, are you sending dtmf inband or out of band? |
| 06:51.17 | jgoo | if it has FXO ports, whichthink of the O as Office, which means you plug office phones into there |
| 06:51.19 | JT | nevyn: and what codec? |
| 06:51.27 | hads|home | ~fxofxs |
| 06:51.29 | jbot | i heard fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 06:51.46 | hads|home | ~book |
| 06:51.47 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 06:51.59 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 06:52.02 | hads|home | norse: The book is a good place to start. |
| 06:53.19 | norse | hads|home can you point me to the book |
| 06:53.20 | norse | LOL |
| 06:53.34 | ^cLuBk|d^ | where can i purchase that book? |
| 06:53.40 | tuck3r | the book store |
| 06:53.40 | jgoo | I am going to be the first person to make a SIP compliant USB tincan/string device. awesome. |
| 06:53.56 | jgoo | ^cLuBk|d^ or free online |
| 06:54.06 | ^cLuBk|d^ | jgoo can u give me the link? |
| 06:54.08 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 06:54.08 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 06:54.29 | jgoo | sure: http://www.google.com/search?q=TFOT |
| 06:55.00 | ^cLuBk|d^ | thanks |
| 06:55.04 | linlin | any way to get more detailed information on why an asterisk server is refusing an IAX@ connection? |
| 06:55.16 | linlin | any way to get more detailed information on why an asterisk server is refusing an IAX@ connection?iax2* |
| 06:55.19 | linlin | oops |
| 06:55.23 | linlin | *iax2 |
| 06:55.44 | linlin | iax2 debug just says connection refused |
| 06:55.55 | EyeCue | asterisk -vvvc |
| 06:55.59 | EyeCue | and read your debug console |
| 06:56.21 | linlin | you mean vvvvr ? |
| 06:56.53 | EyeCue | i dont know, do i? |
| 06:57.12 | linlin | phreakbox:~# asterisk -vvvvvvvvc |
| 06:57.12 | linlin | Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. |
| 06:57.42 | EyeCue | stop asterisk, then restart it using vvvc |
| 06:57.49 | EyeCue | it wont daemonize |
| 06:57.56 | hads|home | or just set verbose 5 |
| 06:58.11 | EyeCue | or that |
| 06:58.31 | hads|home | and set debug 5 if you want debug |
| 06:59.47 | Faithful | Anyone got ideas for integrating a 3G phone with * ? |
| 07:01.08 | *** part/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 07:01.12 | *** join/#asterisk Splat (n=Splat@220-253-105-101.TAS.netspace.net.au) |
| 07:01.57 | fourcheeze | Faithful: sip client on phone? |
| 07:02.10 | linlin | Timestamp: 00013ms SCall: 00019 DCall: 31935 [67.184.230.25:4569] |
| 07:02.10 | linlin | CAUSE : Registration Refused |
| 07:02.10 | linlin | CAUSE CODE : 29 |
| 07:02.20 | linlin | whats "cause code" 29 ? |
| 07:02.42 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:03.29 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:03.29 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 07:04.03 | *** join/#asterisk sob0l (n=sobol@213.173.229.83) |
| 07:06.59 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 07:09.42 | e-ddie | when using blindxfer, asterisk records the cdr data, with src as the one calling in to the person transferring, and dst like the nr transferred to... on the user_id of the person transferring |
| 07:10.45 | e-ddie | is there any way to make this actually record the src nr as the number being transferred from? |
| 07:12.42 | *** join/#asterisk Money5ack (i=moneysac@wer.will.spontanficken.de) |
| 07:13.41 | *** join/#asterisk vgster (n=vgster@217.78.147.238) |
| 07:13.43 | *** join/#asterisk _fa_ (i=faceoff@een.os3.kn.pl) |
| 07:14.48 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:15.53 | *** join/#asterisk Assid (i=assid@203.115.64.10) |
| 07:17.46 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:21.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 07:21.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 07:21.32 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-232-46.telkomadsl.co.za) |
| 07:24.43 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:27.09 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 07:27.30 | *** join/#asterisk saftsack (n=saftsack@p54A7E46D.dip.t-dialin.net) |
| 07:30.09 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 07:30.25 | FlatFoot | anybody awake today ? |
| 07:30.30 | FlatFoot | god morning all |
| 07:30.41 | FlatFoot | good even |
| 07:32.12 | jac]Z[oby | gmornin FlatFoot |
| 07:32.23 | hads|home | good evening |
| 07:32.58 | FlatFoot | once again we have a range of countries logged on then |
| 07:33.04 | FlatFoot | uk for me |
| 07:33.04 | jac]Z[oby | ;) |
| 07:33.07 | jac]Z[oby | austria |
| 07:33.40 | FlatFoot | jac]Z[oby: must be about 09:30 for you then |
| 07:34.04 | jac]Z[oby | yope |
| 07:34.07 | hads|home | 19:30 here in NZ |
| 07:34.11 | jac]Z[oby | 8.30 for you ;) |
| 07:34.12 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 07:34.14 | FlatFoot | yep |
| 07:34.17 | jac]Z[oby | *phew* |
| 07:34.24 | *** join/#asterisk walhala (n=niolou@213.161.196.217) |
| 07:34.41 | FlatFoot | still gets me , been in this world of the internet for 10 years now and it still amazes me |
| 07:34.43 | jac]Z[oby | well btw. i know i am worng in this channel |
| 07:34.59 | jac]Z[oby | but can anyone help me with a number manip in trixbox |
| 07:35.06 | FlatFoot | all this people from all over the world sort of sitting in my office |
| 07:35.16 | jac]Z[oby | somehow the #freepbx has become a bf2k channel |
| 07:35.19 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:35.30 | tuck3r | jac]Z[oby: rofl |
| 07:35.37 | jac]Z[oby | ;) |
| 07:35.44 | JT | short attention span, too short to configure real asterisk |
| 07:35.46 | JT | figures |
| 07:35.48 | JT | games |
| 07:35.55 | jac]Z[oby | ypoe |
| 07:36.13 | tuck3r | hey hey, i configured asterisk and i still play games. what now? :P |
| 07:36.25 | JT | purgatory |
| 07:36.37 | tuck3r | hah |
| 07:36.53 | jac]Z[oby | well im using trixxbox as i have manged to connect an old pbx to ann innovaphone (h323) with an asterisk in a productive environment |
| 07:37.11 | jac]Z[oby | dont want to configure that much in * |
| 07:38.44 | creativx | aagh the pain |
| 07:38.50 | jac]Z[oby | yeees |
| 07:38.54 | creativx | why havent i configured conferencing when i need it |
| 07:42.45 | *** join/#asterisk [Airwolf] (n=airwolf@dsl5402DE87.pool.t-online.hu) |
| 07:44.10 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 07:44.20 | *** join/#asterisk j0 (i=dan@S01060016b6b541be.va.shawcable.net) |
| 07:46.30 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:48.00 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 07:48.08 | *** join/#asterisk daysmen3 (n=primus@host86-138-211-5.range86-138.btcentralplus.com) |
| 07:49.24 | *** join/#asterisk AsteriskAlbania (n=info@217.24.244.130) |
| 07:49.28 | queuetwo | No one knows why .gsm files would play silently over IAX2? Mp3Player, PLayTones, and actual conversations work fine. But when I use Playback or Background, no problem is reported - the console shows the sound playing, but there is no audio. |
| 07:50.10 | queuetwo | Sorry about the tone there - I hit up arrow from a previous conversation. :) |
| 07:56.03 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
| 07:58.49 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 08:01.45 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 08:06.38 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
| 08:11.10 | *** part/#asterisk [Airwolf] (n=airwolf@dsl5402DE87.pool.t-online.hu) |
| 08:12.24 | *** join/#asterisk erwinism (n=erwinpog@203.115.172.243) |
| 08:13.13 | erwinism | hello, how can i write a dialplan script that accepts username and password and checks it on asterisk if it is ok then start the recording? |
| 08:13.52 | erwinism | the username and password is on mysql database |
| 08:14.11 | erwinism | username nad password are actually numeric |
| 08:20.37 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 08:22.46 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 08:25.38 | dlynes_laptop | erwinism, sure |
| 08:25.54 | erwinism | dlynes_laptop: can you help me with that? |
| 08:26.17 | dlynes_laptop | queuetue, it sounds to me like you probably don't have a translation for gsm |
| 08:26.32 | dlynes_laptop | queuetue, do you have format_gsm.so and codec_gsm.so loaded? |
| 08:26.48 | dlynes_laptop | queuetue, do a 'show modules like gsm' from the CLI |
| 08:27.07 | dlynes_laptop | erwinism, i won't do it for you, but i will give you some ideas if you want? |
| 08:27.42 | queuetue | dlynes_laptop, Both show, but with 0 use counts. |
| 08:28.01 | dlynes_laptop | queuetue, now type 'show translation' from the CLI |
| 08:28.08 | dlynes_laptop | queuetue, do you see gsm listed there? |
| 08:28.24 | queuetue | It is. |
| 08:29.08 | queuetue | dlynes_laptop, It loks like there is no g729, g723 or gsm translation, but there is a ulaw one. |
| 08:29.12 | dlynes_laptop | queuetue, ok, now type iax2 show peer xxx, where xxx is the name of the iax2 peer that's having problems iwth the gsm file |
| 08:29.45 | erwinism | dlynes_laptop: thanks, my scenario is... i want the extension to dial 2020 and then "enter 7 digit username" will play then a user should enter, after that, it asks for 4 digit passcode.. i already have the audio |
| 08:30.02 | queuetue | dlynes_laptop, Ok, I see that - what am I looking for in it? |
| 08:30.25 | queuetue | Codecs : 0xfcbe (gsm|ulaw|alaw|g726|adpcm|lpc10|ilbc) |
| 08:30.25 | dlynes_laptop | queuetue, the codecs that the peer supports |
| 08:31.02 | dlynes_laptop | queuetue, ok now, on the remote end, what codecs does it actually support? |
| 08:31.10 | dlynes_laptop | queuetue, i doubt very much it supports all those codecs |
| 08:32.13 | dlynes_laptop | queuetue, generally in your iax.conf file, you should restrict the allow= statements to the codecs that the client/peer actually supports |
| 08:32.49 | queuetue | In the iax.cof they (VoicePulse) supply: disallow=all / allow=ulaw / allow=alaw / allow=gsm / allow=ilbc / allow=g726 / allow=adpcm / allow=lpc10 |
| 08:32.59 | dlynes_laptop | erwinism, the username is allowed to have zeros? |
| 08:33.20 | dlynes_laptop | queuetue, now that's just plain retarded |
| 08:33.45 | queuetue | dlynes_laptop, I'm a newbie - something I did or something they did? :) |
| 08:33.50 | dlynes_laptop | queuetue, i doubt very much they actually support all those codecs :) |
| 08:33.58 | erwinism | dlynes_laptop: yes zero is allowed |
| 08:34.14 | queuetue | dlynes_laptop, What would you recommend? |
| 08:34.54 | erwinism | dlynes_laptop: can i paste here the url of my sample extensions.conf ? so that you can see |
| 08:34.57 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:35.42 | dlynes_laptop | erwinism, in the context where you're looking for username, have an extension '_XXXXXXX,1,SetVar(USERNAME=${EXTEN}) ;\n_XXXXXXX,2,Goto(get_password,s,1)' |
| 08:35.52 | dlynes_laptop | erwinism, replace the '\n' with the enter key |
| 08:36.07 | dlynes_laptop | erwinism, that'll match your 7 digit code |
| 08:36.12 | *** join/#asterisk speekac (n=alwin@60.51.217.58) |
| 08:36.17 | dlynes_laptop | erwinism, it'll also store it as the username variable |
| 08:36.26 | erwinism | ok got it |
| 08:36.26 | dlynes_laptop | erwinism, or whatever variable name you want to use |
| 08:36.32 | dlynes_laptop | erwinism, do the same for the password |
| 08:36.42 | jac]Z[oby | finnish sounds like two alphabets went to nuclear war and the fallout mutated it into something seriously fucked up with lots of i's and e's. |
| 08:36.47 | jac]Z[oby | lol |
| 08:36.50 | jac]Z[oby | sry wrong chan |
| 08:36.54 | MrChimpy | bork bork bork! |
| 08:36.55 | erwinism | in this time ill use the variables you gave me |
| 08:37.03 | dlynes_laptop | erwinism, and use a Set(TIMEOUT(response)=30) or something similar to capture the number |
| 08:37.10 | MrChimpy | vuuud joou like ze cheeeeky? borrk! |
| 08:37.23 | speekac | does anyone know how to set the Agent "on paused" ? |
| 08:37.39 | dlynes_laptop | jac]Z[oby, Kiitos paliios |
| 08:37.52 | jac]Z[oby | yeees |
| 08:38.05 | jac]Z[oby | would you like some coffee with that |
| 08:38.21 | dlynes_laptop | jac]Z[oby, jackass....kiitos paliios means thank you very much :p |
| 08:38.30 | jac]Z[oby | ;)) |
| 08:38.42 | dlynes_laptop | in finnish, of course :) |
| 08:38.59 | dlynes_laptop | erwinism, you could use pastebin |
| 08:39.01 | dlynes_laptop | ~pb |
| 08:39.02 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 08:39.03 | jac]Z[oby | *adohy* |
| 08:39.23 | dlynes_laptop | how's tanappa? |
| 08:39.48 | dlynes_laptop | erm how's the nappa? |
| 08:40.01 | erwinism | dlynes_laptop: ok im making my dialplan now |
| 08:40.29 | dlynes_laptop | erwinism, you might also want to read the TIMEOUT(response) documentation so you can learn about a few gotchas |
| 08:40.50 | dlynes_laptop | erwinism, in particular, make sure you don't have autofallthrough set to yes in your extensions.conf file |
| 08:41.04 | erwinism | dlynes_laptop: ok thanks |
| 08:41.36 | dlynes_laptop | queuetue, brb |
| 08:44.30 | dlynes_laptop | heh |
| 08:44.31 | dlynes_laptop | somebody's an idiot |
| 08:44.43 | dlynes_laptop | they put www.paste-it.net up as an image paster |
| 08:44.55 | JT | yeah? |
| 08:45.12 | dlynes_laptop | jbot should have svn support so we can find out who made changes to his text :) |
| 08:47.48 | queuetue | dlynes_laptop, switching to only ulaw doesn't make any difference. |
| 08:47.56 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 08:48.22 | dlynes_laptop | hrm |
| 08:48.26 | queuetue | What "raw" formats could I create audio in to see if it's gsm that is the issue? |
| 08:48.28 | erwinism | dlynes_laptop: is this correct? http://www.paste-it.net/71 |
| 08:48.31 | dlynes_laptop | and I've got zero experience debugging iax2, too |
| 08:48.58 | dlynes_laptop | queuetue, actually, do you get any messages about formats in your log? |
| 08:49.14 | dlynes_laptop | queuetue, i.e. in /var/log/asterisk/full? |
| 08:49.19 | JT | queuetue: .wav? |
| 08:49.30 | dlynes_laptop | JT, wav's not a raw format |
| 08:49.35 | dlynes_laptop | JT, it has a header |
| 08:49.53 | JT | can asterisk not handle it though? |
| 08:49.57 | dlynes_laptop | yep |
| 08:50.02 | dlynes_laptop | it can handle wav and wav49 |
| 08:50.06 | JT | ok then |
| 08:50.07 | queuetue | dlynes_laptop, I don't have afull, but n /var/log/asterisk/messages , no, I don't see any. |
| 08:50.42 | queuetue | I've tried wavs and asterisk said they were the wrong format. |
| 08:50.43 | Mr-packet | i'm trying to find some reference as to nuts and bolts of the SIP protocol.. anyone got a good URL to such a thing |
| 08:50.55 | dlynes_laptop | queuetue, get out your trusty editor and add another log entry to your logger.conf file: full => debug,verbose,notice,warning,error,dtmf |
| 08:51.03 | dlynes_laptop | queuetue, then save the file |
| 08:51.10 | dlynes_laptop | queuetue, and then do a logger reload from the CLI |
| 08:51.12 | queuetue | Mr-packet, Why not get the SIP RFC? |
| 08:51.15 | queuetue | dlynes_laptop, Ok. |
| 08:51.19 | dlynes_laptop | queuetue, and then try your gsm file again |
| 08:51.32 | dlynes_laptop | queuetue, and then check that log file for strange stuff |
| 08:51.48 | queuetue | dlynes_laptop, Under what heading? |
| 08:52.07 | queuetue | logfiles? |
| 08:52.49 | dlynes_laptop | queuetue, yes |
| 08:52.59 | dlynes_laptop | erwinism, nah...way off |
| 08:53.26 | erwinism | dlynes_laptop: sorry im newbie :) |
| 08:53.33 | erwinism | but im willing to learn |
| 08:56.44 | queuetue | dlynes_laptop, http://www.paste-it.net/72 |
| 08:56.54 | dlynes_laptop | erwinism, try this: http://www.paste-it.net/73# |
| 08:57.54 | dlynes_laptop | queuetue, ah...seems it might be a problem with your dialplan |
| 08:58.09 | queuetue | Really? Let me put that up. |
| 08:58.11 | *** part/#asterisk hellop (n=hellop@udp115314uds.hawaiiantel.net) |
| 08:58.14 | dlynes_laptop | queuetue, can you pb your dialplan, and indicate where in the dialplan it is that you're trying to play the gsm file? |
| 08:58.23 | Dico_ | hello gang |
| 08:58.30 | L|NUX | dlynes_laptop : one question |
| 08:58.34 | L|NUX | dlynes_laptop : if you have time |
| 08:58.35 | dlynes_laptop | Dico_, got tired of #asterisk-dev? :P |
| 08:58.42 | Dico_ | lol no but |
| 08:58.46 | erwinism | dlynes_laptop: ok i will try this |
| 08:58.47 | Dico_ | it's a user question |
| 08:58.50 | dlynes_laptop | L|NUX, Just ask the question :) |
| 08:59.27 | dlynes_laptop | L|NUX, If it's some complicated ass shit that's going to take three hours to answer, then i'll say I don't have time :) |
| 08:59.40 | MrChimpy | cor, i didn't know -dev existed. this could be useful today :) |
| 08:59.47 | Dico_ | I would like to know if any of you have tried to activate the SUBSCRIBE/NOTIFY with many many users ? |
| 08:59.58 | L|NUX | dlynes_laptop : i have setuped chan_jingle and its working perfect but now i want to call using my softphone to any buddy who is in my talk list how can i do this ? |
| 09:00.05 | dlynes_laptop | Dico_, you mean sip presence awareness? i.e. BLF? |
| 09:00.23 | Dico_ | humm, lol |
| 09:00.23 | L|NUX | i have done this exten => s,1,Dial(Jingle/asterisk/${EXTEN}@gmail.com) |
| 09:00.26 | L|NUX | but its not working |
| 09:00.38 | dlynes_laptop | L|NUX, ummmm...when did i become the expert on chan_jingle? I don't even have it installed |
| 09:00.41 | Dico_ | i know there is a talk about BLF and so on |
| 09:00.49 | L|NUX | dlynes_laptop : okay |
| 09:00.58 | Dico_ | but just the sip SUBSCRIBE/NOTIFY |
| 09:01.37 | dlynes_laptop | Dico_, yeah...isn't that the same thing? |
| 09:01.37 | dlynes_laptop | i.e. subscribecontext=....? |
| 09:01.37 | daysmen3 | anyone here use asterisk realTime queue - ive read the wiki but would like to know if anyones using it in production and how to implement it |
| 09:01.39 | Dico_ | dlynes_laptop, not really : |
| 09:01.42 | dlynes_laptop | daysmen3, lots of people using it in production |
| 09:01.51 | dlynes_laptop | daysmen3, maybe not people on atm though |
| 09:01.59 | Dico_ | what you need to do is activate the priority hint on the extensions |
| 09:02.06 | Dico_ | then send subscribe request |
| 09:02.35 | L|NUX | any one have done text messaging using chan_jingle and res_jabber ? |
| 09:02.36 | dlynes_laptop | exten => 101,hint,SIP/101&SIP/101_2&SIP/101_3&SIP/101_4&SIP/101_5 |
| 09:02.46 | queuetue | dlynes_laptop, http://www.paste-it.net/74 - the message is played in [incoming] - at the top, right after [globals] |
| 09:02.49 | dlynes_laptop | subscribecontext=subscriptions |
| 09:02.52 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.16 | Dico_ | MrChimpy, yes there is a -dev chat where you can chat with developpers ^^ it's pretty interesting there, talking about C code ^^ |
| 09:03.50 | dlynes_laptop | Dico_, dood...what'd you go and tell him about it for? now he's gonna go in there and ask how he can configure his @home dialplan |
| 09:04.06 | Dico_ | dlynes_laptop, i mean, yes you can do this but if you don't specify the subscribe context it's not a prob |
| 09:04.34 | Dico_ | looool |
| 09:04.39 | dlynes_laptop | and if you do specify the subscribe context it is a problem? |
| 09:04.46 | MrChimpy | dico: i guessed. i'm doing app_dial changes at the moment |
| 09:04.52 | queuetue | dlynes_laptop, I actually thought I was following the "textbook" formula with my dialplan... did I do something wierd? |
| 09:04.52 | Dico_ | hey he was telling it was for developping ^^ |
| 09:05.11 | Dico_ | dlynes_laptop, not at all |
| 09:05.18 | dlynes_laptop | Dico_, i know...just giving you a hard time...trying to keep the humor in this channel |
| 09:05.23 | Dico_ | the subscribecontext is only from a user perspective |
| 09:05.42 | dlynes_laptop | Dico_, so what seems to be your problem, then? |
| 09:06.00 | Dico_ | lol, my problem , currently is that |
| 09:06.02 | dlynes_laptop | Dico_, and what do you mean by large installations? i.e. how many users do you consider a large install? |
| 09:06.09 | Dico_ | the way the stuff is currently implemented |
| 09:06.19 | daysmen3 | dlynes_laptop: im looking for examples - any leads |
| 09:06.26 | Dico_ | is theorically memory bomber |
| 09:06.51 | dlynes_laptop | Dico_, you mean it thrashes the hell out of asterisk's memory? |
| 09:07.00 | Dico_ | dlynes_laptop, 10 users is already not small for this ^^ |
| 09:07.16 | dlynes_laptop | Dico_, yeah...I've got issues doing it with 6 |
| 09:07.19 | dlynes_laptop | :) |
| 09:07.24 | dlynes_laptop | Even 1 for that matter |
| 09:07.33 | Dico_ | then : what were your results / |
| 09:07.35 | Dico_ | ? |
| 09:07.36 | dlynes_laptop | but my problems stem from the phones I'm trying to use with it |
| 09:07.58 | dlynes_laptop | I keep getting SIP 489's and SIP 500's back from them from the SIP presence capability |
| 09:08.10 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 09:08.15 | dlynes_laptop | and the manufacturer of the phones refuses to admit their phones have a problem |
| 09:08.40 | dlynes_laptop | well, even if asterisk has a problem, and not the phones, they still shouldn't be issuing a SIP 500 |
| 09:08.43 | JT | which mfg would this be? |
| 09:08.47 | dlynes_laptop | Aastra |
| 09:08.50 | Dico_ | humm, ok may be i will have a closer closser look |
| 09:08.55 | Dico_ | but from what i see is |
| 09:09.13 | queuetue | dlynes_laptop, What made you think it was a problem with my dialplan? |
| 09:09.25 | dlynes_laptop | queuetue, haven't finished looking at it yet |
| 09:09.27 | Dico_ | if you activate the SUB/NOT on each extension |
| 09:09.44 | Dico_ | you have something like 6exp6 sip_pvt created |
| 09:09.59 | Dico_ | sorry, a bit less |
| 09:10.09 | Dico_ | 5^6 |
| 09:10.55 | dlynes_laptop | Dico_, ewww...that could explain why my phones went diving then |
| 09:11.01 | Dico_ | BUT |
| 09:11.16 | Dico_ | it's only only a guess |
| 09:11.19 | dlynes_laptop | Dico_, I had six extensions set up, each subscribing to notifications from all six phones |
| 09:11.23 | Dico_ | i need to have a closer look |
| 09:11.36 | Dico_ | yes : you just dont subscribe to yours |
| 09:11.42 | Dico_ | then 5^6 |
| 09:11.47 | dlynes_laptop | Dico_, so extension 101 would subscribe to notifications from 101, 102, 103, 104, 105, and 106 |
| 09:11.50 | Dico_ | or 5^5 |
| 09:12.08 | dlynes_laptop | I did it that way just to make it easier for them to move from one phone to another |
| 09:12.17 | dlynes_laptop | the buttons would be consistent from one phone to the next then |
| 09:12.35 | Dico_ | humm i think this does not create a so big problem |
| 09:12.41 | Dico_ | i thin it's independent |
| 09:12.58 | dlynes_laptop | Dico_, which phones are you using? |
| 09:13.26 | Dico_ | humm softphones and jupitel |
| 09:13.36 | Dico_ | but we have not activated all yet |
| 09:14.08 | dlynes_laptop | queuetue, when the iax extension is connected to the louie-louie song, do a show channel IAX/... whatever its channel name is |
| 09:14.14 | dlynes_laptop | queuetue, see what codec it's using |
| 09:14.18 | Dico_ | ok i need to see if a subscribe to B and B subscribe to A there are 2 pvt created |
| 09:14.20 | Dico_ | but if so |
| 09:14.29 | Dico_ | imagine with 100 users ; |
| 09:14.43 | Dico_ | 99^99 ... |
| 09:14.52 | dlynes_laptop | Dico_, do you have astmm compiled in? |
| 09:15.10 | Dico_ | humm i dont know |
| 09:15.13 | Dico_ | what is astmm ? |
| 09:15.20 | dlynes_laptop | asterisk memory manager |
| 09:15.25 | dlynes_laptop | it's a memory debugging tool for asterisk |
| 09:15.26 | Dico_ | humm no |
| 09:15.35 | queuetue | dlynes_laptop, ulaw |
| 09:16.19 | dlynes_laptop | Dico_, change the line: MALLOC_DEBUG = #-include $(PWD)/include/asterisk/astmm.h |
| 09:16.36 | dlynes_laptop | Dico_, to MALLOC_DEBUG = -include $(PWD)/include/asterisk/astmm.h |
| 09:16.37 | dlynes_laptop | in your asterisk Makefile |
| 09:16.40 | dlynes_laptop | and then remake |
| 09:16.53 | dlynes_laptop | You'll have two new commands |
| 09:17.16 | dlynes_laptop | show memory report(?) and show memory mappings(?) |
| 09:17.20 | dlynes_laptop | something like that |
| 09:17.37 | dlynes_laptop | It'll show you where all the memory resources are allocated |
| 09:17.50 | erwinism | dlynes_laptop: what if i use "please enter your password followed by the pound key", what should i replace with the line exten => _XXXXXXX,1,Set(USERNAME=${EXTEN}) ? |
| 09:17.51 | queuetue | ulaw and gsm seem to have a translation - 3 with gsm on the y axis, 7 with ulaw on the y. |
| 09:18.02 | queuetue | No clue what the numbers mean. |
| 09:18.24 | dlynes_laptop | erwinism, obviously Set(PASSWORD=${EXTEN}) |
| 09:18.43 | dlynes_laptop | queuetue, the number of milliseconds to do the translation |
| 09:19.03 | queuetue | dlynes_laptop, Ok, then 3 and 7 seem pretty good there.:) |
| 09:19.04 | dlynes_laptop | queuetue, I think that's based on 1s of audio |
| 09:19.13 | dlynes_laptop | queuetue, basically anything under 100 is good |
| 09:19.21 | erwinism | dlynes_laptop: i mean onm the _XXXXXXX .. should i change this? |
| 09:19.25 | dlynes_laptop | if it's over 100, your audio quality will start to suffer |
| 09:19.50 | dlynes_laptop | erwinism, yeah... to _XXXX if you have a four digit password |
| 09:20.25 | Dico_ | humm i think i say something wrong here |
| 09:20.26 | queuetue | dlynes_laptop, what is there left to check? I feel I'm getting low on options here. :) |
| 09:20.35 | erwinism | dlynes_laptop: in that case, it automatically goes to next exten after four digit, it will not wait for the pound key |
| 09:20.38 | Dico_ | let me dig in the code |
| 09:22.01 | dlynes_laptop | queuetue, what's your verboseness level set to? |
| 09:22.20 | dlynes_laptop | erwinism, no need for a pound key |
| 09:22.29 | queuetue | dlynes_laptop, I use "asterisk -vvvvr" - is that what you meant? |
| 09:22.40 | dlynes_laptop | erwinism, it'll automatically jump when it gets 7 digits or 4 digits, depending on whether it's looking for username or passwrod |
| 09:22.50 | dlynes_laptop | queuetue, so verboseness level of 4 |
| 09:22.55 | dlynes_laptop | queuetue, set verbose 100 |
| 09:23.06 | dlynes_laptop | queuetue, then try it again, and check your log |
| 09:23.16 | erwinism | dlynes_laptop: ok thanks |
| 09:24.35 | dlynes_laptop | erwinism, however, you can try using the authenticate application for the password |
| 09:25.06 | erwinism | dlynes_laptop: thanks i will check that |
| 09:25.30 | dlynes_laptop | erwinism, you can specify a file as an argument to it |
| 09:25.39 | dlynes_laptop | erwinism, inside that file, you can have your passwords |
| 09:25.40 | queuetue | dlynes_laptop, I see nothing that indicates an error - "chan_iax2.c: Ooh, voice format changed to 4" is the only thing that has even a hint of info, afaict. |
| 09:25.53 | dlynes_laptop | erwinism, so you could have a different password file for each user |
| 09:26.04 | erwinism | dlynes_laptop: what i am working on right now is, a user can authenticate from pstn line and make voice recording .. :) |
| 09:26.05 | dlynes_laptop | queuetue, format 4 is ulaw |
| 09:26.31 | dlynes_laptop | erwinism, yeah it might be more easily done through an AGI script |
| 09:26.33 | Dico_ | ok apologize oej ! If by any chance you read the log of this chat, sorry .... |
| 09:27.07 | erwinism | dlynes_laptop: ok, i will look into that thank you |
| 09:27.07 | dlynes_laptop | erwinism, seeing as how you don't need to measure the length of the call, you might consider using AGI for it |
| 09:27.20 | Dico_ | ok, never mind : it's friday evening here |
| 09:27.34 | dlynes_laptop | erwinism, if you already know perl or PHP it might be easier for you, too |
| 09:27.37 | Dico_ | it's time for me to go out with 2 pretty girls |
| 09:27.41 | Dico_ | see you all |
| 09:27.47 | dlynes_laptop | Dico_, nah...you took too long |
| 09:27.48 | erwinism | dlynes_laptop: i know PHP |
| 09:27.53 | dlynes_laptop | Dico_, they're already leaving with me |
| 09:28.11 | Dico_ | jealous ^^ |
| 09:28.15 | dlynes_laptop | hehe |
| 09:28.25 | erwinism | Dico_: have a good night hehe |
| 09:30.40 | queuetue | sob0l, am I just looking at a buggy asterisk? |
| 09:30.58 | queuetue | sob0l, Sorry about that - automatic nick completion got me. :) |
| 09:32.17 | erwinism | dlynes_laptop: can you help me on checking for the right password? i have this extension.conf working ... please check http://www.paste-it.net/75 |
| 09:32.41 | erwinism | i would concider each user has a file with password written into it |
| 09:38.03 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:38.32 | dlynes_laptop | erwinism, yeah, so after the TIMEOUT(response)=30 for the password, add in an Authenticate(filename.auth) |
| 09:38.54 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:38.55 | dlynes_laptop | queuetue, probably not |
| 09:39.07 | dlynes_laptop | queuetue, maybe RoyK might have some other ideas you can try |
| 09:39.25 | dlynes_laptop | queuetue, he's pretty knowledgable as well |
| 09:41.38 | queuetue | dlynes_laptop, Could you explain my problem to RoyK more conciseley than I would? :) |
| 09:42.41 | queuetue | RoyK, Essentially, gsm sounds are not playing properly - the console and logs indicate they are playing fine, but no audio is heard at the other end of an IAX2 conversation. PlayTones and MP3Player, as well as actual conversation, work fine. |
| 09:43.03 | RoyK | que? |
| 09:43.09 | erwinism | dlynes_laptop: is this possible? Authenticate(/etc/asterisk/users/${USERNAME}.auth) ? in my idea the username is the filename of the password file with the 'auth' file extension |
| 09:43.26 | dlynes_laptop | erwinism, yep, quite possible |
| 09:43.32 | dlynes_laptop | anyways |
| 09:43.37 | dlynes_laptop | i really need to get some sleep |
| 09:43.41 | dlynes_laptop | i need to get up in five hours |
| 09:43.44 | RoyK | I don't use iax2 anymore.... |
| 09:43.45 | l-fy | hey RoyK, dlynes_laptop |
| 09:43.47 | erwinism | dlynes_laptop: thanks for your time helping me |
| 09:43.57 | dlynes_laptop | good morning, l-fy |
| 09:44.08 | l-fy | RoyK > one day i should take that ban out |
| 09:44.10 | dlynes_laptop | good night, l-fy |
| 09:44.14 | queuetue | RoyK, Do you think it's a bug in iax2? |
| 09:44.32 | dlynes_laptop | queuetue, like i said |
| 09:44.41 | dlynes_laptop | queuetue, i think it's a bug in your configuration, not asterisk |
| 09:44.48 | RoyK | l-fy: ban? |
| 09:44.57 | dlynes_laptop | it seems to be a translation issue |
| 09:45.27 | RoyK | it usually is.. |
| 09:45.30 | l-fy | which bug? |
| 09:45.36 | queuetue | dlynes_laptop, Ok, I just can't find any documentation on what kinds of configuration would enhance or prevent this from working. :) |
| 09:45.37 | l-fy | there is no bug around |
| 09:45.40 | l-fy | asterisk is perfect |
| 09:46.08 | dlynes_laptop | queuetue, like i said...I need to get to sleep |
| 09:46.26 | dlynes_laptop | queuetue, i'm sure you'll find someone on here eventually that can help figure out what your problem is though |
| 09:46.29 | queuetue | Ok, I'll start over with the next shift. :) |
| 09:47.05 | queuetue | Wouldn't asterisk report a problem is it could not translate between codecs? |
| 09:47.11 | queuetue | s/is/if/ |
| 09:47.31 | queuetue | Heh, good try jbot. :) |
| 09:47.39 | EyeCue | yeh it does :D |
| 09:48.19 | queuetue | Because no problem is being reported - it's all hunky-dory as far as the logs and console see it. |
| 09:48.52 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 09:49.36 | queuetue | Is there some person I could pay a reasonable amount to to just debug this? |
| 09:51.01 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:51.25 | hads|home | zeedo... I know that name |
| 09:52.05 | zeedo | hads is a nick I recognise from security-forums |
| 09:52.12 | zeedo | ;-) |
| 09:52.23 | hads|home | :) Haven't been over there for a while |
| 09:52.30 | hads|home | How's things mate? |
| 09:52.38 | zeedo | me neither, I left when they merged it with windowssecurity |
| 09:52.46 | hads|home | Me too |
| 09:52.48 | zeedo | great here, been very busy :-) |
| 09:53.07 | hads|home | Busy is always good |
| 09:53.48 | *** part/#asterisk nailbags (n=nailbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
| 09:53.54 | zeedo | hows it going your end ? |
| 09:54.03 | zeedo | I take it your into Asterisk then ? |
| 09:54.07 | hads|home | Yeah, really good actually. |
| 09:54.43 | hads|home | Yeah, been into Asterisk for a bit. I have a few systems out there and I sell Asterisk/VoIP stuff over here. |
| 09:55.26 | zeedo | cool |
| 10:12.58 | *** join/#asterisk Alex (n=hauntedu@about/mandrake/pdpc.student.HauntedUnix) |
| 10:17.47 | Alex | Hello.. |
| 10:17.57 | erwinism | how can i execute a shell program inside my dialplan? such as lame --resample 11.025 -h ${WAVFILE} ${MP#FILE} ? |
| 10:18.22 | Alex | http://server.alexsmith.org/example.txt <-- can anyone see anything glaringly obviously wrong with it? If I remove the user sections from both the extensions.conf and the .agi, it works fine, however if I include the user sections, it appears that the variable is not defined properly? |
| 10:20.56 | Alex | Apologies if I'm being a little vague - I'll happily explain anything else that I've not been clear about.. :) |
| 10:29.35 | *** join/#asterisk stonedyak (n=chatzill@84.45.193.9) |
| 10:38.44 | fourcheeze | Alex: have no fear :-) |
| 10:38.59 | Alex | fourcheeze: Ah - then fix my problem! :D |
| 10:39.41 | fourcheeze | what is $userpieces ? |
| 10:39.44 | fourcheeze | is this php? |
| 10:40.46 | Alex | Yeah - sorry :) That's php (agi/php) |
| 10:41.06 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 10:41.30 | Alex | it's $userpieces as it's exploding the ( and ) - I'm sure there're better ways to do it.. :) Thing is - it works, if you just have $pin, but when I add in $user, it cocks up. I'm guessing it's something to do with Read(), and how that takes in the vars, but I'm really not sure |
| 10:42.06 | l-fy | Alex > what are you trying to do? |
| 10:43.03 | Alex | l-fy: Simple authentication thing, with a lock feature, it reads in a pin (currently), inserts a row into a table in sql saying the system is being used, checks the pin against the database, goes to an "incorrect pin blah blah" context if the pin isn't there, or continues on to the correct context, if the pin is :) |
| 10:43.36 | Alex | l-fy: I thought, by duplicating the pin block of text, changing it over to 'user', I'd just be able to read in the user variable, in the same way, but.. apparently not :) |
| 10:44.36 | fourcheeze | have you tried echoing userpieces to see what you actually get? |
| 10:44.45 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 10:44.47 | fourcheeze | or userresult |
| 10:45.04 | phearless | my IP phone is stuck at "Configuring CM List |
| 10:45.08 | phearless | is it normal ? |
| 10:45.16 | Alex | fourcheeze: Yeah - it's echoed correctly. |
| 10:45.19 | phearless | it is a Cisco 7960 |
| 10:45.35 | fourcheeze | what are you expecting $userpieces[1] to be? |
| 10:46.27 | Alex | Bah - bad copy/paste, that's from a testing version, however, it should be four digits |
| 10:47.03 | phearless | "Configuring CM List" mean that it is looking for Cisco Call Manager ? |
| 10:47.11 | fourcheeze | can you upload the real version? |
| 10:47.41 | phearless | ? |
| 10:49.41 | fourcheeze | so $userresult is something that uses ( as a delimiter? |
| 10:49.49 | fourcheeze | Alex: do you have a sample string of that? |
| 10:51.30 | phearless | "Configuring CM List" |
| 10:51.36 | phearless | it is written this on the phone |
| 10:51.57 | phearless | and I do not know if it is linked with asterisk |
| 10:54.15 | phearless | i wonder if I have to change the firmware of the phone |
| 10:54.35 | queuetue | Should a 1.2.7 -> 1.2.10 upgrade require any changes in configuration? |
| 10:55.06 | hads|home | queuetue: No |
| 10:55.15 | queuetue | hads|home, Thanks. |
| 10:55.42 | phearless | ... |
| 10:55.48 | hads|home | queuetue: You will probably get more help in here rather than -dev |
| 10:56.24 | queuetue | hads|home, I was only asking in there because I had a build issue - which I think I've resolved now. |
| 10:58.18 | queuetue | Can anyone shed any light into my issue? Essentially, gsm sounds are not playing properly - the console and logs indicate they are playing fine, but no audio is heard at the other end of an IAX2 conversation. PlayTones and MP3Player, as well as actual conversation, work fine. |
| 10:58.36 | queuetue | I'm upgrading, because I can't figure out what else to do ... but it feels like a very "windows" solution to the problem. |
| 11:00.35 | hads|home | queuetue: It's probably best to run the latest release anyway. |
| 11:01.11 | hads|home | queuetue: Also, I'd recommend the source from asterisk.org instead of the deb-src |
| 11:01.17 | *** join/#asterisk DSebas (n=sebas@ip-5-112.sn2.eutelia.it) |
| 11:01.55 | queuetue | hads|home, Do you have any idea how to take the "raw" source and shove it into a deb? |
| 11:02.07 | hads|home | checkinstall ? |
| 11:02.10 | queuetue | I'd prefer to not have mixed managed and unmanaged. |
| 11:02.18 | DSebas | Hi everyone, i have a problem with asterisk at work... when i try to connect an extension from the internet i can't logon, i get "wrong password" in the asterisk log, even if password is ok |
| 11:02.22 | DSebas | any hint?? |
| 11:02.32 | hads|home | queuetue: You know about checkinstall? |
| 11:03.16 | queuetue | hads|home, Looking now, thanks. |
| 11:03.58 | hads|home | It's pretty easy, just apt-get checkinstall and then checkinstall rather than make install |
| 11:04.21 | hads|home | It'll make a deb and install it for you. |
| 11:04.42 | queuetue | hads, Ok. |
| 11:04.50 | DSebas | anyone has seen this problem before? I can't connect from internet with softphones, but if a use ipphones y get "wrong password" |
| 11:04.53 | Alex | fourcheeze: "3333" :p |
| 11:06.48 | hads|home | queuetue: But TBH asterisk is pretty tidy with it's installation anyway so it's not a major to install it by hand |
| 11:07.16 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 11:07.27 | fourcheeze | Alex: what does "(" do as the separator then? |
| 11:08.29 | Alex | fourcheeze: That appears to be how asterisk provides the variable to the ag |
| 11:08.31 | Alex | er, agi |
| 11:09.15 | fourcheeze | OK |
| 11:09.34 | fourcheeze | there's got to be a better way of doing that |
| 11:09.45 | Alex | I'd imagine so :) |
| 11:14.33 | fourcheeze | I've personally never tried sending more than one variable, so I don't know how that works |
| 11:15.29 | fourcheeze | Alex: I tend to do it like this |
| 11:15.32 | fourcheeze | ;exten => _0.,2,AGI(setcidmag|${CALLERIDNUM}) |
| 11:16.23 | RoyK | what does setcidmag? |
| 11:16.33 | RoyK | s/?/ do? |
| 11:16.46 | RoyK | jbot: ping |
| 11:16.49 | jbot | pong |
| 11:17.31 | fourcheeze | RoyK: it sets my looks up the right callerid for a number and presents it in the right way |
| 11:17.36 | fourcheeze | ugh |
| 11:17.45 | fourcheeze | you get the idea ;-) |
| 11:17.51 | RoyK | not really :P |
| 11:18.22 | fourcheeze | checks to see if it needs to substitute the current callerid with a different one |
| 11:18.30 | RoyK | ok |
| 11:18.57 | fourcheeze | I tend to pass all my variables in one go and then send them all back together too |
| 11:19.03 | fourcheeze | and use cut to split them up |
| 11:19.06 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 11:20.55 | *** join/#asterisk oadaeh (n=jason@wsip-24-234-160-51.lv.lv.cox.net) |
| 11:32.07 | queuetue | norse, the upgrade has made no difference at all. I still cannot hear any gsm audio over IAX2. Can anyone give me suggestions on what to try? |
| 11:34.23 | queuetue | No - the upgrade has made no difference at all. I still cannot hear any gsm audio over IAX2. Can anyone give me suggestions on what to try? |
| 11:40.14 | l-fy | queuetue > do you have any prefered codec setup? |
| 11:41.01 | queuetue | disallow=all / allow=ulaw / allow=alaw |
| 11:41.06 | queuetue | l-fy: ^^ |
| 11:42.58 | queuetue | Other audio formats seem to work fine - live voice, mp3Player, PlayTones, all fine. gsm playback or background, console reports no issue, but no audio is heard. |
| 11:44.22 | l-fy | this is why dosen't work |
| 11:44.24 | l-fy | add gsm there |
| 11:44.33 | *** join/#asterisk eric-xx (i=Eric@cm83.epsilon192.maxonline.com.sg) |
| 11:44.33 | l-fy | before disallow=all |
| 11:44.47 | queuetue | Before? In iax.conf? |
| 11:45.02 | l-fy | yes |
| 11:45.49 | queuetue | No, that doesn't make any difference. (Wouldn't the disallow override the allow anyway?) |
| 11:46.23 | l-fy | remove the disallow line |
| 11:46.40 | queuetue | So just allow everything? |
| 11:47.26 | queuetue | Or are you suggesting I force this IAX channel to only use the GSM codec? |
| 11:47.44 | hads|home | Asterisk should translate the gsm from the file into the right format from the channel |
| 11:47.57 | queuetue | hads|home: agree, I'm just following advice here. :) |
| 11:48.14 | queuetue | Anyway, forcing gsm does not work either. |
| 11:48.27 | queuetue | forcing ulaw, alaw, or gsm doesn;t make any difference. |
| 11:49.39 | queuetue | Does anyone have any idea what could cause this? Are there likely to be buggy versions of libgsm out there? Is there a way to test that? |
| 11:50.10 | hads|home | I can't think of anything at the moment sorry, that's whay I haven't been making suggestions. |
| 11:52.15 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:01.16 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
| 12:01.40 | *** join/#asterisk lorinc (n=ang@caracas-3788.adsl.interware.hu) |
| 12:06.22 | *** join/#asterisk CtrlAltCa (n=fabio@host42-242.pool80183.interbusiness.it) |
| 12:07.17 | *** join/#asterisk Yanie (n=Dave@82-69-4-3.dsl.in-addr.zen.co.uk) |
| 12:08.09 | *** part/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 12:09.59 | RoyK | http://www.eminent-tech.com/RWbrochure.htm |
| 12:14.26 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 12:15.34 | norse | brb |
| 12:19.43 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 12:22.23 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
| 12:22.26 | hwt | hi. |
| 12:23.00 | hwt | in my realtime setup, howcome sip friends are listed as Nat=N even though i have Nat=yes in the database? |
| 12:25.23 | Pj_ | Can someone enlighten me on the real meaning of rxgain and txgain ? |
| 12:25.39 | Pj_ | I thought it would up the volume, but it doesn't seem to do that |
| 12:25.59 | Pj_ | more compress the sound than really make it louder |
| 12:26.33 | Pj_ | I've found info on how to set it up but I'd rather understand what it is |
| 12:27.43 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 12:27.48 | *** join/#asterisk Vekka (n=chatzill@dsl-146-91-159.telkomadsl.co.za) |
| 12:27.49 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 12:28.15 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:28.43 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-166.usadatanet.com) |
| 12:29.55 | *** join/#asterisk dioedu (n=dioedu@200.207.150.85) |
| 12:30.11 | *** join/#asterisk af_ (n=af@ip-192-212.sn2.eutelia.it) |
| 12:30.57 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 12:31.36 | Ahrimanes | anyone know if there's a cmd postgresql/pgsql like theres a cmd mysql to send queries from the dialplan ? |
| 12:33.54 | hwt | also i have problems with "mysql server has gone away". |
| 12:33.59 | hwt | when inactive for a while. |
| 12:34.05 | hwt | how can i increase the mysql socket timeout? |
| 12:35.10 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 12:39.49 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 12:44.29 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-166.usadatanet.com) |
| 12:46.39 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:48.17 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 12:52.27 | queuetue | Can anyone shed any light into my issue? Essentially, gsm sounds are not playing properly - the console and logs indicate they are playing fine, but no audio is heard at the other end of an IAX2 conversation. PlayTones and MP3Player, as well as actual conversation, work fine. |
| 12:55.22 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 12:55.31 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 12:59.30 | queuetue | No one? Why would mp3 work and gsm not? |
| 13:00.54 | queuetue | Should I just convert all of my audio to mp3 and use it that way? |
| 13:01.30 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 13:02.58 | fourcheeze | hwt: I think N means nat |
| 13:04.53 | fourcheeze | yeah, N means nat, blank means no nat |
| 13:05.00 | toerkeium | guys, if I want to make calls clicking on a item from a web page, should I use AGI ? or it's only to interact withing dialplans? |
| 13:05.15 | toerkeium | can't understand very well how agi works |
| 13:05.45 | fourcheeze | toerkeium: do you understand http and cgi? |
| 13:05.51 | toerkeium | yes |
| 13:05.55 | nortex | toerkeium, I would think that click to dial would use the manager interface. |
| 13:05.58 | fourcheeze | think of agi in the same way |
| 13:06.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.37 | fourcheeze | http: http request results in script execution |
| 13:06.37 | fourcheeze | * : call results in agi execution |
| 13:06.38 | toerkeium | nortex: you mean the Asterisk Manager API ? |
| 13:07.40 | toerkeium | fourcheeze, but from where should I call the agi script? lets say, to catch a extension status for example |
| 13:07.52 | toerkeium | from a web page or inside a dialplan? |
| 13:07.57 | fourcheeze | always a dialplan |
| 13:08.04 | toerkeium | oh, great |
| 13:08.09 | toerkeium | that's what I was not understanding |
| 13:08.21 | fourcheeze | agi is the bridge between * and the scripting world |
| 13:08.30 | nortex | toerkeium, yup. Checkout www.snapanumber.com and the free firefox plugin, tryit and what your cli and you should see the manager events. |
| 13:08.31 | fourcheeze | always coming from a * dialplan |
| 13:08.32 | toerkeium | so, to interact from a web page I should use asterisk manager api as nortex suggested, right? |
| 13:08.42 | fourcheeze | toerkeium: what do you want to do? |
| 13:09.13 | toerkeium | I need to do a listing of users so operatos can click on a "call button" and make the call |
| 13:09.15 | toerkeium | simple |
| 13:09.19 | nortex | toerkeium, I mean watch your cli, not what your CLI :) |
| 13:09.30 | fourcheeze | toerkeium: have you tried the flash operator panel? |
| 13:09.45 | toerkeium | the one that comes with freePBX ? |
| 13:09.50 | fourcheeze | maybe |
| 13:09.56 | fourcheeze | never tried freepbx |
| 13:10.28 | toerkeium | but I need to load several "names, phone number", and make a web page listing, and then operators will click on every row and place the call |
| 13:10.32 | toerkeium | does FOP have that? |
| 13:11.01 | fourcheeze | toerkeium: not sure |
| 13:11.19 | toerkeium | I think it's just to place a queued call to a user, and such things only |
| 13:11.19 | fourcheeze | probably you want to write something yourself |
| 13:11.43 | toerkeium | I will try at least, now that I know where to stat :) |
| 13:11.51 | toerkeium | thanks guys |
| 13:11.57 | *** join/#asterisk Alex (n=hauntedu@about/mandrake/pdpc.student.HauntedUnix) |
| 13:11.58 | nortex | toerkeium, What browser do the operators use? |
| 13:12.03 | Alex | Anyone know if there's a SIP soft phone that can do messaging? |
| 13:12.06 | toerkeium | mostly IE |
| 13:12.18 | fourcheeze | Alex: Eyebeam? |
| 13:12.43 | Alex | fourcheeze: Can that do messaging? |
| 13:12.47 | nortex | Ah, well I know the guy that writes snap is working on a plugin for IE, but that it is not ready yet. |
| 13:13.15 | fourcheeze | Alex: if it doesn't they've made it far too complicated |
| 13:13.17 | toerkeium | reading that web site you gave me |
| 13:13.34 | fourcheeze | toerkeium: can't you make them use the browser of your choice? |
| 13:13.52 | Alex | fourcheeze: Ah - that's the payfor version of xlite - know of any OSS ones? ;) |
| 13:13.55 | queuetue | What other audio formats can asterisk play? Besides gsm, which appears to not work. |
| 13:14.01 | toerkeium | I could fourcheeze |
| 13:14.16 | nortex | Alex check out the xlite 3.0 free version. |
| 13:14.29 | coppice | queuetue: gsm appears to work just fine for everyone else |
| 13:14.44 | queuetue | coppice: And not for me. |
| 13:14.47 | Alex | nortex: Oh, that includes it? |
| 13:14.53 | fourcheeze | toerkeium: personally I would insist on uniformity otherwise you have twice as many bugs to fix |
| 13:15.14 | toerkeium | fourcheeze: you mean I force them to use firefox ? |
| 13:15.22 | fourcheeze | why not? |
| 13:15.28 | toerkeium | is it better? |
| 13:15.30 | Alex | nortex: I can't see it - am I being ridiculously stupid? :) |
| 13:15.31 | fourcheeze | they can use IE for other things if they want |
| 13:15.35 | Alex | (I do mean SIP SMS messaging) |
| 13:15.40 | fourcheeze | or insist they use IE |
| 13:15.58 | fourcheeze | let them think of it as an application |
| 13:16.08 | fourcheeze | the browser is part of that |
| 13:16.11 | toerkeium | gonna try that snap.. I can make them use firefox, you're right |
| 13:16.11 | nortex | Alex, Which os are you looking for? |
| 13:16.33 | *** join/#asterisk DeeJay[2] (n=deejay2@207.134.166.34) |
| 13:16.46 | fourcheeze | IIRC Netmeeting can do messaging ;-) |
| 13:16.49 | *** join/#asterisk Cooltalk (i=io@69.88.14.117) |
| 13:16.51 | Alex | nortex: Windows |
| 13:16.55 | DeeJay[2] | how do you call the feature that makes a fax answer when it detects a specific ring pattern? |
| 13:16.55 | Alex | fourcheeze: True - but I mean specifically SMS :) |
| 13:17.07 | Alex | rather than just 'chat' |
| 13:17.13 | queuetue | What other formats does asterisk know how to play, besides gsm? |
| 13:17.19 | fourcheeze | but message another user, not a cell phone? |
| 13:17.36 | Alex | queuetue: wav! ;) |
| 13:17.38 | Alex | fourcheeze: Either. |
| 13:18.00 | fourcheeze | Alex: if you want to message a real mobile phone you need to think of how your message is going to get there |
| 13:18.05 | queuetue | Alex: I've tried multiple wavs and asterisk reports it cant play them. Do you know what specific format they have to be in? |
| 13:18.19 | Alex | fourcheeze: Already got that handled :) |
| 13:18.21 | Alex | queuetue: 8000Hz |
| 13:18.24 | Alex | Mono |
| 13:18.45 | fourcheeze | Alex: so you have a sip provider that will take messages and send to sms? |
| 13:19.08 | fourcheeze | anyone using tapi drivers with asterisk? |
| 13:19.15 | fourcheeze | does it work? |
| 13:19.15 | toerkeium | seems interesting that snap thing :) |
| 13:19.17 | toerkeium | gonna try it |
| 13:19.41 | nortex | Alex, Have you installed it yet? it expands out just like eyebeam. |
| 13:20.10 | DeeJay[2] | We have a 1124 mediatrix SIP gateway, and we would like to be able to send from asterisk different ring pattern so that a fax can detect a specific one and answer it automatically... |
| 13:20.13 | DeeJay[2] | Does anybody know how? |
| 13:20.23 | Alex | nortex: cool - i'll give it a go - thanks |
| 13:20.27 | DeeJay[2] | does it works thru SIP? |
| 13:20.32 | nortex | fourcheeze, I really like the one I'm using to dial my outlook contacts. |
| 13:21.03 | fourcheeze | how secure is it to allow people to do this with a shared * ? |
| 13:21.34 | nortex | fourcheeze, What do you mean by "shared *" |
| 13:21.39 | *** join/#asterisk creativx (n=creadure@196.82-134-19.bkkb.no) |
| 13:21.46 | fourcheeze | ok, many clients on the same platform |
| 13:21.56 | nortex | ahh |
| 13:21.56 | fourcheeze | of course we run it |
| 13:22.03 | queuetue | Alex: Nope, everything I try to conver thtem to, I get " format_wav.c:153 check_header: Not a wav file 2" |
| 13:22.06 | fourcheeze | but I don't want people to be able to start up calls for each other :-) |
| 13:22.19 | queuetue | Can anyone shed any light into my issue? Essentially, gsm sounds are not playing properly - the console and logs indicate they are playing fine, but no audio is heard at the other end of an IAX2 conversation. PlayTones and MP3Player, as well as actual conversation, work fine. |
| 13:22.36 | hmmhesays | bah this patch is proving to be a huge pain in the arse |
| 13:22.41 | nortex | I would think it would be as secure as any other manager connection, setup specific ones per client. |
| 13:22.48 | Alex | queuetue: Strange :/ it's worked fine here - what does file <file> say? |
| 13:23.21 | queuetue | Alex: /usr/share/asterisk/sounds/cash.wav: RIFF (little-endian) data, WAVE audio, Microsoft ADPCM, mono 8000 Hz |
| 13:24.00 | nortex | fourcheeze, The tapi application we went with was snap, the pro version. |
| 13:24.03 | coppice | I don't think * understands MS ADPCM |
| 13:24.15 | queuetue | What's the downside to just using MP3 for all of my audio? |
| 13:24.37 | Alex | queuetue: wrongpin.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz |
| 13:24.50 | hmmhesays | talkin bout the chine grove whoaoa |
| 13:24.56 | hmmhesays | whooaaa china grove |
| 13:24.56 | queuetue | Alex: Can you put that somewhere for download? |
| 13:25.53 | Alex | queuetue: Sure - made it with audacity, btw |
| 13:26.13 | queuetue | Alex: I don't have a mic that works with my mac, or I'd do the same. :) |
| 13:26.30 | *** join/#asterisk tonys (n=tonys@201.216.253.149) |
| 13:26.38 | Alex | queuetue: You can reformat in audacity, methinks. |
| 13:28.15 | tzanger | in sip.conf, is there a similar directive to zapata.conf's callerid? i.e. a stupid-simple way to set callerID to a value on calls received from the user? I know I can set it up in the dialplan, I'm just wondering if sip.conf has a similar directive |
| 13:28.27 | tzanger | it doesn't seem to, which is fine, This is just for knowledge |
| 13:28.35 | Alex | queuetue: http://alexsmith.org/stuff/goodbye.wav |
| 13:29.04 | fourcheeze | nortex: does the snap thing work with mozilla on linux? |
| 13:29.12 | tzanger | snap rocks |
| 13:29.14 | hmmhesays | snap? |
| 13:29.26 | tzanger | I didn't realize it had an extension for firefox and thunderbird..that's cool |
| 13:30.04 | nortex | fourcheeze, the browser plugin or the tapi? |
| 13:30.17 | fourcheeze | either/both |
| 13:30.34 | fourcheeze | put it this way. I have mozilla on linux - what do I need to try out snap and will it work? |
| 13:30.47 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 13:31.50 | queuetue | Alex: Well, that's interesting - it had no problem with the file, but still no audio. |
| 13:32.02 | Alex | queuetue: How odd. |
| 13:32.12 | queuetue | How do I set the outgoign volume level of playback()? |
| 13:32.38 | nortex | fourcheeze, The tapi is for windows so no there, and the browser plugin for firefox, I am not sure if those plugins work in mozilla. |
| 13:32.40 | hmmhesays | heh joe walsh is doing his first tour with the james gang in 35 years |
| 13:32.53 | [TK]D-Fender | tzanger : Same directive. |
| 13:33.16 | Nugget | I'll be he locks the doors in case they're attacked. |
| 13:33.27 | *** join/#asterisk mocker (n=ks@mocker.org) |
| 13:33.33 | [TK]D-Fender | tzanger : if you don't set it it will accept whatever the client presents (good for using a "friend" as a gateway between PBX's) |
| 13:34.01 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 13:34.04 | queuetue | Is there any way to increase asterisk's volume level? |
| 13:35.37 | Dovid | what is this ? |
| 13:35.38 | Dovid | app_dial.c:1049 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 13:35.49 | hmmhesays | exactly what it says |
| 13:35.57 | Dovid | as of this morning asterisk decided not to make any calls. |
| 13:36.02 | hmmhesays | can't create a channel because there is no one to call |
| 13:36.06 | hmmhesays | sip show peers |
| 13:36.07 | Dovid | it was working 3 hours ago |
| 13:36.33 | queuetue | txgain is only for zapata channels? |
| 13:36.52 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 13:36.55 | [TK]D-Fender | queuetue : yup |
| 13:37.21 | coppice | queuetue: where else would gain make any sense? |
| 13:37.52 | queuetue | Well, I'm trying to increase the sound level of a non-zapata channel. |
| 13:38.06 | queuetue | So, there would make sense, to me. |
| 13:38.16 | [TK]D-Fender | queuetue : blame your endpoints |
| 13:38.16 | coppice | queuetue: but that doesn't make any sense |
| 13:38.24 | queuetue | coppice: Are you angry with me or something? |
| 13:38.48 | coppice | queuetuw: no. its just a meaningless this to do |
| 13:38.55 | queuetue | [TK]D-Fender: Well, I have an issue with playback not producing any audio. I'm just trying to research solutions. |
| 13:39.40 | fourcheeze | nortex: I'm happy using firefox. I installed the plugin but I can't see how to make it work / |
| 13:39.51 | queuetue | If someone could explain to me why actual conversation works fine, why PlayTone works fine, and MP#Player works fine, but Playback and Background don't, I'd love to hear it. ;) |
| 13:40.10 | Dovid | Argh. I keep getting it. but everything is available |
| 13:40.22 | Dovid | sip show registry shows that i am registerd |
| 13:40.31 | nortex | fourcheeze, browse to a page with a number on it and it should become a snap://xxxxxxx hyperlink. |
| 13:40.35 | hmmhesays | Bah I wish dells website would say if the drives are hot swappable or not |
| 13:40.50 | Dovid | sorry. sip show peers |
| 13:40.53 | nortex | hmmhesays, which system? |
| 13:41.03 | hmmhesays | any of the poweredges |
| 13:41.32 | nortex | hmmhesays, the 2850 yes, 1850 no and 850 no for sure :) |
| 13:41.39 | *** join/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 13:41.45 | hmmhesays | where do you see this? |
| 13:41.50 | Dovid | can this be the issue ? |
| 13:41.50 | Dovid | Name/username Host Dyn Nat ACL Port Status |
| 13:41.50 | Dovid | 10305/10305 (Unspecified) D N 0 UNKNOWN |
| 13:42.07 | hmmhesays | yes |
| 13:42.27 | Dovid | can qualify=yes be the issue here ? |
| 13:42.48 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 13:43.21 | nortex | hmmhesays, in my NOC :) |
| 13:43.28 | hmmhesays | haha |
| 13:43.29 | queuetue | [TK]D-Fender: Is there a way to set the gain on a non-zapata channel? Specifically, an IAX one? |
| 13:43.47 | fourcheeze | nortex: ok, I've got that which then takes me to the snap website configuration instructions. Most of the instructions refer to my system tray, which I don't have :-) |
| 13:44.08 | *** join/#asterisk Ne0traX (n=mmm@213.255.65.47) |
| 13:44.16 | Dovid | can qualify be null in asterisk real time ? |
| 13:44.53 | hmmhesays | they call them media bay hard drives |
| 13:45.41 | nortex | fourcheeze, crap, sorry about that I missed that the firefox plugin required the snap program which is windows only right now. Guess that won't help. |
| 13:45.54 | fourcheeze | :/ |
| 13:46.02 | nortex | hmmhesays, Yeah, silly Dell, standards matter. |
| 13:46.30 | hmmhesays | I need something with hot swappable drives to support about 100 users, but don't want overkill |
| 13:46.38 | hmmhesays | looks like this server is going to be about 5k |
| 13:46.59 | [TK]D-Fender | queuetue : Nope. |
| 13:47.18 | queuetue | [TK]D-Fender: OK. Any clue what would cause my situation? |
| 13:47.24 | nortex | hmmhesays, That sounds like overkill :) |
| 13:47.52 | [TK]D-Fender | queuetue : Gains are assumed to be somewhat normalized across enpoints, and only PSTN is considered variable to the point of needing adjustment at the PBX level |
| 13:48.45 | [TK]D-Fender | hmmhesays : Looking for a RAID hot-swap drive chassis? |
| 13:49.08 | queuetue | [TK]D-Fender: Well, if that's not it (some volume setting way too low) then do you have any clue what would cause Playback and Background to act like they are producing audio, but not to make it to the other end of the IAX line? |
| 13:49.20 | hmmhesays | [TK]D-Fender: yeah, preferrable dell |
| 13:49.30 | hmmhesays | looking at a poweredge 2800 here |
| 13:49.49 | nortex | hmmhesays, We did a 2850 with hot swap drives and redundant power 2 gig of ram all aroung 2.5K |
| 13:50.49 | hmmhesays | single or dual proc? |
| 13:51.02 | nortex | hmmhesays, It hardly blinks handeling the 70-80 users on it. Of course we are only using 1 T-1 line so our call volume may be different. |
| 13:51.14 | nortex | hmmhesays, dual |
| 13:51.15 | [TK]D-Fender | queuetue : no idea. |
| 13:51.33 | queuetue | It' snot just IAX, I see now. SIP playback is also silent. |
| 13:51.33 | [TK]D-Fender | hmmhesays : You looking for a full SERVER, or just the drive chassis insert? |
| 13:51.45 | hmmhesays | full server |
| 13:51.52 | nortex | hmmhesays, dual 2.8 Xeons |
| 13:52.04 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 13:52.26 | hmmhesays | nortex what the crap why is this 2800 over 5k |
| 13:52.42 | [TK]D-Fender | hmmhesays : Hot swap will cost you a bit.2850 supports it IIRC, but the RAID & SCSI will cost you. I like SuperMicro Personally. then again nobody ever got fired for buying IBM... |
| 13:53.16 | hmmhesays | 5800 bucks, dual proc, hot swap tower chassis |
| 13:53.36 | [TK]D-Fender | hmmhesays : SCSI RAID = green light to overcharge. I'd say white-box a SuperMicro server w/ SATA hotswap RAID |
| 13:53.46 | [TK]D-Fender | hmmhesays : overkill...... |
| 13:54.02 | nortex | hmmhesays, Which proc? |
| 13:55.14 | [TK]D-Fender | hmmhesays : this is for an * box? What kind of transcoding & PSTN connectivity? |
| 13:55.33 | hmmhesays | dual 3.0ghz's |
| 13:55.35 | *** join/#asterisk Trakkasure (n=Sgemtum@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 13:55.40 | hmmhesays | 100 transcoded calls |
| 13:55.42 | hmmhesays | all ip |
| 13:55.53 | hmmhesays | got gateways to take the PSTN load off |
| 13:56.20 | [TK]D-Fender | hmmhesays : Why all transcoded? |
| 13:56.40 | hmmhesays | because you prepare for that to happen |
| 13:56.43 | queuetue | What would be the downside to doing all of my audio in MP3? |
| 13:56.49 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 13:56.55 | creativx | decoding |
| 13:57.26 | [TK]D-Fender | queuetue : Which GSM's are playing low? |
| 13:57.39 | hmmhesays | because when I'm wakeboarding in the bahamas I don't want a call from a customer saying his * box tanked because it tried to transcode to many calls |
| 13:58.13 | queuetue | [TK]D-Fender: I don't know if they are playing low at all. All I know is that I can't hear them. welcome.gsm, for instance - or any other stock one I have tried. |
| 13:58.54 | [TK]D-Fender | queuetue : Make a script to record one and play it back. |
| 13:59.06 | queuetue | Over IAX or SIP, over a multitide of codecs, both wav and gsm - none make any sound. |
| 13:59.26 | hmmhesays | the base 2850 is 3k |
| 13:59.30 | queuetue | [TK]D-Fender: Ok, can you give me the name of the asterisk command so to record I can find the right page? |
| 13:59.33 | l-fy | queuetue > put something like playback(Welcome) |
| 13:59.39 | l-fy | without .gsm |
| 13:59.53 | queuetue | l-fy: That's how I've got it. |
| 14:00.05 | queuetue | Lower-case w, though - like the file name |
| 14:00.26 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 14:00.46 | [TK]D-Fender | queuetue : I'd start suspecting NAT issues cutting off sounds and reinvited calls being the only thing surviving.... |
| 14:01.43 | hmmhesays | [TK]D-Fender: where do you get your supermicro stuff? |
| 14:01.54 | queuetue | [TK]D-Fender: The same problem exists within sip inside and outside of the NAT, and with IAX to my VOIP provider over PSTN. I don't think that's the issue, but if you can think of a way it could be ... |
| 14:02.23 | hmmhesays | i haven't had to buy hardware in such a long time |
| 14:05.42 | hmmhesays | nortex what os are you running on your poweredges? |
| 14:06.04 | nortex | hmmhesays, I hate to admit it here but CentOS. |
| 14:06.31 | eKo1 | What is wrong with CentOS? |
| 14:06.37 | hmmhesays | nothing wrong with cent |
| 14:06.44 | hmmhesays | Rhel, without the cost |
| 14:06.50 | [TK]D-Fender | hmmhesays : Haven't ordered any personally, but was considering it for a rebuild of my * server. |
| 14:06.58 | [TK]D-Fender | hmmhesays : Can get that stuff anywhere |
| 14:07.02 | queuetue | "RHEL" is the problem with centos. ;) |
| 14:07.07 | nortex | It seems everyone around runs Debian or gentoo. |
| 14:07.14 | hmmhesays | queuetue: huh? |
| 14:07.17 | hmmhesays | I like debian a lot |
| 14:07.32 | eKo1 | nortex: I run gentoo on my home workstation. |
| 14:07.37 | hmmhesays | i run fc5 as my workstation |
| 14:08.09 | eKo1 | but my main * box runs on fc2 :P |
| 14:08.50 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 14:08.52 | hmmhesays | nortex what are you doing with your 2850? |
| 14:09.08 | [TK]D-Fender | Mine work server is on FC3 (it works so I don't care), and all my other systems are Slackware. Most of my clients are on CentOS and I'm taking a liking to it. Will likely convert my home server to it. |
| 14:10.40 | nortex | hmmhesays, Our 2850 is the where all the SIP endpoints are and we run a couple 1000 calls a day on it, then I have an 850 with the T-1 card and IAX trunk between the 2. |
| 14:10.54 | [TK]D-Fender | FC irks me. a LOT. To flakey. |
| 14:10.59 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.9.Dial1.SanJose1.Level3.net) |
| 14:11.42 | DeeJay[2] | We're running all our * on Debian |
| 14:11.45 | ManxPower | I try Redhat once a year. Every time it has issues that we do not find acceptable. |
| 14:12.09 | queuetue | DeeJay[2]: What version? |
| 14:12.47 | nortex | I started with AAH when learning asterisk and so Centos just made sense. So kept it and built a production enviroment on the asterisk source. |
| 14:12.56 | DeeJay[2] | 3.1 |
| 14:13.00 | DeeJay[2] | with a custom kernel |
| 14:13.26 | Alric | ManxPower: Like what? I'm far from a power user, so I probably haven't seen these things. It seems to work alright for me. |
| 14:13.39 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 14:13.51 | DeeJay[2] | We ran one on FreeBSD too |
| 14:14.07 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-2.cybersurf.com) |
| 14:14.08 | DeeJay[2] | but zap drivers are beta on it, so if you don't need it it's good... |
| 14:14.48 | ManxPower | Alex, My clients use Webmin for basic admin. Redhat does not ship with it. That's the one I recall right now, but there were other issues, mostly lack of specific software we use. |
| 14:15.02 | Alex | Manx => The Isle of Man, or ManX? ;) |
| 14:15.13 | ManxPower | All that stuff is included with Mandrake right out of the box, so it saves us a lot of time. |
| 14:15.21 | ManxPower | Alex, Simpsons. |
| 14:15.24 | Alex | aw |
| 14:16.18 | ManxPower | My main client is a $550 million/year realestate company with 19 offices in 2 states with an IT staff of 2 people and a consultant. The #1 highest priority for anything is "it saves us time" |
| 14:16.53 | eKo1 | The business of economy. |
| 14:16.56 | ManxPower | IT = Desktops, WAN, LAN, Telecom |
| 14:17.10 | nortex | Sound familiar. |
| 14:17.10 | Alric | Sounds like they have plenty of job security... |
| 14:17.29 | *** join/#asterisk batphone (n=bugz@cpe-70-123-122-41.houston.res.rr.com) |
| 14:17.43 | ManxPower | Alex, they hired another person about a month ago, and we are all swamped with work. |
| 14:17.45 | batphone | something bad happened today |
| 14:17.53 | ManxPower | sob0l, I guess it's 3 people, 1 consultant. |
| 14:18.00 | batphone | one of our main sipconnect providers made a change to their switch |
| 14:18.05 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 14:18.07 | batphone | to incorporate "symmetric nat" |
| 14:18.29 | batphone | now we are sending back 407 unauthorized on all incoming calls from them |
| 14:18.35 | batphone | "proxy authentication required" |
| 14:18.38 | southtel | Is there a specific escape sequence for RealTime Extensions? |
| 14:18.39 | coppice | its amazing how many people are convinced that 100% of telephony works at 8000 samples/second. some blinkers must be really effective :-) |
| 14:18.53 | batphone | this only affects, oh, every customer we have |
| 14:18.57 | batphone | any ideas anyone? |
| 14:19.25 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 14:19.32 | *** join/#asterisk frenzy (n=frenzy@196.46.104.233) |
| 14:20.00 | frenzy | whenever I reboot my box all files in /var/run become owned by asterisk:asterisk |
| 14:20.07 | frenzy | how do I stop that? |
| 14:20.24 | frenzy | did change the astrundir to /var/run/asterisk |
| 14:20.41 | rob0 | frenzy: don't reboot?? Let me guess ... trixbox? |
| 14:20.47 | frenzy | and also recompiled asterisk but still the same... permissions get overrighted |
| 14:20.48 | frenzy | nop |
| 14:20.52 | frenzy | freepbx |
| 14:21.13 | *** join/#asterisk mirax (n=mirax@ip68-1-148-114.mc.at.cox.net) |
| 14:21.27 | rob0 | What a silly hack they did. |
| 14:21.46 | frenzy | ? |
| 14:22.10 | rob0 | I would suggest you run your * on an unbroken OS. |
| 14:22.15 | frenzy | yup and also changed the Astdir in the Makefile |
| 14:22.25 | rob0 | and see /topic for freepbx questions! |
| 14:23.25 | frenzy | huh |
| 14:27.25 | [TK]D-Fender | frenzy : When you joined the channel ... : Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 14:27.46 | fourcheeze | why would a snom phone generate this: -- Incoming call: Got SIP response 489 "Bad Event" back from 86.141.177.106 |
| 14:28.26 | toerkeium | damn, the snap thing needs xp or 2003 |
| 14:28.27 | frenzy | HUH |
| 14:28.40 | frenzy | well it was a BAD on freepbx |
| 14:28.58 | frenzy | the defaults BREAK the system |
| 14:29.14 | frenzy | unless * is running as root |
| 14:33.09 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 14:35.12 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:36.18 | *** part/#asterisk frenzy (n=frenzy@196.46.104.233) |
| 14:37.05 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 14:37.21 | *** join/#asterisk MikeJ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
| 14:38.26 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 14:38.38 | Dovid | what is this ? |
| 14:38.40 | Dovid | chan_sip.c:11564 sip_poke_noanswer: Peer '10310' is now UNREACHABLE! Last qualify: 0 |
| 14:38.44 | Dovid | what does 0 mean ? |
| 14:39.11 | southtel | I'm trying to use a ";" character in SetCdrUserField via RealTime Extensions, but it isn't working...thoughts anyone? |
| 14:39.17 | *** join/#asterisk SwK (n=Silik0nJ@204.250.115.130) |
| 14:39.38 | fourcheeze | got a weird situation here |
| 14:39.43 | southtel | When I've done this via flat files, I just use "\;" and it works fine, but that doesn't seem to work using Realtime. |
| 14:39.58 | fourcheeze | customer on 2 sites with 2 snom phones at one site and a single snome at the other |
| 14:40.08 | fourcheeze | asterisk out in the internet |
| 14:40.18 | fourcheeze | both sites with nat firewalls (identical routers) |
| 14:40.29 | fourcheeze | identical snom configuration |
| 14:40.33 | southtel | Dovid: I believe that is is the last qualify delay, but I'm not certain. |
| 14:40.43 | Dovid | what do u mean by delay ? |
| 14:40.57 | Dovid | cause the phone is 150 ms away |
| 14:41.14 | Dovid | funny thing it is only happening where i have 2 phones behind NAT |
| 14:41.15 | fourcheeze | however for incoming calls the snoms at site A can answer fine, but if snom at site B answers asterisk logs a 489 "Bad Event" |
| 14:41.19 | Dovid | the same NAT that is |
| 14:41.22 | fourcheeze | and the call dies |
| 14:41.30 | fourcheeze | any ideas where to start with this? |
| 14:42.12 | southtel | Dovid: IIRC, the qualify time when you do a "sip show peers" indicates the amount of delay/lag to the extension. |
| 14:42.45 | Dovid | southtel: i know that. for some reasons the two phones behind the same NAT wont work |
| 14:42.47 | *** join/#asterisk lorinc (n=ang@caracas-2500.adsl.interware.hu) |
| 14:44.33 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 14:46.53 | jbalcomb | OOT: Anyone well versed in postfix with amivis and dbmail? |
| 14:47.20 | xai | We have 2 - PRI T1s which all our inbound calls ride on and IAX |
| 14:47.20 | xai | Terminations for our outbound with Asterlink. We have a lot of dropped calles even under light load, mostly outbound calls. |
| 14:47.58 | xai | We also loose sync occasionally even though we use the verizon source (t1 provider).. they say their side is clean. |
| 14:49.03 | xai | yes, we are running 8 1.2.10. anyone have similar issues? |
| 14:49.11 | Juggie | xai, the first thing i would check would be your system |
| 14:49.46 | xai | Juggle: we've changed the pri cards out twice, and used differnet systems too. what in particular would you check? |
| 14:50.01 | Juggie | eg, what the hell is it, is are any irq's shared... did you disable all un-needed devices... if you run zttool is the card having alot of irq misses, what kind of hdd/storage, did you enable dma on your storage device.. |
| 14:50.42 | coppice | xai: what is in your zaptel.conf file? |
| 14:50.51 | *** join/#asterisk Tili (n=tili@202.133.67.100) |
| 14:50.56 | Juggie | are the calls being dropped because of a network problem? or are they being dropped @ the t1 end |
| 14:51.00 | Juggie | thats what you have to figure out. |
| 14:51.03 | *** join/#asterisk ESCulapio__ (n=ESCulapi@200.88.44.66) |
| 14:51.53 | xai | coppice: we had some * guys look over the configs and they said it was all ok.. I'll take Juggie's advice and start looking at config.. They use a stock kernel which is probably super bloated. |
| 14:52.12 | Juggie | xai, run zttool and check for irq misses |
| 14:52.21 | xai | ok.. i will thanks. |
| 14:52.28 | Juggie | if zttool isnt installed, go to the zaptel source and run 'make zttool' |
| 14:52.35 | Juggie | if it doesnt compile you need libnewt-devel |
| 14:52.37 | coppice | xai: so, what does it say? half the people who offer advice are clueless |
| 14:52.38 | Juggie | and then retry. |
| 14:53.19 | xai | coppice: i'll have to get it.. I don't have access to it now.. |
| 14:54.10 | xai | coppice: i'll show it to you later on. |
| 14:54.11 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 14:54.23 | Samoied | Hello All! |
| 14:54.34 | Samoied | anyone know a good sip client |
| 14:54.39 | Samoied | for MACOSX |
| 14:54.42 | *** join/#asterisk hohum (n=dcorbe@mail.interceltelecoms.com) |
| 14:54.53 | Samoied | that have Video support? |
| 14:56.48 | xai | Does anyone maintain a minimul requirements for linux kernel * support? |
| 14:58.47 | *** join/#asterisk oej (n=oej@65.204.4.227) |
| 14:59.10 | xai | or perhaps a .config would be better.. thanks. |
| 15:00.16 | eKo1 | xai: I don't think so. |
| 15:00.34 | eKo1 | Why would someone bother with that in the first place. |
| 15:00.53 | coppice | E-bola you don't EC SIP channels |
| 15:01.01 | E-bola | whY? |
| 15:01.07 | E-bola | i need to get rid of the echo |
| 15:01.14 | fourcheeze | does the polycom 301 have any programmable keys? |
| 15:01.23 | coppice | You cannot EC SIP |
| 15:01.38 | E-bola | crap i was afraid of that :( |
| 15:02.07 | E-bola | So if our setup on our side is purely sip phones, connected to a * server on the lan, which is connected to a sip->pstn provider over the internet |
| 15:02.18 | coppice | you get the echo out before you put the signal through an RTP route. Once it goes through you cannot remove echo |
| 15:02.19 | E-bola | its the providers fault/task whatever? To do EC? |
| 15:02.46 | E-bola | but everything i have is purely sip/rtp |
| 15:03.06 | eKo1 | Echo is usually due to latency. |
| 15:03.30 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
| 15:03.30 | ManxPower | E-bola, How do you connect to the PSTN? |
| 15:03.38 | E-bola | ManxPower: through a provider |
| 15:03.51 | ManxPower | E-bola, then your PROVIDER is not correctly removing echo. |
| 15:03.53 | E-bola | i reigster with sip on their server |
| 15:04.02 | ManxPower | Since your PROVIDER is where the PSTN/VoIP interface is. |
| 15:04.18 | E-bola | Ok, so to sum it up: Echo only occurs on the transfer from PSTN<>SIP? |
| 15:04.23 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 15:04.28 | E-bola | i cant do anything to improve it on my sip only part of the route? |
| 15:05.16 | [TK]D-Fender | fourcheeze : Depends on your needs. What do you have in mind? |
| 15:05.18 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 15:05.39 | fourcheeze | I need a basic phone that has about 4 blfs |
| 15:05.42 | ManxPower | E-bola, correct. |
| 15:05.50 | E-bola | ManxPower: thanks for clearing that up |
| 15:05.53 | fourcheeze | [TK]D-Fender: wondering if the 301 will do this or if I need a 500 |
| 15:05.56 | E-bola | I couldnt get it confirmed anywhere |
| 15:05.59 | fourcheeze | 501 even |
| 15:06.17 | [TK]D-Fender | fourcheeze : You can get the state in the Buddies screen on any Polycom. If you want it "in your face", the you're looking at an IP 60X |
| 15:06.29 | fourcheeze | ahh ok |
| 15:06.34 | fourcheeze | that might be enoguh - thanks |
| 15:06.39 | ManxPower | E-bola, Echo must ALWAYS be removed before a call becomes VoIP |
| 15:06.45 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 15:08.00 | E-bola | ManxPower: 1 more question if I may. Everytime i call ppl outside my own setup, they complaint hey can barely hear what I am saying, low volume, while i can hear them perfectly |
| 15:08.05 | E-bola | Whats a likely cause for that? |
| 15:08.13 | E-bola | Using a spa922 |
| 15:08.37 | eKo1 | I get that too, but only sometimes. |
| 15:08.47 | iq | Hi |
| 15:08.51 | fourcheeze | E-bola: there's an audio gain control somewhere onthat I think |
| 15:09.00 | eKo1 | On SIP to PSTN calls. |
| 15:09.12 | E-bola | fourcheeze: yep u can +6 gain it |
| 15:09.25 | E-bola | but it seemed to me that made the echo worse? Does that make any sence? |
| 15:09.27 | fourcheeze | is it still quiet then? |
| 15:09.31 | fourcheeze | oh |
| 15:09.40 | E-bola | i think it helps a little on the volume |
| 15:09.46 | fourcheeze | I've never encountered echo on sip except with softphones using speakers |
| 15:10.02 | fourcheeze | which end hears the echo? |
| 15:10.28 | E-bola | only the pstn part |
| 15:10.29 | E-bola | i never hear any echo |
| 15:10.34 | E-bola | everything is basically perfect on my end |
| 15:10.35 | fourcheeze | strange |
| 15:10.38 | E-bola | its the ppl i call that complain |
| 15:10.54 | E-bola | either they can barely hear me, or they can hear their own echo |
| 15:10.57 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 15:11.16 | fourcheeze | E-bola: have you tried a different phone? |
| 15:11.32 | E-bola | fourcheeze: no only of the same model |
| 15:11.37 | E-bola | im gonna try with xlite soon |
| 15:11.43 | E-bola | see if that makes a difference |
| 15:11.48 | fourcheeze | hmm |
| 15:11.55 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:11.57 | fourcheeze | I've not tried the 922 but never had echo on a 841 |
| 15:11.59 | fourcheeze | 941 even |
| 15:12.27 | E-bola | well as manxpower said |
| 15:12.33 | *** join/#asterisk juice (i=1000@mo-71-50-31-176.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 15:12.35 | E-bola | the echo isnt related to my part of the setup |
| 15:12.47 | fourcheeze | where does it come from? |
| 15:13.14 | E-bola | the provider of the pstn<>sip translation |
| 15:13.18 | fourcheeze | ahh ok |
| 15:13.24 | fourcheeze | you've tried the phone with another provider? |
| 15:13.29 | E-bola | no |
| 15:13.34 | websae | you should |
| 15:13.34 | E-bola | that would be a good test too |
| 15:13.38 | fourcheeze | yes, I would |
| 15:14.03 | ManxPower | There are 2 kinds of echo. The first, and most common, comes from the PSTN. The other, less common type, comes from poorly designed phones that cause echo. |
| 15:14.26 | E-bola | ManxPower: is it normal for only 1 end of the conversation to hear the echo? |
| 15:14.31 | E-bola | cuz as i said, everything is perfect on my end |
| 15:15.13 | ManxPower | E-bola, one side echo is pretty common. |
| 15:15.34 | ManxPower | Ebola, also, the echo goes from the analog side -> to the voip side. |
| 15:15.45 | Ebola | =( |
| 15:15.57 | ManxPower | If you are geting echo voip side -> PSTN side then it is caused by your phone. |
| 15:16.15 | E-bola | Mmmm sory dont understand ur arrows hehe |
| 15:16.28 | E-bola | I call a pstn phone form my sip phone, and the pstn phone hears its own voice |
| 15:16.29 | E-bola | while i dont |
| 15:16.39 | E-bola | which one is it then? |
| 15:16.42 | *** join/#asterisk AwgieDawgie (n=awgie@65.168.211.2) |
| 15:16.43 | *** join/#asterisk Ne0traX (n=mmm@213.255.65.47) |
| 15:17.01 | ManxPower | E-bola, that is caused by poorly designed phone |
| 15:17.36 | E-bola | which of them? My sip phone or the person im calling's pstn phone? |
| 15:18.03 | coppice | E-bola: does it change if you hold your hand over the mic of your SIP phone? |
| 15:18.17 | *** join/#asterisk momelod (n=momelod@bas5-toronto12-1128748178.dsl.bell.ca) |
| 15:18.23 | momelod | hello people |
| 15:18.25 | E-bola | coppice: i dont know, but i can try it |
| 15:18.29 | momelod | tgif!! :D |
| 15:18.34 | hmmhesays | hmmm 4k for this poweredge |
| 15:18.47 | E-bola | But is that likely really, thats its bad phoner design? ITs a linksys spa922, i thought that was a pretty decent phone |
| 15:19.26 | fourcheeze | it's ok, I've had people that didn't like them |
| 15:19.57 | momelod | got a questions to the polycom users out there, is anyone willing to share there custom ringtones? or know i place where i can download some sip.cfgs? |
| 15:20.01 | coppice | E-bola: if you use the phone at high volume its quite likely. A good phone echo cancels the handset. many expensive ones don't |
| 15:20.13 | E-bola | ohh i use it at the max volume |
| 15:20.22 | E-bola | because of the low volume problems i was talking about |
| 15:20.24 | *** join/#asterisk salviadud (n=ralfalfa@201.123.130.150) |
| 15:20.49 | momelod | E-bola: low volume on zap channels? |
| 15:21.15 | E-bola | i dont use zap at all |
| 15:21.22 | momelod | never mind then :) |
| 15:21.31 | E-bola | :) |
| 15:21.57 | *** join/#asterisk lorinc (n=ang@caracas-4439.adsl.interware.hu) |
| 15:23.08 | E-bola | Which is smarter? To increase gain on my sip phone or on asterisk? |
| 15:23.33 | ManxPower | Ebola, you can't change the gains on Asterisk for VoIP stuff. |
| 15:24.00 | E-bola | ohh.... |
| 15:24.50 | *** part/#asterisk mirax (n=mirax@ip68-1-148-114.mc.at.cox.net) |
| 15:25.05 | MikeJ | tricky little virus |
| 15:25.14 | E-bola | hehe |
| 15:25.18 | E-bola | copy cat! |
| 15:25.23 | E-bola | Used this nick for ages :) |
| 15:25.26 | momelod | virus's bah, thats for winblows :P |
| 15:25.58 | Ebola | Actually I think the virus has used his nick longer than you on freenode... |
| 15:26.14 | Ebola | Anyway, I don't care as long as people tab out correctly. |
| 15:26.19 | *** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com) |
| 15:27.12 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 15:27.43 | E-bola | mmm possibly true on freenode |
| 15:28.03 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.84) |
| 15:28.10 | salviadud | what's so cool about ebola anyway? |
| 15:28.31 | Ebola | What's so cool about salvia? :) |
| 15:28.38 | file | moo? |
| 15:28.43 | hmmhesays | hmm this 2950 only has pcie slots |
| 15:31.27 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:33.02 | hmmhesays | I guess that means no zaptel hardware |
| 15:33.04 | E-bola | Is there any free programs to messure quality of a voip conversation/link |
| 15:33.11 | E-bola | i could only find commercial ones |
| 15:33.19 | hmmhesays | ethereal |
| 15:33.23 | E-bola | hehe thats a bit low level |
| 15:34.01 | hmmhesays | not really |
| 15:34.08 | hmmhesays | not with the rtp stats |
| 15:34.14 | Alric | Tells you what you need to know. |
| 15:34.43 | E-bola | What about soemthing that could gather stats on it |
| 15:34.47 | E-bola | and maybe graph it out? |
| 15:35.43 | AwgieDawgie | Hello all. Newbie here. I'm curious. Is it still necessary to create sym links between kernels folder and /lib/modules/'uname' for FC4? |
| 15:36.00 | hmmhesays | there is graphing functions |
| 15:36.15 | E-bola | hmmhesays: in? |
| 15:36.24 | *** join/#asterisk FunXioN (n=nunya@63.214.236.169) |
| 15:36.25 | [TK]D-Fender | hmmhesays : Ditch Dell, go IBM |
| 15:36.42 | hmmhesays | where do you order your ibm's from? |
| 15:36.45 | hmmhesays | and ... I like dells |
| 15:36.56 | hmmhesays | has anyone tried to run zaptel hardware in a pci-X slot? |
| 15:36.56 | FunXioN | I like IBMs |
| 15:37.09 | E-bola | HP > * |
| 15:37.14 | [TK]D-Fender | hmmhesays : I got mine from MicroServ (div of Hartco) |
| 15:37.40 | FunXioN | IBM x306 is shorter for shallow racks |
| 15:37.48 | FunXioN | use hp for horsepower applications |
| 15:39.51 | *** join/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 15:39.55 | *** part/#asterisk chandi (n=burni13@modemcable248.1-201-24.mc.videotron.ca) |
| 15:40.44 | *** join/#asterisk MikeJ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
| 15:41.11 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:42.19 | FunXioN | does anyone use asterisk with porta one in here? |
| 15:43.33 | hmmhesays | do I really need a dual processor machine to run asterisk? I think not |
| 15:43.51 | FunXioN | I guess that depends on what your doign with it no |
| 15:43.51 | lunk | i ran it on a celeron 333 for a while |
| 15:43.56 | FunXioN | lol |
| 15:43.59 | lunk | for personal use it wasn't bad at all |
| 15:44.21 | lunk | i did upgrade the box to a 2.8ghz though, connects calls in zero seconds instead of five |
| 15:44.23 | lunk | :D |
| 15:46.13 | E-bola | runs fine here |
| 15:46.17 | E-bola | on a 900mhz p3 |
| 15:46.21 | E-bola | load is constantly 0 |
| 15:46.28 | E-bola | i guess its more demanding to use the zaptel stuff |
| 15:46.52 | lunk | E-bola: with the wildcard cheap cards yea. With the nicer hardware cards, it's not that big of a deal |
| 15:47.19 | E-bola | well i guess for my kind of setups its no deal at all |
| 15:47.31 | E-bola | its like an advanced router nothing more |
| 15:47.39 | lunk | yea |
| 15:48.52 | *** part/#asterisk terje (n=joem@67.41.208.129) |
| 15:49.50 | E-bola | Hmm so if RTP packets tell u everything u wanna knwo about the stats of ur calls, why havent soembody made some sniffer/stats/graph package? |
| 15:52.14 | salviadud | mmmmmm, ethereal? |
| 15:53.01 | E-bola | thats not really targeted at graphing voip call quality as far as i know... |
| 15:53.15 | salviadud | yeah, you're right, just packets in general |
| 15:53.20 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.239.9.Dial1.SanJose1.Level3.net) |
| 15:53.25 | E-bola | i meant a whole web interface thing |
| 15:53.29 | E-bola | like mrtg based or similar |
| 15:53.36 | salviadud | ohhh, you like GUI's then |
| 15:53.44 | E-bola | for watching graphs i do hehe |
| 15:53.45 | salviadud | that's so pinache of you |
| 15:54.01 | salviadud | are you running ubuntu? |
| 15:54.04 | E-bola | no |
| 15:54.06 | E-bola | Debian |
| 15:54.12 | asterboy | ubunpu |
| 15:54.16 | E-bola | without any X stuff installed if u wanna know heh |
| 15:54.20 | asterboy | genpoo |
| 15:54.54 | ManxPower | Well, if you want optimization the RTP wont go thru Asterisk anyway |
| 15:58.19 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 15:58.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:59.19 | hmmhesays | well the dell should be on order |
| 16:01.22 | fourcheeze | hmmhesays: which one did you get? |
| 16:01.51 | *** join/#asterisk cbrake (n=cbrake@cpe-65-25-51-117.neo.res.rr.com) |
| 16:02.06 | E-bola | ManxPower: Its more for the information, to see whats going on and if there are any problems |
| 16:02.24 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 16:03.16 | tanvalmg | hi there...is there a reporting tool for Asterisk? Like getting Peak utilization, etc? |
| 16:03.27 | cbrake | does the AIX secret need to be a number, or can it be characters as well? |
| 16:03.32 | CunningPike | tanvalmg: CDR :) |
| 16:03.54 | tanvalmg | CDR? |
| 16:04.03 | CunningPike | tanvalmg: We import our CDR records into SQL Server and then create PivotTables out of them |
| 16:04.03 | fourcheeze | tanvalmg: I was just about to ask that |
| 16:04.06 | file | cbrake: IAX secret you mean? it can be characters and letters... |
| 16:04.13 | CunningPike | ~cdr |
| 16:04.15 | 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 |
| 16:04.15 | fourcheeze | nah CDR is no good |
| 16:04.17 | hmmhesays | now to try out this SLA |
| 16:04.24 | cbrake | file: yes, thx |
| 16:04.36 | fourcheeze | I want to know things like number of sip registrants, which clients have been down and for how long etc etc |
| 16:04.40 | E-bola | Hmmm |
| 16:04.46 | file | hmmhesays: at Telcomjoshvoxmart we have a 100% SLA, if you're 100% having problems we won't care! |
| 16:04.56 | E-bola | i get a very distinct echo even if i hold my hand over the microphone |
| 16:04.58 | E-bola | :( |
| 16:05.10 | CunningPike | fourcheeze: You would probably need to parse the output from sip show peers to get that |
| 16:05.16 | fourcheeze | yeah |
| 16:05.20 | fourcheeze | I think I'll write a little app |
| 16:05.22 | file | hmmhesays: how goes things? |
| 16:05.26 | CunningPike | fourcheeze: I don't know of any existing tool that provides that |
| 16:05.37 | fourcheeze | mainly so that $boss can get a handle on things while I'm on holiday |
| 16:05.43 | tanvalmg | CunningPike: Can I get the utilization of lines (analog) from the CDR records? |
| 16:06.24 | CunningPike | tanvalmg: We have a PRI, and I have a report that shows how many calls on each channel for any given period |
| 16:07.07 | tanvalmg | CunningPike: Forgive my ignorance, but what's PRI? |
| 16:07.19 | CunningPike | ~pri |
| 16:07.20 | jbot | pri is probably Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 16:08.05 | E-bola | omg! |
| 16:08.09 | tanvalmg | CunningPike: how did you generate the report from pri? |
| 16:08.13 | E-bola | it apears to be the linksys phone thats making the echo |
| 16:08.20 | E-bola | if i use xlite there is allmost no echo at all |
| 16:08.21 | E-bola | :( |
| 16:08.50 | CunningPike | tanvalmg: Each CDR record tells you which Zap channel was used for it - I just aggregate them over a given time period |
| 16:08.57 | CtrlAltCa | does anyone know why my tdm400 fxo module always keeps the line up? (i'm in italy) |
| 16:09.49 | queuetue | Can anyone shed any light into my issue? Essentially, gsm and wav sounds are not playing properly - the console and logs indicate they are playing fine, but no audio is heard at the other end of an IAX2 or SIP connection. PlayTones and MP3Player, as well as actual conversation, work fine. |
| 16:12.22 | file | queuetue: do you have a zaptel timing source? |
| 16:13.05 | queuetue | file: I'm not sure. I have a zaptel card - a digium one with two FXO boards on it. |
| 16:13.13 | file | then you are |
| 16:13.22 | Dovid | is this some thing to be alarmed about ? |
| 16:13.23 | Dovid | chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '72.144.247.22' |
| 16:13.32 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 16:13.33 | Dovid | or just ignore it ? |
| 16:13.38 | CunningPike | queuetue: Could it be a codec issue? |
| 16:13.38 | file | Dovid: chan_sip doesn't support the PUBLISH method, so it just ignores it |
| 16:13.43 | Dovid | thanks |
| 16:14.19 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:14.23 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 16:14.29 | file | queuetue: are you running a 1.2 build? |
| 16:14.29 | queuetue | CunningPike: It certainly could be, but show translations indicates that it could convert between gsm and ulaw, and I tried several different codecs. |
| 16:14.32 | queuetue | file: Yes. |
| 16:14.48 | queuetue | 1.2.10 straight from asterisk.org |
| 16:14.49 | file | I wonder... |
| 16:15.43 | file | queuetue: can I gain access to the box? |
| 16:15.49 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 16:16.02 | queuetue | file: Yes. |
| 16:17.22 | momelod | got a questions to the polycom users out there, is anyone willing to share there custom ringtones? or know i place where i can download some sip.cfgs? |
| 16:20.32 | CunningPike | momelod: We use the standard tones - what sip.cfg info do you need? |
| 16:21.18 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:21.18 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 16:21.46 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 16:22.11 | momelod | i want to have new ringtones |
| 16:22.30 | Qwell[] | momelod: make some with playtones :p |
| 16:22.34 | elg | I'm looking for a local DID on a prepaid pay as you go basis. $20/mo is too much for how much I will use the line. any recommendations? |
| 16:22.47 | momelod | Qwell[]: would that produce waves? |
| 16:22.53 | Qwell[] | no, heh |
| 16:23.08 | momelod | good, i want the polymophic tones |
| 16:23.10 | mog | nufone? |
| 16:23.13 | mog | elg |
| 16:23.16 | momelod | ..if thats what u call them? |
| 16:23.26 | Qwell[] | momelod: I made duophonic, if that counts... |
| 16:23.35 | mog | most places charge 2 -4 dollars for did |
| 16:23.35 | momelod | lol, whats that? |
| 16:23.41 | mog | and some few cents a minute |
| 16:23.43 | Qwell[] | umm...dtmf :p |
| 16:24.15 | FunXioN | what for amt do he ringtones need to be in |
| 16:24.32 | FunXioN | format |
| 16:24.44 | Qwell[] | Verizon stole my idea too...I should sue |
| 16:24.55 | momelod | Qwell[]: how do u turn playtones into ringtones? |
| 16:24.55 | FunXioN | whats that |
| 16:25.04 | Qwell[] | momelod: I added support for midi tones...heh |
| 16:25.14 | momelod | oh, that would be awesome! |
| 16:25.16 | Qwell[] | So, you just make a big ugly string of tones, and feed it to Playtones() |
| 16:25.44 | momelod | but playtones() wouldnt ring the phone would it? |
| 16:25.45 | Qwell[] | it's a hack which is incredibly pointless, but...yeah |
| 16:25.47 | tanvalmg | CunningPike: Have you tried to use Asterisk-Stat (GUI CDR Analyzer)? |
| 16:25.56 | FunXioN | momelod are you looking for ringtones or touch tones? |
| 16:26.00 | Qwell[] | momelod: no, it kills the ringing...it's like the Verizon ringback tones |
| 16:26.04 | momelod | ringtones |
| 16:26.05 | elg | mog: that would be fine. I've been with nufone with a toll-free number but I don't think they have a local number for me |
| 16:26.10 | Juggie | hmmm |
| 16:26.12 | Juggie | file |
| 16:26.18 | mog | you could ask |
| 16:26.22 | mog | they have a lot of did |
| 16:26.23 | mog | s |
| 16:26.28 | E-bola | tanvalmg: i use it |
| 16:26.32 | FunXioN | which polycom phone |
| 16:26.40 | Juggie | question: say i want to match two area codes 613 & 819 |
| 16:26.45 | CunningPike | tanvalmg: No - we went with rolling our own reports. We use queue_stats for our queues though - it's pretty good |
| 16:27.03 | Juggie | do i have to do exten=> _613NXXNXXX and then to another for 819 |
| 16:27.04 | file | erm? |
| 16:27.13 | Qwell[] | Juggie: That'd be best.. |
| 16:27.13 | Juggie | or can i put those both in the exten somehow. |
| 16:27.24 | FunXioN | momelod http://www.voip-info.org/wiki/view/Polycom+Phones&view_comment_id=11555 |
| 16:27.31 | Juggie | i thought _[613,819]NXXXXXX might work |
| 16:27.33 | Juggie | but alas, ti does not |
| 16:27.36 | Juggie | *it |
| 16:27.56 | FunXioN | does anyone use asterisk with porta one in here? |
| 16:28.02 | Juggie | Can anynoe think of a way to combine them? |
| 16:28.04 | momelod | thanx FunXioN |
| 16:28.08 | FunXioN | np |
| 16:28.13 | FunXioN | jsut doa search for ringtone |
| 16:28.13 | CunningPike | Juggie: No - you're stuck with two lines |
| 16:28.20 | l-fy | FunXioN > yate+portaone |
| 16:28.25 | FunXioN | o |
| 16:28.28 | FunXioN | really |
| 16:28.30 | FunXioN | how does that work |
| 16:28.31 | Juggie | damit :) |
| 16:28.36 | FunXioN | l-fy? |
| 16:28.49 | l-fy | look for yate |
| 16:28.53 | Qwell[] | FunXioN: she's trolling |
| 16:29.01 | FunXioN | lol |
| 16:29.02 | l-fy | yes :) |
| 16:29.33 | Qwell[] | and if one were to go into #yate and do the same, they'd very likely be removed |
| 16:29.39 | Qwell[] | so...enough |
| 16:29.59 | CunningPike | Juggie: What we need is for the extensions.conf pattern matching to support regex (hint, hint) |
| 16:30.02 | FunXioN | other than cisco and quitum there is no documentation stating that porta supports anything other than asterisk |
| 16:30.26 | l-fy | Qwell > we never do that |
| 16:30.32 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 16:30.55 | Juggie | maybe i should learn ael or something :) |
| 16:31.29 | *** join/#asterisk intensedrive (n=icechat5@vpn-scott.triton.net) |
| 16:31.55 | Juggie | thought ael is just compiled into reg dialplan isnt it |
| 16:31.59 | Juggie | before being loaded? |
| 16:32.01 | Qwell[] | hmm...accents are funny |
| 16:32.01 | Juggie | or has that changed? |
| 16:32.09 | Qwell[] | "flexgrid" becomes "fleshgrid" |
| 16:32.15 | Qwell[] | completely changes the meaning... |
| 16:32.17 | Juggie | accents? éà? |
| 16:34.12 | intensedrive | where can I find app_conference? |
| 16:34.25 | Qwell[] | intensedrive: google.. |
| 16:36.07 | Juggie | its on a source forge cvs. |
| 16:36.14 | Juggie | google for it |
| 16:36.14 | intensedrive | thanks |
| 16:36.18 | intensedrive | I found it |
| 16:36.29 | intensedrive | Im try to design a system and Im not sure if its possible with Asterisk |
| 16:36.38 | Juggie | what do you need to do. |
| 16:37.03 | tanvalmg | CunningPike: is queue_stats opensource too? Where can I download? |
| 16:37.13 | Juggie | intensedrive? |
| 16:37.33 | intensedrive | Well in a nut shell User dials into our IVR system... They login.. There are voice recordings on the system.. the user can choose to conect to that person. The asterisk would then need to dial out and bridge them together |
| 16:37.55 | Juggie | yes, * can do that. |
| 16:38.02 | Juggie | why do you need app_conference for that? |
| 16:38.11 | intensedrive | I thought that was the best way |
| 16:38.14 | Juggie | no. |
| 16:38.19 | Juggie | not at all |
| 16:38.19 | intensedrive | whats the best way |
| 16:38.25 | Juggie | you are over analyzing it. |
| 16:38.41 | Juggie | sounds like your doing an online dating site :) |
| 16:38.46 | intensedrive | ha |
| 16:38.47 | Juggie | er anyways its rather simple. |
| 16:38.52 | Juggie | you could do like |
| 16:38.52 | intensedrive | good guess |
| 16:39.20 | Juggie | exten=> s,1,Background(play voices) |
| 16:39.20 | CunningPike | tanvalmg: Kinda - I think it's Zend encrypted. www.asterisk-guru.com |
| 16:39.42 | Juggie | exten=> _XXXX,1,.... some db code to look up their number from their 4 digit code. |
| 16:39.48 | CunningPike | tanvalmg: Which would make it not OSS, now that I think |
| 16:39.53 | Juggie | exten => _XXXX,2,some error checking |
| 16:39.54 | tanvalmg | CunningPike: Thanks! yeah I found the link in asterisk-guru |
| 16:39.57 | CunningPike | tanvalmg: But it's free as in beer |
| 16:40.03 | Juggie | exten => _XXXX,3,Dial(zap/blah their number) |
| 16:40.04 | Juggie | thats it. |
| 16:40.07 | Juggie | very simple. |
| 16:40.18 | Juggie | thats obviously psudo code |
| 16:40.19 | intensedrive | Right now its all Sip peering |
| 16:40.20 | Juggie | but app_dial bridges |
| 16:40.25 | Juggie | you dont need to do a conference |
| 16:40.33 | CunningPike | Juggie: Don't forget your Answer() first ;) |
| 16:40.34 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:40.41 | Juggie | :) indeed. |
| 16:40.43 | ManxPower | ~docs |
| 16:40.45 | jbot | well, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation 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 |
| 16:41.06 | Juggie | just play the voices, get the input from the user, check the db to get the number |
| 16:41.08 | tanvalmg | CunningPike: :-) |
| 16:41.11 | Juggie | then either say the id is invalid |
| 16:41.13 | Juggie | or connect |
| 16:41.18 | ManxPower | intensedrive, app_conference and app_meetme are for conference calls, you are not doing conference calls. |
| 16:41.47 | intensedrive | well then how do you get around if one person hangs up it doesnnt disconnect from the system |
| 16:42.06 | Juggie | read app dial my friend :) |
| 16:42.13 | *** join/#asterisk xxoxxx (n=xxoxxx@70.91.242.21) |
| 16:42.25 | Juggie | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 16:42.38 | intensedrive | Last question |
| 16:42.47 | intensedrive | When it connects the call person on other side will have to enter a pin.. |
| 16:42.52 | intensedrive | will timing be a problem to connect the call |
| 16:42.57 | ManxPower | intensedrive, The problem with Asterisk not detecting when the far side hangs up is a common problem and can only be solved by using a digital interface to the PSTN |
| 16:43.19 | Juggie | intensedrive, two pronged solution. |
| 16:43.22 | ManxPower | When you use analog ports Asterisk CANNOT detect when the far side picks up or when the far side hangs up. |
| 16:43.34 | Juggie | #1, enable moh or ringing |
| 16:43.38 | Juggie | preferabally moh |
| 16:43.53 | Juggie | #2, use a macro on the dial |
| 16:43.59 | Juggie | so M(mymacroname) |
| 16:44.02 | Juggie | which gets the pin. |
| 16:44.08 | ManxPower | Juggie, Dial macros cannot really detect DTMF, |
| 16:44.17 | Juggie | they sure can |
| 16:44.22 | ManxPower | Trust me, I tried it. ANY DTMF will jump out of the macro. |
| 16:44.31 | Juggie | they can, i've done it. |
| 16:44.36 | ManxPower | I posted it and was basically told that is how it works. |
| 16:44.52 | Juggie | you cant use pattern matching |
| 16:44.57 | Juggie | you have to use Read() |
| 16:45.00 | Juggie | but it works for me. |
| 16:45.22 | ManxPower | Ah, I didn't use read |
| 16:45.32 | Juggie | i did it in a situation where i connect a call to an agent |
| 16:45.37 | hmmhesays | this is odd |
| 16:45.39 | Juggie | and the macro reads out an id to the agent |
| 16:45.43 | ManxPower | I had to play a file over and over until the user dialed a code. |
| 16:45.46 | Juggie | and tells them to press 1 to accept |
| 16:45.46 | ManxPower | that does not work. |
| 16:45.47 | hmmhesays | Looking for 250 in subscriptions (domain 66.173.103.110) |
| 16:45.47 | hmmhesays | Transmitting (NAT) to 66.92.41.45:5060: |
| 16:45.47 | hmmhesays | SIP/2.0 404 Not Found |
| 16:45.50 | Juggie | and 2 to answer |
| 16:45.58 | Juggie | er, 2 to hangup. |
| 16:46.05 | Juggie | anyways intensedrive, to answer your question |
| 16:46.09 | Juggie | use dial with moh (option m) |
| 16:46.15 | Juggie | use a macro to get the pin |
| 16:46.20 | Juggie | option M |
| 16:46.26 | Juggie | which is used like M(macro name) |
| 16:46.38 | intensedrive | what I would really like to do is be able to have both people on the line, bridge the calls together for a certain amount of time, and then seperate them, and then bridge them again |
| 16:46.40 | *** join/#asterisk apardo (n=apardo@87.217.147.51) |
| 16:46.44 | vader-- | do you guys know if there is a user guide to the asterisk voicemail system? |
| 16:47.02 | intensedrive | I have done the macro thing, but there are a couple of hurdles I can't overcome |
| 16:47.03 | *** part/#asterisk trbldwine (i=troubled@adam.ur.northwestern.edu) |
| 16:47.08 | eKo1 | vader--: check the wiki |
| 16:47.09 | Juggie | intensedrive, that would be S() option. |
| 16:47.11 | Ryushin | Hi all, I'm a heavy Linux administrator (Since '94) and network engineer. I'm about to enter the world of Asterisk and VoIP. I'm going to be building a system for a 30 person company. They haven't purchased any phones or any equipment at this point. I'll be building a PoE network. Which phones should I be looking at? I was thinking of the Cisco phones. Any thoughts or insight would be appreciative at this point. |
| 16:47.20 | Juggie | why do you want to unbridge? |
| 16:47.33 | eKo1 | Ryushin: Polycom |
| 16:47.35 | Juggie | to play advertising? |
| 16:47.44 | intensedrive | no |
| 16:47.52 | intensedrive | so that both people can interact with the ivr |
| 16:47.57 | intensedrive | and then reconnect if need be |
| 16:48.05 | Ryushin | eKol: Any particular model I should be looking at? |
| 16:48.19 | eKo1 | eKo1: the PoE model |
| 16:48.34 | Juggie | you can use h,H to allow both users to hang up on the call |
| 16:48.38 | Juggie | by pressing * |
| 16:48.40 | Juggie | what else do you need? |
| 16:48.54 | intensedrive | I think I have to use asterisk managment and send the users to a conferance, take them both out and back again |
| 16:49.15 | *** join/#asterisk beyond (n=beyond@200.192.160.100) |
| 16:49.38 | intensedrive | like the typical call would be, users connect, log in, it connects two of them together for a minute |
| 16:50.04 | intensedrive | then they both interact with the prompts, and it might reconnect them |
| 16:50.19 | intensedrive | I don't want people to have to hangup and then reconnect |
| 16:50.32 | l-fy | FunXioN > still here? |
| 16:50.53 | hmmhesays | hint extension seems to be broken |
| 16:51.10 | CunningPike | Ryushin: We are using IP501s in a government environment and they have been great |
| 16:51.42 | CunningPike | Ryushin: Much better value than Cisco, imho |
| 16:51.59 | E-bola | they arent as flashy designed though :) |
| 16:52.09 | CunningPike | E-bola: Yet :) |
| 16:52.25 | intensedrive | basically a typical call might include one user talking to 5 differant people |
| 16:53.00 | *** join/#asterisk DFiber (n=bwarner@65.113.208.31) |
| 16:53.17 | intensedrive | the user would request a to talk to one person, the system calls them, and verifies it is the right person and then connects them for a minute or 5 minutes |
| 16:53.41 | DFiber | I've just installed asterisk but cannot start it from cli...reasons why?? |
| 16:53.51 | intensedrive | then the connection is interupted, both people interact with the ivr, and they might reconnect to eachother or they might move on and talk to someone else |
| 16:54.50 | intensedrive | I think sending two people to a conferance, pulling them back out and sending them back again is the only way to do it |
| 16:55.06 | l-fy | o god |
| 16:56.29 | vader-- | does asterisk@home use the same voicemail as regular asterisk? |
| 16:56.48 | vader-- | im finding guides for asterisk@home voicemail but nothing for just plain asterisk's voicemail guide |
| 17:00.58 | mocker | Ryushin: I like the Linksys SPA942 |
| 17:00.59 | l-fy | i love my linksys |
| 17:00.59 | l-fy | it works flowlest |
| 17:00.59 | mocker | l-fy: I don't know what that means, but it sounds awesome. |
| 17:01.02 | mocker | :) |
| 17:01.03 | l-fy | flowless |
| 17:01.06 | l-fy | sorry |
| 17:01.15 | l-fy | damn, i should look for that word |
| 17:01.49 | tzanger | flawless |
| 17:01.52 | tzanger | without flaw |
| 17:01.52 | intensedrive | Any IDeas? |
| 17:02.02 | hmmhesays | bah, why isn't my hint extension working |
| 17:02.17 | tzanger | hmmhesays: don't be so subtle... it's obviously not getting the hint |
| 17:02.18 | l-fy | thanks tzanger |
| 17:02.23 | l-fy | tzanger > long time |
| 17:03.11 | tzanger | indeed |
| 17:03.38 | tzanger | hmmhesays: try exten => 500,OBVIOUS_GLARE :-) |
| 17:03.47 | hmmhesays | hahaha |
| 17:04.05 | hmmhesays | I have exten => 250,HINT,SIP/250 |
| 17:04.09 | intensedrive | any ideas juggie |
| 17:04.17 | Juggie | sorry on the phone. |
| 17:04.22 | intensedrive | k |
| 17:04.24 | tzanger | try without the HINT... all lowercase instead |
| 17:04.29 | hmmhesays | oops |
| 17:04.32 | hmmhesays | it is lowercase |
| 17:04.42 | tzanger | so much for that idea |
| 17:04.47 | hmmhesays | and I still get 404 not found on the subscribe request |
| 17:04.47 | tzanger | exten was reloaded? |
| 17:04.57 | tzanger | phone's context= is that context? |
| 17:05.26 | tzanger | sip.conf was reloaded? |
| 17:05.29 | hmmhesays | whoa |
| 17:05.37 | hmmhesays | whoops, I see something funneh |
| 17:05.45 | *** join/#asterisk SwK (n=Silik0nJ@70.232.183.130) |
| 17:06.19 | CunningPike | vader--: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain. GITF |
| 17:06.21 | file | SwK: airport? |
| 17:06.27 | CunningPike | s/GITF/GIYF/ |
| 17:06.45 | hmmhesays | Found user '250' |
| 17:06.46 | hmmhesays | Looking for s in subscriptions (domain x.x.x.x) |
| 17:06.46 | hmmhesays | Transmitting (NAT) to x.x.x.x:5060: |
| 17:06.46 | hmmhesays | SIP/2.0 404 Not Found |
| 17:07.03 | hmmhesays | [subscriptions] |
| 17:07.03 | hmmhesays | exten => s,hint,SIP/250 |
| 17:07.09 | cbrake | I'm having difficulting getting an asterisk box to connect to several IAX providers NuFone and FWD. I have the asterisk system behind a Linksys firewall. Is there anything I need to worry about with this setup or should it just work. |
| 17:07.27 | l-fy | cbrake > not with iax |
| 17:07.42 | l-fy | except if you have some weird firewall but by default it should work |
| 17:08.01 | cbrake | l-fy: ok, don't think I'm doing anything non-standard w/ the fw |
| 17:08.13 | l-fy | cbrake > i use fwd with sip |
| 17:08.15 | l-fy | so.... |
| 17:08.27 | cbrake | l-fy: what are some reasons registration fails and I get "no authority found" when trying to dial? |
| 17:08.51 | l-fy | i have no idea |
| 17:09.34 | tzanger | hmmhesays: ok? |
| 17:09.38 | cbrake | hmmm, this should not be this difficult ... time to read some more ... |
| 17:10.50 | hmmhesays | tzanger: that SHOULD be working |
| 17:10.53 | hmmhesays | I dunno wtf is up |
| 17:11.05 | tzanger | oh |
| 17:11.11 | tzanger | try exten => 250,hint,SIP/250 |
| 17:12.33 | *** join/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 17:15.02 | hmmhesays | it is not looking for 250 though |
| 17:15.07 | hmmhesays | it is looking for s |
| 17:15.13 | *** join/#asterisk ThePlot (i=ThePlot@202.164.38.211) |
| 17:15.33 | file | hmmhesays: grab me a sip debug, and what version of Asterisk? if from SVN what revision |
| 17:15.47 | CunningPike | hmmhesays: What type of phone? |
| 17:16.10 | queuetue | I am so happy to discover this was an actual bug.... I've been chasing my tail for 48 hours. :) file, and even happier you were here to fix it for me. |
| 17:16.22 | ThePlot | Hi, I am having a weird problem loading zaptel.. modprobe zaptel results in FATAL: Error inserting zaptel (/lib/modules/2.6.9-34.ELsmp/extra/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg).. dmesg shows.. zaptel: Unknown symbol copy_from_user; RTC is enabled so is CONFIG_CRC_CCITT |
| 17:16.24 | hmmhesays | hey file |
| 17:16.32 | file | queuetue: it's more complex then you think... I will be back though |
| 17:16.32 | ThePlot | am in desperate need for help, been pulling my hair out since last 5 hours |
| 17:17.05 | Juggie | theplot, type uname -a |
| 17:17.06 | file | hmmhesays: hola |
| 17:17.06 | Juggie | paste. |
| 17:17.29 | ThePlot | Linux voip.test.net 2.6.9-34.ELsmp #1 SMP Wed Mar 8 00:27:03 CST 2006 i686 athlon i386 GNU/Linux |
| 17:17.45 | ThePlot | its a Dual Core Opteron 265x2 machine with 4GB ram |
| 17:17.53 | Juggie | centos or rhel? |
| 17:17.56 | ThePlot | centos |
| 17:18.20 | Juggie | ~centosbug |
| 17:18.26 | jbot | methinks centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 17:18.27 | ThePlot | tried kernel recompile, everything in the wiki's but am so frustated now |
| 17:18.40 | ThePlot | I think I did that already |
| 17:18.41 | Juggie | you dont need to kernel recompile, you should use stock. |
| 17:18.43 | ThePlot | it was in one of the wikis |
| 17:18.50 | ThePlot | let me try again |
| 17:18.55 | ThePlot | should I recompile zaptel after that? |
| 17:19.03 | Juggie | my concern now is that your kernel source is mismatched frmo your compiled kernel |
| 17:19.04 | Juggie | in config. |
| 17:19.11 | *** join/#asterisk anti-uce (n=anti-uce@210.213.198.89) |
| 17:19.33 | hmmhesays | http://pastebin.ca/115505 |
| 17:19.43 | ThePlot | hrmm |
| 17:19.49 | hmmhesays | 1.2.7.1 |
| 17:19.55 | hmmhesays | patched with bristuff |
| 17:20.01 | file | ah, better |
| 17:20.09 | Juggie | ThePlot, you could remove kernel-version-devel and re-add it using yum |
| 17:20.12 | file | it IS looking for 250 fyi |
| 17:20.12 | Juggie | or just force install over top. |
| 17:20.19 | firejon | has anyone here had any audio quality problems with the Grandstream GXP-2000? |
| 17:20.56 | hmmhesays | ok |
| 17:21.04 | ThePlot | [root@voip src]# sed -i s/rw_lock/rwlock/ /usr/src/linux/include/linux/spinlock.h |
| 17:21.18 | hmmhesays | ok, now it is looking for 250 |
| 17:21.21 | ThePlot | Juggie: will do that, let me a recompile of zaptel |
| 17:21.33 | hmmhesays | so i have exten => 250,hint,SIP/250 in the correct context |
| 17:21.35 | hmmhesays | and get nuffin |
| 17:21.36 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 17:21.39 | TripleFFFF | : whats the standart on phone numbers.. is it .. 011blah 11blah or blah ? |
| 17:21.48 | file | and you did a reload naturally |
| 17:21.51 | *** join/#asterisk watchy (n=gweg@office2.gwhsi.com) |
| 17:21.55 | watchy | hey mog you there? |
| 17:21.56 | Juggie | ThePlot, make sure when you do, you make the centos bug mod again. |
| 17:21.57 | *** join/#asterisk vlrk (n=vlrk@202.65.134.119) |
| 17:22.01 | hmmhesays | hehe yes |
| 17:22.07 | ThePlot | ok |
| 17:22.12 | TripleFFFF | got like no friggin ISO on anything.. all providers have differnet formats its driving me NUTS ! |
| 17:22.16 | file | hmmhesays: what about show hints |
| 17:22.16 | mog | yup |
| 17:22.23 | TripleFFFF | ~centosbug |
| 17:22.25 | jbot | from memory, centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 17:22.25 | TripleFFFF | btw |
| 17:22.31 | TripleFFFF | REALLY needed to make zaptel |
| 17:22.47 | hmmhesays | CLI> show dialplan subscriptions |
| 17:22.47 | hmmhesays | [ Context 'subscriptions' created by 'pbx_config' ] |
| 17:22.47 | hmmhesays | <PROTECTED> |
| 17:22.47 | hmmhesays | <PROTECTED> |
| 17:23.10 | hmmhesays | 1 dialplan hint registered |
| 17:23.12 | watchy | mog: we moved some phone lines around on the tdm400ps, could that cauase our echo to come back? |
| 17:23.32 | mog | if the lines have different impediances |
| 17:23.40 | file | okay so that's good, and yet it's still coming up with the 404 Not Found? |
| 17:23.40 | mog | which is not unlikely |
| 17:23.45 | mog | you might just run fxotune again |
| 17:23.56 | hmmhesays | yep |
| 17:24.03 | watchy | mog: the guy at digium told me all he ran last time was fxotune, so i ran it and they seem to still have a little echo left but lasttime they didnt ever hear any |
| 17:24.50 | hmmhesays | file: if I add a noop to the dialplan like exten => 250,1,NoOp(blargh); exten => 250,hint,SIP/250; then it comes back with 489 bad request |
| 17:26.07 | ThePlot | Juggie: FATAL: Error inserting zaptel (/lib/modules/2.6.9-34.ELsmp/extra/zaptel.ko): Invalid module format ..... zaptel: disagrees about version of symbol struct_module |
| 17:26.21 | ThePlot | Juggie: any idea? :/ |
| 17:26.32 | ThePlot | this is after the centos bug fix |
| 17:26.40 | Juggie | rpm -qa|grep kernel |
| 17:26.44 | queuetue | If I set a timeout(response) of 5 and then play a background that lasts 10 seconds, the timeout will still kick in at 5, right? |
| 17:26.47 | watchy | mog: does the system take a bit to fix echo? |
| 17:26.47 | Juggie | paste it to www.pastebin.ca |
| 17:26.53 | Juggie | and tell me the link |
| 17:27.09 | ThePlot | doing |
| 17:27.36 | ThePlot | Juggie: http://pastebin.ca/115521 |
| 17:27.54 | hmmhesays | this is craziness |
| 17:27.55 | mog | nope |
| 17:27.56 | hmmhesays | dag nabit |
| 17:28.13 | Juggie | hmmmmmm |
| 17:28.21 | Juggie | theplot, go into zaptel, make clean; make; |
| 17:28.25 | Juggie | and paste the output to pastebin |
| 17:28.28 | ThePlot | ok |
| 17:28.39 | watchy | mog: imma look at it a bit more, if it dont go away you mind having that guy look at my box again? |
| 17:28.54 | mog | yeah no problem |
| 17:30.07 | ThePlot | Juggie: http://pastebin.ca/115527 |
| 17:30.37 | Juggie | ohh kay. |
| 17:30.44 | Juggie | per your rpm -qa|grep kernel |
| 17:30.50 | Juggie | kernel-devel isnt installed |
| 17:30.53 | Juggie | however |
| 17:31.03 | Juggie | it is obviously there as zaptel is compiling against it. |
| 17:31.12 | ThePlot | I built it out from the .bz2 file in the linux directory of redhat/sources |
| 17:31.18 | Juggie | ahhhhh |
| 17:31.20 | Juggie | bad bad bad |
| 17:31.25 | ThePlot | heh really? :/ |
| 17:31.35 | Juggie | yeah, i'm sure its a dif config, who knows. |
| 17:31.39 | Juggie | simplify your life. |
| 17:31.45 | Juggie | 'yum install kernel-devel' |
| 17:31.50 | ThePlot | so basically I nuke the kernel stuff I have right now and use yum to do everything? |
| 17:32.00 | ThePlot | ok *salute*, will let you know result in couple of minutes |
| 17:32.02 | Juggie | yes, it makes things much nicer trust me ;) |
| 17:32.14 | Juggie | the kernel config from the source |
| 17:32.19 | Juggie | doesnt match the actual compiled binary |
| 17:32.24 | Juggie | thats most likey the problem |
| 17:32.25 | Ryushin | Okay, so I guess the two main phones that everyone likes is the Linksys and the Polycom. But I think I'm going with the Polycom because it has the attendent phones and such. So, can I assume that IP601 and IP501 phones will work fine with Asterisk? |
| 17:32.39 | ThePlot | I was under the impression it did, lesson #1.. stop assuming stuff :) |
| 17:32.53 | Juggie | well, i could be proven wrong |
| 17:32.57 | Juggie | but that would be my first guess. |
| 17:33.00 | watchy | Ryushin: yeap thats what i use |
| 17:33.14 | Juggie | there was a time when i did everything from source too |
| 17:33.23 | Juggie | now i simplify my life and use yum whenever i can. |
| 17:33.29 | *** join/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 17:33.33 | hmmhesays | i hate yum |
| 17:33.37 | ThePlot | you have no idea how frustating the last 4 hrs have been for me heh |
| 17:33.39 | hmmhesays | apt puts the smack down on yum |
| 17:33.54 | Ryushin | I assume a single core AMD64 system with a gig of ram will be powerful enough to run Asterisk for at max 40 people? |
| 17:33.54 | watchy | i like my gentoo juarez |
| 17:33.59 | hmmhesays | Ryushin: if you know what the hell you are doing |
| 17:34.06 | CunningPike | Ryushin: There are some phone features that don't work yet - ACD login/logout, SLA/BLA are examples - but there are workarounds for most of them |
| 17:34.07 | matteof | hi folks, how to configure asterisk to get sip info from mysql? |
| 17:34.12 | Juggie | the eight ball says: yes |
| 17:34.19 | Juggie | matteof, RTFM, asterisk realtime |
| 17:34.29 | Ryushin | hmmhsays: Well, I don't know what the hell I'm doing. I've never implemented this. |
| 17:34.30 | matteof | ok |
| 17:34.47 | hmmhesays | Ryushin: then prepare for a hard time |
| 17:34.53 | hmmhesays | unless you got the dinero to pay someone to help you |
| 17:35.12 | CunningPike | Ryushin: Lots of people have, so stick around in here and you;ll get lots of help |
| 17:35.29 | hmmhesays | over a long period of time |
| 17:35.46 | hmmhesays | be prepared for lots of off topic discussions |
| 17:35.50 | Juggie | no one wants to help anyone who doesnt try |
| 17:35.53 | Ryushin | Well, the thing is, I'm very good at implementing new technologies. I am a network engineer, and I've been using Linux since '94. |
| 17:35.53 | Juggie | thats the main thing. |
| 17:35.56 | hmmhesays | about boobies and such |
| 17:36.01 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 17:36.06 | Ryushin | I also only have 30 days to get this thing done. Or at least, working. |
| 17:36.09 | Juggie | Ryushin, then you are in a very good position. |
| 17:36.12 | Juggie | www.voip-info.org |
| 17:36.13 | Nivex | boobies? |
| 17:36.13 | Nivex | :) |
| 17:36.17 | Juggie | and start he learning process :) |
| 17:36.24 | hmmhesays | 94 eh? so the sun and vaginas are near mythical to you |
| 17:36.33 | CunningPike | Ryushin: There is a good book, too |
| 17:36.37 | CunningPike | ~thebook |
| 17:36.38 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:37.10 | *** part/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 17:37.29 | Juggie | i'm eagerly awaiting theplot's results :) |
| 17:37.31 | Ryushin | hmmhesays: Yea, well, I'm married and motorcyclist. So I do get expose to both. :) |
| 17:37.53 | hmmhesays | I'm just giving you a little n00b shiat |
| 17:38.01 | jac]Z[oby | re |
| 17:38.01 | Juggie | do you have a penguin on your ride? :) |
| 17:38.35 | CunningPike | Ryushin: Can't be married that long then ;) |
| 17:38.38 | Ryushin | Well, I am a n00b at this. So it's time to start ordering hardware then. I have a whole network to build as well. |
| 17:38.56 | hmmhesays | send me a couple spa-942's while you're at it |
| 17:38.58 | Ryushin | CunningPike: 10 years. |
| 17:39.10 | Ryushin | Juggie: Nah, Honda Wings for the cbr929rr. |
| 17:39.10 | Juggie | Ryushin, whats the project goal? |
| 17:39.18 | CunningPike | Ryushin: Hey, me too |
| 17:39.41 | Ryushin | To build a a new network for a 30 person office, and a new phone system for said office. |
| 17:39.47 | Juggie | ahhh |
| 17:39.53 | Juggie | thats small beans |
| 17:39.57 | Juggie | thats like 1-2 switches and an * box. |
| 17:40.15 | hmmhesays | purdy much |
| 17:40.23 | hmmhesays | get out your dual proc pIII |
| 17:40.29 | hmmhesays | $100 bucks on ebay |
| 17:40.32 | *** join/#asterisk SwK_ (n=Silik0nJ@70.232.183.130) |
| 17:40.40 | Juggie | 30 person office your looking @ 10ppl max on the phone at once probally. |
| 17:40.48 | Ryushin | Well, that's what I'm hoping. But it will be a good way to get my feet wet with Asterisk. |
| 17:40.55 | Juggie | your obviously going to do ulaw straight through |
| 17:40.56 | Nugget | unless you exclusively hire 16 year old girls. |
| 17:40.59 | Juggie | so no cpu load. |
| 17:41.08 | hmmhesays | lets say 18 for jail time sake |
| 17:41.48 | Juggie | Ryushin, all sip phones? |
| 17:41.50 | Ryushin | Okay, now I have to find out how CBeyond hands off the circuit so I know what card I need to buy from digicom. |
| 17:41.56 | Ryushin | Juggie: That's the plan. |
| 17:42.03 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 17:42.07 | Juggie | Ryushin, how are you getting PSTN termination? |
| 17:42.21 | Juggie | analog/t1? |
| 17:42.32 | Egonis | If I use Wait,# in my dialplan, it doesn't allow extensions to be dialed at that time -- am I using the wrong command? |
| 17:42.33 | Ryushin | Juggie: That's what I have to find out from CBeyond. |
| 17:42.52 | Rowter | how could I get a variable set before queue() ? ImportVar(newvar=channelname|variable) |
| 17:42.52 | Rowter | ? |
| 17:42.55 | Juggie | Ryushin, cbeyond is who, local telco? |
| 17:42.58 | wunderkin | Egonis, show applications.... use waitexten |
| 17:43.06 | Juggie | ImportVar? wtf? |
| 17:43.14 | Ryushin | I was planning on using Debian for this. Since I use it for pretty much everything else as well. |
| 17:43.41 | Juggie | thats cool. |
| 17:43.44 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 17:43.54 | Juggie | cbeyond is a local telco? |
| 17:43.57 | wunderkin | Rowter, importvar is to get a variable from another channel, just use ${var} if it is already there |
| 17:44.06 | Ryushin | CBeyond is a VoIP company that provides a data T1, and hands off voice pieces from their cisco router. |
| 17:44.23 | Juggie | via sip? |
| 17:44.39 | Rowter | wunderkin, when you enter queue it creates a nother channel, and I can't seem to get it |
| 17:44.45 | Ryushin | Juggie: No, via POTS or hopefully via a T1. |
| 17:44.53 | Ryushin | I have to go look. |
| 17:45.11 | Juggie | Ryushin, if they are doing voice over a data t1 then whats the protocol they are using on the data. |
| 17:45.28 | Juggie | sip/mgcp? |
| 17:45.37 | wunderkin | Rowter, no it doesnt, use show channel and dumpchan to see variables |
| 17:46.01 | hmmhesays | damnit |
| 17:46.04 | Juggie | Ryushin, idealy you want a voice t1. |
| 17:46.32 | Juggie | putting voice ontop of data offers no benefit unless you expect to use part of the t1 to provide internet for the office. |
| 17:46.55 | Juggie | it also certainly cant be cheaper. |
| 17:47.10 | Juggie | the only benefit would be more channels |
| 17:47.26 | Juggie | but, you only have 30ppl i'm pretty sure 23 channels is plenty :) |
| 17:48.16 | Juggie | also, they must be using voice compression of some sort, which would be nice to avoid. |
| 17:48.31 | Ryushin | Juggie: Well, the way it works, is CBeyond gets the T1 installed and terminates it to their Cisco router. They then hand off a Ethernet for Internet access, and either POTS or voice T1 circuit. |
| 17:49.10 | Juggie | are you really going to use them for internet |
| 17:49.21 | Juggie | i mean i could see if it was an E10 w/ voice&data |
| 17:49.34 | Juggie | but T1 isnt much of an internet pipe these days, espically among 30 users |
| 17:49.40 | Juggie | and espicalyl when it has to provide voice. |
| 17:49.46 | Rowter | wunderkin, after entering queue I do a Local/ so its a diferent chan.. mmh |
| 17:52.27 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 17:52.59 | Ryushin | Juggie: Yea, it is gong to do Internet as well. 1.54 Mb isn't all that great. Only 48K is used from the 1.54 Mb for each person using the phone, and when they hang up, that bandwidth goes back for data. |
| 17:53.33 | Ryushin | It would be nice if there was a cheap DS3 circuits to upgrade to from T1's, but they don't exist. |
| 17:53.39 | Juggie | Ryushin, if they provide a voicet1 its not 48k |
| 17:53.47 | Ryushin | I'll have to see what kind of DSL offerings there are. |
| 17:53.47 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 17:53.47 | Juggie | its 64k |
| 17:53.49 | Juggie | unless its compressed |
| 17:53.57 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 17:53.57 | Juggie | Ryushin, E10 |
| 17:54.12 | Ryushin | Juggie: It's compressed, and it's their own internal stuff. |
| 17:54.27 | Juggie | sounds expensive and dumb :) |
| 17:54.55 | Juggie | you would be 10x better off with a voicet1 and a 50$/mo dsl lnie. |
| 17:54.57 | Juggie | *line. |
| 17:55.03 | Ryushin | It doesn't use pieces of the T1. The T1 just has data on it. The cisco router handles the data versus voice. |
| 17:55.39 | Juggie | if they provide you a t1 w/ voice at the other end |
| 17:55.39 | Ryushin | Well, they also have a couple of servers that need to send speed up. 864K isn't enough. I have to see if they can get 1.5 MB up on business class DSL. |
| 17:56.23 | Juggie | then its either a partial t1 w/ part voice & part data or they are using the cisco router to trunk over ip and then provide a t1 on the other side. |
| 17:56.27 | Ryushin | That's were I think I'm going to move them to. Get a true voice T1 and business class DSL. |
| 17:56.30 | Juggie | which sounds expensive & un-necessairy. |
| 17:56.52 | Juggie | Ryushin, i dunno where you live but business class dsl is 10mbit/1mbit here. |
| 17:57.19 | Ryushin | That would be nice. I'll have to see what Qwest can provide to their building. |
| 17:57.27 | Ryushin | Have to go check on some things. |
| 17:59.31 | Ryushin | Now I just have to find out the costs associated with the T1 and business class DSL. |
| 17:59.56 | Juggie | a voicet1 costs me 600$/mo |
| 17:59.59 | Juggie | cdn |
| 18:00.17 | *** join/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 18:01.43 | System010 | Hey. I've configured our phone system to talk with asterisk over a T card, works great. However, i would like to transfer calls back to the phone system without tying up 2 channels. Is this possible? |
| 18:02.36 | charles___ | System010: no dude |
| 18:02.47 | System010 | didn't think so. |
| 18:02.50 | System010 | thanks for the info |
| 18:02.54 | charles___ | System010: if you want to terminate the call yo gotta create another channel and bridge |
| 18:03.46 | hmmhesays | bah |
| 18:03.49 | hmmhesays | what the crap is going on here |
| 18:04.19 | System010 | charles___: so i have to use 2 channels no matter what |
| 18:04.36 | charles___ | System010: on a T yes |
| 18:04.43 | hmmhesays | can someone pastebin me a successful subscribe to a hint extension? |
| 18:04.46 | hmmhesays | please |
| 18:04.51 | charles___ | System010: or you can play a message to the user to call another number :P |
| 18:04.59 | System010 | ah, how nice. more T cards! |
| 18:05.01 | charles___ | hmmhesays: hint ? |
| 18:06.13 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 18:07.01 | *** join/#asterisk test34 (n=test34@unaffiliated/test34) |
| 18:10.04 | *** join/#asterisk jeanmi_ (n=besnard@mic92-2-82-67-216-218.fbx.proxad.net) |
| 18:10.10 | hmmhesays | ok did hints get broken somewhere along the way? |
| 18:10.38 | jeanmi_ | I have added "exten => s,n,Set(LANGUAGE()=fr)" to the [demo] after installing asterisk |
| 18:10.53 | jeanmi_ | and even though I have installed the french prompts it keeps using the english ones |
| 18:11.11 | dlynes_laptop | hmmhesays: I don't think it was ever fully working |
| 18:11.23 | dlynes_laptop | hmmhesays: it was just a quick hack, afaik |
| 18:11.28 | hmmhesays | I had it working fine back in the * 1.0 days |
| 18:11.56 | intensedrive | hi Juggie |
| 18:12.01 | dlynes_laptop | hmmhesays: yeah...every time I've tried it, it crashes my aastra phones, big time |
| 18:12.09 | dlynes_laptop | hmmhesays: i gave up on trying to use it |
| 18:12.33 | file | did hints even exist in 1.0? |
| 18:12.38 | dlynes_laptop | file: yes |
| 18:12.42 | file | scary |
| 18:12.55 | dlynes_laptop | file: I just don't think it was documented then |
| 18:13.30 | hmmhesays | yeah |
| 18:13.33 | hmmhesays | now I just get 404 not found |
| 18:13.53 | dlynes_laptop | hmmhesays: I get spammed with 489's and 500's from the phones |
| 18:13.56 | jeanmi_ | and in the CLI I see : |
| 18:13.57 | jeanmi_ | <PROTECTED> |
| 18:13.57 | jeanmi_ | <PROTECTED> |
| 18:14.00 | dlynes_laptop | hmmhesays: asterisk doesn't seem to have a problem |
| 18:14.07 | jeanmi_ | but still it is in english and not french ..... |
| 18:14.19 | hmmhesays | if you have another extension in that context that matches the hint priority extension you will get a 489 |
| 18:14.32 | hmmhesays | if you have your hints in a context alone you will get a 404 |
| 18:15.39 | dlynes_laptop | hmmhesays: you mean if I have exten => 101,1,... and exten => 101,hint,... in the same context, i'll get a 489, but just exten => 101,hint,... will give me a 404? |
| 18:15.53 | hmmhesays | yes |
| 18:15.56 | dlynes_laptop | ah |
| 18:16.04 | dlynes_laptop | yeah...never tried the hints in their own context |
| 18:16.23 | *** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net) |
| 18:16.34 | dlynes_laptop | didn't anthony minnessale write that code? |
| 18:17.02 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 18:17.12 | kavit | has anyone managed to use asterisk with Call Manager Express and Unity over sip and managed to hairpin SIP to SIP calls? |
| 18:17.13 | hmmhesays | a clue I have not |
| 18:17.23 | kavit | it is giving me too much grief :( |
| 18:17.36 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 18:17.50 | dlynes_laptop | Unity is some kinda Lucent garbage, is it not? |
| 18:18.06 | kavit | nah Unity is cisco's voice mail platform |
| 18:18.10 | dlynes_laptop | ah |
| 18:18.15 | dlynes_laptop | still garbage then :) |
| 18:18.19 | kavit | ocourse |
| 18:18.26 | *** join/#asterisk linagee (n=linagee@cpe-66-75-136-115.san.res.rr.com) |
| 18:18.28 | kavit | expensive garbage |
| 18:18.33 | CunningPike | But it's the best garbage money can buy |
| 18:18.34 | linagee | voip over mesh networking... heheh |
| 18:18.41 | linagee | free phone service |
| 18:19.08 | dlynes_laptop | linagee: you're on google's free internet? |
| 18:19.23 | kavit | well if anyone can give me a few pointers, i'd be glad |
| 18:19.43 | jeanmi_ | how can I have the CLI otr the logs to show what prompt files are being played ? |
| 18:19.51 | dlynes_laptop | kavit: dump cisco, and go polycom? :P |
| 18:20.21 | kavit | Cisco like to use 3XX sip messages, especially 302 Moved Temporarily |
| 18:20.30 | kavit | asterisk doesn't like that |
| 18:20.39 | Nugget | How could you tell I was on google's free internet? [ _Shell_Servers_Cheap_ : Enjoy IRC from our shell accounts! Ads by Gooooooogle ] |
| 18:20.45 | linagee | dlynes_laptop: i am just saying that it would be cool. google has free internet?? |
| 18:20.46 | dlynes_laptop | kavit: have you tried using sccp? |
| 18:20.54 | kavit | I did promiscredir=yes but it always gets caught in a loop |
| 18:21.08 | dlynes_laptop | linagee: yeah...somewhere in the san francisco area, i thought? |
| 18:21.24 | dlynes_laptop | kavit: i.e. chan_skinny? |
| 18:21.27 | kavit | dlynes_laptop: the asterisk box belongs to a telco |
| 18:21.27 | linagee | dlynes_laptop: grr. i am in sd |
| 18:21.32 | kavit | i cant touch it |
| 18:21.40 | kavit | they provide sip only |
| 18:21.42 | dlynes_laptop | oh |
| 18:22.09 | dlynes_laptop | kavit: so shouldn't you be on the #cisco$$$$$$$ channel, instead then? |
| 18:22.25 | kavit | dlynes_laptop: I am |
| 18:22.31 | kavit | it is dead |
| 18:22.32 | dlynes_laptop | kavit: seeing as how you can't do anything with the asterisk box? |
| 18:22.37 | kavit | not much of a community |
| 18:22.53 | dlynes_laptop | kavit: dunno...I was on there a while back |
| 18:23.10 | kavit | I can get them to add extensions et al accordingly to deal with SIP 302 |
| 18:23.12 | dlynes_laptop | kavit: and I got an answer right away to a problem I was having getting bay networks to talk to cisco gear |
| 18:23.30 | kavit | asterisk just doesnt like 302 |
| 18:23.44 | dlynes_laptop | kavit: are you proficient in C? |
| 18:24.03 | dlynes_laptop | kavit: oh yeah...nvm...you don't have access to the asterisk box |
| 18:24.12 | *** join/#asterisk ulkeshkosh (n=ulkeshko@71-12-171-82.dhcp.gsvl.ga.charter.com) |
| 18:24.14 | kavit | 482 "Loop Detected" |
| 18:24.31 | kavit | dlynes_laptop: I was about 6-8 years ago |
| 18:24.39 | dlynes_laptop | heh |
| 18:24.54 | dlynes_laptop | C's one of those things you never forget, though |
| 18:24.59 | kavit | i have let myself slip |
| 18:25.03 | dlynes_laptop | Java on the other hand |
| 18:25.08 | dlynes_laptop | I'd love to forget |
| 18:25.11 | kavit | heh |
| 18:25.19 | kavit | i don't mind netbeans |
| 18:25.23 | dlynes_laptop | Oh Lord how I hate that language |
| 18:25.27 | dlynes_laptop | Yeah...I love netbeans |
| 18:25.29 | dlynes_laptop | I just hate Java |
| 18:25.49 | dlynes_laptop | netbeans is an excellent ide |
| 18:25.50 | kavit | i am actually writing an interface for using asterisk with Linux-HA in java |
| 18:26.01 | dlynes_laptop | cool |
| 18:26.19 | dlynes_laptop | There's one cool thing about Java though |
| 18:26.22 | dlynes_laptop | JBoss :) |
| 18:26.34 | dlynes_laptop | It's got a cool architecture |
| 18:26.38 | kavit | cisco charge an arm and a leg for fail over and redundancy |
| 18:26.43 | kavit | yeah |
| 18:27.15 | dlynes_laptop | but all the rest of the crud is so bloody annoying...especially struts and having to memorize 50 million acronyms to be able to get any work done |
| 18:27.30 | kavit | and their solution uses similiar technology.... heartbeats |
| 18:27.33 | kavit | yeeah |
| 18:27.38 | wunderkin | adelphia and comcat are part of time warner now.. |
| 18:27.46 | kavit | and I hate gui programming |
| 18:27.49 | dlynes_laptop | any HA solution will probably employ heartbeats of some description |
| 18:28.04 | dlynes_laptop | I'm having to deal with heartbeats myself with a foreign exchange trading bot that I'm writing |
| 18:28.14 | hmmhesays | bah |
| 18:28.25 | kavit | well have a look at linux-ha v2 |
| 18:28.39 | dlynes_laptop | I don't have the time |
| 18:28.48 | CunningPike | kavit: Polycoms use 302 as well - and we haven't had any problems with it? |
| 18:28.49 | dlynes_laptop | Hell, I don't have the time to work on what I need to work on :( |
| 18:29.05 | dlynes_laptop | CunningPike: yeah..the sysadmin at his telco is probably just incompetent |
| 18:29.12 | CunningPike | dlynes_laptop: You spend too much time in here ;) |
| 18:29.24 | *** part/#asterisk System010 (n=jgargano@hide247.cybergnostic.com) |
| 18:29.35 | kavit | CunningPike: does your trunk hairpin? as in the same gateway for |
| 18:29.38 | dlynes_laptop | CunningPike: nah...how much have you seen me in here in the last couple of weeks? |
| 18:29.40 | CunningPike | dlynes_laptop: Telco sysadmins incompetent? Surely you are mistaken :/ |
| 18:29.51 | kavit | the incoming and outgoing call |
| 18:29.52 | dlynes_laptop | CunningPike: heh...we just inked a deal with Telus today |
| 18:30.00 | kavit | thats what messes it up with a loop |
| 18:30.04 | CunningPike | dlynes_laptop: Actually, not as much as usual, now that you mention it |
| 18:30.10 | dlynes_laptop | CunningPike: we're going to start reselling their pri's, t1's, analog lines, and bri's |
| 18:30.21 | CunningPike | dlynes_laptop: Really?? Very interesting |
| 18:30.25 | dlynes_laptop | CunningPike: yeah |
| 18:30.36 | CunningPike | dlynes_laptop: Congrats |
| 18:30.40 | dlynes_laptop | CunningPike: so i'm hoping to get into channellized t1's |
| 18:30.57 | dlynes_laptop | CunningPike: so i can start selling internet and phones all on one t1 to the customer |
| 18:31.03 | CunningPike | dlynes_laptop: Will you be offering fractional T1s? |
| 18:31.21 | dlynes_laptop | and then we'll kick telus' ass |
| 18:31.41 | dlynes_laptop | CunningPike: i'd like to, yeah |
| 18:31.49 | kavit | CunningPike: do you actually use Transfer() or multiple entries in sip.conf or extensions.conf dial plan logic? I will have to email the sysadmin with some sort of a solution |
| 18:31.55 | dlynes_laptop | CunningPike: and then upgrade the customer to a full t1 when they get big enough |
| 18:32.10 | CunningPike | dlynes_laptop: Cool - the market needs more competition |
| 18:32.22 | CunningPike | dlynes_laptop: You'll be competing with Allstream too |
| 18:32.28 | dlynes_laptop | CunningPike: now that we're in their backyard, we've got access to some people that actually know what the hell they're doing |
| 18:32.35 | CunningPike | kavit: For what? |
| 18:32.53 | kavit | CunningPike: dealing with SIP 302 messages |
| 18:32.53 | *** join/#asterisk ZPertee (n=zach@cpe-65-25-51-117.neo.res.rr.com) |
| 18:33.21 | CunningPike | dlynes_laptop: Yes, they do exist - there are some really good people at Telus, but they like to keep them tucked away. It lends them an air of competence that the rest of the organization can't live up to |
| 18:33.44 | ZPertee | anyone ever heard of no authority found cause 3 error |
| 18:33.59 | CunningPike | kavit: We don't really deal with them - it just happens. If someone forwards their phone, it sends a 302, and asterisk just redirects the call |
| 18:34.01 | bcnl | < dlynes_laptop> CunningPike: so i can start selling internet and phones all on one t1 to the customer |
| 18:34.07 | dlynes_laptop | CunningPike: you know that normally when you get call forwarding put on your line, you can have one call going through simultaneously on the call-forward, right? |
| 18:34.10 | linlin | is there any way to get more information as to why asterisk is refusing an iax2 softphone connection? |
| 18:34.17 | bcnl | /msg me when you can do a T1 in Surrey with 6 voice channels and some data |
| 18:34.28 | dlynes_laptop | bcnl: ok |
| 18:34.34 | CunningPike | dlynes_laptop: That is a very good question - I don't know |
| 18:34.55 | dlynes_laptop | bcnl: once we get our billing system finished, that's where we'll be concentrating our business development efforts |
| 18:34.57 | intensedrive | anyone use app_conf |
| 18:35.09 | CunningPike | dlynes_laptop: You mean on a POTS line |
| 18:36.10 | kavit | CunningPike: as it should be... i don't know who to be pissed off at, Cisco for not allowing sip to sip hairpin connections natively on call manager express or the telco for being totally lame |
| 18:37.25 | dlynes_laptop | CunningPike: yes |
| 18:37.33 | CunningPike | kavit: How does Cisco enter the picture (forgive me if you've explain this already) |
| 18:37.43 | CunningPike | dlynes_laptop: Ya - no idea |
| 18:38.03 | CunningPike | dlynes_laptop: I think it would depend on the telco and how they route the call |
| 18:39.28 | dlynes_laptop | CunningPike: well, anwyays...everyone i talked to at telus said it couldn't be done |
| 18:39.44 | dlynes_laptop | CunningPike: then I talk to these guys today, and find out it can be done, and quite cheap, too |
| 18:40.09 | CunningPike | dlynes_laptop: Yup - most people in there haven't a clue |
| 18:40.29 | dlynes_laptop | yeah, no kidding |
| 18:40.41 | dlynes_laptop | but at least Telus is a hell of a lot more responsive than group telecom |
| 18:40.57 | dlynes_laptop | group telecom seems to have this we don't care if we get your business or not type of attitude |
| 18:41.11 | kavit | CunningPike: With incoming calls through the SIP trunk cisco doesn't bridge call fowards to voicemail or any other number natively if the outgoing trunk is SIP or the voicemail link is SIP |
| 18:41.40 | CunningPike | dlynes_laptop: Telus are going that way too. I think they'll end up 100% wholesale |
| 18:42.38 | CunningPike | kavit: Well, that bites. Are you migrating to asterisk from Cisco? |
| 18:42.42 | kavit | so instead of just bridging the call it uses SIP 3XX messages to re-establish the call |
| 18:43.16 | kavit | CunningPike: no a customer purchased cisco 3485 and spent 70K on cisco gear |
| 18:43.38 | CunningPike | kavit: Ah - buy first and figure it out later :D |
| 18:43.57 | CunningPike | kavit: You should talk to Dr-Linux|work |
| 18:44.01 | CunningPike | :) |
| 18:44.20 | kavit | he was using E1 trunks but now he has moved to a pure VoIP solution but still wants to use the cisco voicemail stuff |
| 18:44.39 | kavit | and call manager express and what not.... |
| 18:44.43 | kavit | CunningPike: thanks |
| 18:45.29 | CunningPike | kavit: Sucks to be you ;) |
| 18:45.42 | dlynes_laptop | CunningPike: they seem to want our business big time...complete opposite of group telecom |
| 18:45.50 | dlynes_laptop | CunningPike: they're quite happy about it |
| 18:46.09 | CunningPike | dlynes_laptop: As resellers. As end customers, they couldn't care less about us |
| 18:46.21 | CunningPike | dlynes_laptop: All part of their 100% wholesale direction |
| 18:47.24 | kavit | CunningPike: Completely.... I also have to support the thing for 3 months atleast |
| 18:47.33 | kavit | the juice is not worth the squeeze |
| 18:47.34 | kavit | :( |
| 18:47.44 | dlynes_laptop | CunningPike: yeah...totally different attitude towards consumers |
| 18:48.02 | CunningPike | kavit: Take heart my friend - at least it's SIP....... |
| 18:48.37 | kavit | CunningPike: yeah and they don't have a nat |
| 18:49.21 | Juggie | ~centosbug |
| 18:49.23 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 18:56.56 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 18:58.37 | *** join/#asterisk edorsey (n=root@216-63-192-139.ded.swbell.net) |
| 19:00.36 | hohum | lol @ http://washingtondc.craigslist.org/nva/mis/190082533.html |
| 19:01.20 | *** join/#asterisk crich1999 (n=crich@port-212-202-210-134.dynamic.qsc.de) |
| 19:02.28 | *** join/#asterisk dan__t (n=dant@72.232.74.146) |
| 19:02.33 | dan__t | 'morning |
| 19:02.35 | dan__t | er, afternoon, rather. |
| 19:03.16 | dan__t | I had a question that could possibly be semi-OT, however, I am stuck using this system until I'm a bit more comfortable with Asterisk. I have an Intertel IPRC card, which I'd like to reset to factory defaults. |
| 19:03.42 | dan__t | My vendor, who is a moron, says Intertel needs to be contacted. I don't know that I believe that, but I don't know where else to ask. |
| 19:03.59 | trelane_ | dan__t, use the "factory reset tool" (ball-peen hammer) and beat the crap out of the card to reset it. |
| 19:04.13 | dan__t | Do you have a part number? |
| 19:04.29 | dan__t | har ar ar ar. |
| 19:04.30 | trelane_ | offhand no, but what you're looking for is a round cylinder on the top of the card |
| 19:04.35 | trelane_ | make sure you hit that |
| 19:04.48 | dan__t | It's not a button - simply apply force to jar it, or something? |
| 19:05.07 | trelane_ | it's a capaciter and it has enough charged capacitance gel to give you a nice second degree burn |
| 19:05.24 | dan__t | Oh, good call. |
| 19:05.42 | file | please try NOT to give the #asterisk inhabitants second degree burns |
| 19:05.43 | trelane_ | glad I could help, why not try calling Intertel, and tell them you need them to help you reset their card |
| 19:05.50 | dan__t | I thought you were serious for a minute, then I figured this is payback for asking a semi-OT question heh. |
| 19:06.04 | trelane_ | dan__t, to some extenst, I would recommend following your co-worker's advice |
| 19:06.20 | dan__t | Whihc is completely undertandable :) |
| 19:06.33 | trelane_ | file, what about devs? would you mind if I subject you to a bit of battery acid, or some concentrated bleach for the good chemical burn effect? |
| 19:06.56 | file | trelane_: I think I would, might impair my ability to code |
| 19:07.04 | trelane_ | file, fair enough point |
| 19:07.12 | trelane_ | perhaps another time then? |
| 19:07.21 | dan__t | I appreciate the help regardless, thank you. |
| 19:07.43 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 19:07.45 | Ryushin | So is a PRI and a Voice T1 the same thing? |
| 19:08.17 | file | trelane_: perhaps! |
| 19:08.54 | trelane_ | Ryushin, we could split hairs, but for all intents and purposes the CPE functions in a similar manner |
| 19:09.03 | *** join/#asterisk jlewis (n=jlewis@solo.atlantic.net) |
| 19:09.43 | jlewis | I think I just found a bug in 1.0.11.1...or a silly feature |
| 19:09.54 | dlynes_laptop | Ryushin: no. a voice T1 has 24 voice channels; a PRI has 23 b channels and 1 d channel; a voice T1 can have one DID per channel; a PRI can have multiple DIDs, and the telco may choose to send them down any arbritrary channel, and you can even have a DID sent down more than one channel, simultaneously |
| 19:09.59 | file | jlewis: in 1.0? wouldn't surprise me |
| 19:10.51 | Ryushin | Will the Digium Wildcard TE110P work for a PRI or a T1? |
| 19:11.03 | dlynes_laptop | Ryushin: a PRI in North America is also known as a T1 PRI |
| 19:11.04 | jlewis | from the CLI, if I show channel SIP... and show channel IAX2... for different legs of the same call, the SIP channel reports a non-zero elaped time...but the IAX channel always reports an elapsed time of 0h0m0s |
| 19:11.07 | file | Ryushin: yes. |
| 19:11.12 | dlynes_laptop | Ryushin: It'll work with both |
| 19:11.23 | Ryushin | Okay then that is what I need to buy. |
| 19:11.26 | file | jlewis: you do realize we're at 1.2.10 now and 1.4 is on the horizon? |
| 19:11.41 | dlynes_laptop | file: but 1.0 keeps life interesting :) |
| 19:11.57 | file | this is true |
| 19:12.16 | jlewis | yeah...but at least this is cosmetic...we had too many "it just doesn't work" bugs in 1.2.x (for several early x) and haven't tried it again recently |
| 19:12.37 | file | did you file bugs and follow through with them? |
| 19:12.56 | jlewis | yep...even had people from digium in our boxes trying to figure it out |
| 19:13.12 | jlewis | never got it resolved |
| 19:13.24 | file | k, if you had just said nope I would have been sad - but if you gave it effort, it makes me happier |
| 19:13.28 | ZPertee | where are we suppossed to put our config files, so that we don't take up lots of lines on this irc channel? |
| 19:13.37 | Nivex | ~pastebin |
| 19:13.42 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts |
| 19:13.51 | ZPertee | ok thanks |
| 19:13.55 | Nivex | You're welcome. |
| 19:14.15 | jlewis | I do have one box running 1.2.x...leme test and see if it does the same thing |
| 19:15.56 | jlewis | looks like its cosmetic, and IAX2 just doesn't give Elapsed Time in a CLI show channel |
| 19:16.15 | jlewis | in 1.2.x, instead of all 0's it's N/A |
| 19:16.28 | file | yay cosmetic |
| 19:16.36 | Rowter | its possible to pause queue members from manager_ |
| 19:16.36 | Rowter | ? |
| 19:16.58 | jlewis | still seems odd |
| 19:17.47 | Corydon-w | Rowter: from the CLI: show manager commands |
| 19:19.05 | Rowter | Corydon-w, ahh queuepause great >) |
| 19:19.22 | *** join/#asterisk frenzy (n=frenzy@196.46.104.134) |
| 19:20.23 | frenzy | Trying to call from * to * using IAX2 the second box is behind NAT when calling it I get CAUSE : No authority found |
| 19:21.07 | intensedrive | Do you have to use IAX for app_conf? |
| 19:26.15 | hmmhesays | BORED |
| 19:26.39 | frenzy | uhh |
| 19:26.47 | intensedrive | I cant get app_conf working |
| 19:26.50 | intensedrive | Crap |
| 19:26.51 | intensedrive | grr |
| 19:26.53 | frenzy | pls help with iax2 |
| 19:26.59 | Qwell[] | intensedrive: Why not use meetme? |
| 19:27.05 | intensedrive | not zaptel |
| 19:27.12 | intensedrive | no hardware |
| 19:27.15 | Qwell[] | ztdummy |
| 19:27.19 | intensedrive | I heard its a pain |
| 19:27.21 | intensedrive | recompile |
| 19:27.22 | Qwell[] | it's simple |
| 19:27.24 | frenzy | tyring to make incoming work using IAX2 between two boxes |
| 19:27.28 | Qwell[] | takes all of 2 minutes |
| 19:27.35 | intensedrive | you dont have to recompile the kernel? |
| 19:27.38 | Qwell[] | ...no |
| 19:27.45 | intensedrive | everything I read said you do |
| 19:27.49 | Qwell[] | unless you're missing important stuff, but that's unlikely |
| 19:28.07 | Qwell[] | get zaptel, run `make install`, and if it yells at you...then well...yeah |
| 19:29.37 | intensedrive | What makes meetme so Much better Qwell? |
| 19:29.52 | Qwell[] | intensedrive: it works ;) |
| 19:29.57 | intensedrive | HA |
| 19:31.24 | *** join/#asterisk razu_ (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 19:32.47 | intensedrive | Are zaptel and ztdummy different files |
| 19:32.52 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 19:33.03 | CunningPike | intensedrive: ztdummy is a zaptel module |
| 19:33.25 | smackus | so my first phase of asterisk is finally up and stable. |
| 19:33.36 | smackus | i have about 100 people connecting to two different servers. |
| 19:33.39 | intensedrive | you install zaptel then ztdummy |
| 19:33.40 | smackus | I am really excited. |
| 19:34.17 | Alric | smackus: Great :) |
| 19:34.22 | *** join/#asterisk jeanmi_ (n=besnard@mic92-2-82-67-216-218.fbx.proxad.net) |
| 19:34.25 | smackus | thanks to all of those who helped |
| 19:34.28 | frenzy | need some dummy help .... I have a reg string for iax 567:pwd@host |
| 19:34.45 | jeanmi_ | when I call my asterisk, the DTMF are working fine |
| 19:34.49 | frenzy | inorder to receive calls what will be the correct conext configuration |
| 19:34.59 | jeanmi_ | however when asterisk is calling me the dtmf wont work |
| 19:35.04 | jeanmi_ | what should I look at ? |
| 19:35.12 | frenzy | I want calls to go to context host extension 567 |
| 19:35.21 | Qwell[] | jeanmi_: sip? check your dtmfmode |
| 19:35.34 | Qwell[] | frenzy: add a per, with context=host |
| 19:35.46 | Qwell[] | peer too |
| 19:35.58 | jeanmi_ | Qwell[] I am just not too sure how to set it, is it global or per trunk ? |
| 19:36.06 | Qwell[] | per peer |
| 19:36.21 | *** join/#asterisk DFiber (n=bwarner@65.113.208.19) |
| 19:36.42 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 19:37.11 | jeanmi_ | but there is no peer in that very case |
| 19:37.25 | jeanmi_ | it is just asterisk calling my mobile phone |
| 19:37.33 | frenzy | <PROTECTED> |
| 19:37.33 | frenzy | type=peer |
| 19:37.33 | frenzy | context=host |
| 19:37.33 | frenzy | user=567 |
| 19:37.33 | frenzy | secret=pwd |
| 19:37.33 | frenzy | is that correct? |
| 19:37.46 | *** join/#asterisk Tall-guy (n=noway@207-195-103-110.regn.static.sasknet.sk.ca) |
| 19:37.49 | websae | that's fine |
| 19:37.52 | jeanmi_ | I connect to the manager through a php script |
| 19:37.53 | Qwell[] | jeanmi_: so how is asterisk connecting to your provider? |
| 19:37.53 | websae | it should work |
| 19:38.11 | jeanmi_ | the php script say =asterisk should call a given number@my outbound trunk |
| 19:38.11 | websae | setup extensions.conf to use that context now |
| 19:38.27 | jeanmi_ | qwell[] with a login/passwd ... |
| 19:38.42 | Qwell[] | jeanmi_: add a peer.. |
| 19:38.49 | jeanmi_ | oh, u mean the provider is a peer ? |
| 19:39.33 | dlynes_laptop | I'm wondering if anyone's experienced this issue on tdm400p's or tdm2400p's |
| 19:39.49 | dlynes_laptop | the problem is when a call comes in, and someone decides to make an outgoing call at the same time |
| 19:40.08 | dlynes_laptop | the two calls get bridged (asterisk doesn't detect the line as being in use) |
| 19:40.30 | jeanmi_ | qwell good it works, I have set dtmfmode=inband in my registration settings |
| 19:40.32 | dlynes_laptop | I'm experiencing this on a sangoma a200d, and don't know if it affects tdm400p's and tdm2400p's or not |
| 19:40.39 | Qwell[] | yuck |
| 19:40.52 | Qwell[] | jeanmi_: it works with inband? |
| 19:40.57 | frenzy | Is this dial string correct (sorry just not too iax friendly yet) exten => 567,1,Dial(IAX2/567,50,tr) |
| 19:41.13 | Qwell[] | frenzy: why are you uising r? |
| 19:41.15 | dlynes_laptop | frenzy: yeah, that's fine...but why are you adding the 'r'? |
| 19:41.38 | smackus | now I am scoping out for phase 2... expansion! I do not want to put more than a 4 port t1 card in a system. But I will have one dial plan that will need more than 4 t1s. And I would like to start growing this out to have 4 servers... some redundancy would be nice. putting a trunk group of T1s spanned across the 4 machines. so that I could lose up to 3 servers and still have all of my numbers get some sort of response. |
| 19:41.53 | smackus | any advice before I start trying to set that up? |
| 19:43.27 | frenzy | <PROTECTED> |
| 19:44.02 | anti-uce | frenzy: yes it correct 567 is the peer account on the other side |
| 19:44.35 | dlynes_laptop | frenzy: 'r', means generate ringing where it is not appropriate to do so |
| 19:46.21 | frenzy | remove r |
| 19:46.26 | frenzy | removed** |
| 19:46.39 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 19:46.46 | frenzy | I have friend on the box it is registering |
| 19:46.55 | frenzy | and peer on the box it has regstered |
| 19:47.22 | frenzy | peer from the box it is registering *** |
| 19:48.00 | frenzy | from the box first box (where it is registered to) is have exten => 567,1,Dial(IAX2/567,50,t) |
| 19:48.14 | Tall-guy | [TK]D=fender: you alive? |
| 19:48.27 | frenzy | but when I call it CAUSE : No authority found |
| 19:48.27 | frenzy | <PROTECTED> |
| 19:48.43 | file | registration does not mean you have authority to call something |
| 19:48.44 | anti-uce | frenzy: do you have 567 entry on your iax.conf? |
| 19:48.45 | ArkonaDev-Mike | I'm trying to call a BASH script from the Dialplan, (tcp ping) but it's not seeing the command switches in my script . . .however running it from the command line works fine. |
| 19:49.49 | frenzy | yes |
| 19:49.51 | ArkonaDev-Mike | I get a 510 Invalid or Unknown Command back from the AGI |
| 19:50.14 | *** join/#asterisk babyju (n=babyju@151.202.195.132) |
| 19:50.39 | anti-uce | frenzy: iax2 show registry what's d output? |
| 19:51.04 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 19:51.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 19:51.53 | Tall-guy | Help! Polycom IP501 and Asterisk...getting sip registration errors....(Xlite config works fine)...too many damn boxes to fill in on this Polycom! :( |
| 19:51.54 | frenzy | REMOTEIP:4569 567 ROUTERIP:50100 60 Registered |
| 19:54.17 | anti-uce | frenzy: pastebin some entry on both left and right leg of asterisk iax.conf just replace ip and username and password obviously |
| 19:57.07 | *** part/#asterisk babyju (n=babyju@151.202.195.132) |
| 19:58.04 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 19:59.52 | SplasPood | ugh, one of my peers is letting certain numbers go with ring no answer and not returning CONGESTION, etc so my stuff never fails over |
| 20:00.24 | Qwell[] | SplasPood: failing is bad anyways |
| 20:01.20 | SplasPood | Qwell[]: hrm? |
| 20:02.12 | SplasPood | 800-369-0996, I can call that via a number of my peers, but not my primary.. problem is, it just rings and rings via the primary, so I never get the chance to try another peer as I normally would on CONGESTIOn/CHANUNAVAIL/etc |
| 20:05.06 | frenzy | here it is... http://pastebin.ca/115751 |
| 20:05.12 | *** join/#asterisk edorsey (i=edorsey@216-63-192-130.ded.swbell.net) |
| 20:05.42 | anti-uce | frenzy: w8 |
| 20:06.28 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 20:10.10 | frenzy | anti-uce: ? |
| 20:10.29 | *** join/#asterisk wunderkin (n=wunderki@216-19-202-6.getnet.net) |
| 20:10.32 | rob0 | mog: When you go home today would you try http://pricewatch.com/ ? I know you're on Comcast (so am I), and I'm being blocked. |
| 20:10.47 | rob0 | I wrote to them but doubt they'll reply. |
| 20:10.48 | mog | lol |
| 20:10.53 | mog | thats kinda creapy |
| 20:10.55 | mog | ill look |
| 20:11.27 | anti-uce | frenzy: could you please change type=peer |
| 20:11.28 | mog | 403 forbidden |
| 20:11.34 | mog | by iis |
| 20:11.45 | anti-uce | frenzy: type=friend since your local server is registering on the remote right |
| 20:11.52 | mog | if i go to www.pricewatch.com it works |
| 20:12.00 | anti-uce | frenzy: then you initiate a call from remote server right? |
| 20:12.02 | rob0 | 403.6 ... oh haha ok, I will try that |
| 20:12.07 | mog | i think pricewatch is just dumb |
| 20:12.11 | frenzy | anti-uce: yap |
| 20:12.21 | rob0 | They use IIS, 'nuff said. :) |
| 20:12.33 | rob0 | thanks for checking it out |
| 20:12.53 | *** part/#asterisk firejon (n=jellis@206-169-49-105.static.twtelecom.net) |
| 20:13.48 | anti-uce | frenzy: change type=friend also on your local server |
| 20:14.43 | mog | works for ya rob0? |
| 20:15.23 | frenzy | same |
| 20:15.54 | frenzy | mybad ... CAUSE : No such context/extension |
| 20:16.37 | anti-uce | frenzy: check your dialplan the call should be go to s or specific extensions |
| 20:16.38 | anti-uce | w8 |
| 20:17.19 | anti-uce | frenzy: on the local server if there is an inboud call where should it go by default? |
| 20:17.20 | dlynes_laptop | What does the immediate=no/yes really do for you in the zapata.conf file? |
| 20:17.27 | anti-uce | frenzy: on what context? |
| 20:17.39 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.36) |
| 20:18.01 | dlynes_laptop | ; Specify whether the channel should be answered immediately or if the simple |
| 20:18.01 | dlynes_laptop | ; switch should provide dialtone, read digits, etc. |
| 20:18.09 | dlynes_laptop | That really doesn't say much, to me |
| 20:18.11 | frenzy | ok changes to s |
| 20:18.18 | frenzy | now works |
| 20:18.26 | rob0 | mog: yes, I can't believe I didn't try that first :) |
| 20:18.28 | frenzy | how can I send it to specific channel? |
| 20:18.55 | anti-uce | frenzy: exten => 212,1,Goto(blah. blah.) should be enough |
| 20:19.34 | frenzy | doesnt work |
| 20:19.50 | frenzy | unles itss exten => s |
| 20:20.37 | anti-uce | frenzy: it should much w8 let me check |
| 20:20.56 | anti-uce | frenzy: should match * |
| 20:20.59 | frenzy | or is it becoze i'm missing regexten |
| 20:21.12 | *** join/#asterisk Yasha (i=daemon@69.15.218.218) |
| 20:21.53 | anti-uce | frenzy: ask them while i'll try to figure it out thou |
| 20:23.13 | SplasPood | Qwell[]: Hrm, this is interesting.. if I don't supply local ring, that number works fine |
| 20:23.13 | Yasha | Please Help: Is it possible to place a long distance call to an Asterisk server located in the same local exchange, then have that server place the call over the Internet to another Asterisk server located in that long distance area code and have it placed locally? Let me know if this does not make sense? |
| 20:23.31 | Qwell[] | SplasPood: That's why you aren't supposed to use r... |
| 20:23.31 | dlynes_laptop | Yasha: yes, it's possible; yes, it makes sense |
| 20:24.00 | Yasha | dlynes_laptop: thank you so much, can you please point me in the right direction? |
| 20:24.07 | SplasPood | Qwell[]: I'm begining to see that... Why not tho.. Shouldn't it bridge when it gets an answer? |
| 20:24.24 | dlynes_laptop | Yasha: in the right direction for what? |
| 20:24.39 | Rowter | anyone know what values are requiered for QueuePause on manager? |
| 20:24.48 | frenzy | how do I send IAX2 incoming calls to a specific extension ? |
| 20:25.06 | CunningPike | Yasha: It's called toll bypass |
| 20:25.19 | Yasha | dlynes_laptop: sorry... :) basically, I already have 1 * server running using VOIP and would like to have another one setup in a different city. How do I configure that? |
| 20:25.22 | dlynes_laptop | frenzy: context=blahblah, [blahblah] exten => _X.,1,Dial(SIP/specificextension) |
| 20:25.35 | *** join/#asterisk elg (n=fugalh@falcon.fugal.net) |
| 20:25.35 | dlynes_laptop | Yasha: try iax2 |
| 20:25.41 | dlynes_laptop | Yasha: then you can do call trunking |
| 20:25.51 | dlynes_laptop | ~iax |
| 20:25.53 | jbot | it has been said that iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 20:26.05 | file | EEEEEEEEEKS |
| 20:26.06 | dlynes_laptop | ~trunking |
| 20:26.13 | dlynes_laptop | ~trunk |
| 20:26.15 | jbot | extra, extra, read all about it, trunk is my trunk my trunk; my lovely asterisk trunk (check it out) |
| 20:26.52 | frenzy | dlynes_laptop: I meant instead of exten => s how I make it go to exten => 123 |
| 20:27.04 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 20:27.16 | frenzy | so I can Have many incoming friends under the same context |
| 20:27.29 | Yasha | dlynes_laptop: how will the call be placed when it arrives to the other * server? Sorry, I am thinking through this and still not following... |
| 20:27.48 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 20:28.10 | anti-uce | Yasha: 1st you should figure it out how can you send the call to 2nd asterisk server on the same local exchange |
| 20:28.42 | dlynes_laptop | Yasha: you need to learn how to send calls between two asterisk boxes using iax2 first |
| 20:28.49 | dlynes_laptop | Yasha: so you need to read up on iax2 channels |
| 20:28.59 | dlynes_laptop | frenzy: I already told you |
| 20:29.08 | dlynes_laptop | frenzy: context=blahblah, [blahblah] exten => _X.,1,Dial(SIP/123) |
| 20:29.48 | anti-uce | dlynes_laptop: problem is exten => 212,1,Goto(blah) doesn't much on his server if there's incoming call |
| 20:29.49 | Tall-guy | Ok, Polycom IP 501, not a bad phone to configure....once the FIRMWARE has been upgraded :) |
| 20:29.59 | anti-uce | dlynes_laptop: but exten => s is working |
| 20:30.21 | dlynes_laptop | anti-uce: s/doesn't much/doesn't do much/? |
| 20:30.39 | Yasha | sure... I am not actually doing anything right nowm just trying to follow the logic. So, yes, I do understand that two servers will have to talk to each other. No problem. So, if I pick up the phone and place a long distance phone call, I can have it routed to the second * server in that exchange, then ????? - how does this call gets out on pots? hwo does that work? |
| 20:31.06 | anti-uce | dlynes_laptop: exten => s,1,Goto(frenzy_ivr) is working while exten => 212,1,Goto(frenzy_ivr) the call doesn't go thru |
| 20:31.12 | dlynes_laptop | Yasha: the other asterisk box will have some logic in it to route that call out onto some pstn hardware |
| 20:31.32 | dlynes_laptop | Yasha: such as an fxo gateway, or a tdm400p card, or an a200d card |
| 20:31.33 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 20:31.55 | anti-uce | YashaL: or have it route to another SIP/IAX2 gateway |
| 20:31.59 | *** join/#asterisk oadaeh (n=jason@las-static-208.57.199.83.mpowercom.net) |
| 20:32.35 | dlynes_laptop | frenzy: I take it call's coming in on an analog line? |
| 20:32.44 | frenzy | no |
| 20:32.44 | dlynes_laptop | s/call's/the call's/g |
| 20:32.50 | frenzy | remote IAX |
| 20:33.08 | dlynes_laptop | and is the remote IAX not passing along the extension it's wanting to dial? |
| 20:33.27 | frenzy | how do I go about doing that? |
| 20:33.41 | Yasha | Got it.. starting to follow it now. Ok, what hardware would you use if you need to make hundreds of calls? |
| 20:33.42 | frenzy | I have regexten defined but same |
| 20:33.50 | dlynes_laptop | frenzy: the switch statement, or Dial(IAX2/iax2peername/${EXTEN}) |
| 20:34.36 | dlynes_laptop | Yasha: pri hardware (sangoma a104d, sangoma a108d, or a digium te410p with hardware echo canceller) |
| 20:35.05 | dlynes_laptop | Yasha: a104d and te410p both handle 4 pri ports; an 108d can handle 8 pri ports |
| 20:35.29 | frenzy | phew |
| 20:35.29 | frenzy | finally |
| 20:35.36 | anti-uce | frenzy: what did you do? |
| 20:35.47 | frenzy | thanks anti-uce & dlynes_laptop |
| 20:35.50 | Yasha | awesome..thank you SO MUCH to everyone. |
| 20:36.02 | frenzy | changed the dial string to Dial(IAX2/iax2peername/${EXTEN}) |
| 20:36.16 | anti-uce | frenzy: ok |
| 20:36.26 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 20:36.57 | anti-uce | frenzy: so that it will pass ${EXTEN} to another * box to make judgment of the call ok |
| 20:37.44 | *** part/#asterisk anti-uce (n=anti-uce@210.213.198.89) |
| 20:38.10 | dlynes_laptop | frenzy: yeah...if you don't pass that extension along, it'll dump it into the default extension of 's' |
| 20:45.13 | *** join/#asterisk ArkonaDev-Mike (n=somewher@65.203.186.131) |
| 20:46.02 | *** join/#asterisk turbo- (i=miles@S01060060979907a2.ed.shawcable.net) |
| 20:48.08 | turbo- | couple easy questions (i hope) - i plan to use a TDM400P card (1 FXO module) - does it matter if i use kernel 2.4 or 2.6? i know that changes to scheduling an interrupts can affect quality (like clicking, etc) |
| 20:48.32 | frenzy | how is the TDM2400P ? |
| 20:48.43 | frenzy | I need good echo cancellation |
| 20:48.44 | *** join/#asterisk MikeJ (n=vircuser@c-24-13-240-121.hsd1.il.comcast.net) |
| 20:48.47 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 20:48.57 | frenzy | trying the wildcard X100P but it was aweful |
| 20:49.34 | *** join/#asterisk RaYmAn-Bx (i=rayman@skumler.dk) |
| 20:50.06 | Nugget | WWhhaatt''ss eecchhoo ccaanncceellaattiioonn?? |
| 20:50.42 | Qwell[] | Nugget: That probably took far too long to type |
| 20:51.01 | Nugget | so you're saying my latency is bad, too? dang. |
| 20:51.10 | Qwell[] | quite |
| 20:51.14 | file | Nugget: your mooability level is VERY high |
| 20:51.21 | Nugget | ]:8) Mooooooo |
| 20:51.28 | frenzy | I dont know if its the hardware but I tried just about all echo cancellers but want happy with x100p |
| 20:51.37 | frenzy | wasnt** |
| 20:52.39 | SplasPood | Can someone explain to me why supplying local ring (r option to Dial) is a bad idea (tm) ? |
| 20:53.04 | mog | yes |
| 20:53.10 | mog | lots of things supply early media |
| 20:53.15 | mog | r over rides it |
| 20:53.32 | file | which is why it should only be used when you know what you are doing |
| 20:53.33 | SplasPood | When you say early media, what exactly do you mean? |
| 20:55.08 | *** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net) |
| 20:55.31 | Nugget | isn't it clear from the context? |
| 20:56.50 | SplasPood | Nugget: No.. I applogize for asking what must be an imbecilic question .... |
| 20:56.57 | SplasPood | but if someone could explain, I would appreciate it.. |
| 20:58.08 | Nugget | when you supply your own "r" you are displacing the "r" that is likely being provided by the channel and consequently make it much harder to detect or diagnose problems. |
| 20:58.31 | SplasPood | Nugget: what's odd is I have an 800 #, that works fine without local ring, but with local ring the channels never get bridged |
| 20:58.43 | SplasPood | yet 99% of other numbers are fine.. |
| 20:58.57 | Nugget | e.g. the channel is providing a fast busy, which is useful information, but the caller never hears it because they just hear the ringing you're generating |
| 20:58.57 | SplasPood | I mean, an 800# i'm calling.. |
| 20:59.08 | dlynes_laptop | frenzy: that's because the x100p is a huge pos |
| 20:59.13 | SplasPood | Nugget: right, but shouldn't I be getting CONGESTION from the remote side in that case? |
| 20:59.35 | noname32 | hey guys is there a way to make it so when some one is leaving a vm it makes it louder because it is realy soft right now |
| 20:59.50 | SplasPood | noname32: there's an option to Voicemail() I believe |
| 20:59.54 | SplasPood | noname32: to control the gain |
| 21:00.48 | Tall-guy | splas: REALLY????? hell, I've been trying to balance my RXGAINS on my ZAP channels to make voicemail better, which was hooping my eeeechooo cancellation (yeah, mooo).... |
| 21:01.04 | dlynes_laptop | SplasPood: not afaik |
| 21:01.48 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 21:01.51 | SplasPood | Tall-guy: I *think*.. |
| 21:01.58 | CunningPike | Tall-guy: VoiceMail(1234@default|ug(20)) |
| 21:02.02 | SplasPood | <PROTECTED> |
| 21:02.07 | dlynes_laptop | oh yeah...nvm |
| 21:02.09 | dlynes_laptop | 'g' option |
| 21:02.23 | CunningPike | Tall-guy: That's REAL LOUD |
| 21:05.31 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 21:05.48 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 21:09.06 | turbo- | anyway, zaptel drivers... will i notice any difference on kernel 2.4 as opposed to 2.6 using a TDM400P? |
| 21:10.48 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 21:14.13 | *** join/#asterisk AJaymn (n=Ya@70.59.126.206) |
| 21:16.49 | toerkeium | guys, what's a high latency from where I should have start having problems or delays? any idea' |
| 21:16.55 | toerkeium | ? |
| 21:17.05 | carrar | 150ms |
| 21:17.09 | carrar | well |
| 21:17.11 | carrar | actually |
| 21:17.17 | carrar | couple seconds I guess |
| 21:17.26 | carrar | problembeing hard to talk because of dealy |
| 21:17.26 | De_mon | I'm receiving a call, but it's not being answered.. http://pastebin.ca/115821 |
| 21:17.29 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 21:17.42 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:17.47 | carrar | I would think 20 seconds is long ;) |
| 21:17.50 | foo | mitcheloc: hey |
| 21:17.54 | toerkeium | carrar: not voice interrumptions? |
| 21:18.03 | carrar | interrumptions from what? |
| 21:18.15 | toerkeium | I mean, when the voice you hear cuts from time to time |
| 21:18.16 | De_mon | line 32 says it found the peer 'broadband' which i defined in sip.conf where I set context=inbound |
| 21:18.22 | carrar | packet loss? |
| 21:18.32 | toerkeium | something like that |
| 21:18.40 | toerkeium | but caused by high latency |
| 21:18.45 | toerkeium | could it be possible? |
| 21:18.50 | carrar | yes |
| 21:18.57 | carrar | if it's not consistant latency |
| 21:19.13 | De_mon | but line 39 says it cant find the extension in local domain ??? |
| 21:19.16 | toerkeium | so if I have to use a VOIP provider, I better use one in my country, right? |
| 21:19.19 | carrar | like if latency pops up to 20 seconds |
| 21:19.25 | carrar | or route changes |
| 21:19.31 | Tall-guy | <PROTECTED> |
| 21:19.35 | carrar | packet ordering is important |
| 21:19.41 | carrar | AKA QoS |
| 21:19.54 | toerkeium | even if I have available bandwidth for sure? |
| 21:20.08 | carrar | sounds like you need qos |
| 21:20.17 | carrar | so voice packets have priority? |
| 21:20.36 | toerkeium | no, but I am sure I have available bandwidth |
| 21:20.40 | mitcheloc | foo, que pasa? |
| 21:20.41 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 21:20.41 | carrar | sounds like you need qos |
| 21:20.46 | foo | mitcheloc: PM? |
| 21:20.53 | carrar | available bandwidth does not gurantee packet timing |
| 21:20.58 | mitcheloc | foo, okay |
| 21:21.40 | toerkeium | carrar: oh, ok, so I better implement packet priority |
| 21:21.43 | carrar | available bandwidth is subject to interpitation without values |
| 21:21.51 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-a3d7cf053b2c4ed8) |
| 21:21.58 | toerkeium | carrar: it's all RTP what I should consider right? |
| 21:22.08 | carrar | toerkeium, "implement packet priority" is QoS, yes |
| 21:22.25 | carrar | the voice stream is RTP yes |
| 21:22.41 | toerkeium | should I give priority to the SIP protocol too? |
| 21:22.52 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 21:22.59 | carrar | just give priority to everything from the device |
| 21:23.09 | carrar | yes |
| 21:23.42 | toerkeium | excellent |
| 21:23.49 | toerkeium | thanks carrar |
| 21:23.53 | carrar | np |
| 21:24.08 | carrar | besure to do that both on the routers and switches |
| 21:24.32 | toerkeium | hmm.. my switches can't prioritize packets |
| 21:24.35 | carrar | CoS |
| 21:24.43 | carrar | most switches can do class of service |
| 21:24.57 | carrar | and set it on the switch for voice devices |
| 21:25.00 | toerkeium | they are all catalyst 1900, pretty old |
| 21:25.11 | carrar | disable honoring of cos on data ports |
| 21:25.21 | carrar | ah |
| 21:25.33 | carrar | hope for the best then :) |
| 21:25.56 | toerkeium | I could probably dedicate one switch for this devices, that would fix the priorization from the switch? |
| 21:26.13 | carrar | should be better yes |
| 21:26.22 | carrar | If it's all just voice devices |
| 21:26.41 | toerkeium | yes, will be all softphones |
| 21:26.47 | carrar | (assuming you are having conge4stion on the switch in the first place) |
| 21:27.03 | carrar | but good QoS is end to end |
| 21:28.00 | toerkeium | going to check my switches to be sure completely |
| 21:32.59 | toerkeium | carrar: just to be sure, I just have to make sure that rpt and sip packets goes first, regardless of bandwidth availability.. right? |
| 21:35.48 | carrar | yes |
| 21:36.05 | *** join/#asterisk Nitrus^ (n=ntisog@72-34-76-86.skyriver.net) |
| 21:36.40 | toerkeium | do you know how to apply a g729 licence in * ? no idea how it works, not even where to buy one |
| 21:37.00 | carrar | there are instructions on the voip page |
| 21:37.08 | Nitrus^ | im having a problem transfering calls with asterisk.. i specify t on the dial command but the caller cant transfer me by pressing #1.. i can do an attended transfer but not unattended.. |
| 21:37.27 | Nitrus^ | callee* |
| 21:38.21 | toerkeium | thanks again carrar |
| 21:38.34 | carrar | http://www.digium.com/en/supportcenter/documentation/viewdocs/G729 |
| 21:39.55 | turbo- | based on the lack of answers, i assume having kernel 2.4 and 2.6 dont make any difference? :) |
| 21:39.58 | *** join/#asterisk oej (n=oej@65.204.4.227) |
| 21:40.00 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 21:41.05 | hads|home | turbo-: It should work fine on both |
| 21:42.39 | *** join/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 21:43.04 | matteof | hi, I cannot call from a CCME to * |
| 21:43.13 | *** join/#asterisk turbo- (i=miles@S01060060979907a2.ed.shawcable.net) |
| 21:43.14 | matteof | can anyone help? |
| 21:43.17 | turbo- | stupid cable modem |
| 21:43.29 | Qwell[] | matteof: If you actually ask a question, sure |
| 21:44.10 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 21:44.12 | matteof | I'am trying to integrate a Cisco CCME to an asterisk testing network |
| 21:44.39 | matteof | now i can call from * to a Cisco 7960 that is registered on the CCME |
| 21:44.54 | matteof | but I cannot call from the 7960 to an * extension |
| 21:45.02 | matteof | I receive a 404 error |
| 21:45.07 | matteof | can you help me? |
| 21:45.12 | Qwell[] | matteof: turn on verbose, and see what it's trying to call |
| 21:45.34 | matteof | on a callmanager or on *? |
| 21:45.37 | Qwell[] | * |
| 21:46.08 | matteof | how to turn on verbose on *? |
| 21:46.31 | Qwell[] | set verbose 10 |
| 21:46.31 | matteof | I have debug on |
| 21:48.22 | matteof | ok, but I cannot see anything |
| 21:48.47 | toerkeium | guys, I am planning to have 10 sip lines, and this provider wors with g.729.. so I need 10 g.729 licences or I could need more? no idea where I would need more, not sure how it counts |
| 21:49.19 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:49.37 | matteof | Qwell[]: this is the debug output |
| 21:49.37 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: Allocating new SIP dialog for CF3CC99E-233911DB-8185DF86-A07C247@192.168.1.1 - INVITE (With RTP) |
| 21:49.37 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE |
| 21:49.37 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: * SIP extension value: 7 for call CF3CC99E-233911DB-8185DF86-A07C247@192.168.1.1 |
| 21:49.37 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: Checking SIP call limits for device |
| 21:49.38 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: Updating call counter for incoming call |
| 21:49.38 | Qwell[] | ~paste |
| 21:49.40 | jbot | from memory, paste is see http://paste.husk.org |
| 21:49.40 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: Updating call counter for incoming call |
| 21:49.42 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: = Found Their Call ID: CF3CC99E-233911DB-8185DF86-A07C247@192.168.1.1 Their Tag 38579490-6BE Our tag: as3592b0c2 |
| 21:49.43 | *** join/#asterisk eBody (n=ehernand@207.71.51.162) |
| 21:49.45 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: **** Received ACK (6) - Command in SIP ACK |
| 21:49.49 | matteof | Aug 4 23:47:48 DEBUG[20357] chan_sip.c: Stopping retransmission on 'CF3CC99E-233911DB-8185DF86-A07C247@192.168.1.1' of Response 101: Match Found |
| 21:49.49 | file | argh |
| 21:49.50 | matteof | Aug 4 23:47:52 DEBUG[20357] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) |
| 21:49.52 | matteof | ops....sorry |
| 21:50.03 | eBody | hey guys, when i try to leave a voice mail or even record my voicemail away messages, it cuts me off right when it prompts me to start recording. any ideas? |
| 21:50.22 | toerkeium | I will repeat my question just in case it wasn't readed |
| 21:50.40 | hads|home | eBody: What does it show in the consloe when this is happeneing? |
| 21:50.44 | Qwell[] | toerkeium: You could need more - it depends |
| 21:51.01 | eBody | hads, which console? |
| 21:51.04 | toerkeium | Qwell[]: what would be the case? |
| 21:51.05 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-33-32.losaca.adelphia.net) |
| 21:51.07 | carrar | licesnse are for transcoding |
| 21:51.14 | hads|home | eBody: The Asterisk console |
| 21:51.26 | Qwell[] | toerkeium: If you wanted your devices to do g729 on the lan...various things |
| 21:51.34 | matteof | Qwell[]: http://paste.husk.org/6429 <--- here you can find the debug |
| 21:51.58 | Qwell[] | matteof: except that I asked for verbose, not debug... |
| 21:52.02 | file | out of curiosity, why do you have debug on? |
| 21:52.25 | toerkeium | Qwell[]: I could use any codec from "SIP Device > asterisk", the issue is between asterisk and sip provider.. how does it apply ? |
| 21:52.27 | eBody | hads, : Recording the message |
| 21:52.27 | eBody | <PROTECTED> |
| 21:52.27 | eBody | <PROTECTED> |
| 21:52.27 | eBody | <PROTECTED> |
| 21:52.27 | eBody | <PROTECTED> |
| 21:52.28 | matteof | Qwell[]: verbose doesn't display anything |
| 21:52.31 | De_mon | I'm receiving a call, but it's not being answered.. http://pastebin.ca/115821 |
| 21:52.45 | De_mon | line 32 says it found the peer 'broadband' which i defined in sip.conf where I set context=inbound |
| 21:53.09 | De_mon | but line 39 says it cant find the extension in local domain, which I have defined in the inbound context |
| 21:53.16 | carrar | toerkeium, good rule of thumb is one g729 lic per phone |
| 21:53.38 | Qwell[] | carrar: in this case, that's incorrect... |
| 21:53.52 | Rowter | dialing with an agi will end the agi just after an answer any idea how could I hang there after native bridge? |
| 21:54.21 | carrar | well if even gets on a conf |
| 21:54.24 | carrar | everyone |
| 21:54.25 | skac | anyone know any free sip providers in Canada, for Canada calls? |
| 21:54.27 | Rowter | I was able to do this with a macro, but on this new version it drops agi.. |
| 21:54.30 | hads|home | eBody: Looks OK |
| 21:54.38 | Qwell[] | carrar: then they'd be using ulaw or something |
| 21:54.42 | eBody | hads, no idea then? |
| 21:54.49 | eBody | it just doesn't give any time. |
| 21:54.51 | carrar | not if g729 is only set |
| 21:54.53 | Tall-guy | skac: Skype? :) |
| 21:54.59 | skac | indeed... |
| 21:55.28 | matteof | Qwell[]: any idea? |
| 21:55.35 | carrar | we don;t have enough details |
| 21:55.42 | toerkeium | so, if I set my softphones to lets say ulaw, it will count only all out/incoming calls encoded in any other codec than g729 ? |
| 21:55.44 | hads|home | eBody: No idea from that output sorry. |
| 21:55.50 | eBody | thanks hads. |
| 21:56.43 | carrar | toerkeium, if you are paying for 10 channels from your sip provider get 10 lics |
| 21:57.39 | toerkeium | well.. licences are cheap, just don't wanted to buy unnecesary things :) |
| 21:57.42 | Tall-guy | skac: you looking for cheap calls to Grandma? or business use? |
| 21:57.45 | carrar | and run ulaw inside |
| 21:57.50 | toerkeium | will take your suggestiong carrar |
| 21:57.51 | intensedrive | where do you guys go to get your voice prompts professionally recorded |
| 21:58.37 | toerkeium | I should avoid having the softphones in a nated lan right? |
| 21:58.53 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
| 21:59.02 | carrar | unless you need it nated for specific network reasons |
| 21:59.11 | carrar | but I wouldn't |
| 21:59.14 | tzanger | toerkeium: I have polycom ip501s behind nat (many) without issue |
| 21:59.27 | tzanger | no fancy config on the polycom, and asterisk has host=dynamic and nat=yes |
| 21:59.28 | tzanger | that's it |
| 21:59.45 | Tall-guy | tzanger: can I bug you in private re: the 501 config (not nat, just plain old config w/asterisk) |
| 21:59.52 | skac | Tall-guy: Gramdma ;) |
| 21:59.57 | tzanger | I'm just popping in here for a moment |
| 22:00.00 | tzanger | heading to the cottage |
| 22:00.03 | Tall-guy | skac: what province? |
| 22:00.08 | Qwell[] | intensedrive: thevoice.digium.com |
| 22:00.09 | toerkeium | great, as you can see I am trying to find the most secure configuration |
| 22:00.11 | skac | Tall-guy: i am moveing to Canada in 2 weeks, and i would allso like to ring home (the UK) |
| 22:00.12 | Tall-guy | tzanger: no sweat, talk to you later :) |
| 22:00.15 | skac | Ontario. |
| 22:00.43 | Tall-guy | skac: ah, sorry, Sask here. |
| 22:00.53 | skac | was born in sask hehe. |
| 22:00.55 | skac | where abouts? |
| 22:01.04 | matteof | has anyone some experience in integration between cisco ccme and *? |
| 22:01.36 | intensedrive | Thanks Qwell |
| 22:01.42 | Tall-guy | <PROTECTED> |
| 22:01.46 | skac | lol |
| 22:01.52 | skac | where i was born! |
| 22:01.53 | skac | hehe |
| 22:02.37 | *** part/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 22:02.41 | Qwell[] | intensedrive: looks like it changed, actually |
| 22:02.43 | Qwell[] | http://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=IVRPROMPT |
| 22:06.17 | *** join/#asterisk topping (n=topping@ppp-67-124-89-235.dsl.pltn13.pacbell.net) |
| 22:06.22 | kavit | matteof: are you using sip? |
| 22:06.56 | kavit | if your version of cme is less than 3.4 don't bother |
| 22:07.20 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 22:08.14 | kavit | matteof: allow connections sip to sip came to be in CCME 3.4, without that forwarding calls to voicemail (Unity) is a royal pain in the derriere |
| 22:09.56 | kavit | matteof: call forwarding and transfers are done using SIP 302 Moved Temporarily by default for SIP. 3.5 on CCME supports hairpin sip to sip bridging |
| 22:12.18 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 22:13.29 | matteof | kavit: I am here...sorry... |
| 22:13.49 | matteof | kavit: yes, i am using sip |
| 22:14.02 | matteof | and connection from * to ccme is ok |
| 22:14.14 | matteof | connection from ccme to * is KO |
| 22:14.41 | kavit | matteof: to make calls from ccme to * you have to allow anonymous calls in your context |
| 22:14.51 | matteof | ccme is version 3.3 |
| 22:15.34 | kavit | matteof: you will have to put something in from-sip and firewall your asterisk box |
| 22:16.07 | matteof | kavit: how can i allow anonymous call? |
| 22:16.11 | kavit | matteof: you can use sip-ua to register |
| 22:16.57 | matteof | kavit: I don't want to use sip-ua, I want to set up a trunk |
| 22:20.31 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 22:20.36 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:20.42 | Dr-Linux | hi all |
| 22:21.34 | *** join/#asterisk oceanlan|dustin (n=info@cpe-24-210-253-66.woh.res.rr.com) |
| 22:21.49 | oceanlan|dustin | Hey all, I need some help with ring Groups. Anyone got a sec? |
| 22:22.23 | matteof | kavit: I can't find hints (by google) on how to allow anonymous call, can you help me? |
| 22:22.47 | oceanlan|dustin | I am having troubles with more than 9 ring groups, does anyone know why I cant use 10+ for ring groups? |
| 22:23.42 | eKo1 | Funny how I read the words ring and groups and just though about abstract algebra. |
| 22:23.48 | oceanlan|dustin | exten => 12,1,Macro(rg-group,18,,9374401692) ; |
| 22:23.48 | oceanlan|dustin | exten => 12,2,Goto(ext-group,14,1) ; |
| 22:24.04 | oceanlan|dustin | is this not working because I am using 12??? |
| 22:24.17 | eKo1 | Time for some math. |
| 22:31.16 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 22:34.54 | Dr-Linux | i have problem with one of my cisco 7940 |
| 22:35.37 | *** join/#asterisk Kumba_ (n=kumba@office.crashsys.com) |
| 22:35.50 | Dr-Linux | it's on spanDSP, mean on the same lan where asterisk server located, when it calls out site, voice breaks after sometime |
| 22:35.55 | Dr-Linux | what can be the issue? |
| 22:36.37 | matteof | how to allow anonymous call on a context? |
| 22:36.43 | *** join/#asterisk wvtrixbox (n=roger@216.115.192.22) |
| 22:38.43 | *** join/#asterisk linlin (i=linlin@c-67-184-230-25.hsd1.il.comcast.net) |
| 22:39.46 | *** join/#asterisk chexum (i=chexum@gateway/tor/x-7ce238cc56834b72) |
| 22:39.50 | *** join/#asterisk Delta239 (n=blablabl@200.124.18.171) |
| 22:41.22 | Delta239 | hey if i have a tdm400p with 4 fxo channels.. and my second channel is like open and if i call the number attached to that channel it will give me a busy tone |
| 22:42.04 | Kumba_ | Say someone calls in on Zap/1... that context is passed through an automated attendant script/etc, and eventually rings to an extension... within the context for that extension, is it possible to flash the channel and then dial a number on that channel? |
| 22:42.30 | Kumba_ | I'm setting up a SOHO for myself, and by flashing the line, I can dial my cell phone, then hang up, and it'll forward the phone call to me... |
| 22:43.06 | *** join/#asterisk budmang (n=budman@12.206.134.162) |
| 22:43.11 | *** join/#asterisk apardo (n=apardo@87.217.145.80) |
| 22:43.23 | budmang | anyone use vitelity.net? |
| 22:45.15 | matteof | how to allow anonymous call on a context? |
| 22:46.26 | Kumba_ | So a priest, a sadist, and a pedophile walk into the bar... and that was just the first guy! |
| 22:47.06 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:48.30 | *** join/#asterisk turbo- (i=miles@S01060060979907a2.ed.shawcable.net) |
| 22:49.02 | turbo- | anyone know off hand if a TDM400P would fit into a Shuttle small form factor case? ;) |
| 22:49.15 | De_mon | heh |
| 22:49.19 | De_mon | off hand, I'd say no |
| 22:49.57 | turbo- | well, some of the shuttle cases that have pci support full heigh cards, so its really a matter of length i think |
| 22:50.21 | hads|home | turbo-: Should do, it's just a half length card |
| 22:50.44 | hads|home | Interrupts would be the thing to worry about |
| 22:50.53 | turbo- | yea, that was my other concern |
| 22:52.08 | turbo- | im just running out of places to put full size towers ;) |
| 22:52.17 | De_mon | half length? I must be thinking of the 2400P |
| 22:52.31 | turbo- | yea, 2400P are considerably larger |
| 22:52.35 | De_mon | ^_^ |
| 22:52.39 | Delta239 | hey if i have a tdm400p with 4 fxo channels.. and my second channel is like open and if i call the number attached to that channel it will give me a busy tone |
| 22:52.39 | Kumba_ | anyone have any clue behind the "Ring/Off-Hook in strange state 6" error messages? or know what usually triggers it? |
| 22:53.19 | Kumba_ | Would my particular linux distro effect that? (Slack 10.2) |
| 22:54.03 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 22:54.06 | Nugget | Asterisk doesn't care what Linux you're running. Asterisk doesn't even care if you're running Linux at all. |
| 22:54.14 | hads|home | telnet! |
| 22:54.14 | Nugget | telnet is eeeeeeevil! |
| 22:54.24 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:54.33 | turbo- | i could always do a mini-atx machine instead of mini-itx i suppose |
| 22:54.40 | Kumba_ | Zaptel cares tho... and that would be where my ring/off-hook errors would be coming from right? |
| 22:54.47 | turbo- | should reduce possible interrupt issues |
| 22:54.57 | turbo- | Kumba_: zaptel cant tell your distro either. just your kernel version |
| 22:55.15 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 22:56.30 | Nitrus^ | if i have 4 incoming zap lines that i dial out on how do i get asterisk to select the next one in line that isn't busy/ |
| 22:56.52 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 22:57.00 | hads|home | Nitrus^: Put the in a group and Dial(Zap/g1 or whatever |
| 22:57.07 | Nitrus^ | ahh |
| 22:57.09 | Nitrus^ | ok thanks alot |
| 22:57.20 | Kumba_ | hmm... if I do a ztcfg -vvv without changing any settings, is it normal for all spans to get a red alarm then come back up? |
| 22:57.27 | Nitrus^ | i setup groups in zapata.conf correct? |
| 22:57.59 | Nitrus^ | nvm i got it |
| 22:58.49 | Kumba_ | I'm trying to trouble shoot a problem... |
| 22:58.57 | *** join/#asterisk Trakkasure (n=Sgemtum@189.sub-70-222-221.myvzw.com) |
| 22:59.19 | Kumba_ | With my channel bank plugged into my T1... everything works fine... but when I plug my TE205p into it, the span will drop red, then come back up instantly... |
| 22:59.24 | Kumba_ | at randomn intervals... |
| 22:59.28 | Nitrus^ | how do i get call transfer to work with the # key? do i just push #<EXT> while in call and it will transfer? this is for unattended xfer |
| 23:00.11 | hads|home | Nitrus^: Look at the t and T options to the dial command |
| 23:00.32 | Nitrus^ | i have them all set Dial(Zap/X,,tTr) |
| 23:00.44 | hads|home | Ew. r |
| 23:00.48 | mountainm2k | OK, so I have auto-attendant, and options for several departments "press 1 for accounting"... Any good way to make it obvious to the user that it came via that option? |
| 23:01.44 | hads|home | Nitrus^: That stuff is setup in features.conf |
| 23:02.26 | mountainm2k | SetCIDName works, but then I don't have the CID name anymore... Even worse for transfer... |
| 23:02.59 | Nitrus^ | yeah i uncommented everything for blindxfer and it doesnt do anything |
| 23:03.07 | Nitrus^ | i tried doing a hook flash then typing the crap in but no go |
| 23:03.20 | De_mon | Looking for +18638774181 in local (domain 66.192.107.225) |
| 23:04.06 | De_mon | if the peer is setup with context=inbound why is asterisk sending back a 404 not found? |
| 23:05.13 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.13.Dial1.SanJose1.Level3.net) |
| 23:07.14 | hads|home | Nitrus^: What phones are you using? |
| 23:10.24 | Nitrus^ | some non ADSI phones |
| 23:10.40 | Nitrus^ | they have multi lines and all that crap and built in transfer |
| 23:10.49 | Nitrus^ | im switching to cisco phones as soon as they arrive |
| 23:11.14 | Nitrus^ | but hte built in transfer is part of their integrated PBX features |
| 23:11.28 | Nitrus^ | it's not the same as the transfer sequence dialed by a phone with a normal transfer button |
| 23:12.16 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
| 23:14.01 | De_mon | hrm, its getting it from the [general] domain= settings.. grr how to change it |
| 23:15.50 | De_mon | OF COURSE! now it works.. O_o i give up. atleast it works now comment out the context, then uncomment it and walah |
| 23:19.34 | De_mon | okay.. now its not working again, i know it worked the 1st time i reloaded.. maybe I didn't save. Boo |
| 23:21.51 | Nitrus^ | sweet, it works now.. thanks hads! |
| 23:22.41 | Nitrus^ | the only thing i dont like is the caller hears then #1 sequence |
| 23:29.31 | Nitrus^ | is there a way to stop the caller from hearing tones that are dialed on the callee's end? |
| 23:33.39 | *** join/#asterisk adorah (n=Administ@87.68.146.78.cable.012.net.il) |
| 23:34.55 | Kumba_ | Is there a way to look at the caller Id on a zap channel from asterisk console? |
| 23:38.38 | Rowter | when I dial using an agi, it ends the agi after calling, any ideas how could I wait to the call ends so the agi gets the dialstatus? |
| 23:39.56 | *** join/#asterisk kramba (n=icechat5@216-15-64-119.c3-0.eas-ubr9.atw-eas.pa.cable.rcn.com) |
| 23:40.11 | *** join/#asterisk tempest1 (n=asf@adsl-150-198-149.chs.bellsouth.net) |
| 23:44.17 | kramba | Inbound PSTN calls are having trouble with DTMF tones not working very well, sometimes a caller has to press the same key 4-5 times before asterisk hears it. Also I'm using a softphone and the callers DTMF tones sound really crackly if they press a key when I"m talking to them, might just be the softphone, but not sure if this is what the FXO card is hearing? If so that's probably the problem, if not I don't know th |
| 23:44.30 | Un1x | heh |
| 23:44.39 | Un1x | i wanan get a DTMF tone decoder :D |
| 23:45.07 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 23:48.43 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 23:49.08 | *** join/#asterisk angom_w (n=angom@red-corp-200.79.158.218.telnor.net) |
| 23:49.50 | kramba | anybody? |
| 23:52.28 | toerkeium | does anyone know why could be the reason that my php script isn't getting the returned headers from asterisk? could it be a buffer problem? |
| 23:52.45 | toerkeium | it keeps executing the script, without result |
| 23:59.46 | kramba | wow this channel is dead |
| 23:59.52 | puppet | not at all |
| 23:59.54 | puppet | just late |