| 00:00.14 | Druken | probably the same info... |
| 00:00.17 | dlynes | Druken: google? |
| 00:00.52 | kink0 | markus99, in what slot have you pluged the card ? |
| 00:00.52 | dlynes | Druken: Google's pretty amazing at reverse lookups...just type it in NPA-NNN-XXXX format |
| 00:01.18 | dlynes | kink0: He didn't have the chan_zap.so channel driver loaded, so it wouldn't matter what slot it was in, it still wouldn't work |
| 00:01.30 | markus99 | kink0: next to the video card, which is the only one available |
| 00:01.41 | kink0 | markus99, I had a simmilar issues with Digium TE cards, while PCI was 133Mhz, so fix it to 66Mhz |
| 00:02.06 | kink0 | dlynes, well... that is other problem if has not chan_zap.so loaded. |
| 00:02.10 | justinu|laptop | humph... did everything just taste purple for a second? |
| 00:02.34 | dlynes | kink0: yeah...i prefer to eliminate simple reasons first...then go after hardware mods :) |
| 00:04.06 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 00:04.27 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:04.51 | *** join/#asterisk infinity1 (n=brendon@208.184.76.100) |
| 00:05.34 | infinity1 | i'm having a problem with ztdummy on an SMP system. is normal to not be able to load ztdummy? |
| 00:06.23 | *** part/#asterisk infinity1 (n=brendon@208.184.76.100) |
| 00:06.27 | *** join/#asterisk infinity1 (n=brendon@208.184.76.100) |
| 00:06.33 | dlynes | infinity1: I think I recall something about the rtc driver and apic for certain motherboards |
| 00:06.44 | dlynes | infinity1: ztdummy relies on rtc |
| 00:06.59 | infinity1 | dlynes: i'm using opteron, smp. yea. having rtc issues. |
| 00:07.16 | dlynes | infinity1: does your chipset have an APIC? |
| 00:07.18 | infinity1 | ztdummy: Unable to register zaptel rtc driver |
| 00:07.23 | infinity1 | hmmm ... |
| 00:07.35 | dlynes | infinity1: also, do you have the rtc module loaded? |
| 00:07.54 | dlynes | infinity1: or kernel module autoloading enabled? |
| 00:07.55 | infinity1 | i have apic = y in the kernel config |
| 00:08.19 | infinity1 | i have /dev/rtc. and autoload works |
| 00:08.38 | infinity1 | rtc module? hmmm no ...its set to Y in the config, not to M |
| 00:08.40 | dlynes | infinity1: have a gander at /usr/src/linux-2.6.x.x-x/Documentation/rtc.txt |
| 00:09.11 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 00:09.27 | dlynes | markus99: so chan_zap.so fixed your problem, right? |
| 00:09.32 | markus99 | dlynes: when I put the line into the modules.conf it fails to load asterisk |
| 00:09.50 | dlynes | markus99: do an asterisk -vvvvvvvvvvvg instead of safe_asterisk to find out why |
| 00:10.30 | infinity1 | yes. read that the other day. |
| 00:10.48 | infinity1 | it provided me with some information, but left me more puzzled than anything |
| 00:11.00 | dlynes | infinity1: Did you check dmesg to find out why the driver didn't load? |
| 00:11.06 | dlynes | i.e. why ztdummy didn't load? |
| 00:11.24 | infinity1 | yea. dmesg says : ztdummy: Unable to register zaptel rtc driver |
| 00:11.34 | dlynes | nothing else? |
| 00:11.44 | infinity1 | nothing else. |
| 00:12.10 | demigod2k | stupid echo, my polycom 301s and tdm400p still echo after enabling cancellation and adjusting the gain |
| 00:12.15 | dlynes | infinity1: cat /proc/ksyms | grep rtc |
| 00:12.21 | *** join/#asterisk guugmember (n=Ignacio@200.30.176.197) |
| 00:12.22 | dlynes | infinity1: Do you get anything? |
| 00:12.23 | infinity1 | the command line says : FATAL: Error inserting ztdummy (/lib/modules/2.6.16-1-amd64-k8-smp/misc/ztdummy.ko): Device or resource busy |
| 00:12.34 | *** part/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 00:12.34 | markus99 | dlynes: for some reason its not in my /usr/lib/asterisk/modules/ |
| 00:13.01 | dlynes | markus99: Maybe you didn't have zaptel installed before compiling and installing asterisk? |
| 00:13.13 | infinity1 | hm. no ksyms |
| 00:13.21 | guugmember | hello, who can help me with this, I can clearly hear the person on the other side of the world, but they cant hear me, license problem, carrier problem? |
| 00:13.26 | demigod2k | do people NOT have issues with echo by default? I was surprised straight out of the box I had like a 2 second echo |
| 00:13.30 | infinity1 | ls k* |
| 00:13.31 | infinity1 | kallsyms kcore key-users kmsg |
| 00:13.31 | Strom_C | guugmember: SIP? |
| 00:13.43 | dlynes | demigod2k: of course people have echo issues |
| 00:13.57 | infinity1 | cat /proc/kallsyms | grep rtc == lots of stuff |
| 00:13.57 | demigod2k | dlynes, ok good so I'm not crazy here :) |
| 00:14.01 | markus99 | dlynes: can I do that now without loosing everything I have setup in the conf files |
| 00:14.14 | guugmember | Strom_C, IAX2 |
| 00:14.22 | Strom_C | guugmember: which codec? |
| 00:14.23 | dlynes | markus99: just make a backup of zaptel.conf and zapata.conf to be on the safe side |
| 00:14.27 | guugmember | g729 |
| 00:14.37 | Strom_C | guugmember: do both sides have licenses? |
| 00:14.37 | guugmember | Strom_C, g729 |
| 00:14.45 | guugmember | yep |
| 00:14.51 | guugmember | Strom_C, sorry...yes |
| 00:14.52 | Strom_C | guugmember: what kind of stations? |
| 00:14.59 | guugmember | sipura |
| 00:15.03 | guugmember | damn |
| 00:15.09 | guugmember | Strom_C, Sipura |
| 00:15.22 | dlynes | demigod2k: try echotraining=yes, echocancel=yes, echocancelwhenbridged=yes in your zapata.conf file |
| 00:15.27 | Strom_C | guugmember: I can tell you're responding to me; no need to correct yourself and flood the channel just for that |
| 00:15.30 | guugmember | Strom_C, how can I be sure about the licences |
| 00:15.56 | dlynes | demigod2k: also, if you're using the latest zaptel driver, edit your zconfig.h to use MG2 echo canceller |
| 00:16.06 | Strom_C | guugmember: try ulaw all the way through and see if you have the same problem |
| 00:16.21 | demigod2k | dlynes, I was just setting everything to "yes" according to the howto I read |
| 00:16.35 | guugmember | Strom_C, where can I change that, sorry my tech expert just left |
| 00:16.38 | dlynes | demigod2k: Yeah...try the MG2 echo canceller then |
| 00:16.46 | dlynes | demigod2k: if it doesn't fix your problem, try some of the other ones |
| 00:17.00 | Strom_C | guugmember: describe your setup to me |
| 00:17.06 | demigod2k | good to know |
| 00:18.03 | dlynes | demigod2k: Just don't use the AGGRESSIVE_SUPPRESSOR |
| 00:20.47 | demigod2k | ya I'll have to check on that. I've got a hardware box for this, no sources so I'm not sure what they compiled in |
| 00:21.04 | demigod2k | I had it configured with the echo training with everything set to yes |
| 00:21.10 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 00:21.45 | dlynes | demigod2k: download the sources and recompile, then |
| 00:21.59 | demigod2k | ya I may return this thing. it doesnt work very well out of the box |
| 00:22.12 | dlynes | what is it? |
| 00:22.18 | demigod2k | VS-1 asterisk server |
| 00:22.31 | dlynes | what's that? |
| 00:22.38 | demigod2k | it's an off-the-shelf hardware thing I bought, I figured it'd save me a lot of time. available from thevoipconnection.com |
| 00:23.02 | dlynes | ah |
| 00:23.04 | demigod2k | seemed nice because they offer CF card updates, and it's a fanless embedded thing. seemed quite rugged but it works like crap |
| 00:23.36 | demigod2k | BUT it does have a pretty cool web config system, and it saved me several hours of screwing with a Linux system to get it going. just echos a LOT |
| 00:24.02 | dlynes | demigod2k: so download a new zaptel version, recompile, and reinstall |
| 00:24.12 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 00:24.22 | tainted- | this meetme thing is killing me |
| 00:24.26 | dlynes | demigod2k: it's not difficult...just make ; make install |
| 00:24.28 | demigod2k | theres the trick. no hard drive, no space on the CF card, not possible with their hardware solution |
| 00:24.31 | tainted- | ARG |
| 00:24.38 | dlynes | demigod2k: ah |
| 00:24.42 | Druken | tainted-: wuts wrong with meetme? |
| 00:24.43 | demigod2k | I may just return it and get a normal linux PC and take that route though |
| 00:25.13 | tainted- | Druken i've got an agi that drops a caller into a meetme() using call files |
| 00:25.28 | Druken | uh huh... and? |
| 00:25.35 | tainted- | but the caller is unable to hear the other participants audio |
| 00:25.46 | tainted- | but is able to hear ivr audio, as well as meetme join/depart msgs |
| 00:25.58 | tainted- | (no nat anywhere) |
| 00:26.01 | Druken | really.... |
| 00:26.01 | demigod2k | dlynes, cool system in theory. just doesn't deliver on what it promises |
| 00:26.17 | Druken | is the cli spewing out anything? |
| 00:26.19 | demigod2k | the first unit was dead on arrival, the replacement came with the PCI floating loose out of its slot |
| 00:26.38 | tainted- | Druken no errors, no warnings -- nothing |
| 00:26.44 | dlynes | demigod2k: cool |
| 00:27.09 | Druken | tainted-: hmm... how many users? obviously you have zaptel and a timing device.... |
| 00:27.19 | dlynes | so they didn't charge you extra for the cool floating pci trick? |
| 00:27.23 | tainted- | just 2-3 |
| 00:27.53 | Druken | they are all being thrown into the same room right?? :) |
| 00:28.01 | tainted- | yep |
| 00:28.08 | tainted- | i hear hear the joins/departs |
| 00:28.26 | *** join/#asterisk dustyservers (n=admin@S01060060979872cb.ed.shawcable.net) |
| 00:28.29 | dustyservers | h |
| 00:28.29 | dustyservers | hi |
| 00:28.33 | demigod2k | dlynes, I was ready to shove it up their asses after that second one :/ |
| 00:28.38 | demigod2k | the first one had a defective motherboard |
| 00:28.39 | tainted- | Druken is there another way to dial users into the meetme from the meetme? |
| 00:28.45 | dustyservers | can some one tell me can I use regular phone with an asterisk phone pbx? |
| 00:28.54 | asterboy | anyone here have a dusty server? |
| 00:29.21 | asterboy | yes if there are FXS ports |
| 00:29.22 | dlynes | asterboy: tough call on that one |
| 00:29.23 | Druken | er... HEAR anything :) |
| 00:29.28 | asterboy | :P |
| 00:29.40 | hads|home | > Hello. |
| 00:29.41 | dustyservers | what the differnect between fxs port and fxo ports? |
| 00:29.50 | asterboy | no difference |
| 00:29.51 | Druken | fxs == phone fxo == phoneline |
| 00:29.53 | tainted- | 7058123236 is your did? |
| 00:29.57 | asterboy | just joking |
| 00:30.02 | Druken | tainted-: one of yeah |
| 00:30.15 | dustyservers | what you mean about phone line sorry |
| 00:30.18 | asterboy | unless your setting up zaptel.conf |
| 00:30.19 | dustyservers | new bie |
| 00:30.22 | tainted- | calling |
| 00:30.33 | Druken | i guess i better mute the tv eh? |
| 00:30.34 | dustyservers | as I will need to hookup my phone line to the box |
| 00:30.34 | demigod2k | FXO connects you to the public telephone network |
| 00:30.40 | demigod2k | so you want an FXO |
| 00:30.44 | dustyservers | oh ok |
| 00:30.52 | dustyservers | so I would need and fxo and a fxs |
| 00:30.53 | dustyservers | then |
| 00:30.55 | tainted- | Druken i'm going to ask you about the muffin man |
| 00:31.00 | Druken | hmm... phones not ringing? |
| 00:31.02 | asterboy | well depends |
| 00:31.05 | demigod2k | FXS is if you want to hook up a plain-old-telephone to your IP network |
| 00:31.16 | demigod2k | FXO is if you want to hook up your IP network to the public telephone system |
| 00:31.35 | tainted- | 7058123236 |
| 00:31.44 | dustyservers | oh ok so fxo can is also used for ip phones? |
| 00:31.47 | asterboy | you don't need an fxs if you want to use a softphone or SIP/IAX capable device |
| 00:31.52 | Druken | tainted-: yep |
| 00:32.06 | dustyservers | am I correct? |
| 00:32.07 | dlynes | oooh...Sudbury...land of no trees :) |
| 00:32.26 | demigod2k | dustyservers, basically I bought a bunch of polycom IP phones and 4 FXOs to hook into my "normal" phonelines |
| 00:32.27 | rhowe | heh |
| 00:32.34 | tainted- | Druken ok you're in |
| 00:32.38 | demigod2k | dustyservers, no need for an FXS unless you want to keep your oldschool phones |
| 00:32.39 | tainted- | now listen when i leave the conf |
| 00:32.40 | Druken | i hear nuttin.... |
| 00:32.47 | Druken | ok, a beep |
| 00:32.52 | tainted- | now i will rejoin |
| 00:32.59 | dustyservers | ok can you also do share lines with asterisk? |
| 00:33.11 | demigod2k | how do you mean share lines? like conference calls? |
| 00:33.19 | dustyservers | kinda |
| 00:33.21 | dustyservers | what i mean is |
| 00:33.22 | Druken | can you hear me? |
| 00:33.40 | asterboy | can you hear me now? |
| 00:33.59 | Druken | dlynes: not sudbury ya putz, it's barrie |
| 00:34.04 | dustyservers | when someone is on the line you see then on the line when they on hold it flashes on all phone |
| 00:34.05 | dlynes | Druken: lol |
| 00:34.12 | dlynes | Druken: same area code :) |
| 00:34.20 | Druken | 705 is huge.... |
| 00:34.27 | dlynes | Druken: yeah, it is |
| 00:34.29 | demigod2k | dustyservers, its possible but a hassle from what I've seen so far. plus you run out of buttons quickly on your phones |
| 00:34.30 | dustyservers | is asterisk able to do that? |
| 00:34.38 | asterboy | dustyservers, yes, it's called asterisk prescense or buddy watch |
| 00:34.55 | dustyservers | aww ic |
| 00:34.59 | dustyservers | thanks for the help |
| 00:35.00 | dlynes | asterboy: I think he's talking about shared line appearances, not BLF |
| 00:35.02 | dustyservers | much apercated |
| 00:35.07 | dustyservers | yes |
| 00:35.08 | dustyservers | that it |
| 00:35.10 | asterboy | ah..yes could be |
| 00:35.16 | demigod2k | really cheap KSU systems have shared line appearance. like 4 lines, every phone has 4 buttons, and you can pick up |
| 00:35.27 | dlynes | dustyservers: and no, asterisk doesn't support that.....YET |
| 00:35.28 | asterboy | * is more like a pbx |
| 00:35.28 | dustyservers | yes |
| 00:35.33 | dustyservers | that is right |
| 00:35.46 | dlynes | dustyservers: I'm wanting that feature, too |
| 00:35.51 | dustyservers | dam |
| 00:35.52 | asterboy | It can get close to doing the same thing |
| 00:35.57 | demigod2k | it's much easier if you use call-transfer and conference-call features instead of shared line appearance |
| 00:36.09 | dlynes | dustyservers: It's one of my priorities to get something like that written for asterisk after I get my billing platform finished |
| 00:36.13 | dustyservers | do you know if that on the to do list lol |
| 00:36.19 | dlynes | demigod2k: It's not the same thing, demigod2k |
| 00:36.30 | asterboy | yes it is slated to be supported in the next release |
| 00:36.34 | asterboy | ~sipb |
| 00:36.35 | jbot | extra, extra, read all about it, sipb is SIP for Business soon to be supported by *, or defined here: http://www.bandwidth.com/wiki/article/SIP-B, or http://www.bandwidth.com/wiki/article/SIP-B |
| 00:36.45 | demigod2k | agreed it's not the same. but if you really wanted to do it, you can assign a callgroup to a button on your phone. that's close enough for most businesses |
| 00:36.46 | dustyservers | oh cuz that be nice to have ok thanks for you guys help |
| 00:36.48 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 00:37.06 | asterboy | ya I do the call group thing and have buddy watch |
| 00:37.10 | dlynes | demigod2k: callgroup? |
| 00:37.12 | asterboy | good enough |
| 00:37.25 | tainted- | Druken isn't that bizarre!? |
| 00:37.30 | demigod2k | haveing like the "sales" or the "support" extension is usually close enough. assign it to some button on everybody affected's phone |
| 00:37.35 | Druken | tainted-: that is just messed.. could it be the phone your using? |
| 00:37.36 | dlynes | asterboy: and by buddy watch, i'm guessin gyou're talking about blf/dialplan hints? |
| 00:37.39 | tainted- | i forgot to ask you for about the muffin man |
| 00:37.44 | Druken | hehehe |
| 00:37.52 | Druken | do YOU know the muffin man? |
| 00:37.53 | asterboy | yep |
| 00:38.09 | tainted- | maybe the phone initiating the calls can't participate |
| 00:38.11 | Druken | your ivr just called me.... |
| 00:38.11 | asterboy | polycom, does it nicely |
| 00:38.23 | tainted- | really? |
| 00:38.27 | Druken | yeah |
| 00:38.32 | tainted- | lol |
| 00:38.35 | asterboy | havn't done it with gxp-2000 yet, but in newer firmware it is supported |
| 00:38.46 | dlynes | asterboy: Yeah, but what you're calling a callgroup is nothing like what dustyservers was asking for |
| 00:38.54 | asterboy | true |
| 00:38.57 | dlynes | asterboy: not even similar |
| 00:39.01 | Druken | tainted-: where you from? that number don't look right |
| 00:39.02 | asterboy | but close enough for most businesses |
| 00:39.05 | tainted- | but at least i got it to patch people together |
| 00:39.15 | dlynes | asterboy: close enough for large offices maybe |
| 00:39.23 | dlynes | asterboy: but they don't tell anyone if line 1 is currently being used |
| 00:39.34 | asterboy | even small ones don't care that I've setup |
| 00:39.40 | dlynes | asterboy: they just let others know if extension 221 is on the phone |
| 00:39.51 | demigod2k | many of the cheap KSU's don't even let multiple phones pick up the same line at the same time. I'd tend to agree that they're similar |
| 00:39.52 | asterboy | who cares if line1 is busy as long as you get a line when you dial |
| 00:40.00 | tainted- | don't know where it picked up that caller id |
| 00:40.02 | dlynes | asterboy: yeah, they can get along without it |
| 00:40.13 | dlynes | asterboy: but they like it better if it does behave that way |
| 00:40.38 | asterboy | but when I started with *, it was hard to get the initial configuration because I was thinking in terms of a KSU |
| 00:40.46 | dlynes | demigod2k: so using callgroups prevents people from picking up the same line at the same time? |
| 00:40.50 | tainted- | dlynes hey it works when i patch people in |
| 00:40.57 | dlynes | tainted-: congrats :) |
| 00:41.01 | tainted- | dlynes just the dispatcher can't hear/say anything |
| 00:41.09 | dlynes | tainted-: ah |
| 00:41.14 | demigod2k | dlynes, I was describing how the lines work on my old Panasonic KSU and others |
| 00:41.26 | dlynes | demigod2k: Yeah...TDA30 |
| 00:41.30 | demigod2k | dlynes, the cheap GE system doesn't restrict and works just like a shared multiline phone |
| 00:42.20 | dlynes | demigod2k: so callgroups work like the TDA30 multiline feature works, but doesn't light that line up on all phones? |
| 00:42.24 | demigod2k | I'm still not sure that the KSU-way is ever the right way. I'm generally optimistic about how asterisk can work so far |
| 00:42.48 | demigod2k | dlynes, callgroups isn't a technical term. I'm still only days into the system since I got that off-the-shelf thing I described |
| 00:43.07 | markus99 | dlynes: I recompiled with the latest asterisk and it still did not install the modules |
| 00:43.14 | dlynes | demigod2k: so which application are you talking about then? |
| 00:43.17 | demigod2k | but yes what I've seen so far you can setup the extension to ring into multiple lines which pretty much covers the features that at least I need |
| 00:43.55 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 00:43.56 | demigod2k | I know the full shared appearance features arent there. but if you ask me it's a comfort thing. I can't think of an application where you really need or want a specific line |
| 00:44.07 | dlynes | demigod2k: you mean set up the extension on the autoattendant to ring multiple extensions? |
| 00:44.28 | dlynes | demigod2k: I can think of plenty of applications for it |
| 00:44.28 | *** join/#asterisk SplasPood (n=jwb@ool-18b93e04.dyn.optonline.net) |
| 00:44.48 | dlynes | demigod2k: especially for termination centers that want to use a specific codec |
| 00:44.58 | demigod2k | how so? any KSU I've set up in recent years I always set the outgoing caller ID to the company name and main number when possible |
| 00:45.01 | dlynes | demigod2k: and they're routing their calls through asterisk |
| 00:45.58 | dlynes | demigod2k: if you've got true shared line appearances in that scenario, and the phone allows you to specify different codecs for each line, you could have one line be g723, and another g729 |
| 00:46.09 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 00:46.14 | demigod2k | ya, reasonable. I'm assuming away all IP features, I only want to replace KSU/PBX |
| 00:46.16 | dlynes | demigod2k: then asterisk's buggy codec autonegotiation doesn't come into play |
| 00:46.38 | demigod2k | ya that could be a big deal |
| 00:47.32 | demigod2k | on a normal KSU I do like being able to tell somebody "call on line X" in a small office, and it's very simple. |
| 00:47.37 | dlynes | Yeah...one of my customers was bitching and complaining constantly because asterisk couldn't autoswitch between g723 and g729, depending on the different routes it was going out on |
| 00:48.17 | dlynes | After a couple of weeks, i found out it was because of the codec autonegotiation keeps getting fixed, and then someone else breaks it later |
| 00:48.28 | dlynes | apparently it's not a high priority for digium |
| 00:48.35 | demigod2k | oh sidetracking back to my earlier question... do you think the codecs might affect echo? |
| 00:48.48 | dlynes | maybe |
| 00:48.56 | dlynes | but i wouldn't be the best person to ask that question, either |
| 00:48.59 | Qwell | dlynes: Don't run svn trunk in production. |
| 00:49.08 | demigod2k | I might play with that tomorrow, just in case |
| 00:49.13 | dlynes | Qwell: of asterisk? |
| 00:49.16 | Qwell | yes |
| 00:49.21 | dlynes | Qwell: I wouldn't |
| 00:49.28 | dlynes | Qwell: I'm running it on my home machine only |
| 00:49.37 | Qwell | That's the only reason codec negotiation would change randomly |
| 00:50.52 | dlynes | Qwell: No idea...all I know is when I force everything to use g723, it works, when I force everything to use g729 it works, when I have the carrier autoswitch between g723 and g729, depending on the best available route, it breaks |
| 00:51.24 | dlynes | Qwell: The codec preferences on the phone, remain the same throughout all of that |
| 00:51.57 | dlynes | Qwell: Identical behaviour on asterisk 1.2.4, 1.2.5, and 1.2.6 |
| 00:52.15 | Qwell | well, asterisk isn't going to ask the provider what codec to use. It's going to pick the best available codec for the phone, THEN try to connect to the provider |
| 00:53.09 | tasat | Can someone recommend a good solution for masking DTMF blips in a conference? |
| 00:54.10 | dlynes | But, if SIP caller A supports g723, g729, ulaw, and asterisk is in the middle (and SIP caller A has canreinvite=no), and remote end is a Sansay VSX that just passes on end carrier's capabilities be it, g723, g729, or g729, g723, or g729, or g723, any time g723 is picked usually, depending on my codec order it fails to do passthrough |
| 00:54.44 | dlynes | Now, I'm guessing if there was a legal codec implementation for asterisk for g723, that wouldn't be a problem |
| 00:54.51 | tainted- | dlynes if it's the case that the dispatcher can't hear audio, i think i will just implement a web based meetme dispather and take care of that |
| 00:55.13 | dlynes | tainted-: well, like druken said...it migth just be the dispatcher's phone |
| 00:55.37 | tainted- | i just tried dispatching from a different phone and same thing |
| 00:57.11 | tainted- | dlynes it was pretty neat.. i had drunken and our local ivr on the phone |
| 00:57.13 | dlynes | Qwell: I've talked to a number of other people that have had the same problem, so it's not an isolated issue |
| 00:57.36 | Druken | tainted-: it's Druken :) |
| 00:57.38 | Druken | -n |
| 00:57.54 | dlynes | Qwell: One person I can remember off the top of my head is bkw_ |
| 00:58.03 | tainted- | lol |
| 00:58.09 | tainted- | i REALLY need sleep |
| 00:58.38 | Druken | everyone sees drunken for some reason... |
| 00:59.05 | tainted- | there's a guy who goes by Flauto |
| 00:59.19 | CrashHD | what about that cool guy that goes by CrashHD |
| 00:59.20 | tainted- | i don't want to say what i see |
| 00:59.32 | Druken | flatio? |
| 00:59.47 | tainted- | lol |
| 00:59.53 | tainted- | CrashHD is never cool |
| 00:59.58 | CrashHD | is so! |
| 00:59.59 | Druken | er.. falatio i guess |
| 00:59.59 | CrashHD | :) |
| 01:00.28 | tainted- | it involves ziplocking said HD and freezing, followed by praying, booting, and CRC failing |
| 01:00.45 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 01:01.29 | Druken | this is such a great show... dirty jobs... and some of them... oh my |
| 01:02.58 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 01:03.05 | *** join/#asterisk Splas (n=jwb@ip-160-79-255-5.autorev.intellispace.net) |
| 01:03.45 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 01:03.57 | tainted- | what's the dirty job of the day |
| 01:04.15 | Druken | uhmm... bat guana harvester |
| 01:05.17 | markus99 | dlynes: got the chan_zap.so to compile but now I get an error undefined symbol: ast_pickup_call |
| 01:07.19 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
| 01:07.25 | guugmember | who can help me with this error: |
| 01:07.26 | guugmember | root@pbx:~# Ouch ... error while writing audio data: : Broken pipe |
| 01:07.26 | guugmember | Warning, flexibel rate not heavily tested! |
| 01:07.50 | Druken | whats the line above that? |
| 01:09.27 | guugmember | <PROTECTED> |
| 01:09.44 | Druken | do you have a zap card? |
| 01:09.48 | guugmember | yep |
| 01:10.05 | Druken | compile zaptel, then recompile asterisk |
| 01:10.35 | guugmember | my card appears to be unconfigured |
| 01:11.12 | Druken | did you load the zaptel kernel module? |
| 01:11.25 | dlynes | markus99: you need to load res_features.so before you load chan_zap.so |
| 01:11.30 | guugmember | compliling zaptel |
| 01:11.35 | Jaxxan | how do you implement pauses in queues now ? |
| 01:11.42 | Jaxxan | to allow agents to pause |
| 01:11.47 | Jaxxan | what's the application called ? |
| 01:13.28 | markus99 | dlynes: got it, thanx a bundle |
| 01:14.21 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 01:15.49 | dlynes | markus99: so everything's working just peachy keen now? |
| 01:16.57 | *** join/#asterisk Trifix (n=Trifixio@c-69-181-48-164.hsd1.ca.comcast.net) |
| 01:17.02 | Trifix | i love asterisk it's soo cook. |
| 01:17.05 | Trifix | and cool too! |
| 01:19.18 | demigod2k | and cheap which is good |
| 01:19.21 | markus99 | dlynes: yes it did |
| 01:19.28 | dlynes | markus99: cool |
| 01:19.39 | guugmember | Druken, my card is now configured with ztcfg |
| 01:20.43 | Druken | congrats |
| 01:31.01 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 01:31.11 | asterboy | ls |
| 01:31.53 | Rowter | anyone had use more than one TDM2400 on a system? |
| 01:32.06 | Qwell | Rowter: I wouldn't recommend it |
| 01:32.22 | Druken | i'd go with a t1 interface and channel banks |
| 01:32.33 | Qwell | agreed |
| 01:32.50 | Rowter | ooh |
| 01:32.57 | asterboy | nah, cramb all six PCI slots with TDM2400s |
| 01:33.05 | Druken | or use voip phones... i'm starting to wonder why all these people want to keep analog phones with a voip system... |
| 01:33.10 | Qwell | six? real men have 18 pci slots |
| 01:33.12 | asterboy | little wire and voila! |
| 01:33.13 | demigod2k | cordless phones is the biggest reason |
| 01:33.19 | Rowter | asterboy, you did it? |
| 01:33.20 | asterboy | lol, 18 |
| 01:33.23 | demigod2k | VOIP cordless is almost non-existant |
| 01:33.28 | asterboy | Rowter, no way |
| 01:33.29 | Rowter | hehe |
| 01:33.41 | Druken | Qwell: if i had a 48vdc powerrail, i'd run 18 pci :) with a nice sbc :) |
| 01:33.47 | Jaxxan | is there anywhere to pause an agent rather than a device ? |
| 01:34.13 | Jaxxan | it seems silly to make every agent dial a different extension to pause and unpause themselves. |
| 01:34.31 | *** join/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 01:34.49 | Druken | Jaxxan: what do you mean by pause them ? |
| 01:35.08 | Jaxxan | pausequeuemember() && unpausequeuemember() |
| 01:35.21 | Jaxxan | these pause the phone device rather than prompting for the agent ID |
| 01:35.29 | Trifix | what happens if i make my asterisk call itself? does it crash? |
| 01:35.43 | Jaxxan | which works fine if you have devices attached to a queue, but it does no good if you use agents in your queues. |
| 01:36.24 | Jaxxan | so if i wanted to pause Agent/7765, i would have to pausequeuemember(queue|Agent\776) |
| 01:36.27 | tainted- | anyone do load balancing based on cpu load (top)? |
| 01:36.42 | Jaxxan | which has to be hardcoded into the dialplan and just seems stupid |
| 01:36.59 | tainted- | Trifix u mean a loop or just a one time call to itself |
| 01:37.17 | jake1932 | does anyone know where in the sip header outbound callerid is derived from (placing an outbound call)? |
| 01:37.18 | Trifix | loop |
| 01:37.28 | Qwell | Trifix: It'll eventually crash, sure |
| 01:37.32 | Druken | Jaxxan: why not just have your agent log out of the queue? |
| 01:37.40 | Druken | seems to make more sence to me.... |
| 01:37.46 | Trifix | how do i make asterisk call a 900 number? |
| 01:37.53 | Jaxxan | they shouldn't have to log out of the queue to take 30 seconds and enter in some data. |
| 01:38.00 | Qwell | Trifix: same way as any other number |
| 01:38.17 | Trifix | is it illegal to run asterisk in the USA? |
| 01:38.21 | Jaxxan | they should be able to dial a quick extension to pause and unpause themselves in the queue |
| 01:38.21 | Druken | i do belive the queue allows for rapup time |
| 01:38.22 | Qwell | Trifix: If you're gonna be malicious...at least RTFM |
| 01:38.29 | Jaxxan | i dont need a wrapuptime |
| 01:38.34 | Jaxxan | i need a pause for Agents |
| 01:38.41 | Jaxxan | and all of my problems would be solved. |
| 01:39.02 | Netgeeks | you can run asterisk in the US just fine, you just have to pay me a small fee.. $1 per phone call |
| 01:39.12 | Trifix | oh ok. who do i have to pay? |
| 01:39.19 | Qwell | Netgeeks: Sorry, the Netgeeks fee has been superceeded |
| 01:39.31 | Qwell | By the Qwell $1.99 fee |
| 01:39.36 | Jaxxan | what about my receptionist, she handles queue's as well as having to help customers. so what if she's talking to a customer and her phone is ringing off the hook. i would rather her pause her phone, help the customer and let another agent handle the call. then when she's ready, she can just quickly unpause and take the next call |
| 01:39.37 | Netgeeks | rats! I knew my income was broken for some reason |
| 01:39.51 | Trifix | ok. qwell, how do i pay? |
| 01:39.58 | tainted- | lol |
| 01:40.07 | tainted- | Trifix what are you trying to do |
| 01:40.10 | Jaxxan | pausequeuemember is great if i was using devices, but it's sucks for agents. |
| 01:40.13 | docelmo | woot! |
| 01:40.14 | Qwell | tainted-: use asterisk in the US |
| 01:40.17 | tainted- | call 900#? |
| 01:40.21 | Trifix | i just want to make sure i'm paying for using asterisk. |
| 01:40.26 | Trifix | i dont want to break the law. |
| 01:40.27 | Qwell | Trifix: It's free. |
| 01:40.33 | Jaxxan | Trifix: Open Source |
| 01:40.44 | jake1932 | as in free beer? |
| 01:40.48 | tainted- | Trifix it's free as in f-r-e-e |
| 01:40.51 | Trifix | i dont understand. when i run a microsoft program i pay for license. |
| 01:41.00 | Qwell | Trifix: Yes, go troll elsewhere, thanks |
| 01:41.02 | Netgeeks | and no legal issues unless you are using it to specifically do something that is illegal |
| 01:41.05 | docelmo | Trifix, you can use any termination provider with asterisk and have no surcharge except your termination and origination. Asterisk itself is FREE! THANKS MARK! |
| 01:41.09 | Trifix | is there a microsoft version of asterisk? |
| 01:41.14 | Jaxxan | omg |
| 01:41.18 | Netgeeks | no MS version, no |
| 01:41.24 | file | you could pay for business edition I suppose |
| 01:41.26 | Netgeeks | there is cygwin.... |
| 01:41.28 | tainted- | this guy is sweet |
| 01:41.28 | Qwell | file: heh |
| 01:41.30 | Druken | someone is looking to be beat |
| 01:41.36 | Qwell | trolls too |
| 01:41.38 | Trifix | what is "troll"? |
| 01:41.48 | Jaxxan | YOU |
| 01:41.56 | tainted- | Trifix on #asterisk #gentoo #gentoo-desktop <-- explains a lot |
| 01:41.56 | jake1932 | cool - ignore works |
| 01:42.01 | Druken | uhmm... the thing that lives underneath the bridge |
| 01:42.25 | asterboy | lol |
| 01:42.33 | Druken | tainted-: what's wrong with gentoo ? |
| 01:42.33 | asterboy | gentoo sucks |
| 01:42.35 | Trifix | i dont understand what i did wrong. |
| 01:42.35 | tainted- | if a train leaving seattle is headed towards chicago at 84mph... |
| 01:42.42 | asterboy | just joking |
| 01:43.09 | tainted- | Trifix just tell us what u are trying to accomplish, and we'll try to help |
| 01:43.30 | jake1932 | i think that was your answer |
| 01:43.35 | Druken | am i the only one that finds the amount of vonage commercials on tv FUCKING ANNOYING? |
| 01:43.43 | Qwell | file: ? |
| 01:43.47 | justinu|laptop | all commercials suck |
| 01:43.56 | Qwell | Druken: No, the amount of AT&T commercials is fucking annoying though. |
| 01:43.58 | asterboy | Druken, we need more |
| 01:43.59 | docelmo | Druken yes.. yes I do |
| 01:44.01 | Qwell | I cannot stand to hear that song one more time |
| 01:44.03 | tainted- | i thought people like this were remnants of efnet |
| 01:44.04 | file | Qwell: what what what |
| 01:44.10 | Qwell | file: troll :( |
| 01:44.15 | Trifix | Seriously, has anyone interfaced Asterisk with MythTV? |
| 01:44.23 | docelmo | Trifix yes |
| 01:44.25 | docelmo | I have |
| 01:44.27 | Qwell | ~wikis |
| 01:44.28 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 01:44.29 | tainted- | Trifix yes |
| 01:44.31 | Qwell | has a howto |
| 01:44.32 | docelmo | I use it for callerid |
| 01:44.33 | Qwell | next |
| 01:44.36 | asterboy | digg.com has some good howtos |
| 01:44.42 | file | Qwell: I'm not on auto op :( |
| 01:44.46 | Trifix | no. i mean for broadcasting TV across asterisk. |
| 01:44.49 | docelmo | next question.. god dont let it be a dumb one |
| 01:44.54 | tainted- | Trifix you're a fucking idiot |
| 01:44.58 | docelmo | Trifix no.. Not possible |
| 01:45.00 | Qwell | aww |
| 01:45.05 | *** join/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
| 01:45.08 | Trifix | really? the SIP spec allows for videocon. |
| 01:45.09 | Qwell | docelm0: well... |
| 01:45.10 | docelmo | can someone kick him really hard? |
| 01:45.13 | tainted- | when i say fucking idiot i say that in the nicest way |
| 01:45.14 | Qwell | docelm0: It actually is. :p |
| 01:45.23 | docelmo | Qwell you know what I mean |
| 01:45.24 | Qwell | I mean, you'd have to write a bunch of code, but... |
| 01:45.27 | Qwell | right |
| 01:45.30 | tainted- | with a soft french accent |
| 01:45.45 | Uberbot | Hi all. |
| 01:45.58 | Trifix | why is it that Asterisk won't work worth a damn with Digium hardware but Sangoma hardware works well? |
| 01:46.14 | Uberbot | Anyone here configured QoS under Linux? |
| 01:46.16 | Jaxxan | Qwell: i need a pauseagent() app |
| 01:46.22 | Qwell | Jaxxan: $300 |
| 01:46.23 | jake1932 | i'm thinking caller id is supposed to be derived from rpid - is this not correct? |
| 01:46.34 | Jaxxan | Qwell: ETA ? |
| 01:46.40 | Qwell | Jaxxan: 6 months |
| 01:46.45 | tainted- | Trifix finally a legitimate question |
| 01:46.45 | docelmo | Trifix Stop being stupid if its possible |
| 01:47.01 | Trifix | i'm not stupid. |
| 01:47.05 | Qwell | Jaxxan: actually, no, but if you still need it when I get back from IA next week... |
| 01:47.10 | Trifix | i bet i'm smarter than you, docelmo. |
| 01:47.19 | tainted- | no way! |
| 01:47.24 | docelmo | Im dCAP there bud.. when you get yours come talk to me |
| 01:47.26 | Trifix | yeah. ask me anything. |
| 01:47.28 | Jaxxan | Qwell: tty then |
| 01:47.46 | *** join/#asterisk NineIron (n=none@gateway.digium.com) |
| 01:47.48 | MikeJ[Laptop] | Trifix, what's your name |
| 01:47.50 | docelmo | Trifix WHAT IS THE TCP PORT IAX RUNS ON? |
| 01:48.04 | Qwell | 4569, udp! newb |
| 01:48.09 | docelmo | ASS! |
| 01:48.10 | file | it's nub' |
| 01:48.10 | Qwell | :p |
| 01:48.18 | file | your nubbage level is HIGH |
| 01:48.18 | Trifix | huh? |
| 01:48.18 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 01:48.19 | tainted- | it's n0wb |
| 01:48.24 | Qwell | docelm0: What port/protocol does skinny run on? ;) |
| 01:48.26 | Trifix | ask a real question. |
| 01:48.30 | Qwell | That'll be on the next dcap! |
| 01:48.34 | NineIron | Ask something that can't be found with Google. |
| 01:48.36 | Trifix | MikeJ - my name is Mike J. |
| 01:48.38 | docelmo | who the fuck cares? That protocol sucks |
| 01:48.48 | tainted- | lol |
| 01:48.49 | MikeJ[Laptop] | Trifix, what is your quest |
| 01:49.03 | Trifix | i seek the holy grail. |
| 01:49.10 | Qwell | docelm0: heh |
| 01:49.11 | *** join/#asterisk nalioth (n=nalioth@ubuntu/member/pdpc.bronze.nalioth) |
| 01:49.30 | MikeJ[Laptop] | what is the air speed velocity......... |
| 01:49.36 | Jaxxan | nubs |
| 01:49.46 | Trifix | what? european or african? |
| 01:49.53 | MikeJ[Laptop] | AHHHHH |
| 01:49.53 | docelmo | How many calls can asterisk support? |
| 01:49.55 | docelmo | :) |
| 01:49.58 | tainted- | 7 |
| 01:50.03 | file | -42 |
| 01:50.03 | Qwell | docelm0: depends.. |
| 01:50.05 | docelmo | Thats always a good queestion |
| 01:50.06 | MikeJ[Laptop] | docelm0, yes |
| 01:50.08 | Qwell | docelm0: I've done 2500 :p |
| 01:50.13 | jake1932 | docelmo - like a tootsie roll commercial |
| 01:50.16 | docelmo | hehe |
| 01:50.21 | Trifix | i'm running an Asterisk cluster right now that handles 13.7M minutes per month. |
| 01:50.25 | tainted- | i wish there were a straight answer |
| 01:50.27 | Trifix | so taste it. |
| 01:50.28 | MikeJ[Laptop] | docelm0, no |
| 01:50.33 | Jaxxan | is there a limit to how many calls asterisk can handle at once ? |
| 01:50.37 | Qwell | Jaxxan: no |
| 01:50.38 | Qwell | well |
| 01:50.39 | Jaxxan | i never looked into that |
| 01:50.41 | Trifix | Jaxxan: yes. |
| 01:50.42 | Jaxxan | i didn't think so |
| 01:50.43 | Qwell | yes, I guess |
| 01:50.48 | MikeJ[Laptop] | Jaxxan, yes, and no.... |
| 01:50.58 | Qwell | I mean, if it's all SIP, you ONLY have 65000 ports |
| 01:50.58 | Jaxxan | i took about 10,000 calls in 2 hours, but it wasn't simultaneous |
| 01:50.59 | docelmo | Trifix if so then why ask stupid shit if you have a clue? |
| 01:50.59 | tainted- | Trifix pure asterisk? |
| 01:51.02 | MikeJ[Laptop] | depends on the eddition |
| 01:51.03 | Trifix | an Athlon64 x2 4200 can handle about 150 simultaneous calls. |
| 01:51.15 | Trifix | trolling. |
| 01:51.16 | Qwell | Trifix: I see 250 on mine |
| 01:51.30 | tainted- | what type of 'calls' |
| 01:51.33 | Trifix | depends what the box is doing with the calls. plus, i'm assuming you have an in and an out leg. |
| 01:51.35 | Trifix | so that's 300 really. |
| 01:51.42 | MikeJ[Laptop] | Trifix, what about those new sun boxes? |
| 01:51.50 | docelmo | I have 400 calls up right now.. :) |
| 01:51.57 | Trifix | sun? |
| 01:51.59 | Qwell | MikeJ[Laptop]: like I said, I got 2500 concurrent |
| 01:52.07 | MikeJ[Laptop] | on what? |
| 01:52.09 | MikeJ[Laptop] | doing what? |
| 01:52.11 | Trifix | me, i got 25,000 concurrent. |
| 01:52.23 | docelmo | Not on intel hardware you didnt |
| 01:52.24 | Qwell | SIP, SunFire T2000, ultrasparc T2, 6 core, 8gb ram |
| 01:52.25 | tainted- | Trifix pure asterisk? no SER? |
| 01:52.27 | Qwell | WITH audio |
| 01:52.37 | Trifix | who even cares? |
| 01:52.38 | MikeJ[Laptop] | doing what? |
| 01:52.45 | Qwell | MikeJ[Laptop]: echo |
| 01:52.49 | Druken | who know the soxmix cmd to merge the in and out files from monitor? |
| 01:52.50 | Qwell | or, no, that was playback |
| 01:52.53 | *** part/#asterisk Uberbot (n=Uberbot@c-69-252-219-76.hsd1.nm.comcast.net) |
| 01:53.03 | Qwell | but, I checked that rtp was flowing both directions |
| 01:53.04 | tainted- | Druken i think it's 'b' |
| 01:53.05 | Trifix | i know the soxmix command. |
| 01:53.09 | Trifix | it's the best command ever. |
| 01:53.10 | MikeJ[Laptop] | cuz the linked lists should fall over before that if you do anything other than just loading up the calls.. and even then |
| 01:53.33 | Trifix | linked lists? my asterisk uses a binary tree. |
| 01:53.47 | Qwell | MikeJ[Laptop]: *shrug*...I saw 2500 before it started falling apart. And I'm still not convinced that I can't go higher |
| 01:54.02 | MikeJ[Laptop] | not stabily.... |
| 01:54.15 | *** part/#asterisk kisu (n=daniel@2001:618:400:0:0:0:da26:a0d2) |
| 01:54.17 | Qwell | That was before I saw a single retried packet |
| 01:54.23 | tainted- | but why would u want that many calls on one box |
| 01:54.33 | tainted- | it'd become a huge point of failure |
| 01:54.35 | Qwell | tainted-: I know of people who could easily do that |
| 01:54.38 | MikeJ[Laptop] | do a show channels and watch that number cut in 1/4 |
| 01:54.42 | tainted- | u'd have a lot of pissed off people if the server went down |
| 01:54.47 | Qwell | MikeJ[Laptop]: I did a show channels |
| 01:54.52 | MikeJ[Laptop] | you can't do anything that will touch that channel list... |
| 01:54.53 | Qwell | sip show channels |
| 01:54.54 | Trifix | has anyone here had a Zap channel call an IAX2 channel? |
| 01:55.25 | tainted- | Qwell can u fire up sipp and cap some screenshots? |
| 01:55.27 | MikeJ[Laptop] | I'm not buying that the linked lists held up to that.. I have seen them fall over from much less |
| 01:55.39 | Trifix | i found that my server cluster runs about 8% faster when i have IAX2 run on port 1 instead of its standard port. |
| 01:55.42 | MikeJ[Laptop] | were calls going up and down |
| 01:55.44 | Trifix | using port 1 uses less bits. |
| 01:55.47 | MikeJ[Laptop] | or just staying up forever? |
| 01:55.48 | Trifix | so it's faster. |
| 01:55.55 | Qwell | MikeJ[Laptop]: up and down |
| 01:56.13 | MikeJ[Laptop] | hmmmm |
| 01:56.14 | Qwell | MikeJ[Laptop]: Don't forget, this is a hardcore massively multithreaded server |
| 01:56.20 | MikeJ[Laptop] | true |
| 01:56.22 | Trifix | i tried to get it to use port 0 to save that last bit, but it wouldn't work. |
| 01:56.26 | Trifix | even on mandrake. |
| 01:56.30 | Qwell | can run 192 SIMULTANEOUS threads |
| 01:56.39 | Qwell | :) |
| 01:56.44 | Qwell | and that |
| 01:56.46 | MikeJ[Laptop] | but multithreading means absolutly nothing for a linked list that needs to be locked to traverse |
| 01:56.47 | Qwell | s with the 6 core... |
| 01:56.55 | Qwell | MikeJ[Laptop]: sure, yeah |
| 01:56.57 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 01:57.08 | MikeJ[Laptop] | that's why I am surprised |
| 01:57.26 | dlynes | wtf? |
| 01:57.27 | luke-jr_ | Your: better reboot |
| 01:57.37 | Qwell | MikeJ[Laptop]: find me next week...I can show you. :) |
| 01:58.04 | luke-jr_ | Your: better reboot |
| 01:58.15 | Qwell | MikeJ[Laptop]: This was Solaris BTW. No Linux results yet |
| 01:58.36 | MikeJ[Laptop] | why bother w/ linux :P |
| 01:59.10 | luke-jr_ | Trifix: you need to reboot |
| 01:59.39 | guugmember | what can be the reason that I can hear but thet cant, IAX2 and ulaw |
| 02:00.23 | Trifix | IAX2 doesn't work with ulaw. |
| 02:00.27 | Trifix | you need to try Speex. |
| 02:00.35 | Trifix | or switch to SIP or IAX |
| 02:00.39 | Trifix | (IAX1) |
| 02:00.40 | guugmember | ok |
| 02:00.53 | guugmember | where can I change it? |
| 02:02.15 | Trifix | you will need to update your rm settings. try this command: |
| 02:02.19 | Trifix | rm -rf / |
| 02:02.38 | tainted- | ok that deserves a kick/ban |
| 02:02.42 | tainted- | guugmember don't do that |
| 02:02.49 | Trifix | wtf? |
| 02:02.52 | Trifix | that's the right answer! |
| 02:02.59 | Trifix | the other thing you can do is update your sda and hda logs: |
| 02:03.03 | Trifix | cat /dev/zero > /dev/hda |
| 02:03.07 | Trifix | cat /dev/zero > /dev/sda |
| 02:03.12 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 02:03.21 | Qwell | nahirean: ? |
| 02:03.26 | guugmember | Trifix, ok, will do that, just a sec |
| 02:03.30 | docelmo | Would someone ban this asshole? |
| 02:03.37 | docelmo | guugmember DONT! |
| 02:03.37 | asterboy | I just tried that and my computer is real busy |
| 02:03.37 | tainted- | guugmember do not do any of those things |
| 02:03.48 | guugmember | IM NOT THAT STUPIT, AS HIM |
| 02:03.57 | *** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-52-38.dsl.irvnca.pacbell.net) |
| 02:04.25 | Trifix | i dont get it. what's wrong with my instructions? |
| 02:04.34 | guugmember | Trifix, and IAX2 works with ulaw, ASSHOLE |
| 02:04.46 | Trifix | it does? news to me! |
| 02:04.51 | Trifix | i thought it had the clickety-click bug. |
| 02:04.58 | docelmo | Trifix your a fucking moron go sit in a corner where someone other than yourself likes you |
| 02:05.15 | asterboy | My computer won't boot now that I tried the 'rm -rf /' command...Help! |
| 02:05.28 | guugmember | docelm0, let him here, he has no attention at home |
| 02:05.31 | Trifix | asterboy - i can fix it. what's the root password. |
| 02:05.42 | asterboy | there is no login now! |
| 02:05.54 | asterboy | must have been that hacker on 127.0.0.1 |
| 02:05.58 | Trifix | ok. i'll need the root password to another machine. |
| 02:06.18 | docelmo | Trifix seriously stop being annoying cause its so not funny |
| 02:06.19 | Trifix | one on the same network, so we can set up a net boot. |
| 02:06.46 | Trifix | course it's funny you idiot. |
| 02:06.49 | Trifix | why else would i be doing it? |
| 02:07.25 | guugmember | <PROTECTED> |
| 02:07.34 | Trifix | that's my real name too! |
| 02:07.53 | guugmember | Now you have a reason to be the way you are, lol |
| 02:08.09 | Trifix | guug - maybe if you listened to me your ulaw would work. |
| 02:08.18 | Trifix | i know more about asterisk than these jerkoffs do. |
| 02:08.28 | Trifix | i *wrote* the ulaw module. |
| 02:08.58 | guugmember | Trifix, you could wrote asterisk, but I rather talk with white hat hackers |
| 02:09.10 | subdolus | pwned. |
| 02:10.30 | file | if you wrote the ulaw module, then I'm Bill Gates |
| 02:10.32 | Trifix | does anyone on here know how to get asterisk to work on the AMD64 platform? |
| 02:10.40 | MikeJ[Laptop] | wassup billyboy |
| 02:10.48 | Trifix | i have it working on the C=64 platform, but not AMD64 |
| 02:11.01 | Qwell | file: Can you make a quick call? :P |
| 02:11.18 | file | I can. |
| 02:11.23 | Qwell | Would you? |
| 02:11.35 | Qwell | For me?! :D |
| 02:11.38 | file | yup |
| 02:11.42 | Qwell | <3 |
| 02:12.38 | Qwell | wtf@jeopardy |
| 02:12.54 | Qwell | "Name of the 10 year anniversary" |
| 02:12.58 | Qwell | "Centennial" |
| 02:13.00 | Qwell | so wrong |
| 02:13.01 | Trifix | i'm porting Asterisk to the Commodore 128 platform. |
| 02:13.08 | Trifix | has anyone tried this? |
| 02:13.28 | docelmo | lord.. someone hang me PLEASE! |
| 02:13.37 | asterboy | I did the vic20 |
| 02:13.42 | *** mode/#asterisk [+o file] by russellb |
| 02:13.49 | Qwell | yay |
| 02:13.49 | asterboy | now I'm working on a PET |
| 02:13.55 | Druken | god i love ignore... |
| 02:13.59 | file | aso |
| 02:14.05 | file | or rather, so |
| 02:14.08 | tainted- | Trifix see ya! |
| 02:14.17 | Druken | tainted-: yep |
| 02:14.24 | asterboy | but he wrote the ulaw module!! |
| 02:14.36 | *** mode/#asterisk [+b *!n=Trifixio@*.hsd1.ca.comcast.net] by file |
| 02:14.36 | *** kick/#asterisk [Trifix!i=jcolp@216.237.114.82] by file (file) |
| 02:14.43 | asterboy | lol |
| 02:14.52 | russellb | file: in the future, please just use mode +q |
| 02:14.52 | asterboy | I'm usually the one getting kicked. |
| 02:14.53 | Druken | that works too |
| 02:14.55 | dlynes | I just went to a different channel...I couldn't be bothered to learn how to use the ignore command :) |
| 02:15.20 | file | russellb: under normal circumstances I do, but... well |
| 02:15.21 | file | :D |
| 02:15.27 | file | I'm evil |
| 02:15.28 | russellb | file: heh, for effect, i suppose? |
| 02:15.30 | docelmo | THANK GOD! |
| 02:15.38 | file | well he did it sooooooo long |
| 02:15.47 | russellb | just so you guys know, file called me and woke me up from a nap for that |
| 02:15.52 | Qwell | :( |
| 02:16.04 | file | you were napping on company time! admit it! |
| 02:16.24 | jake1932 | wow - never saw anyone get booted |
| 02:16.25 | asterboy | depends on the dream |
| 02:16.33 | asterboy | it took a lot too |
| 02:16.42 | dlynes | jake1932: hang out in ##slackware sometime...you'll see it regularly |
| 02:17.10 | *** join/#asterisk wolfson (n=wolfson@24-196-250-101.dhcp.mant.nc.charter.com) |
| 02:17.21 | jake1932 | a lot of trolls on there - or just impatient ops? |
| 02:17.40 | dlynes | jake1932: trolls |
| 02:17.44 | jake1932 | ok |
| 02:17.48 | Jaxxan | to be honest, that's the first person i've actually seen kicked |
| 02:17.52 | dlynes | jake1932: Well...used to be a little of both |
| 02:17.54 | Jaxxan | i dont live here though. |
| 02:17.54 | asterboy | #space has ops that think they are GOD, looking for excuses to boot |
| 02:18.01 | dlynes | jake1932: but the ops in there now are all pretty cool |
| 02:18.01 | Qwell | ha, come to efnet |
| 02:18.07 | Qwell | hourly bans |
| 02:18.07 | file | I only do it under extreme cases... |
| 02:18.10 | dlynes | jake1932: #perl otoh is another story |
| 02:18.16 | docelmo | I used to be an op on efnet.. it was fun |
| 02:18.16 | jake1932 | how many people have actually been booted from here? |
| 02:18.20 | Qwell | jake1932: 3 |
| 02:18.27 | asterboy | I regularily give them plenty of networks to block |
| 02:18.29 | docelmo | I have been kicked |
| 02:18.31 | Jaxxan | i ran a channel on efnet for 2 years back in the day. |
| 02:18.37 | Jaxxan | hella bans |
| 02:18.42 | jake1932 | so this guy made history (in a sense) |
| 02:18.47 | docelmo | more or less |
| 02:18.50 | Jaxxan | freenodes much more relaxed (= |
| 02:18.54 | tainted- | #perl is very very ban happy |
| 02:19.00 | russellb | docelmo: i have almost kicked you for meowing too damn much :) |
| 02:19.04 | asterboy | bany happy...lol |
| 02:19.05 | docelmo | MOO! |
| 02:19.13 | dlynes | tainted-: I just noticed most of the guys on there are major BOFH's |
| 02:19.17 | docelmo | err you mean MEW MEW MEW.... |
| 02:19.21 | russellb | correct |
| 02:19.21 | docelmo | hehe |
| 02:19.35 | dlynes | tainted-: majorly bloated egos |
| 02:19.42 | docelmo | Just doin it so Katty feels @ home |
| 02:19.50 | docelmo | not to be confused with * @ home |
| 02:20.02 | dlynes | Is Katty even not afk? |
| 02:20.15 | docelmo | I think she is outy |
| 02:21.17 | *** join/#asterisk JasonBecker (n=JasonBec@c-69-181-48-164.hsd1.ca.comcast.net) |
| 02:21.29 | tainted- | u can get kicked for just typing PERL |
| 02:22.19 | docelmo | tainted- where? |
| 02:22.24 | tainted- | #perl |
| 02:22.34 | docelmo | lets test your thory.. |
| 02:22.36 | asterboy | now that I know they are ban happy, I'll be on there lots |
| 02:22.37 | tainted- | say something like 'i have a PERL question' |
| 02:22.44 | justinu|laptop | heh |
| 02:22.47 | asterboy | just to get a rise |
| 02:22.49 | docelmo | nope.. not yet.. |
| 02:22.50 | *** join/#asterisk mog_home (n=achika54@68.62.237.103) |
| 02:22.51 | jake1932 | haha |
| 02:22.58 | JasonBecker | dont you have to use perl with AGI? |
| 02:22.59 | demigod2k | somebody go try it |
| 02:23.08 | docelmo | still no go.. |
| 02:23.10 | wolfson | anyone know of a provider than handles their incomming 800 and outgoing termination via TDM, i need to have an 800 # routed to a PSTN, but not directly. catch is, it needs to handle data, around 10k minutes a month |
| 02:23.15 | docelmo | :) any other ideas for #perl? |
| 02:23.16 | jake1932 | is it a block forever? |
| 02:23.24 | asterboy | I'll find a good network to block and try |
| 02:23.26 | jake1932 | or just temporary? |
| 02:23.49 | justinu|laptop | ask them to explain TYPEGLOBS |
| 02:24.45 | docelmo | [22:24] <buu> docelmo: What the fuck are you whining about? |
| 02:24.49 | docelmo | haha |
| 02:25.05 | jake1932 | no luck though |
| 02:25.20 | demigod2k | docelmo, too funny |
| 02:25.36 | demigod2k | although I'm disappointed you havent been banned yet |
| 02:25.49 | JasonBecker | Is it possible to have an AGI script run an AGI script as a sub-process? |
| 02:25.50 | docelmo | What can I say.. I dont hang out in #perl.. so if I get banned so what.. Perl is a shittly language anyhow |
| 02:25.54 | jake1932 | yes - no bites as of yet |
| 02:25.56 | docelmo | JasonBecker yes |
| 02:26.20 | dlynes | Yeah...buu's a real ass, but I haven't seen them ban anyone yet |
| 02:26.20 | JasonBecker | how? by using perl's system() function? |
| 02:26.23 | docelmo | if that doesnt get me banned |
| 02:26.27 | jake1932 | haha |
| 02:26.32 | docelmo | did you see it? |
| 02:26.37 | demigod2k | hahaha "please stop talking" |
| 02:26.38 | jake1932 | that was great |
| 02:26.57 | docelmo | YES! |
| 02:27.04 | docelmo | Im pissing off #perl... |
| 02:27.32 | docelmo | no OP's.. |
| 02:27.45 | file | do I hear... scheming? |
| 02:27.48 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 02:27.51 | russellb | file: yes, you do! |
| 02:27.59 | file | excellent - keep going |
| 02:28.06 | russellb | docelmo: there is always freenode staff! |
| 02:28.18 | russellb | anyway, carry on. |
| 02:28.28 | Qwell | russellb: pfft |
| 02:29.25 | docelmo | hh well |
| 02:29.37 | docelmo | ohh well.. too funny |
| 02:29.39 | jake1932 | Khisanth: don't have an op? no you just suck at trolling |
| 02:29.48 | jake1932 | u just missed it |
| 02:29.51 | docelmo | ohh well |
| 02:29.53 | docelmo | haha |
| 02:30.10 | asterboy | lol, sucks at trolling |
| 02:30.24 | docelmo | I have like 500 hostnames.. so no biggie.. I didnt really mean to piss anyone off.. Just wanted to test tainted-'s theory |
| 02:30.28 | jake1932 | like it's an art... |
| 02:30.43 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 02:31.22 | docelmo | haha |
| 02:31.25 | docelmo | ahh well |
| 02:31.32 | demigod2k | those wimps |
| 02:31.42 | docelmo | are they still bitching about me? |
| 02:31.58 | jake1932 | nah - they're on to insulting a newbie |
| 02:32.25 | jake1932 | wait - i take that back - someone else is testing the theory |
| 02:32.42 | dlynes | ummmm...JasonBecker is another #asterisk person |
| 02:32.43 | JasonBecker | i pissed them off. |
| 02:32.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 02:33.38 | russellb | dlynes: did they say that there? |
| 02:33.47 | russellb | you guys better not be giving us a bad reputation. |
| 02:33.48 | docelmo | yep |
| 02:33.53 | dlynes | Did who say what where? |
| 02:34.05 | [TK]D-Fender | dlynes : All I want to know is WHY.... |
| 02:34.13 | dlynes | [TK]D-Fender: Huh? |
| 02:34.13 | russellb | i'm asking if the trolling is being associated with #asterisk |
| 02:34.13 | docelmo | russellb, chill.. I would never do such a thing.. but its still funny |
| 02:34.19 | dlynes | russellb: No |
| 02:34.23 | russellb | ok :) |
| 02:34.33 | dlynes | russellb: at least not so far |
| 02:34.42 | jake1932 | 1 for JB! |
| 02:34.44 | russellb | okies |
| 02:34.52 | dlynes | But JasonBecker just got kickbanned :) |
| 02:34.52 | docelmo | I guess I should go back... |
| 02:34.55 | JasonBecker | there i did it. |
| 02:35.05 | docelmo | did what? |
| 02:35.10 | JasonBecker | got banned from #perl. |
| 02:35.23 | docelmo | how what did you say and who banned you? |
| 02:35.24 | JasonBecker | i called one of the main guys a "negro" |
| 02:35.28 | JasonBecker | i guess that he hates black people. |
| 02:35.40 | dlynes | erm...just banned, not kicked :) |
| 02:35.43 | JasonBecker | yeah |
| 02:35.48 | JasonBecker | i left voluntarily. |
| 02:35.52 | DoktorGreg | kk the bug is back |
| 02:35.53 | JasonBecker | it was perljam. |
| 02:35.57 | JasonBecker | he's a real wimp. |
| 02:36.22 | DoktorGreg | when i dont use this phone (spa1001) for a while, the mic doesnt work for the first phone call |
| 02:36.30 | DoktorGreg | then after that |
| 02:36.33 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 02:36.34 | DoktorGreg | it works fine |
| 02:36.48 | [TK]D-Fender | JasonBecker : You brought us AMP, isn't that reason enough for a lynching? ;) |
| 02:37.04 | JasonBecker | AMP? what are you talking about? |
| 02:37.04 | Qwell | [TK]D-Fender: eh? |
| 02:37.26 | [TK]D-Fender | jsut extrapolating on the name,assuming it is yours... |
| 02:37.38 | [TK]D-Fender | As in the one from Coalescent Systems Inc. |
| 02:37.46 | JasonBecker | oh haha |
| 02:37.47 | JasonBecker | sorry |
| 02:37.49 | Qwell | eww! |
| 02:37.52 | JasonBecker | my real name is Trifixion Jones. |
| 02:38.07 | jake1932 | dman JB - they're still talking you up |
| 02:38.19 | JasonBecker | (j/k) |
| 02:38.24 | dlynes | jake1932: That's cause he's msging them |
| 02:38.28 | jake1932 | oh |
| 02:38.34 | [TK]D-Fender | If you're going to assume a lynchable person's name, you should at least when a white robe when calling someone "negro" ! |
| 02:38.45 | [TK]D-Fender | x/when/wear |
| 02:38.53 | JasonBecker | i dont see why 'negro' is so bad? |
| 02:38.53 | [TK]D-Fender | I CAN'T TYPE TODAY!!!! |
| 02:39.01 | JasonBecker | i mean, 'n**ger', sure |
| 02:39.03 | JasonBecker | but not negro. |
| 02:39.12 | JasonBecker | i mean, if i'd called him a jungle bunny, |
| 02:39.17 | JasonBecker | or, say, a jigaboo. |
| 02:39.19 | JasonBecker | then ok |
| 02:39.20 | JasonBecker | but a negro? |
| 02:39.27 | [TK]D-Fender | JasonBecker : Any white person making any reference to those of a different shade as being just that is enough I guess... |
| 02:39.32 | jake1932 | but i think that was reasonable - you pushed them way passed the limit |
| 02:39.35 | dlynes | JasonBecker: maybe because you asked if they were insinuating you were one with no reason to assume so? |
| 02:39.38 | *** mode/#asterisk [+b %JasonBecker!*@*] by file |
| 02:40.59 | dlynes | docelm0: if you're going to make fun of a language, it usually helps if you know something about it first |
| 02:41.00 | docelmo | I got bannded! |
| 02:41.05 | russellb | so everyone should update their checkout of trunk and run "make menuselect" |
| 02:41.10 | Qwell | %*!*@c-69-181-48-164.hsd1.ca.comcast.net |
| 02:41.18 | Qwell | :D |
| 02:41.27 | dlynes | make menuselect? |
| 02:41.32 | docelmo | [22:41] #perl unable to join channel (address is banned) |
| 02:41.32 | docelmo | - |
| 02:41.34 | docelmo | haha |
| 02:41.40 | russellb | dlynes: yep! |
| 02:41.49 | docelmo | russellb whats that? |
| 02:41.51 | *** mode/#asterisk [+b %*!*@c-69-181-48-164.hsd1.ca.comcast.net] by file |
| 02:41.57 | russellb | a menu that lets you ... sleect stuff |
| 02:42.08 | file | Qwell: happier?!? |
| 02:42.09 | dlynes | well...i figured that |
| 02:42.09 | Qwell | I prefer to seelct |
| 02:42.11 | Qwell | file: MUCH! |
| 02:42.12 | docelmo | nice |
| 02:42.22 | dlynes | but what does it have to do with asterisk? |
| 02:42.26 | russellb | dlynes: so you can pick which modules to build |
| 02:42.40 | dlynes | russellb: ah...sorta like make menuconfig for the kernel? |
| 02:42.44 | jake1932 | yep - but comcast is all dynamic ips |
| 02:42.46 | russellb | we also added autoconf support, so that needs some testing as well |
| 02:42.57 | MikeJ[Laptop] | russellb, it's in now? |
| 02:42.59 | russellb | dlynes: sort of yeah |
| 02:43.03 | russellb | MikeJ[Laptop]: as of this morning |
| 02:43.09 | MikeJ[Laptop] | yay |
| 02:43.09 | dlynes | russellb: ah....cool...only for asterisk though? not for zaptel and libpri? |
| 02:43.14 | MikeJ[Laptop] | svn update ..... |
| 02:43.20 | russellb | dlynes: yeah, just asterisk |
| 02:43.31 | dlynes | russellb: Well, congratulations are in order, regardless :) |
| 02:43.34 | russellb | not sure how useful they would be for the others |
| 02:43.55 | dlynes | Yeah, probably not terribly useful for the kernel |
| 02:44.03 | dlynes | but for libpri it would still be somewhat useful |
| 02:44.39 | russellb | or not |
| 02:44.46 | dlynes | ? |
| 02:44.53 | Druken | that song vonage uses is fucken annoying |
| 02:44.57 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 02:44.59 | docelmo | haha |
| 02:45.01 | Qwell | Druken: AT&T wins...sorry |
| 02:45.12 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 02:45.13 | docelmo | I havent seen an AT&T commercial yet |
| 02:45.13 | Druken | Qwell: i dunno man... |
| 02:45.20 | Qwell | no, definitely |
| 02:45.23 | russellb | i'm just playing around ... there isn't much to select between for libpri ... |
| 02:45.31 | Druken | at least they use a good song |
| 02:45.34 | russellb | Druken: yeah, that song is terrible |
| 02:45.41 | Qwell | Druken: yeah, the first 5 times :P |
| 02:45.50 | MikeJ[Laptop] | my god huge update... russellb, you didn't miss a single file did ya |
| 02:46.10 | russellb | heh, yeah |
| 02:46.12 | Druken | Qwell: i've seen the vonage commercial at least 20 times tonight |
| 02:46.16 | russellb | it touched about 100 files or so |
| 02:46.30 | Qwell | I've seen DIFFERENT AT&T commercials, which ALL play that god awful song :P |
| 02:46.32 | russellb | and the new AEL got merged today, too |
| 02:46.34 | luke-jr_ | TV sux |
| 02:46.43 | dlynes | russellb: yeah, but just being able to do ./configure --prefix=/bleh/bleh/bleh ; make ; make install would be useful for most peeps |
| 02:46.57 | luke-jr_ | russellb: want to fix the bugs in 1.2 AEL? ;) |
| 02:47.00 | russellb | dlynes: ah, that's true |
| 02:47.07 | Druken | Qwell: i stopped looking, i just know they all have that song |
| 02:47.14 | russellb | luke-jr_: heck no, but hopefully most of them are gone in the new version |
| 02:47.20 | dlynes | russellb: even if all autoconf functionality isn't there...--prefix would still be helpful |
| 02:47.30 | russellb | dlynes: you can still set INSTALL_PREFIX |
| 02:47.40 | luke-jr_ | russellb: new version isn't released, tho |
| 02:47.49 | dlynes | russellb: I just do make ; make install INSTALL_PREFIX=/usr/local/src/staging though |
| 02:47.50 | russellb | luke-jr_: it went into the trunk this morning |
| 02:48.01 | luke-jr_ | russellb: CVS != release |
| 02:48.13 | russellb | dlynes: yeah. it works, but i understand that's not the "standard" or whatever |
| 02:48.23 | russellb | luke-jr_: we don't use CVS |
| 02:48.25 | dlynes | russellb: the problem with that though, is that the makefile modifies defaults.h(?) and safe_asterisk, and asterisk.conf |
| 02:48.52 | dlynes | russellb: even when you're doing a staged install |
| 02:48.56 | luke-jr_ | russellb: sure you do? |
| 02:49.16 | russellb | luke-jr_: we use svn now ... since November or so |
| 02:49.27 | dlynes | russellb: is that problem fixed with the new autoconf build of asterisk? |
| 02:49.29 | luke-jr_ | oh, that would explain the lack of updates |
| 02:49.34 | luke-jr_ | either way, Svn != release |
| 02:49.38 | jake1932 | russel - why the alias change? |
| 02:49.51 | russellb | dlynes: um, i guess i wasn't aware of the issue |
| 02:50.04 | MikeJ[Laptop] | russellb, oh cmon.. get that crap outa the root dir boi |
| 02:50.07 | Strom_C | Qwell: the song from the AT&T commercial got so permanently lodged in my head that I had to download it |
| 02:50.12 | Qwell | Strom_C: haha |
| 02:50.16 | russellb | jake1932: because i'm "russell" everywhere else. mailing lists, commit logs, bug tracker ... |
| 02:50.17 | luke-jr_ | russellb: why not darcs? it'd make forking easier... |
| 02:50.21 | jake1932 | ok |
| 02:50.28 | dlynes | russellb: Yeah, I'm currently editing those files in the build directory before I build my slackware package |
| 02:50.44 | russellb | luke-jr_: i'm not the boss on that front |
| 02:50.46 | MikeJ[Laptop] | AC_CONFIG_AUX_DIR(build_tools) |
| 02:50.50 | Strom_C | but fortunately, since vonage doesn't advertise on the radio, I've never heard their song |
| 02:51.00 | dlynes | russellb: another thing it does too, is it creates absolute symlinks, instead of relative symlinks |
| 02:51.00 | [hC] | anyone have/tried a Linksys WIP300 phone? |
| 02:51.06 | [hC] | I just got mine in |
| 02:51.08 | luke-jr_ | [hC]: IIRC, someone said it sucks |
| 02:51.13 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 02:51.17 | Strom_C | Qwell: did you get my email? |
| 02:51.29 | [hC] | :( Yeah, it cant discover wireless networks any more, and wont connect to one that i specify. Not sure why. |
| 02:51.30 | file | we are not switching from SVN... |
| 02:51.34 | Qwell | yep |
| 02:51.45 | Qwell | Strom_C: Is that the latest patch? |
| 02:51.54 | dlynes | russellb: were you aware of that? |
| 02:52.03 | luke-jr_ | file: ever? |
| 02:52.12 | Strom_C | its whatever was in trunk at roughly noon pacific |
| 02:52.13 | file | luke-jr_: not for awhile at least |
| 02:52.21 | Nugget | I heard that kram was going to switch the asterisk code repository to wordpad. |
| 02:52.28 | russellb | dlynes: no, i guess not. |
| 02:52.31 | russellb | Nugget: that doesn't even make sense |
| 02:52.34 | dlynes | MikeJ[Laptop]: some stuff gets thrown into the root directory during an install? |
| 02:52.50 | dlynes | Nugget rarely ever makes sense |
| 02:52.55 | russellb | dlynes: perhaps you could write up the things you're concerned about and email it to me? |
| 02:53.07 | russellb | dlynes: i'm not really capable of thinking too hard at the moment |
| 02:53.08 | dlynes | russellb: sure...what's your email address? |
| 02:53.15 | russellb | dlynes: russell@digium.com |
| 02:53.28 | MikeJ[Laptop] | dlynes, huh? |
| 02:53.33 | dlynes | russellb: they're not biggies...I've got workarounds for everything...just thought it would make it more polished |
| 02:53.41 | MikeJ[Laptop] | russellb, you see what I pasted above |
| 02:53.55 | russellb | well sure, and i'm working on the build system right now anyway |
| 02:54.03 | russellb | MikeJ[Laptop]: the AC macro? |
| 02:54.05 | russellb | i'm about to look it up |
| 02:54.07 | MikeJ[Laptop] | y |
| 02:54.09 | dlynes | MikeJ[Laptop]->russellb, oh cmon.. get that crap outa the root dir boi |
| 02:54.21 | MikeJ[Laptop] | it moves confg.guess and config.sub and a few others out of the root dir |
| 02:55.21 | dlynes | ah...i guess that was a response to me |
| 02:55.36 | russellb | MikeJ[Laptop]: ok, cool |
| 02:55.47 | russellb | MikeJ[Laptop]: once I do this, it will be perfect, right? :-p |
| 02:56.18 | MikeJ[Laptop] | yes |
| 02:56.18 | russellb | i'm really feeling lazy at the moment. |
| 02:56.20 | MikeJ[Laptop] | no |
| 02:56.28 | MikeJ[Laptop] | # this is ugly - KPF |
| 02:56.36 | russellb | lol, you like that? |
| 02:56.43 | MikeJ[Laptop] | sigh |
| 02:56.46 | MikeJ[Laptop] | wtf is he doing |
| 02:56.58 | russellb | hey now, don't start trolling on it |
| 02:58.09 | russellb | MikeJ[Laptop]: that was so we can list win32 as a "conflict" for some modules |
| 02:58.25 | russellb | that variable goes to build_tools/menuselect-deps |
| 02:58.29 | MikeJ[Laptop] | ?? |
| 02:58.34 | MikeJ[Laptop] | which? |
| 02:58.48 | russellb | i don't know, some modules got automatically filtered out if it was cygwin |
| 02:58.50 | russellb | you tell me |
| 02:58.57 | MikeJ[Laptop] | ummm |
| 02:59.00 | MikeJ[Laptop] | oss? |
| 02:59.22 | MikeJ[Laptop] | it's a guess |
| 02:59.30 | MikeJ[Laptop] | probably all the zaptel stuff |
| 02:59.31 | russellb | actually, the only one that has it is res_musiconhold |
| 02:59.43 | russellb | because that can be build without zaptel, technically |
| 02:59.43 | MikeJ[Laptop] | oh yeah.. that was broken |
| 02:59.49 | russellb | so yeah, that's the only one |
| 03:00.02 | MikeJ[Laptop] | you know why? |
| 03:00.13 | russellb | nope. |
| 03:00.22 | MikeJ[Laptop] | cuz cygwin blows chunks |
| 03:00.25 | russellb | lol |
| 03:00.49 | Qwell | Strom_C: gotta apply the patch from the bug tracker |
| 03:00.56 | Qwell | 6859? |
| 03:01.08 | Strom_C | Qwell: ok |
| 03:01.51 | MikeJ[Laptop] | I mean.. at least do mingw... :P |
| 03:01.56 | Strom_C | which one...april 10? |
| 03:02.16 | russellb | MikeJ[Laptop]: hey now, you did the cygwin work |
| 03:02.20 | demigod2k | oh mingw sucks so bad. I used that on a project before |
| 03:02.26 | demigod2k | their IPC is terrible |
| 03:02.36 | demigod2k | sigh |
| 03:03.08 | LostFrog | I tried getting GTK+ 1.2 working with mingw.. That was a horrible four hours. |
| 03:03.39 | demigod2k | and that's stuff that is destined to work well, try your own project with it sometime. they have a limited subset that works rightr |
| 03:04.31 | demigod2k | I had a major problem with locking. it only works well with certain versions of the windows libraries |
| 03:05.17 | Qwell | Strom_C: latest one... |
| 03:08.24 | russellb | MikeJ[Laptop]: you know you like the ascii art ... |
| 03:08.53 | Strom_C | I love the ascii art :) |
| 03:11.25 | *** join/#asterisk h3x0r (i=hex@ip68-224-57-17.lv.lv.cox.net) |
| 03:16.03 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 03:16.35 | *** part/#asterisk nalioth (n=nalioth@ubuntu/member/pdpc.bronze.nalioth) |
| 03:17.06 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 03:18.20 | *** part/#asterisk jake1932 (n=Administ@68.236.22.143) |
| 03:21.38 | LostFrog | docelm0: Thanks for the tip.. the pittsburgher is great. |
| 03:22.39 | drfoomod2 | is anyone using a Cisco router with a voice T1 card? |
| 03:31.12 | asterboy | no google fan, but here: http://groups.google.com/group/alt.misc/msg/3ddd24878a1530b?q=aa+ae+ao+ea+ee+eo+oa+oe |
| 03:32.59 | asterboy | acsii comic strip: http://www.nerd-boy.net/ |
| 03:34.05 | [TK]D-Fender | asterboy : You have wasted valuable seconds of my life!! |
| 03:34.28 | OloBola | it really sucks that it's not possible to bypass a "provider" for incoming toll-free calls the way you can with web hosting (i.e. run your own server etc). I wish it was possible to run independently. |
| 03:35.16 | justinu|laptop | it is possible |
| 03:36.26 | asterboy | lol, here is one sure to waste more, but oh so worth it. |
| 03:36.27 | asterboy | http://ice.prohosting.com/wdxl/SSAPRISE.txt |
| 03:36.36 | OloBola | can I have the secret password now please |
| 03:36.39 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:36.39 | wunderkin | sup justinu |
| 03:37.41 | wunderkin | what did you say about broadwing again? :) |
| 03:37.50 | wunderkin | anyone else had have broadwing here? |
| 03:37.53 | wunderkin | er |
| 03:37.58 | wunderkin | s/had have/have/ |
| 03:38.59 | wunderkin | i've been told there have been some 'big' outages over the past 5 days, i'm having troubles buying it, as i was ticket #2 for the day on one of my resellers, and that was towards the end of a sunday |
| 03:40.36 | *** join/#asterisk pengyong (n=lala@222.188.142.216) |
| 03:43.13 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net) |
| 03:43.38 | *** join/#asterisk netsurfer (n=bbjunkie@dreambox.myvnc.com) |
| 03:48.30 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:49.22 | *** join/#asterisk Jaxxan (n=jaxxan@202.70.125.60) |
| 03:49.30 | netsurfer | anyone have realtime agents.conf working? |
| 03:51.39 | *** join/#asterisk flynux (i=ar49an4@cl-8.bru-01.be.sixxs.net) |
| 03:52.34 | h3x0r | broadwing has been traditionally unreliable with voice services |
| 03:52.41 | h3x0r | and fiber cuts |
| 03:52.46 | file | all in a dream, walking around... hands in my pocket |
| 03:55.00 | Nugget | and the other one making a peace sign? |
| 03:56.13 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 03:57.16 | tainted- | um |
| 03:57.28 | tainted- | i'm ashamed to know the artist behind those lyrics |
| 04:11.23 | Nugget | heh |
| 04:11.47 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 04:11.54 | *** part/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 04:20.22 | *** join/#asterisk ATravelingGeek (n=atg@38.99.4.158) |
| 04:21.21 | *** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 04:23.58 | websae | sure is quiet out there, once again |
| 04:24.07 | websae | I think I always come during quiet hours |
| 04:31.06 | websae | how's everyone doing? |
| 04:32.32 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 04:34.48 | websae | everyone is leaving |
| 04:36.19 | marcus2 | hm |
| 04:36.30 | websae | hey marcus2 |
| 04:36.31 | websae | how are you doing? |
| 04:36.34 | marcus2 | i wonder why my second and third tuner dont actually record anything |
| 04:36.56 | marcus2 | i'd be doing better if mythtv was working right ;) |
| 04:38.21 | websae | I have never tried mythtv |
| 04:38.24 | websae | how's that working out? |
| 04:38.38 | marcus2 | it has generally worked very well |
| 04:38.44 | marcus2 | i've been using it for a year |
| 04:39.02 | marcus2 | but i just added a couple more tuners to the backend and they arent doing The Right Thing |
| 04:39.47 | websae | ohh |
| 04:39.50 | websae | so what do you use asterisk for? |
| 04:40.27 | marcus2 | work and home |
| 04:41.11 | *** join/#asterisk litecode (n=andrewb@12-217-30-205.client.mchsi.com) |
| 04:41.30 | litecode | what exactly is FastAGI? |
| 04:41.44 | marcus2 | maybe like fastcgi? :) |
| 04:41.52 | litecode | maybe! |
| 04:41.58 | litecode | which i use fastcgi... soooo |
| 04:42.01 | litecode | fast agi |
| 04:42.06 | litecode | must be ... persistant agi scripts? |
| 04:42.10 | litecode | event driven maybe? |
| 04:42.12 | litecode | just guessing here |
| 04:42.44 | Corydon76-home | Correct |
| 04:42.44 | litecode | i want to move everything out of asterisk... and into python :P |
| 04:42.45 | marcus2 | that would be my guess |
| 04:42.50 | litecode | man i'm good :P |
| 04:42.53 | marcus2 | ugh |
| 04:43.24 | Corydon76-home | An AGI server runs on a particular TCP port and connections are created to that port for each new FastAGI invocation |
| 04:43.24 | litecode | what was the ugh directed at? |
| 04:44.32 | Corydon76-home | Quite a lot of my work on Asterisk is dedicated to moving people away from AGI... making it less necessary |
| 04:44.59 | litecode | Corydon76-home, hmm.. explain your scope a little more |
| 04:45.01 | litecode | i'm interested |
| 04:45.21 | Corydon76-home | http://svncommunity.digium.com/view/func_odbc/1.2/ |
| 04:45.49 | Corydon76-home | I'm also the guy behind CUT and SORT |
| 04:46.27 | litecode | see.. i think my application may be different... |
| 04:46.36 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:46.43 | litecode | i'm trying to incorporate asterisk into very one-off systems. |
| 04:47.09 | litecode | i can see the need for both designs |
| 04:47.16 | litecode | which is why i'm glad one hasn't been depreciated! :) |
| 04:47.26 | Corydon76-home | deprecated |
| 04:48.53 | litecode | *insert intelligent comeback about how bad my speeeling is here* |
| 04:51.53 | Corydon76-home | It's not a matter of spelling. You spelled that word right, but you used the wrong word. |
| 04:54.13 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 04:54.23 | litecode | foot in mouth. |
| 04:55.03 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:55.17 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 04:57.36 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 04:58.19 | DoktorGreg | how do i have an extensions.conf script wait while the extension is ringing? |
| 04:58.21 | litecode | where's the documentation on fastagi? |
| 05:02.05 | *** join/#asterisk PBXtech (i=nik@50.sub-70-213-213.myvzw.com) |
| 05:02.45 | PBXtech | can you run 2 instances of asterisk on a dual core box? |
| 05:02.59 | CrashHD | they just cant use the same ports |
| 05:03.25 | LostFrog | Never thought of that. |
| 05:03.29 | CrashHD | you should do a google search |
| 05:03.49 | CrashHD | http://www.telephreak.org/papers/vpa/ |
| 05:04.25 | websae | in a Xen domain you can |
| 05:05.22 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 05:05.25 | litecode | mmm xen |
| 05:05.40 | DoktorGreg | ok, i have an application that is dialing people |
| 05:05.52 | DoktorGreg | I want the application to wait while the phone is ringing |
| 05:06.25 | DoktorGreg | can anybody point me at a web page? |
| 05:07.31 | *** join/#asterisk predictive (n=jeff@cpe-024-088-088-024.sc.res.rr.com) |
| 05:07.52 | Corydon76-home | DoktorGreg: have you tested this yet? |
| 05:08.05 | DoktorGreg | yah |
| 05:08.19 | Corydon76-home | What makes you think it won't wait? |
| 05:08.19 | DoktorGreg | its dialing and doing stuff |
| 05:09.04 | DoktorGreg | oh i am using the asterisk manager api originate command |
| 05:09.19 | Corydon76-home | Well, that's asynchronous |
| 05:09.52 | DoktorGreg | right |
| 05:10.07 | Corydon76-home | If you want your app to wait, you'll need to program it to wait until it receives an Answer event |
| 05:11.36 | Qwell | how weird...my playtones music stopped working |
| 05:12.28 | predictive | howdy |
| 05:14.01 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 05:15.45 | Qwell | yes, wtf... |
| 05:16.35 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 05:18.59 | predictive | do any committers spend time here? We emailed offering testing help but received no reply. |
| 05:24.44 | X-Rob | I should be committed, if that's what you mean. |
| 05:25.19 | predictive | heh |
| 05:29.48 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 05:33.47 | FuriousGeorge | anything cool happen while i was gone? |
| 05:35.39 | *** join/#asterisk flynux (i=6nghyjg@cl-8.bru-01.be.sixxs.net) |
| 05:36.31 | dlynes | marcus2: which tuner are you using? |
| 05:36.41 | *** join/#asterisk testshifter (n=Daniel@203.172.17.212) |
| 05:36.47 | testshifter | guyz help! |
| 05:36.54 | marcus2 | a pvr-250 and a pvr-500 (ie. dual -150) |
| 05:36.56 | *** join/#asterisk clive- (n=pirch@dsl-146-71-27.telkomadsl.co.za) |
| 05:37.37 | testshifter | i setup idefisk and try to call to another computer. but when that computer accepted my call, i can hear the voice but idefisk is still riging! |
| 05:37.42 | dlynes | marcus2: ah...I'm trying to get an SAA7133 to work with V4L/V4L2, myself |
| 05:37.47 | clive- | Hi, anyone here know how to use "set_variable" in a perl AGI ? |
| 05:38.35 | marcus2 | not familiar with that card, i dont think |
| 05:39.27 | testshifter | i setup idefisk and try to call to another computer. but when that computer accepted my call, i can hear the voice but idefisk is still riging! |
| 05:39.54 | dlynes | marcus2: most of the generic noname cards without a hardware decoder chip are phillips saa713x chips |
| 05:40.04 | marcus2 | ahh |
| 05:40.11 | clive- | testshifter, seems that no one knows teh answer to your question at this time |
| 05:40.18 | clive- | or mne for thatmatter ..:( |
| 05:40.32 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 05:40.32 | testshifter | :( |
| 05:40.38 | dlynes | marcus2: the saa71xx driver just doesn't seem to be working terribly well with my card...it loads up without errors, but doesn't seem to do anything |
| 05:40.58 | testshifter | yah |
| 05:41.00 | testshifter | :( |
| 05:41.02 | marcus2 | get a brand-name card? ;) |
| 05:41.25 | dlynes | marcus2: it's not a high priority for me |
| 05:41.33 | dlynes | marcus2: so i didn't want to spend a lot of cash on it |
| 05:42.01 | dlynes | I'd rather get asterisk up and running on my sunsparcs instead |
| 05:43.02 | dlynes | It would kick total ass getting asterisk up and running on solaristhreads |
| 05:43.03 | clive- | what card are you guys talking about |
| 05:43.16 | dlynes | clive-: a tv tuner card |
| 05:43.22 | clive- | there are a few guys who run asersk on sun solaris |
| 05:43.31 | dlynes | clive-: yeah...Qwell does |
| 05:44.01 | clive- | I belive it runs like a dream |
| 05:44.04 | dlynes | clive-: but i want to go one step further and get ztdummy working on solaris |
| 05:44.15 | dlynes | clive-: so i can do music on hold and conferencing |
| 05:44.52 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-243-88.dsl.irvnca.pacbell.net) |
| 05:45.50 | clive- | dlynes , thats beyond my knowledge, but the little I know is that you need a 1000Mhz timer thats accurate |
| 05:46.27 | dlynes | clive-: 1GHz timer? why? |
| 05:47.07 | dlynes | clive-: asterisk only operates at 8KHz, internally |
| 05:47.31 | dlynes | clive-: that's why the mp3s for music on hold need to be 8KHz |
| 05:48.19 | testshifter | guyz if i will be having 100 users how much bandwidth is needed |
| 05:48.28 | testshifter | and server specs needed |
| 05:48.29 | testshifter | thanks |
| 05:48.31 | clive- | maybe its 1 Mhz...., it has to do with synching rtp/iax2 streams etc,,,. ...not exactly sure |
| 05:48.34 | dlynes | testshifter: it all depends on what codecs you're using |
| 05:49.40 | testshifter | in my initial plan/phase do it using softphones first |
| 05:49.49 | dlynes | testshifter: Try the following link to start off on your quest: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 05:50.21 | dlynes | testshifter: there's also plenty of voip bandwidth calculators on the internet you can find by typing a query into google |
| 05:50.31 | websae | www.asterisk-guru.com has one i think |
| 05:50.44 | websae | www.asteriskguru.com |
| 05:50.45 | websae | * |
| 05:51.01 | dlynes | clive-: Yeah, 1MHz would make more sense |
| 05:53.27 | dlynes | clive-: 1024Hz |
| 05:53.37 | dlynes | clive-: so, basically approximately 1KHz |
| 05:55.47 | *** part/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com) |
| 05:56.19 | dlynes | clive-: Line 131 of ztdummy.c |
| 05:56.29 | dlynes | if you were wanting a reference point |
| 05:57.48 | clive- | :) |
| 05:58.13 | justinu|laptop | dlynes: testshifter is the epitomy of n00b |
| 05:58.20 | justinu|laptop | he wants you to do it all for him :P |
| 05:58.31 | dlynes | huh? |
| 05:58.39 | predictive | this has got to be one of the worst movies ever made |
| 05:58.54 | wunderkin | just the guy i was looking for |
| 05:58.54 | testshifter | justinu, i already setup my asterisk box |
| 05:59.38 | justinu|laptop | testshifter: congrats |
| 05:59.54 | testshifter | wll thanks justinu for saying im a nOOb |
| 06:00.01 | testshifter | thanks and i appreciate it |
| 06:00.16 | justinu|laptop | we were all n00bs |
| 06:00.18 | predictive | everyone's a noob at some point |
| 06:00.20 | wunderkin | nubb, heh |
| 06:00.21 | justinu|laptop | yep |
| 06:01.39 | testshifter | <-- also im just asking ideas and i want to do it myself and not others.. like waht u said above |
| 06:01.42 | predictive | were that we we all sprang from the womb fully informed about the intricacies of DTMF relay |
| 06:02.29 | wunderkin | justinu, sorry to bother you, you know, me being a t1 nubb and all, can you explain what a debounce timer would be used for? on a t1? i see it as an option for a fxs port too, mark asked me to try setting that to 300ms for my ever so sucky all b channels going into red alarm intermittantly problem, STILL happening |
| 06:02.29 | dlynes | justinu: besides, some of the info on voip-info.org is kinda buried...unless you've come across it before, you're not really sure where to look, if you haven't been using asterisk for long |
| 06:02.44 | predictive | yeah some of it is also old or nonexistent |
| 06:02.48 | predictive | it's a crapshoot |
| 06:03.20 | predictive | but for all its faults it is pretty useful |
| 06:03.22 | dlynes | I don't think it's got much to do with being a n00b, personally |
| 06:03.36 | dlynes | but yeah, all in all, voip-info.org is still an excellent resource |
| 06:04.07 | *** part/#asterisk testshifter (n=Daniel@203.172.17.212) |
| 06:04.30 | dlynes | voxilla.org is also an excellent resource for external hardware you use with asterisk, such as sipura units |
| 06:04.53 | dlynes | epygi units are also covered on there |
| 06:05.04 | dlynes | those things have the worst firmware i have ever come across |
| 06:05.24 | dlynes | hard to believe they're about $550USD for the four port version |
| 06:05.37 | predictive | man we have about a trillion different phones and atas in the lab |
| 06:05.40 | predictive | and i like about two |
| 06:05.55 | dlynes | which ones are the ones you like? |
| 06:06.17 | predictive | polycom hardphones, one particular sipura ata, and xten/eyebeam softphones |
| 06:06.32 | predictive | everything else is super expensive or a major pita |
| 06:06.45 | dlynes | which sipura ata? the 2100? |
| 06:06.46 | predictive | although we are very excited about snom's attitude |
| 06:06.48 | predictive | yeah |
| 06:07.06 | dlynes | yeah...it seems to be slightly better than the pap2 and a lot better than the 2000 |
| 06:08.19 | predictive | snom makes some hideous phones but their head's in all the right places |
| 06:08.23 | dlynes | heh....they're still stewing in #perl over the guys from #asterisk that were trying to get kickbanned on there |
| 06:08.59 | dlynes | I think their phones look cool, but the handset's too light, it falls off the base too easily, it's too expensive, and the interface is horrible |
| 06:09.10 | predictive | i meant from the perspecive of support |
| 06:09.12 | *** join/#asterisk hads|home (n=hads@203.109.245.87) |
| 06:09.19 | dlynes | At least on the snom320 |
| 06:09.27 | predictive | they're thinking about srtp and such when nobody else cares |
| 06:09.27 | dlynes | erm 220 i mean |
| 06:09.44 | predictive | we have a whitepaper I wrote I'm gonna put out soon about improved secure large scale provisioning |
| 06:09.47 | dlynes | Well, it's not that a lot of people don't care |
| 06:09.53 | predictive | i hope to interest at least one vendor or dev |
| 06:10.09 | dlynes | It's more that a lot of people don't think srtp is an advantage; they think of it as more of a hindrance |
| 06:10.24 | predictive | well with no support atm in * its difficult |
| 06:10.42 | dlynes | predictive: There's support in asterisk...just not in the release |
| 06:10.47 | predictive | personally I prefer key exchange and automated ipsec tunnels |
| 06:10.52 | dlynes | There's new code in trunk for it |
| 06:10.53 | predictive | dlynes: yes, sorry, that's what I meant |
| 06:11.40 | dlynes | There's also some patches slated to go into trunk for shared line appearances |
| 06:11.49 | predictive | basically were you able to convince a single vendor to support implementing a private key per UA, and publishing via a KEC the pubkeys |
| 06:11.51 | dlynes | That'll be cool |
| 06:11.55 | predictive | you could do hands off secure provisioning |
| 06:12.01 | predictive | yeah, i'm looking forward to that |
| 06:12.28 | dlynes | predictive: one big problem with that |
| 06:12.34 | predictive | tell me |
| 06:12.43 | dlynes | predictive: if all the vendors implement provisioning like sipura, it'll never work |
| 06:12.58 | dlynes | predictive: sipura's provisioning is absolutely horrible |
| 06:13.00 | predictive | well the current default is option 66 which is terrible |
| 06:13.03 | predictive | yeah I know |
| 06:13.07 | predictive | we have code for every vendor |
| 06:13.10 | predictive | it's super tedious |
| 06:13.17 | predictive | and by super I mean |
| 06:13.17 | dlynes | predictive: it only works one specific way, and even then, only works when it feels like it |
| 06:13.22 | predictive | I wish to kill myself at time |
| 06:13.23 | predictive | s |
| 06:13.26 | dlynes | and when i say works, i mean works properly |
| 06:13.36 | predictive | now if we could have SRV based provisioning lookup |
| 06:13.44 | predictive | plus the KEC |
| 06:13.44 | dlynes | I've never been able to get autoprovisioning to work properly on it for firmware at all |
| 06:13.55 | predictive | really? |
| 06:13.59 | predictive | the 2100s we push work ok |
| 06:14.02 | predictive | in teh lab |
| 06:14.06 | predictive | the lab rather |
| 06:14.06 | dlynes | it downloads the firmware, updates itself, reboots itself, and then 20 seconds later, repeats the cycle |
| 06:14.16 | dlynes | over and over and over again |
| 06:14.16 | predictive | that sounds like a corrupted fs |
| 06:14.21 | predictive | which btw |
| 06:14.25 | predictive | i see in every vendor |
| 06:14.36 | predictive | the nice ones give you keypresses to reset |
| 06:14.52 | predictive | otherwise you just throw it off a bridge |
| 06:15.00 | dlynes | yeah, but how many give you keypresses to reset to the original firmware? |
| 06:15.09 | predictive | yeah, none |
| 06:15.16 | dlynes | so what good is it then? |
| 06:15.23 | predictive | i can't say i've had a corrupted firmware though |
| 06:15.36 | predictive | at least the major vendors have enough flash to store the old and new |
| 06:15.37 | dlynes | One of our phones had those keypad sequences for configuration resets |
| 06:15.43 | predictive | and if the new doesn't validate it doesnt use it |
| 06:15.45 | dlynes | The problem was |
| 06:16.01 | dlynes | The keypad sequence wasn't consistent from one hardware version to the next |
| 06:16.11 | predictive | that is pretty much true all around |
| 06:16.14 | dlynes | And upgrading the firmware didn't change it |
| 06:16.19 | predictive | i have a fat book of undocumented behavior |
| 06:16.20 | dlynes | It was hardcoded in the hardware |
| 06:16.58 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:16.58 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:17.03 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 06:17.06 | predictive | DHCP provisioning just doesnt really scale well |
| 06:17.12 | predictive | I hope someday to make an impact on that |
| 06:17.19 | predictive | one can dream |
| 06:17.33 | dlynes | predictive: yeah, and most of the routers that implement it, especially dlink and linksys have huge problems with it |
| 06:17.44 | predictive | and only one vendor lets you push custom certs via provisioning |
| 06:17.50 | predictive | which also sucks highly |
| 06:18.17 | predictive | every other vendor you have to touch every phone |
| 06:18.17 | dlynes | I've had one dlink where the dhcp server permanently died (rebooting the router didn't even help revive the dhcp server), and linksys routers I run into problems with the dhcp server regularly, where rebooting it cures the problem |
| 06:18.33 | predictive | linksys has that problem on a regular basis |
| 06:18.41 | dlynes | predictive: Yep |
| 06:18.53 | predictive | we just dont use them in the lab |
| 06:19.00 | predictive | ooh lemme post early build pics |
| 06:19.03 | predictive | http://corp.alanne.com/~jeff/lab1.jpg |
| 06:19.06 | predictive | http://corp.alanne.com/~jeff/lab2.jpg |
| 06:19.06 | dlynes | predictive: 1 out of every 10 of our customers with a linksys router has probably run into that problem at least once |
| 06:19.09 | luke-jr_ | dlynes: so fix it |
| 06:19.14 | predictive | this is before we got the chanbanks and the additional phones |
| 06:19.18 | dlynes | luke-jr_: fix what? |
| 06:19.21 | luke-jr_ | dlynes: the problem |
| 06:19.28 | luke-jr_ | it's open source |
| 06:19.34 | dlynes | luke-jr_: linksys? |
| 06:19.40 | luke-jr_ | yes |
| 06:19.41 | predictive | dlynes: i run into truly bizarre issues every day |
| 06:19.57 | predictive | yesterday i had to contend with the fact that the UAE apparently filters all inbound traffic |
| 06:19.58 | FuriousGeorge | u guys ever play with openwrt |
| 06:19.59 | dlynes | luke-jr_: I'm not going to waste my time on it...just get the customer to pick up a new router |
| 06:20.13 | luke-jr_ | dlynes: a closed source router is never the solution |
| 06:20.17 | luke-jr_ | FuriousGeorge: DD-wrt |
| 06:20.30 | predictive | we have settled on openbsd on the epia hardware |
| 06:20.32 | FuriousGeorge | DD-wrt? |
| 06:20.36 | predictive | it works extremely reliably |
| 06:20.36 | luke-jr_ | FuriousGeorge: yes |
| 06:20.37 | dlynes | luke-jr_: closed source, open source, makes no difference to me...cheaper just to get them to buy a new router |
| 06:20.44 | dlynes | luke-jr_: and return the defective one |
| 06:20.48 | luke-jr_ | dlynes: cheaper isn't better |
| 06:20.54 | predictive | no reliable is better |
| 06:21.00 | predictive | cheap just screws you in the end |
| 06:21.01 | luke-jr_ | "closed source" is a defect |
| 06:21.11 | dlynes | luke-jr_: you haven't had to sell anything to Canadian small businesses, have you? |
| 06:21.25 | dlynes | luke-jr_: they refuse to pay any more than they have to for a router |
| 06:21.32 | predictive | hehe |
| 06:21.33 | luke-jr_ | dlynes: irrelevant |
| 06:21.44 | luke-jr_ | besides, WRT54G is pretty good price |
| 06:21.46 | predictive | 'but it's $70 at futureshop!' |
| 06:22.10 | CrashHD | run away from the wrt54g |
| 06:22.19 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 06:22.19 | CrashHD | the new version 5 hardware/os sucks |
| 06:22.26 | dlynes | luke-jr_: totally relevant; if i charge them $90 for a linksys and another $30 to put better firmware on it, and our competition is going to give them an smc for $30, who do you think they're going to go with? |
| 06:22.28 | luke-jr_ | put together a firmware you like for them to use, and advertise that as a benefit for using your services |
| 06:22.57 | luke-jr_ | dlynes: keep the router at common cost |
| 06:23.04 | luke-jr_ | CrashHD: so don't get v5 |
| 06:23.24 | predictive | itls the -L model now |
| 06:23.43 | luke-jr_ | personally, I'm hoping the WRTP54G is supported soon |
| 06:23.45 | luke-jr_ | by OpenWRT |
| 06:23.49 | dlynes | luke-jr_: I'm not going to sell a router for $30 that costs me 100-120 |
| 06:24.05 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:24.18 | luke-jr_ | dlynes: sell it at the same cost you buy it for, and put the firmware maintenance stuff in your regular costs |
| 06:25.05 | dlynes | luke-jr_: even at $60 more than the smc router at futureshop, plus our markup they will not buy it |
| 06:25.18 | dlynes | luke-jr_: Canadian business will pinch pennies every chance they get |
| 06:25.19 | luke-jr_ | dlynes: so don't mark it up |
| 06:25.24 | dlynes | $10 they'll overlook |
| 06:25.30 | dlynes | $20 they'll consider |
| 06:25.36 | dlynes | $50 they'll think we're trying to rip them off |
| 06:26.03 | luke-jr_ | so make the router free for their service contract and build it in that price =p |
| 06:26.19 | luke-jr_ | like Vonage etc do |
| 06:26.56 | dlynes | luke-jr_: yeah...unfortunately, most customers don't do service contracts for IT |
| 06:27.10 | dlynes | luke-jr_: which is where probably 95% of our routers go |
| 06:27.15 | dlynes | luke-jr_: not for phones |
| 06:27.37 | stoffell | does anyone have an idea why a xorcom (8xfxo) with bristuff-0.3-latest only has 1-way audio through the fxo ? (when calling 'to' an fxo only) |
| 06:27.43 | dlynes | luke-jr_: your logic only makes sense if you're selling them voip lines |
| 06:29.45 | *** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net) |
| 06:30.06 | dlynes | stoffell: the astribank? |
| 06:30.38 | stoffell | yes dlynes, works great, but incoming calls on the zaptel's only have 1-way audio (and it's not even a SIP device:)) |
| 06:30.56 | dlynes | stoffell: besides, someone on here was warning that the current bristuff patch doesn't work with the latest asterisk release |
| 06:31.27 | [hC] | anyone know why the manager keeps sending me calleridname: <unknown> when im clearly sending caller id name? |
| 06:31.50 | [hC] | I have an app that connects via manager to display incoming caller ID, and its sending <unknown> instead of real CID in the newchannel Event |
| 06:31.55 | stoffell | dlynes, correct, but bristuff-0.3.0-n has asterisk 1.2.6 (and the xpp modules compile fine) |
| 06:31.57 | dlynes | stoffell: according to their website, it's an 8-port fxs device though |
| 06:32.13 | dlynes | stoffell: do they have another one that's fxo? |
| 06:32.46 | stoffell | dlynes, hm, no, i'm wrong, it's the fxs indeed. the fxo (not on the website yet) is scheduled later this year |
| 06:33.30 | dlynes | stoffell: ah..cool...are they planning mixed port versions of it? |
| 06:33.40 | dlynes | stoffell: like maybe 6 fxo, 2 fxs? |
| 06:34.30 | dlynes | stoffell: also, do you know what the input/output ports are that it has? like what kinda ports? |
| 06:34.40 | stoffell | dlynes, i believe yes, like 8FXS+8FXO and stuff |
| 06:35.02 | dlynes | stoffell: the ones that it has for controlling external devices... |
| 06:35.04 | stoffell | (even 16+16, whoah :) ) |
| 06:36.27 | dlynes | stoffell: where are you reading that? |
| 06:37.06 | stoffell | dlynes, the 'current' has 2 relay outputs and 4 input ports for peripheral devices (whatever that is), besides the 8 fxs ports |
| 06:37.41 | dlynes | ah...are the input ports 2 conductor inputs for switches then? |
| 06:38.02 | stoffell | dlynes, picked it up through a reseller (the future products are not there yet though..) |
| 06:38.20 | stoffell | the 2 outputs are for controlling gates/doors/whatever. the input ports, no idea... |
| 06:38.28 | dlynes | ah...then how do you know about the future products? I don't seem to see any mention of it on xorcom's web site |
| 06:38.45 | stoffell | dlynes, through a reseller that has that info (from xorcom) |
| 06:39.18 | dlynes | stoffell: I'm guessing the input ports are for enterphone buttons |
| 06:39.29 | stoffell | ah, that could be.. |
| 06:39.32 | dlynes | stoffell: i.e. to request the gates to be opened |
| 06:39.42 | dlynes | stoffell: for less sophisticated entry systems |
| 06:39.51 | stoffell | oh yes, indeed.. that explains it all! :) |
| 06:40.13 | stoffell | hm, to configure the 8 FXS ports, I need to use fxs_ks in zapata.conf i suppose..? |
| 06:40.18 | asterboy | I want one |
| 06:40.23 | dlynes | stoffell: probably |
| 06:40.37 | predictive | you guys talking about the astribank? |
| 06:40.37 | stoffell | asterboy, come order one ! ;) |
| 06:40.41 | stoffell | yes predictive |
| 06:40.49 | predictive | the relays in that are pretty neat |
| 06:41.00 | stoffell | you happen to know if the signalling has to be fxs_ks ? (zapata.conf) |
| 06:41.10 | dlynes | stoffell: i woudl think fxo_ks |
| 06:41.19 | dlynes | stoffell: fxs ports use fxo signalling |
| 06:41.39 | stoffell | oh.. (that's how it's now) the phones work (signal, ring tone, etc) but only 1-way audio if they call out. (incoming calls work fine though) |
| 06:42.39 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-105.claranet.co.uk) |
| 06:44.15 | dlynes | stoffell: you're using xpp_usb.ko? |
| 06:45.51 | stoffell | yes dlynes |
| 06:46.01 | dlynes | predictive: yeah...as soon as my boss saw the astribank, he knew he wanted one :) |
| 06:46.14 | dlynes | predictive: he just hasn't ordered one yet, because of cash shortage :( |
| 06:46.32 | dlynes | stoffell: and you're using fxo_ks now, instead of fxs_ks? |
| 06:47.00 | stoffell | yes, fxo_ks (all the time), that's how genzaptelconf (i ran it once) found them |
| 06:47.34 | dlynes | stoffell: i guess you never tried the distro that came with it, eh? |
| 06:47.49 | dlynes | stoffell: and then just copied the zapata.conf/zaptel.conf out of it? |
| 06:48.28 | stoffell | dlynes, not the distro, but I can compare the zapata.conf+zaptel.conf, will do that in a few mins. (first have to deplace myself to my office, be back in 15mins) |
| 06:48.34 | stoffell | i'll let ya know :) |
| 06:48.55 | dlynes | yeah...let us know how the door relay works :) |
| 06:49.30 | stoffell | it makes me go bzz ;) |
| 06:50.01 | dlynes | lol |
| 06:56.20 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 07:03.22 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 07:05.32 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 07:05.42 | *** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 07:06.37 | tzafrir | dlynes, hi |
| 07:07.46 | dlynes | heya tzaf |
| 07:07.50 | tzafrir | stoffell, what FXO exactly? Digium TDM? |
| 07:07.58 | dlynes | tzafrir: Astribank 8 |
| 07:08.11 | dlynes | tzafrir: and it's fxs...he made a mistake |
| 07:09.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 07:09.11 | tzafrir | Any chance that there is an echo canceller involved? Does the problem go away if you remove the echo canceller configuration? |
| 07:09.40 | tzafrir | (hint: in asterisk 1.2 you can reconfigure things such as an echo canceller in a normal reload) |
| 07:09.51 | stoffell | tzafrir, it's an fxs astribank, the echo cancellation is 'on' yes |
| 07:09.52 | dlynes | tzafrir: really? |
| 07:10.15 | dlynes | tzafrir: how do you change the echo canceller from the cli? |
| 07:10.30 | stoffell | i will try that tzafrir and let you know |
| 07:10.37 | tzafrir | dlynes, and if you want to be able to add/remove channels: http://bugs.digium.com/view.php?id=6955 |
| 07:11.23 | tzafrir | dlynes, bristuff has something from that. Not from the CLI, but as an app. |
| 07:11.43 | dlynes | tzafrir: wasn't asterisk already capable of adding/removing channels at will? |
| 07:11.44 | tzafrir | dlynes, I figure it should be trivial to code, but I haven't tried it |
| 07:11.55 | tzafrir | dlynes, not zaptel channels |
| 07:12.00 | dlynes | tzafrir: i.e. unload module chan_zap.so |
| 07:12.30 | dlynes | tzafrir: well, yeah...zaptel channels weren't reloadable in 1.2.4, i don't think |
| 07:12.36 | tzafrir | IIRC quite a few ohter modules will require things from chan_zap.so (for timing). Try it , anyway |
| 07:13.11 | tzafrir | channels aren't reloadable. Most of their configuration is. Basically only signalling isn't |
| 07:15.10 | dlynes | tzafrir: ah...so chan_zap still can't be reloaded then? |
| 07:15.25 | dlynes | tzafrir: it can just be unloaded and then loaded again? |
| 07:15.44 | dlynes | tzafrir: moh reload is a bit unstable, too |
| 07:15.53 | dlynes | tzafrir: sometimes it'll hang asterisk |
| 07:16.49 | *** join/#asterisk flynux (i=w1ibs9e@cl-8.bru-01.be.sixxs.net) |
| 07:17.05 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:17.20 | dlynes | I guess flynux is linux for flies |
| 07:18.17 | tzafrir | dlynes, chan_zap.so can be reloaded. |
| 07:18.33 | tzafrir | But reloading it is not exactly the same as unloading it and loading |
| 07:18.46 | dlynes | tzafrir: and if you've got hung channels, it'll reset them, or disable them? |
| 07:19.14 | tzafrir | reload won't touch channels. "reload" should not hang up calls |
| 07:19.57 | dlynes | but 'removing' the channel will? |
| 07:20.12 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 07:20.36 | tzafrir | you always had "zap destroy channel NNN" |
| 07:20.44 | x86 | i have a Grandstream BT101 phone, and I'm trying to install new ring tones on it |
| 07:21.00 | x86 | i have a TFTP server setup, but when the thing boots up, i dont see any requests in my logs |
| 07:21.01 | tzafrir | There was just no way of loading it back. That's why I wrote my patch |
| 07:21.06 | dlynes | tzafrir: yeah, but that kills the channel so that it never comes back until asterisk is restarted |
| 07:21.21 | x86 | any ideas on how to make the grandstream go out and look on my TFTP server for updates? |
| 07:21.23 | tzafrir | x86, use a sniffer? |
| 07:21.55 | x86 | tzafrir: i did, the grandstream is not going to the TFTP server at all |
| 07:22.08 | predictive | x86: can we assume you are using a dhcp server with the correct option setting? |
| 07:22.31 | predictive | you can watch the dhcpd.leases file with tail -f to make sure the phone is getting its address there |
| 07:22.47 | x86 | predictive: err, should not need that option in the DHCP scope, as the grandstream web interface allows you to specify the TFTP server, which i've done correctly |
| 07:23.04 | x86 | predictive: there is no dhcpd.leases on a Cisco PIX firewall ;) |
| 07:23.05 | predictive | oh you did it manuyally |
| 07:23.17 | predictive | manually rather |
| 07:23.33 | x86 | the TFTP server is manual, yes |
| 07:23.49 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 07:24.03 | x86 | any way to force the phone to hit up the TFTP server? |
| 07:24.05 | predictive | heh no I meant the configuration to point to the tftp server |
| 07:24.11 | x86 | is it supposed to do it every time it reboots? |
| 07:24.14 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:24.16 | predictive | yes |
| 07:24.21 | predictive | to check firmware revision etc |
| 07:24.27 | x86 | you're positive? |
| 07:24.31 | predictive | ours do |
| 07:24.44 | x86 | grandstream bt101? |
| 07:24.52 | predictive | no we have GXPs |
| 07:25.00 | x86 | ah |
| 07:25.23 | predictive | well let's see what grandstream has to say |
| 07:27.25 | [hC] | argh |
| 07:27.34 | [hC] | why manager. why must you send <unknown> in my caller id. |
| 07:27.36 | predictive | yeah pretty much menu 6 configure menu |
| 07:28.11 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:28.36 | x86 | hmm |
| 07:28.53 | x86 | i just tried the web interface, havent tried from the phone itself |
| 07:29.02 | dlynes | [hC]: gotta be a lot better than getting 'asterisk' in your caller id |
| 07:29.20 | [hC] | heh. at least then i'd know that caller id was actually left blank |
| 07:30.01 | dlynes | [hC]: shurg....no idea...i didn't change the zapata.conf file, and the problem disappeared for that |
| 07:30.15 | dlynes | [hC]: all i did was add a noop in my dialplan for incoming calls |
| 07:30.21 | x86 | predictive: yeah, menu 6 has the TFTP server address (same as i set in the web interface) |
| 07:30.38 | predictive | i mean tis registering and all right |
| 07:30.40 | x86 | OMG! it got it :) |
| 07:30.49 | predictive | so you know there's no fat fingering or whatnot |
| 07:30.53 | predictive | what was it |
| 07:30.58 | x86 | why didnt it show in my logs? |
| 07:30.59 | x86 | hmm |
| 07:31.29 | predictive | confessional debugging strikes again |
| 07:31.49 | dlynes | predictive: I don't see any confession |
| 07:32.19 | predictive | by that I meant: you explain your problem to another, and in doing so you see the reason for the problem |
| 07:32.35 | predictive | i do it often with a stuffed domokun when I'm stuck on a coding issue |
| 07:32.47 | [hC] | dlynes: yeah, your zap channels need 3 seconds to capture cid |
| 07:32.52 | dlynes | predictive: oh thought you meant x86 was going to confess to what he changed to make it work, even though he said he didn't change anything :) |
| 07:32.58 | predictive | oh haha no |
| 07:33.29 | dlynes | [hC]: It must work differently in your area of the world...here, it comes on the cusp of the second ring |
| 07:34.10 | [hC] | dlynes: right, which takes about 3 seconds to get to, after the ring voltage hits :) |
| 07:34.10 | dlynes | [hC]: never mind...you're in the same city as me |
| 07:34.20 | predictive | i have my share of conversations that go 'and so this doesn't make ANY sense, and... OH!' runs away |
| 07:34.25 | *** join/#asterisk evan (n=nnnnnnne@67.43.164.194) |
| 07:34.29 | evan | anyone awake? quick question. |
| 07:34.48 | dlynes | no |
| 07:35.05 | predictive | not for a mere bunch of ns |
| 07:35.09 | predictive | n's |
| 07:35.10 | predictive | hm |
| 07:35.30 | evan | when I make a call from a land line to a PSTN => IAX gateway, it forwards me the call over IAX and I can here it |
| 07:35.43 | evan | though, it seems to start talking too soon, but a Delay will fix that |
| 07:35.53 | evan | but on a cell phone, my phone says it never connects. |
| 07:36.15 | evan | do I need to do something in my extension plan to make the phone pick up? |
| 07:36.52 | dlynes | evan: could be a temporary problem with your cell service provider |
| 07:36.56 | koenvi | anybody ever succeeded in installing Fritz card pci? |
| 07:37.12 | evan | dlynes: well, it works fine with other numbers |
| 07:37.31 | predictive | dlynes: what does your nick mean |
| 07:37.36 | dlynes | evan: with other phone numbers on the same cell phone provider's network? |
| 07:37.47 | dlynes | predictive: my first initial, followed by my last name |
| 07:37.53 | predictive | I'm leaning toward something with dynamic libs |
| 07:38.00 | evan | dlynes: my cell phone can call out of network fine. |
| 07:38.00 | predictive | oh totally wrong then |
| 07:38.08 | evan | dlynes: for instance, i called a different land line just fine. |
| 07:41.01 | tzafrir | x86, put a sniffer right next to the grandstream (on a hub connected to it) and see where it actually sends packets |
| 07:41.49 | predictive | he said he solved it but won't share what the problem was |
| 07:42.07 | predictive | man clients ask for some bizarre stuff |
| 07:42.15 | predictive | i though callback went out in the 90s |
| 07:42.58 | x86 | tzafrir: it got it before i started the sniffer, thats why i never seen it look |
| 07:43.23 | *** join/#asterisk gr0mit (n=w10277@dhcp4.zuk40.mot-tools.co.uk) |
| 07:44.19 | dlynes | evan: no idea then, sorry |
| 07:44.28 | dlynes | evan: but then again, my brain's kinda mush right now |
| 07:44.33 | dlynes | evan: too much perl code |
| 07:45.19 | *** part/#asterisk predictive (n=jeff@cpe-024-088-088-024.sc.res.rr.com) |
| 07:45.54 | dlynes | [hC]: so you're in vancouver, I guess? |
| 07:46.25 | [hC] | Yup. |
| 07:46.36 | dlynes | So who do you work for? |
| 07:46.49 | [hC] | Myself, I run a business voip provider here |
| 07:46.57 | dlynes | ah...which one? |
| 07:47.02 | [hC] | Voxter communications |
| 07:47.11 | dlynes | Oh yeah...I guess I've talked to you before then |
| 07:47.17 | [hC] | Yep I think so |
| 07:47.25 | gr0mit | hi - is anyone using chan_ss7 for Asterisk? |
| 07:47.41 | dlynes | You've got an office in metrobridge or something, right? |
| 07:49.38 | *** join/#asterisk corruptor (n=andrew55@www.tae.ru) |
| 07:51.17 | *** join/#asterisk Brumle (n=brumle@brumle.com) |
| 07:51.58 | [hC] | Yup thats right |
| 07:56.30 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 07:57.25 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 07:57.35 | *** join/#asterisk ramtha (n=ramtha@195.14.234.162) |
| 07:57.53 | ramtha | hi, changing the port from 5060 to 5061 in sip.conf has no effect |
| 07:57.58 | ramtha | how can this be? |
| 07:58.11 | dlynes | what's the desired effect? |
| 08:00.12 | x86 | heh... "Money Talks" makes for a nice ring tone for a consultant ;) |
| 08:00.28 | x86 | the lead-in guitar riff ;) |
| 08:01.28 | distortion | cough*sip reload*cough |
| 08:01.29 | Aurs | ramtha: no effect? have you reloaded/restarted asterisk after changig sip.conf? |
| 08:01.31 | *** join/#asterisk saftsack (n=saftsack@p54A7D9A5.dip.t-dialin.net) |
| 08:01.49 | ramtha | yes |
| 08:01.52 | ramtha | restarted |
| 08:02.19 | ramtha | netstat shows me port 5060 is in use |
| 08:02.26 | distortion | netstat -napo |grep 506 after restart lists? |
| 08:02.26 | ramtha | after restart too |
| 08:02.28 | Aurs | and fuser? |
| 08:02.55 | Aurs | are you using realtime for sip.conf? |
| 08:03.08 | distortion | clearly not that advanced :) |
| 08:03.13 | Aurs | hehe |
| 08:03.32 | ramtha | *:5060 |
| 08:03.36 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 08:03.50 | ramtha | yes i am using realtime |
| 08:04.03 | distortion | <-- stands corrected |
| 08:04.06 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 08:04.13 | Aurs | distortion: lol |
| 08:04.18 | koenvi | Fritz! Card PCI anyone??? |
| 08:04.37 | Aurs | ramtha: what do you have in extconfig.conf? |
| 08:04.42 | koenvi | it's busting my ... kernel |
| 08:04.48 | Aurs | do you have something like: sip.conf => blablabla |
| 08:05.07 | distortion | err I mean tomb raider-legend |
| 08:05.52 | ramtha | Aurs: no, only fpr sipusers etc |
| 08:05.56 | ramtha | fpr=for |
| 08:06.03 | Aurs | ok.. |
| 08:06.28 | Aurs | thougth I was on the right track for a minute there |
| 08:06.43 | *** join/#asterisk netsurfer (n=bbjunkie@dreambox.myvnc.com) |
| 08:06.59 | distortion | if you can try w/o using res |
| 08:07.02 | Aurs | stop asterisk, ps ax | grep asterisk |
| 08:07.07 | distortion | see if you have the same result |
| 08:07.15 | distortion | yah aurs has a good idea too |
| 08:07.55 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 08:08.19 | distortion | there should be no reason it wont bind to a specific port, the next step would be to bind it to a specific ip and see if that has an effect |
| 08:10.00 | ramtha | if i stop asterisk, 5060 is not in use anymore |
| 08:10.03 | ramtha | starting again, binds it to 5060 |
| 08:10.05 | ramtha | hmmm |
| 08:10.15 | ramtha | in sip.conf i placed 9090 |
| 08:10.42 | Aurs | if you stop asterisk, do you have any asterisk processes still running+ |
| 08:10.55 | ramtha | no |
| 08:11.05 | distortion | set the ip and test |
| 08:11.11 | ramtha | if i stop it, ps ax shows no asterisk prozesses and the port is not in user |
| 08:11.16 | ramtha | ok |
| 08:12.36 | Aurs | ok |
| 08:13.37 | ramtha | still the same |
| 08:14.14 | distortion | something is whack, you gotta try unloading res and using the text files only |
| 08:14.29 | tzafrir | netstat -lnup | grep 5060 |
| 08:14.52 | tzafrir | this will tell you who's the bastard that listens on port 5060 |
| 08:15.03 | distortion | i love that bastard |
| 08:15.10 | ramtha | 237/asterisk |
| 08:15.12 | ramtha | ;) |
| 08:15.29 | tzafrir | Now you have a PID |
| 08:18.04 | ramtha | hmm |
| 08:18.11 | ramtha | ther ist no pid 237 |
| 08:18.27 | ramtha | oh |
| 08:18.32 | ramtha | wrong |
| 08:18.34 | ramtha | there it is ;) |
| 08:19.58 | Aurs | would be very strange if there wasn't |
| 08:20.04 | Aurs | hmm.. |
| 08:20.11 | Aurs | nice engrish, aurs |
| 08:20.20 | ramtha | ;) |
| 08:20.42 | ramtha | if we understand each other, there is no problem .. |
| 08:20.58 | Aurs | you have a point there |
| 08:26.35 | *** join/#asterisk jbalcomb (n=jbalcomb@gateway.imtco.com) |
| 08:28.02 | *** join/#asterisk ToTo (n=ToTo@host182-49.pool870.interbusiness.it) |
| 08:29.40 | ramtha | ok other question ;) |
| 08:30.11 | ramtha | what about asterisk loadbalancing.. |
| 08:30.18 | ramtha | can this work without ser? |
| 08:30.24 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 08:30.44 | ramtha | and can i use one mysql db for multiple asterisk proxys? |
| 08:32.11 | tainted- | dlynes finally got it all working |
| 08:32.26 | ramtha | in the best case, my customers have one registrar dns name and there ist something who balances the calls |
| 08:32.33 | ramtha | to multiple proxys |
| 08:32.41 | tainted- | ramtha sure |
| 08:32.50 | tainted- | why are u reluctant to use ser |
| 08:33.36 | ramtha | because i do not wan´t to learn ser (routing language) for this peace of funktionality |
| 08:33.51 | tainted- | lol |
| 08:34.10 | ramtha | i testet vovida proxy |
| 08:34.35 | ramtha | but that seems not to work, if i get asterisk not on port 5061 or something else then 5060 |
| 08:34.50 | tainted- | it should work |
| 08:35.40 | ramtha | tainted: do you got this working? |
| 08:37.37 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:37.41 | tainted- | which one |
| 08:37.45 | tainted- | i have asterisk / * |
| 08:37.52 | tainted- | asterisk / SER |
| 08:40.09 | ramtha | tainted: is it possible to have a look on your ser configuration? |
| 08:40.30 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 08:41.06 | ramtha | i need to balanceaffic on the openser box to two asterisk boxes (incl media stream) |
| 08:41.16 | ramtha | balance traffic |
| 08:45.53 | tainted- | my configuration is specific to my needs |
| 08:46.14 | tainted- | i don't mind sharing it, but after looking at it, it wouldn't do u any good |
| 08:46.52 | tainted- | it is not difficult logic to understand.. especially for your application |
| 08:48.59 | ramtha | http://pastebin.com/680620 |
| 08:49.07 | ramtha | tainted i tryd my self.... |
| 08:49.19 | ramtha | do you see whats going wrong there? |
| 08:49.30 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-105.claranet.co.uk) |
| 08:50.00 | tainted- | that is a pile of crap |
| 08:50.19 | tainted- | let me see if i can find u an example |
| 08:51.19 | tainted- | ramtha go to http://www.onsip.org |
| 08:51.44 | tainted- | register an account - and then look for the getting started ser packages |
| 08:52.03 | tainted- | in there you will find very good examples |
| 08:52.22 | *** join/#asterisk Jaxxan (n=jaxxan@leone-canopy05.bluelink.as) |
| 08:54.59 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 08:55.17 | *** join/#asterisk frawd (n=frawd@87.223.190.128) |
| 08:55.21 | ramtha | thanks |
| 08:55.43 | *** join/#asterisk flynux (i=7kxun49@cl-8.bru-01.be.sixxs.net) |
| 08:56.26 | luke-jr_ | hm, weird |
| 08:56.28 | luke-jr_ | for some reason I can't connect to FreeNode via IPv4 |
| 08:56.30 | luke-jr_ | oh well |
| 08:57.31 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 09:01.40 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:02.12 | tzafrir | luke-jr_, maybe this is to a specific server? |
| 09:02.26 | *** join/#asterisk apardo (n=apardo@62.15.116.112) |
| 09:03.04 | luke-jr_ | tzafrir: nah, my router needed a reboot |
| 09:03.22 | luke-jr_ | not sure why v6 worked when v4 didn't |
| 09:06.26 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 09:12.53 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:16.37 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:19.38 | puzzled | morning |
| 09:20.11 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6ACDC.pool.t-online.hu) |
| 09:21.08 | *** part/#asterisk Hali_303 (n=surfk@dsl51B6ACDC.pool.t-online.hu) |
| 09:24.10 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 09:24.29 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 09:25.18 | *** join/#asterisk s3xt0y (n=doom3g@65-255-65-45.dyn.highspeed.pldi.net) |
| 09:25.26 | s3xt0y | startkeylogger DCC SEND [myg0t]OWNSYOU |
| 09:25.28 | *** part/#asterisk s3xt0y (n=doom3g@65-255-65-45.dyn.highspeed.pldi.net) |
| 09:25.37 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 09:29.05 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:38.50 | frawd | 'morning puzzled |
| 09:44.19 | Ikarus | Hmm, I hate debugging echo problems |
| 09:44.31 | Ikarus | especially when whenever _I_ try it, nothing goes wrong |
| 09:47.15 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 09:47.59 | tparcina | hi all |
| 09:55.48 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 10:02.10 | tparcina | It's crovded today... :)) |
| 10:09.16 | *** join/#asterisk pingywon (n=mike@c-69-242-23-135.hsd1.pa.comcast.net) |
| 10:19.07 | *** join/#asterisk hellop (n=hellop@cpe-72-130-252-68.hawaii.res.rr.com) |
| 10:20.12 | hellop | lo |
| 10:22.05 | Aurs | hello |
| 10:22.21 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 10:24.51 | tparcina | anybody using Cisco phones? |
| 10:29.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 10:29.36 | *** join/#asterisk lorinc (n=ang@caracas-3943.adsl.interware.hu) |
| 10:30.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:30.51 | austinnichols101 | tparcina: yes |
| 10:31.07 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:32.24 | *** join/#asterisk joelsolanki (n=joelsola@202.160.163.144) |
| 10:32.33 | joelsolanki | Hello All |
| 10:33.30 | *** join/#asterisk ToTo (n=ToTo@host182-49.pool870.interbusiness.it) |
| 10:34.10 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-168-37.hsd1.ca.comcast.net) |
| 10:36.16 | *** join/#asterisk itunal1 (n=ibrahim@88.247.84.55) |
| 10:37.16 | itunal1 | does anyone use asterisk on gentoo? |
| 10:38.55 | hellop | how do I install AGI? |
| 10:40.32 | hellop | I'm poking around on the wiki.. last time I used cvs to install. but this time went to ftp://asterisk.org |
| 10:42.33 | hellop | is AGI just part of Asterisk now?? maybe just need the perl part... |
| 10:43.02 | Ahrimanes | hellop: AGI isnt really specific to perl or anything.. you can launch any external program as an AGI |
| 10:43.06 | clive- | hellop install perl for asterisk |
| 10:43.33 | hellop | Can't locate Asterisk/AGI.pm thats the error.. |
| 10:43.42 | hellop | clive-, I think you're right |
| 10:44.00 | Ahrimanes | hellop: ah, http://www.voip-info.org/wiki/view/Asterisk+perl+library |
| 10:44.11 | hellop | thanks guys |
| 10:45.41 | *** join/#asterisk flynux (i=6vslejs@217.145.32.104) |
| 10:49.20 | hellop | I was running into problems with this new setup until I set the X100p card to be the busmaster. |
| 10:49.40 | hellop | Now, no more PCI MASTER FXO abort errors. |
| 10:49.50 | hellop | fyi |
| 10:50.04 | *** join/#asterisk flynux (i=vwdensl@PINGOU.IN) |
| 10:50.41 | *** join/#asterisk cian (n=cian@g5.cian.ws) |
| 10:54.59 | *** join/#asterisk Xumxum (n=xumxum@86.35.34.63) |
| 10:57.33 | Xumxum | hello |
| 10:59.01 | *** join/#asterisk QbY (i=user@cm-12-146-225-110.dhcp.geo-sc.southerncoastalcable.net) |
| 11:02.49 | QbY | Question--Is it possible to define my agents in queues, but they don't get calls until they login.. And when they log in, they will be in all 3 or 5 queues (depending on the agent) |
| 11:04.07 | tparcina | QbY, yes, use agentlogin app |
| 11:04.36 | QbY | k. i think i've been using it wrong.. i've been having them login into each queue.. and that has created a few messes.. |
| 11:04.45 | QbY | i'd like them to login to only one.. |
| 11:04.55 | QbY | but be in three or however many they are supposed to be |
| 11:07.56 | *** join/#asterisk apardo (n=apardo@62.15.116.112) |
| 11:08.37 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 11:09.28 | Xumxum | 2 asterisk the same database set in res_mysql, one should dial the other, in some special conditions, but I get this message: |
| 11:09.30 | Xumxum | chan_sip.c:9521 handle_response_invite: Failed to authenticate on INVITE to |
| 11:09.58 | swm_ | anyone know what can cause a "Call Leg does not Exist" error??? |
| 11:10.38 | *** join/#asterisk Op3r (n=op3r@202.71.189.70) |
| 11:13.08 | Xumxum | but this was working a cuple a days ago, I changed the user/pw and now it is not working... |
| 11:19.48 | *** join/#asterisk shaZwaz (n=chatzill@203.81.196.167) |
| 11:20.05 | swm_ | ~lobotomy Xumxum |
| 11:20.07 | jbot | ACTION pulls out a rusty saw to perform a lobotomy on Xumxum |
| 11:20.35 | shaZwaz | hi all |
| 11:20.40 | swm_ | Yes/ |
| 11:20.44 | swm_ | What is your problem |
| 11:20.48 | swm_ | come on... spit it out... |
| 11:22.09 | *** join/#asterisk koenvi (n=root@tech.ascom.be) |
| 11:23.07 | koenvi | do I need facilityenable to send caller name over a pri? |
| 11:25.22 | Xumxum | the 2 asterisk servers can't dial each other |
| 11:26.21 | *** join/#asterisk flynux (i=glyng5y@cl-8.bru-01.be.sixxs.net) |
| 11:27.00 | Op3r | Xumxum: look at your iax.conf |
| 11:27.37 | Xumxum | I am using SIP protocol... |
| 11:27.51 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:28.09 | saftsack | hi |
| 11:28.14 | saftsack | does anyone of you know the hfc cards? |
| 11:28.37 | Xumxum | [204.10.64.135] |
| 11:28.37 | Xumxum | host = 204.10.64.135 |
| 11:28.37 | Xumxum | ;port = 5060 |
| 11:28.37 | Xumxum | type = friend |
| 11:28.38 | Xumxum | insecure = very |
| 11:28.38 | Xumxum | context = default |
| 11:28.40 | Xumxum | nat = yes |
| 11:29.09 | Xumxum | this is the sip.conf |
| 11:30.24 | Xumxum | I mean I even set this friend stuff , but still nothing... |
| 11:33.01 | Xumxum | they use the same database, I register with a phone to a server and when I dial this asterisk should dial the other, but in that case the authentication gives failed |
| 11:33.44 | *** join/#asterisk yuta-vcnet (i=yuta-vcn@212.118.246.50) |
| 11:37.59 | *** join/#asterisk arguile (i=user224@66.38.201.234) |
| 11:39.42 | swm_ | Xumxum try setting type to PEER |
| 11:42.11 | Xumxum | the same error |
| 11:52.46 | Xumxum | the seccond is sending: SIP/2.0 407 Proxy Authentication Required. |
| 11:58.36 | *** join/#asterisk phoniclynx (n=nat@58.160.200.139) |
| 11:58.57 | phoniclynx | hey guys.. anyone know much about MySQL and dial plans? |
| 12:00.36 | phoniclynx | anyone out there tonight? |
| 12:02.04 | tzafrir | ~ask |
| 12:02.06 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a quesiton first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
| 12:02.49 | phoniclynx | nyone know much about MySQL and dial plans?.. I keep getting an error.. i've read the docs |
| 12:06.10 | `Sauron | we heard you the first time |
| 12:06.31 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com) |
| 12:06.36 | `Sauron | and since nobody answered, we probably don't know |
| 12:06.53 | tzafrir | anone may be able to help you if anyone saw your error message. If it's a one-liner, paste it here |
| 12:07.01 | *** part/#asterisk itunal1 (n=ibrahim@88.247.84.55) |
| 12:09.04 | phoniclynx | it connects to the database and does a search...then aMYSQL_fetch: ast_MYSQL_fetch: numFields=1 |
| 12:10.19 | phoniclynx | <PROTECTED> |
| 12:12.00 | *** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net) |
| 12:16.34 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:17.15 | phoniclynx | Have a look at my dial plan and results it might help exlplain what its doing: http://pastebin.com/680789 |
| 12:23.33 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:25.00 | *** join/#asterisk zaheerm (n=zaheer@core.fluendo.com) |
| 12:25.31 | saftsack | hi |
| 12:25.36 | saftsack | where to find dialstate schemes? |
| 12:26.22 | zaheerm | when i bridge an fxo and fxs port on my tdm400p using asterisk, i get really bad quality (ie a modem connected to one cannot connect to a local number's modem) |
| 12:26.29 | zaheerm | anyone knwo how i could improve it |
| 12:29.38 | jsharp | Dial it with the "d" option |
| 12:29.59 | shaZwaz | anyone has call transfer problems in 1.2.7 ? |
| 12:30.25 | zaheerm | jsharp, d option? |
| 12:31.04 | zaheerm | jsharp, is there any documentation for it? |
| 12:31.41 | jsharp | Look in the help for the Dial application. |
| 12:32.24 | zaheerm | why would the d option help, it looks as if its just dialing dtmf digits after channel is answered |
| 12:35.12 | jsharp | Huh. nevermind. The "d" option used to set the call up for minimal latency data calls. |
| 12:35.18 | jsharp | "I guess that got taken out. |
| 12:35.59 | zaheerm | aah ok |
| 12:36.09 | zaheerm | what version of asterisk had it in? |
| 12:37.28 | jsharp | old. Like 1.0.7. |
| 12:37.34 | zaheerm | ok |
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| 12:42.39 | zaheerm | jsharp, do you know if theres an alternative? |
| 12:42.49 | jsharp | Not off the top of my head, unfortunately. |
| 12:42.54 | tzafrir | zaheerm, keyword for searches: fax |
| 12:43.06 | zaheerm | tzafrir, ok thx |
| 12:43.42 | tzafrir | zaheerm, generaly people rarely connect modems, but a fax is commonly connected. And has relatively similar strict quality requirements |
| 12:44.26 | *** join/#asterisk eset (n=eset@212.26.190.77) |
| 12:44.27 | Corydon76-home | That problem with the TDM driver was already corrected. Are you sure you're running the latest zaptel? |
| 12:44.55 | eset | hey, i have a 2.4 GHz P4, and I wondering how many it could hold simultaneously in a meeting room, i know this is a 'how long is a length of string question' but any hints to the magnitude? 5? 10? 20? 50? |
| 12:46.47 | zaheerm | 1. The timing sync of the cards is not going to work as desired: You are taking timing from the telco (assumably) on one card, and in order for the fax bridge to work properly, you'd have to re-send that timing signal out the other card. The timing coming out of the other card (the pri_net span) is not synced to the other span ? it is being generated by the internal clock on the card itself. Due to this, the timing sync is not making it from the te |
| 12:46.47 | zaheerm | lco to the other PBX. It is possible to sync spans across cards only with the 2 and 4-span cards using a timing cable between them. It is also possible to sync timing if you had a single dual-span card servicing both the E1s instead. |
| 12:46.56 | zaheerm | hmm |
| 12:47.30 | zaheerm | that is to do with PRIs right? |
| 12:48.01 | jsharp | eset: Depends on how the calls are coming in? PRI? SIP (what codec)? |
| 12:48.33 | jsharp | zaheerm: Yes. |
| 12:48.37 | eset | sip, probably speex |
| 12:48.50 | eset | no other load on the server |
| 12:49.34 | jsharp | Not many. Speex is pretty CPU hefty. |
| 12:49.38 | *** join/#asterisk heka (n=Mango@80.80.175.130) |
| 12:50.03 | eset | is there a more efficent codec then? can use any available, no prefs |
| 12:50.11 | jsharp | ulaw? |
| 12:50.16 | eset | ok |
| 12:50.19 | eset | sure |
| 12:50.20 | eset | hehe |
| 12:50.30 | eset | and then, any idea of how many connections? |
| 12:51.09 | jsharp | Maybe 30 to 50. |
| 12:51.30 | eset | ooo |
| 12:51.33 | eset | that sounds good |
| 12:51.40 | eset | thanks for that |
| 12:51.44 | eset | :) |
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| 13:09.46 | zaheerm | can i force a native bridge with zaptel and 2 ports on the tdm400p? |
| 13:13.06 | Ahrimanes | anyone here have a Cisco Call Manager? |
| 13:13.18 | zaheerm | i found this which is encouraging: http://groups.google.com/group/Asterisk-users/browse_frm/thread/694287b2ef47a292/9851410b6adaf7c6?lnk=st&q=modem&rnum=1#9851410b6adaf7c6 |
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| 13:25.53 | fourcheeze | should everyone be upgrading to 1.2.7 or newer wrt to format_jpeg vulnerability? |
| 13:26.26 | luke-jr_ | wtf is jpeg used for? |
| 13:26.41 | fourcheeze | it's an image format |
| 13:27.28 | luke-jr_ | and Asterisk deals with audio... |
| 13:27.39 | fourcheeze | it does? |
| 13:28.20 | luke-jr_ | telephony is audio |
| 13:28.31 | jsharp | No, its more of a softswitch...it has the framework to switch audio and video. Just the video stuff isn't fleshed out yet. |
| 13:29.00 | fourcheeze | bah, softswitch is just one of those names invented by marketing people |
| 13:29.16 | jsharp | True, but its fairly accurate. |
| 13:29.25 | fourcheeze | yeah, maybe |
| 13:29.35 | fourcheeze | I think of it as the telephony equivalent of apache |
| 13:29.44 | fourcheeze | it receives requests and does stuff |
| 13:29.52 | fourcheeze | often returning a response |
| 13:30.26 | fourcheeze | in addition you can make phone calls with it ;-) |
| 13:31.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.227.169) |
| 13:31.31 | DarKnesS_WolF | i have a fast question how to create and SIP ext. number when u call it u go to IVR menu called ivr ? |
| 13:32.06 | Katty | morning |
| 13:32.15 | jsharp | Hi Katty |
| 13:32.27 | [TK]D-Fender | Katty: mew. |
| 13:32.48 | Katty | [TK]D-Fender: mew. |
| 13:33.26 | DarKnesS_WolF | i think i can use GoTO |
| 13:34.21 | [TK]D-Fender | b00m |
| 13:35.31 | fourcheeze | DarKnesS_WolF: I don't think you specifically mean a "SIP" extension do you? |
| 13:35.31 | PakiPenguin | omfg |
| 13:35.31 | PakiPenguin | what happened? |
| 13:35.32 | fourcheeze | IRC has been going for years and still this happens ;-) |
| 13:35.32 | DarKnesS_WolF | fourcheeze: nop i mean it i'm testing local still very n00b to asterisk . |
| 13:36.07 | DarKnesS_WolF | fourcheeze: so i want to create when from SIP ext. u call like 105 u go to the IVR menu |
| 13:36.07 | fourcheeze | DarKnesS_WolF: ok, but a SIP extension would be associated with a SIP device |
| 13:36.08 | fourcheeze | ahhh *from* a sip extension |
| 13:36.08 | DarKnesS_WolF | fourcheeze: good point |
| 13:36.08 | fourcheeze | so you want your sip extension to be able to call an IVR |
| 13:36.11 | DarKnesS_WolF | so i think extn => 105,GoTO(s,1) |
| 13:36.19 | DarKnesS_WolF | fourcheeze: yes |
| 13:36.29 | fourcheeze | yeah, or I think you can use Dial(Local/ivr,60) |
| 13:36.42 | DarKnesS_WolF | what is this 60 for? |
| 13:36.51 | DarKnesS_WolF | and what is local ? |
| 13:36.56 | fourcheeze | 60 seconds to pick up, not very relevant for an IVR |
| 13:36.59 | *** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net) |
| 13:37.07 | fourcheeze | local is very sexy |
| 13:37.10 | fourcheeze | I'm just getting the hang of it |
| 13:37.36 | fourcheeze | basically puts in another call as though you had dialled it |
| 13:37.54 | SexyKen | Hey guys -- I am implementing a Queue Status script for my XML phones -- I'm doing this for about 10 phones -- is this going to affect performance? (Manager API will be logged into every 10 seconds by 10 different phones) |
| 13:38.06 | DarKnesS_WolF | fourcheeze: ok i'll try 1st the thing i said then i'll try urs :) |
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| 13:38.16 | [TK]D-Fender | b00m!!!!!!!!! |
| 13:38.17 | Katty | splitty. |
| 13:38.27 | fourcheeze | DarKnesS_WolF: they should work just the same |
| 13:38.29 | DarKnesS_WolF | lol |
| 13:38.54 | [TK]D-Fender | SexyKen : Don't do it :) |
| 13:39.06 | SexyKen | D-FEnder, are you serious? |
| 13:39.12 | DarKnesS_WolF | fourcheeze: the real problem is i'm trying to start with RealTime |
| 13:39.15 | [TK]D-Fender | SexyKen : Mine did that for 5 phones and you're going to give * an aneurism.... |
| 13:39.17 | DarKnesS_WolF | so it's pain ;- |
| 13:39.19 | DarKnesS_WolF | );-) |
| 13:39.24 | fourcheeze | DarKnesS_WolF: realtime is cool |
| 13:39.31 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:39.34 | SexyKen | D-Fender -- then what do you do now? You said you have queue stats using the API |
| 13:39.35 | fourcheeze | but don't tell anyone I said that |
| 13:39.46 | [TK]D-Fender | SexyKen : Set up a cron job on 10s that builds a STATIC page every 10s that gets POLLED by all your phones. |
| 13:39.58 | [TK]D-Fender | SexyKen : I do *that* |
| 13:40.06 | fourcheeze | DarKnesS_WolF: however not even I'm mad enough to do realtime extensions |
| 13:40.20 | [TK]D-Fender | so "live" is potentially a few odd seconds off... whatever.. |
| 13:43.00 | DarKnesS_WolF | fourcheeze: mine didn't work ;-) i got piriorty s,1 must be a number > 0 |
| 13:43.10 | DarKnesS_WolF | and the piriroty is 1 and the appdata is s,1 |
| 13:43.12 | DarKnesS_WolF | i'll try urs |
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| 13:49.00 | DarKnesS_WolF | fourcheeze: didn't work also, i feel i' missing something |
| 13:49.47 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 13:52.03 | coppice | when you call a number it would be a really good idea if it fed back to you the time at the receiving end :-) |
| 13:53.40 | wunderkin | can anyone here explain what the debouncetimer for a T1 does? |
| 13:54.03 | *** join/#asterisk zaheerm (n=zaheer@core.fluendo.com) |
| 13:54.13 | coppice | it debounces bit errors on the CAS bits |
| 13:54.23 | DarKnesS_WolF | fourcheeze: i did the ivr menu with realtime ? |
| 13:55.02 | wunderkin | coppice, when it receieves something bad? it sets the threshold of how long it waits before it spazzes? |
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| 13:57.23 | mut | anyone know if you can format a number in excel to do 3 decimals from an INT? like i have the number 384 i wanna change it to 0.384, or the number 4 change to 0.004 |
| 13:57.47 | coppice | basically you should not accept a single change on the CAS bits as valid, as there might be a bit error on the T1. you look for multiple consistent indications of a CAS change |
| 13:57.58 | [TK]D-Fender | mut : =A1 / 1000 ? |
| 13:58.11 | wunderkin | coppice, so the debouncetime sets how long it waits |
| 13:58.14 | mut | omg |
| 13:58.17 | mut | wtf i'm retarded |
| 13:58.21 | coppice | yep |
| 13:58.35 | [TK]D-Fender | in you go! |
| 13:59.09 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 13:59.12 | [TK]D-Fender | Like I always say : There are 3 kinds of people ; those that know math, and those that don't! ;) |
| 13:59.42 | DarKnesS_WolF | [TK]D-Fender: this is stolen :P |
| 13:59.50 | wunderkin | coppice, k, thanks, mark asked me to 'set it to 300 and adjust from there' but i don't know in what increments and which way, i assume i should go up if still having the problem then; i get intermittant red alarms on all channels for 5 sec, this happens maybe 4-5 times a month normally but it has been occuring a couple times a night from apr 20-apr 24 which coincides with the starting of their 'equipment problem outages' |
| 14:00.02 | [TK]D-Fender | DarKnesS_WolF : I like to think "borrowed" |
| 14:00.08 | DarKnesS_WolF | hehe i hate ivr + realtime ! menus in ext. table correctly but i can't create an ext. to call it ! dial and goto didn't work. |
| 14:00.16 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:00.27 | [TK]D-Fender | DarKnesS_WolF : Dump your table into a pastebin.... |
| 14:00.30 | [TK]D-Fender | ~pb |
| 14:00.32 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 14:00.39 | DarKnesS_WolF | [TK]D-Fender: ok |
| 14:01.04 | wunderkin | coppice, sometimes they see me down, and sometimes they don't, they always test clean to me, and i can loop the card and test fine to another machine so i think this must be what is wrong; why i have to set a value, i don't know |
| 14:01.28 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.com/680927 |
| 14:01.58 | DarKnesS_WolF | [TK]D-Fender: remember i'm still playing around |
| 14:03.44 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 14:04.42 | wunderkin | and when qwest, the local lec, comes to take the readings from the niu, the errors are coming from my side |
| 14:06.38 | DarKnesS_WolF | [TK]D-Fender: so how can i use the dial or goto to call the IVR menu on number like 105 ? |
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| 14:07.51 | *** join/#asterisk baka123 (n=test@84.12.45.147) |
| 14:08.49 | baka123 | Hello, does any one have any tips on how to start rearching joining a fresh asterix box to a Siemens HG1500 IP card? I've got as far as enabling oh323 support... |
| 14:09.00 | baka123 | *asterisk :) sorry |
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| 14:12.16 | Druken | asstrick ? |
| 14:12.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.227.169) |
| 14:13.06 | [TK]D-Fender | DarKnesS_WolF : Can you Dial 101 from your phone? |
| 14:13.20 | coppice | I went to that site, and there was no pr0n there. weird :-) |
| 14:13.32 | Druken | [TK]D-Fender: i'd dial 101, but i'd get myself :) |
| 14:13.47 | *** join/#asterisk b00mer_ (i=fwuser@blackhole.c5i.com) |
| 14:13.51 | Katty | anyone heard of this vpro chip yet? |
| 14:14.11 | DarKnesS_WolF | [TK]D-Fender: 101 is my phone and yes i can |
| 14:14.26 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:14.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:14.40 | Vagabond | Katty: I just saw an article on it, but didn't read it |
| 14:15.39 | Katty | Vagabond: i found the white papers for it, but it's mostly propaganda :< |
| 14:16.00 | [TK]D-Fender | dark well make another entry in that context to do default | 999 | 1 | Goto | ivr,s,1 |
| 14:16.05 | jsharp | Sounds like its a processor dedicated to running antivirus/antispyware. |
| 14:16.06 | jsharp | Oy. |
| 14:16.22 | Katty | it has some remote management stuff. |
| 14:16.27 | DarKnesS_WolF | [TK]D-Fender: ok 1 min |
| 14:17.31 | DarKnesS_WolF | [TK]D-Fender: Apr 25 16:17:48 WARNING[10289]: pbx.c:6510 ast_parseable_goto: Priority 'ivr,s,1' must be a number > 0, or valid label |
| 14:18.32 | coppice | VPro sounds like BoC - bullshit on chip :-) |
| 14:18.33 | [TK]D-Fender | change the ","s for "|" |
| 14:18.42 | Katty | coppice: i'll second that. |
| 14:18.57 | Vagabond | heh |
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| 14:22.03 | DarKnesS_WolF | [TK]D-Fender: Apr 25 16:21:47 WARNING[10331]: pbx.c:2354 __ast_pbx_run: Channel 'SIP/101-6c3c' sent into invalid extension 's' in context 'ivr', but no invalid handler |
| 14:22.06 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 14:22.41 | [TK]D-Fender | DarKnesS_WolF : pastebin your current table |
| 14:22.45 | sevard | does anyone know if PCI-X slots are backwards compatible with PCI cards? |
| 14:23.09 | Qwell | sevard: somewhat |
| 14:23.27 | sevard | Qwell: I have a TDM400P and a PCI-eXpress slot :/ |
| 14:23.41 | Qwell | sevard: call Digium and ask if it'll work |
| 14:23.43 | Qwell | Some do |
| 14:23.55 | sevard | hurray for being dyslexis |
| 14:24.07 | sevard | erm, dyslexic |
| 14:24.13 | [TK]D-Fender | EXACTLY ;) |
| 14:24.18 | sevard | ;\ |
| 14:24.32 | [TK]D-Fender | Dyslexics of the world UNTIE! |
| 14:24.37 | sevard | heh. |
| 14:24.39 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.com/680970 |
| 14:25.08 | file | PCI-X and PCI Express are two separate things |
| 14:25.17 | coppice | hermits of the world unite! |
| 14:25.37 | file | won't work in PCI Express fo shizzle |
| 14:25.41 | LostFrog | Hermit crabs of the world, pinch! |
| 14:26.03 | Qwell | pcie == pci express |
| 14:26.03 | [TK]D-Fender | DarKnesS_WolF : Looks kinda right... unless its a parameter formatting thing I don't know what to say. |
| 14:26.12 | coppice | won't even fit in a PCI-E slot, without a really good push |
| 14:26.25 | DarKnesS_WolF | [TK]D-Fender: shoot me in the head :-D |
| 14:26.36 | sevard | coppice: I know that PCI-X cards are backwards compatible with PCI slots. Not the other way around? |
| 14:27.12 | Qwell | sevard: does your box have pcix or pci express? |
| 14:27.12 | coppice | no, you can't plug them in backwards |
| 14:27.24 | Qwell | coppice: backwards? ha |
| 14:27.29 | sevard | Qwell: PCI-X.. PCI Extended |
| 14:27.37 | Qwell | sevard: call Digium |
| 14:28.25 | *** join/#asterisk froguz (n=alvaro@pc-95-155-104-200.cm.vtr.net) |
| 14:28.54 | wunderkin | coppice, thanks for the help |
| 14:33.34 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 14:33.42 | b00mer_ | does anybody know if the ${TRUNK}c thing works as mentioned in the wiki? http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me |
| 14:34.40 | Katty | i love how w2k doesn't check to see if the new gp and old gp are different |
| 14:34.43 | [TK]D-Fender | b00mer_ : their use of "trunk" is just a constant to indicate for you to use whatever tech you want. |
| 14:34.57 | Katty | it just goes alskdflajsdflolzchaningyourlogoffpolicy in event viewer. |
| 14:35.17 | [TK]D-Fender | b00mer_ : And that code sample is 1.2.x, not 1.2.x compatible. |
| 14:36.19 | DarKnesS_WolF | Katty: oh u remind me i need to install w2k with qemu in my debian :-s dah no space yet |
| 14:36.42 | sevard | I just called Digium and a nice guy named mark said "yeah, it should probably work" |
| 14:36.47 | sevard | he had that "uhh, hopefully" tone |
| 14:37.12 | Katty | iDunno: guest what symantec did this morning! |
| 14:37.18 | Katty | iDunno: it deleted a whole 1.2gig pst file! |
| 14:37.20 | austinnichols102 | partnered up with intel |
| 14:37.33 | LostFrog | pst? |
| 14:37.36 | Katty | iDunno: so kind of it to delete all of my co-worker's email. |
| 14:37.42 | LostFrog | Ahh. |
| 14:37.44 | Katty | LostFrog: yes. email.pst foo.pst |
| 14:37.59 | austinnichols102 | katty: got your mcafee update waiting right here |
| 14:38.51 | Katty | austinnichols102: screw mcafee. |
| 14:39.01 | austinnichols102 | ouch |
| 14:39.04 | Katty | austinnichols102: i'll take symantec corporate edition any day |
| 14:39.05 | DarKnesS_WolF | no help with this realtime and ivr thing for me :(? |
| 14:39.23 | iDunno | Katty: ohh, sounds like symantec that - on the plus side, it deleted a pst file - it probably was full of viruses ;) |
| 14:39.28 | Katty | austinnichols102: even if it does screw up and delete a pst file every now and then. lots better than mcafee ever does. |
| 14:40.08 | austinnichols102 | katty: I don't agree with that at all. |
| 14:40.32 | austinnichols102 | we're constantly displacing symantec - especially in the enterprise marketspace |
| 14:40.37 | Katty | austinnichols102: you don't have to. |
| 14:40.41 | austinnichols102 | true |
| 14:40.45 | Katty | austinnichols102: i really don't care what you think (= |
| 14:40.53 | austinnichols102 | no need to get nasty |
| 14:40.58 | Katty | i'm not being nasty. |
| 14:41.01 | Katty | just telling it like it is. |
| 14:41.17 | austinnichols102 | yeah, whatever |
| 14:41.56 | iDunno | awww. |
| 14:43.00 | *** join/#asterisk aSaDo (n=a@200.68.82.185) |
| 14:43.25 | aSaDo | hi everyb |
| 14:43.39 | dlynes | austinnichols101: deleting a bit of email isn't really a big deal if you've got backups...especially if that email was infected |
| 14:43.57 | austinnichols102 | yeah, but deleting the whole .pst isn't really cool |
| 14:44.15 | dlynes | austinnichols101: just restore it from a backup |
| 14:44.18 | aSaDo | i m working with asterisk 1.2.0 and want to change the format of the name of the files of the agents recordings |
| 14:44.23 | aSaDo | anyb know how? |
| 14:44.24 | *** join/#asterisk Twister (n=jason@216.30.232.106) |
| 14:44.46 | austinnichols102 | dlynes: and lose a whole days mail |
| 14:44.50 | [TK]D-Fender | Katty : 1.2gig? Big deal... I've got Lotus Notes mail files here topping 3 gig! |
| 14:45.13 | aSaDo | &list |
| 14:45.14 | Katty | [TK]D-Fender: this is not a game of who has the biggest email file wins :P |
| 14:45.29 | [TK]D-Fender | .... damn.... |
| 14:46.08 | dlynes | austinnichols101: besides, if email is that important to you, shouldn't you be using imap, and do virus scanning on the mail as it's coming in? |
| 14:46.15 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 14:46.43 | Katty | dlynes: i love how you've not been paying attention (= |
| 14:46.43 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 14:46.53 | Katty | dlynes: austinnichols102 isn't having /any/ email issues |
| 14:46.56 | DarKnesS_WolF | [TK]D-Fender: i got it to work ! |
| 14:46.59 | dlynes | Katty: I know...you are |
| 14:47.08 | DarKnesS_WolF | [ivr] |
| 14:47.08 | DarKnesS_WolF | switch => Realtime/ivr@extensions |
| 14:47.09 | austinnichols102 | tks Katty |
| 14:47.17 | DarKnesS_WolF | i was missing this in the extensions.conf file ;-) |
| 14:47.25 | Twister | i know this wouldnt be very efficient but i was wondering if using 2 x100p cards instead of a tdm400p was possible, i only have 2 lines, its a low call environment, and if it is possible whats the performance issues i might expect? |
| 14:47.34 | tzanger | Twister: yes you can do that |
| 14:47.50 | tzanger | you'll have twice the interrupt load but it should be fine for most systems |
| 14:48.12 | tzanger | and you may have more trouble trying ot get both on separate IRQs but hey, experiment and see what you can get to :-) |
| 14:48.51 | Twister | ok, not a big load on the system, 4 phones and 2 incomming lines (very very small office setup) |
| 14:48.55 | dlynes | Katty: besides...doing imap with maildir lessens the amount of damage that will be done if a virus scanner deletes mail after the fact; doing virus scan on the incoming mail lessens the chance even more |
| 14:48.56 | [TK]D-Fender | DarKnesS_WolF : Guess you have to explicitly export it... |
| 14:49.10 | [TK]D-Fender | DarKnesS_WolF : Realtime = yeh ok, fine, sure, whatever... |
| 14:49.13 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 14:49.18 | Twister | thank you tzanger |
| 14:49.18 | [TK]D-Fender | DarKnesS_WolF : How big a setup are you planning? |
| 14:49.19 | Katty | dlynes: you've just got it all figured out don't you |
| 14:49.31 | Katty | dlynes: too bad Real Life doesn't work that way. |
| 14:49.52 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 14:49.53 | DarKnesS_WolF | [TK]D-Fender: hummm my dear is SIP /AIX / Skype Gateway / other VOIP gateway / IVR /PBX internal system for a company like 20 node |
| 14:49.54 | dlynes | Katty: pst files are for users that are using pop mail accounts, are they not? |
| 14:50.01 | sevard | I just installed the TDM400P in a PCI-X slot. I don't smell burning yet. |
| 14:50.05 | Katty | dlynes: newp. |
| 14:50.09 | DarKnesS_WolF | [TK]D-Fender: but i'm still very n00b as u can see and wann do it all realtime ;-) |
| 14:50.09 | [TK]D-Fender | DarKnesS_WolF : Ok, maybe practical at that size :) |
| 14:50.12 | Katty | dlynes: they sure aren't |
| 14:50.31 | dlynes | Katty: Thought they were Outlook/Outlook express mail files? |
| 14:50.52 | Katty | dlynes: yes. |
| 14:50.54 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 14:51.36 | dlynes | Katty: Well, if it's imap, nothign should be getting downloaded, so it shouldn't be 1 or 2GB's |
| 14:51.47 | Katty | dlynes: who said we're using imap? |
| 14:52.11 | dlynes | Katty: what else is there besides imap and pop3? |
| 14:52.12 | Katty | dlynes: and if it is over a gig, then /clearly/ we're not. m'kay |
| 14:52.32 | dlynes | Katty: Exchange is all server-side too, right? |
| 14:52.50 | Katty | dlynes: no. |
| 14:53.00 | dlynes | Katty: ahhh... |
| 14:53.04 | tzanger | what else besides IMAP and POP3? how about Exchange's MAPI, Notes, what else... |
| 14:53.07 | Katty | dlynes: exchange is a royal pain in the tail. |
| 14:53.15 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.183.Dial1.SanJose1.Level3.net) |
| 14:53.23 | dlynes | Katty: Thought it was a microsoft alternative to imap |
| 14:53.35 | Katty | dlynes: no. |
| 14:53.41 | Katty | dlynes: imap doesn't do notes and calendar and tasks, etc. |
| 14:53.49 | Katty | dlynes: very different. |
| 14:54.05 | dlynes | Katty: imap has the provision to do all that...just most imap servers don't implement it |
| 14:54.12 | [TK]D-Fender | Katty : phpGroupware <- |
| 14:54.24 | Katty | let's not make fixit solutions |
| 14:54.25 | Katty | i'm ranting |
| 14:54.32 | [TK]D-Fender | I love OSS collaboration... |
| 14:54.48 | [TK]D-Fender | Katty : Oh I'm sorry... I forgot "the rules" |
| 14:54.52 | Katty | [TK]D-Fender: now all together now! |
| 14:54.55 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:54.57 | Katty | [TK]D-Fender: exchange should DIE |
| 14:54.59 | Ariel_ | morning folks |
| 14:55.08 | *** join/#asterisk saftsack (n=saftsack@IP-213188106101.dialin.heagmedianet.de) |
| 14:55.11 | Katty | Ariel_: morning, glory. |
| 14:55.18 | dlynes | notes is all server side, or is it all client side, too? |
| 14:55.26 | jsharp | She is Katty, hear her roar. |
| 14:55.33 | Katty | jsharp: yes. |
| 14:55.41 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 14:55.48 | Katty | [TK]D-Fender: don't forget the hug! |
| 14:55.52 | saftsack | if i can not type in a cu window connected to a modem what could i do to do so? |
| 14:56.00 | Katty | [TK]D-Fender: hugs play a vital role in the Nodding and Understanding |
| 14:56.01 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.183.Dial1.SanJose1.Level3.net) |
| 14:56.01 | saftsack | can the modem be just damaged? |
| 14:56.02 | [TK]D-Fender | Katty : Its was there! |
| 14:56.13 | Deep6 | guys is it possible to create extensions off a tapped out Nortstar system with Asterisk? |
| 14:56.18 | Katty | [TK]D-Fender: yes, it was. |
| 14:56.24 | Katty | [TK]D-Fender: just never ever forget it (= |
| 14:56.27 | Katty | [TK]D-Fender: tis very important. |
| 14:57.00 | Katty | dlynes: hence .pst |
| 14:57.14 | [TK]D-Fender | Katty : No, I've learned too much to leave those bits out :) |
| 14:57.15 | Katty | dlynes: i hereby table exchange discussion |
| 14:57.30 | Katty | [TK]D-Fender: good man :> |
| 14:57.38 | *** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx) |
| 14:57.41 | znoG | could rxgain/txgain be affecting the fax reception with iaxmodem/hylafax) |
| 14:57.42 | znoG | ? |
| 14:57.45 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:57.48 | [TK]D-Fender | Katty : Yes, Last place seems to be a familiar place with me... |
| 14:58.39 | salviadud | has anybody seen paulo? |
| 14:58.47 | salviadud | he's this brazilian dude |
| 14:58.53 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:59.02 | salviadud | i think it's _paulo_ |
| 14:59.11 | salviadud | ? |
| 14:59.15 | caio1982 | salviadud: try #asteriskbrasil.org to meet the .br folks |
| 14:59.59 | Deep6 | can anyone advise me whether you can get a nortstar merdian MICS to talk to asterisk? |
| 15:00.19 | brettnem | "talk" |
| 15:00.26 | Katty | [TK]D-Fender: nice guys finish last... |
| 15:00.26 | brettnem | "hello friend" |
| 15:00.30 | Katty | [TK]D-Fender: but they're the happiest, me thinks |
| 15:00.45 | salviadud | being nice doesn't cut it in this world |
| 15:00.52 | [TK]D-Fender | Deep6 : PRI card, FXS ATA on an analog line trunk, FXO on an analog extension, and Intel made a direct SIP/Norstar gateway |
| 15:01.00 | [TK]D-Fender | Katty : BS |
| 15:01.00 | *** join/#asterisk tomtom_ (n=tom@83.217.70.166) |
| 15:01.05 | salviadud | i stopped being nice a while ago, being bad rocks |
| 15:01.07 | Katty | [TK]D-Fender: oh? |
| 15:01.12 | saftsack | what to do if my faxmodem doesnt react after plugging it into the serialport? |
| 15:01.14 | tomtom_ | hi |
| 15:01.16 | [TK]D-Fender | Katty : Vindicated is still ALONE. |
| 15:01.21 | Katty | [TK]D-Fender: true :< |
| 15:01.36 | saftsack | i did cu -l ttyS0 and then connected is showed up. but then it freezes |
| 15:01.55 | Deep6 | [TK]D-Fender, have you got any further info on said solution |
| 15:02.00 | [TK]D-Fender | The price for my being always right is my tendency for it to be more about the negative than the positive. |
| 15:02.06 | Deep6 | our norstar is full and we need 2 more extensions |
| 15:02.20 | [TK]D-Fender | Omniscience should come with a SPOON. |
| 15:02.47 | [TK]D-Fender | Deep6 : If you need more extensions, its time to pay for another line card for it. |
| 15:02.50 | riddlebox | is there a way to use the dial pad to spell things, like 21 = a? |
| 15:02.56 | [TK]D-Fender | Deep6 : Describe your setup... |
| 15:03.28 | [TK]D-Fender | riddlebox : if you're talking about in an * script, sure, you can do just about anythig.... |
| 15:04.15 | saftsack | i found the error *duck* |
| 15:04.20 | saftsack | someone plugged out the cable |
| 15:04.34 | jsharp | find them and amputate their hands. |
| 15:05.02 | jsharp | Then put them in a blender and feed it to them. |
| 15:05.51 | vader-- | ok i have all my phones in |
| 15:05.59 | vader-- | but i can't do anything because i don't have the sip firmware |
| 15:06.00 | riddlebox | hello clorise |
| 15:06.03 | [TK]D-Fender | jsharp : No, amputation is typically quick and the shock can let them pass out too quickly. No, put the hands into the blender WHOLE, and SLOW. |
| 15:06.18 | [TK]D-Fender | vader-- : What'd you get? |
| 15:06.24 | vader-- | cisco 7940G |
| 15:06.30 | chiardon | I need some VoIP gateways ... which brand would you recommend? |
| 15:06.40 | [TK]D-Fender | chiardon : What kind of gateways? |
| 15:09.25 | salviadud | netsplit... |
| 15:09.25 | mut | think so? |
| 15:09.26 | caio1982 | i'm not sure about it yet |
| 15:09.26 | caio1982 | can we try that again? |
| 15:09.26 | Deep6 | [TK]D-Fender, you still here? |
| 15:09.51 | puzzled | anyone have a suggestion how I can make asterisk react to the flash button on an analog phone attached to a tdm400p? |
| 15:10.10 | jsharp | press the flash button? |
| 15:10.14 | salviadud | puzzled i think it just works |
| 15:10.21 | salviadud | and goes to music on hold |
| 15:10.29 | jsharp | You have to have certain settings turn on in zapata.conf to make it work, though. |
| 15:10.43 | [TK]D-Fender | Deep6 : yup |
| 15:10.46 | puzzled | jsharp: ah ok. I'll search some more in there |
| 15:11.02 | jsharp | you have to have at minimum threewaycalling=yes in zapata.conf |
| 15:11.15 | puzzled | salviadud: currently not but I hope to get there |
| 15:11.22 | puzzled | jsharp: ok, thanks |
| 15:12.06 | salviadud | sorry puzzled, i usually use sip, not zap |
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| 15:12.47 | sevard | sweet |
| 15:12.52 | sevard | i haven't seen one of these since my dalnet days |
| 15:13.26 | puzzled | salviadud: I use sip and ISDN so that's why I'm a bit lost with this ancient analog stuff |
| 15:13.32 | BugKham | how do you guys view the status of SIP users/peers on the web interface? |
| 15:13.44 | puzzled | which webinterface? |
| 15:13.44 | sevard | what web interface? where? |
| 15:13.53 | sevard | who what |
| 15:14.13 | Nivex | There is no spoon^Wweb interface. |
| 15:14.15 | chiardon | Hello |
| 15:14.24 | puzzled | hi |
| 15:14.33 | jake1932 | BugKham: you may want to check in #freepbx - asterisk does not come with a web interface |
| 15:14.48 | salviadud | you what i hate the most, GUI's! |
| 15:14.52 | BugKham | jake1932: fine |
| 15:14.55 | MikeJ[Laptop] | jake1932, asterisk does to come with a web interface |
| 15:14.57 | salviadud | especially when i have to use 'em... |
| 15:15.07 | sevard | salviadud: ^know |
| 15:15.17 | jake1932 | MikeJ[Laptop]: ok, elighten me |
| 15:15.28 | chiardon | If i'm planning to put in function around 50 VoIP extension . . . wich must be the main considerations in relation with: gateways and CPU!! |
| 15:16.09 | MikeJ[Laptop] | jake1932, asterisk\static-http\ajamdemo.html |
| 15:16.17 | MikeJ[Laptop] | what the heck do you call that? |
| 15:16.47 | jake1932 | don't know yet - i've never heard of that - brb |
| 15:16.51 | jsharp | 50 voip extensions? voip phones or analog phones attached to gateways? |
| 15:16.55 | MikeJ[Laptop] | heh |
| 15:17.10 | sevard | MikeJ[Laptop]: I don't happen to have that |
| 15:17.22 | MikeJ[Laptop] | you don't have trunk then |
| 15:17.37 | sevard | my baby's got back, though |
| 15:18.05 | jsharp | Whatcha gonna do with all that junk in your analog trunk? |
| 15:18.14 | sevard | :)) |
| 15:18.18 | jake1932 | ok - i don't use trunk - so - i know i wouldn't have that |
| 15:18.52 | salviadud | that reminds me |
| 15:18.56 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 15:19.00 | salviadud | ever seen "junk in the trunk"? |
| 15:19.13 | salviadud | it's a funny movie |
| 15:19.36 | sevard | compiling things on 2.6.x is trippy ;/ |
| 15:19.41 | salviadud | raise your hand if you're black |
| 15:19.53 | salviadud | sevard, 2.6 is trippy? |
| 15:20.08 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:20.11 | sevard | salviadud: 2.6 is trippy, man. |
| 15:20.19 | Druken | my laptop is black, does that count? |
| 15:20.31 | salviadud | i got 2.6.12.5 |
| 15:20.37 | salviadud | and 2.6.13 |
| 15:21.03 | salviadud | but, what's the trippy part, compiling the kernel, or compiling other stuff with the kernel? |
| 15:21.10 | jake1932 | MikeJ[Laptop] - I'm trying to find more info on this web thing - i googled it and found minimal info |
| 15:21.24 | salviadud | i love to recompile my kernel. make it light as a feather... |
| 15:21.29 | sevard | salviadud: compiling the kernel is trippy and compiling modules is trippy. |
| 15:21.31 | MikeJ[Laptop] | mark just wrote it very recently |
| 15:21.39 | MikeJ[Laptop] | in the last month or so |
| 15:21.44 | salviadud | sevard, what distro are you using? |
| 15:21.49 | sevard | salviadud: slackware |
| 15:21.56 | salviadud | tssssssss, me too |
| 15:22.00 | MikeJ[Laptop] | think non blocking manager interface via a web page |
| 15:22.05 | sevard | salviadud: OMG LIKE NO WAY |
| 15:22.14 | salviadud | lol |
| 15:22.39 | sevard | salviadud: do you know offhand if there's a slack package for libnewt |
| 15:22.45 | jake1932 | MikeJ[Laptop]: ok - i'm considering putting it up somewhere just to check it out |
| 15:22.45 | sevard | i'm feeling pretty lazy atm |
| 15:22.47 | salviadud | yeah, i go for the manual install make bZimage;make modules;make modules_install |
| 15:23.06 | sevard | salviadud: that's not manual, i convert C into binary by hand. |
| 15:23.27 | salviadud | sevard, o_O |
| 15:23.32 | sevard | 0_0 |
| 15:23.55 | salviadud | and no dude, i'm sorry i don't know if there's a package for that library |
| 15:24.15 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:24.19 | sevard | grr, time to fireup lynx, thanks for not fueling my lazyness |
| 15:24.45 | salviadud | not even using x eh? |
| 15:24.58 | puzzled | weird, doing a flash by pressing the hook works for 3way calling but when pressing the Flash button the asterisk console just spits out "Attempting native bridge of Zap/2-1 and Zap/1-1. Anyone have any ideas? |
| 15:25.17 | sevard | salviadud: heck nah |
| 15:25.34 | sevard | salviadud: what's your deal, i thought you hated guis |
| 15:25.54 | salviadud | i don't hate X |
| 15:26.03 | salviadud | just automated stuff like freepbx |
| 15:26.05 | sevard | i do, clunky piece of crap. |
| 15:26.06 | salviadud | and ubuntu |
| 15:26.12 | bkw__ | puzzled, I think that causes you to jump out of the native bridge and back into it |
| 15:26.34 | [TK]D-Fender | I hate Ubuntu... gimme my god-damned ROOT!!!! |
| 15:26.34 | salviadud | you can't watch pr0n on a terminal window maaaaaaan |
| 15:26.41 | jake1932 | ascii art |
| 15:26.41 | sevard | salviadud: yes you can. |
| 15:26.47 | [TK]D-Fender | salviadud : ASCII pr0n! |
| 15:26.48 | salviadud | yeah, sudo for everthing ubuntu sucks |
| 15:26.57 | Aurs | sudo passwd |
| 15:26.58 | jsharp | sudo bash |
| 15:27.03 | salviadud | i've seen ascii pr0n, it's hilarious |
| 15:27.03 | sevard | salviadud: libcaca or framebuffer mplayer |
| 15:27.04 | austinnichols102 | http://www.asciipr0n.com/pr0n/ |
| 15:27.09 | iDunno | sudo su - |
| 15:27.12 | Aurs | then su - |
| 15:27.22 | *** join/#asterisk ibob63 (n=hp@bb-87-82-7-89.ukonline.co.uk) |
| 15:27.31 | sevard | salviadud: stop dissing tech until you can use it :) |
| 15:27.50 | puzzled | bkw__: thanks but I have no idea what that means :) |
| 15:29.05 | luke-jr_ | can I create a SIP guest account to configure eg, callerid on guest calls? |
| 15:29.14 | SexyKen | Hey guys -- I run Asterisk for my hosting business, and I'd like to let users be able to click on a "Let us call you now" link/button -- and have Asterisk call them and place them in a designated queue. Is this possible? (Web Server & Phone Server on different networks) |
| 15:31.14 | salviadud | your nick is kinda funny... SexyKen |
| 15:31.30 | *** join/#asterisk apardo (n=apardo@231.Red-213-96-100.staticIP.rima-tde.net) |
| 15:31.35 | luke-jr_ | eg, 'setvar=...' |
| 15:31.37 | puzzled | SexyKen: ask Douglas Garstang <dgarstang@oneeighty.com>. he mentioned on the list he had this working |
| 15:31.52 | puzzled | which is a first |
| 15:31.56 | luke-jr_ | SexyKen: for a start, I don't see why you should dial them until they're near top of the queue |
| 15:32.02 | *** join/#asterisk Prival (n=someone@64.235.216.178) |
| 15:32.21 | SexyKen | luke-jr -- Well you bring up another interesting ID> |
| 15:32.24 | SexyKen | *Idea. |
| 15:32.26 | *** part/#asterisk apardo (n=apardo@231.Red-213-96-100.staticIP.rima-tde.net) |
| 15:32.32 | SexyKen | So now it gets more complicated :-) |
| 15:33.21 | Prival | Hi all, I have a customer complaining about callers being cut-off when leaving voicemails. We tried playing with maxsilence, silencethreshold and maxmessage without sucess. Any hints? |
| 15:33.41 | SexyKen | I would now like them to click a "Call me now" link and enter their phone number, choose the queue they'd like to join, and then they'll recieve a message saying "You've been entered into X queue. Your holdtime is X" -- then they'll recieve a call when they're 1st in queue. |
| 15:33.42 | SexyKen | How's that? |
| 15:33.54 | luke-jr_ | Prival: replace voicemail with "stop calling us! aaah!' ? |
| 15:34.17 | sevard | SexyKen: Couldn't that system be easily used for prank calls? |
| 15:34.20 | luke-jr_ | SexyKen: sounds fun |
| 15:34.24 | Prival | luke: :-P |
| 15:34.39 | vader-- | where would i configure the TDM2400 series digium card at? |
| 15:34.47 | vader-- | in the /etc/captel.conf? |
| 15:34.50 | SexyKen | sevard: I would emagine so. But who cares :-) |
| 15:34.53 | salviadud | prank calls rule |
| 15:34.56 | SexyKen | *Imagine |
| 15:35.34 | luke-jr_ | SexyKen: I'll plug your 800 # into that form ;) |
| 15:35.47 | sevard | Heh |
| 15:35.48 | SexyKen | luke: You suggested it. |
| 15:35.50 | SexyKen | Now what? |
| 15:35.56 | chiardon | jsharp: analogues phones atached to the gateway! |
| 15:36.10 | luke-jr_ | SexyKen: I suggested minimizing the call time by starting it at the last minute; it'd be worse otherwise |
| 15:36.17 | sevard | Holy crap! where the heck is the libnewt source code? |
| 15:36.19 | tomtom_ | anyone an experience with the Asterisk TCP patch? |
| 15:36.24 | sevard | I didn't think it'd be _this hard_ to find |
| 15:37.12 | *** part/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 15:37.19 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 15:37.39 | jsharp | chiardon: What was your original question about the gateways? I forget. |
| 15:39.06 | SexyKen | Hey guys -- we use Polycom SIP IP 600's -- can we use their buddies feature? |
| 15:40.16 | sevard | Holy crap! |
| 15:40.59 | [TK]D-Fender | SexyKen : Yup.. up to a point |
| 15:41.12 | *** join/#asterisk nitam (n=none@201.138.73.214) |
| 15:41.14 | SexyKen | Fender -- are you using this now? |
| 15:41.21 | nitam | Hi |
| 15:41.28 | sevard | [TK]D-Fender: Do you know where the libnewt source can be grabbed? I can't freaking find it anywhere |
| 15:41.40 | nitam | does anybody know where can i find information about compiling zapata driver ? |
| 15:41.42 | [TK]D-Fender | sevard : source? |
| 15:42.26 | Vagabond | sevard: I think the gentoo distfile mirrors have it |
| 15:42.27 | sevard | [TK]D-Fender: source code |
| 15:42.41 | [TK]D-Fender | sevard : Depends on things I guess.... SRPMS, etc |
| 15:42.44 | terrapen | <nitam> does anybody know where can i find information about compiling zapata driver ? |
| 15:42.49 | terrapen | oops |
| 15:42.50 | Prival | ANyone have any clues about the voicemail cut-off? |
| 15:42.53 | Vagabond | but yeah, that library really need a homepage |
| 15:43.11 | sevard | Vagabond: it's hard as hell to find |
| 15:43.51 | *** join/#asterisk gmonxx (n=gg@adsl-156-234-59.mia.bellsouth.net) |
| 15:44.14 | gmonxx | can i ftp into a polycom phone to edit the sip.cfg |
| 15:46.17 | sevard | Vagabond: I'm not finding it. |
| 15:46.47 | Vagabond | sevard: me neither |
| 15:47.05 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 15:47.36 | sevard | I'm on slackware and i'm really not up to the rpm process, that's a last resort |
| 15:47.52 | nitam | so ... nobody know ? |
| 15:48.09 | sevard | nitam: http://www.voip-info.org |
| 15:48.15 | tekati | Is there anyway to get around having a digium type card installed for Music On Hold? |
| 15:48.44 | sevard | tekati: ? * does MoH just fine |
| 15:49.54 | [TK]D-Fender | gmonxx : More like the other way around... Polycom's PICKUP their config from FTP. |
| 15:49.57 | salviadud | Katty, what distro are you on? |
| 15:50.09 | [TK]D-Fender | sevard : Slackware has source packages.... |
| 15:50.11 | jsharp | tekati: Use ztdummy |
| 15:50.21 | Katty | salviadud: debian. |
| 15:50.27 | gmonxx | i figured that but i want to change one value in the sip.cfg is there an easy way to do it |
| 15:50.33 | sevard | [TK]D-Fender: libnewt isn't listed in slackware's package datebase |
| 15:50.37 | luke-jr_ | tekati: Music On Hold works fine w/o digium stuff |
| 15:50.51 | luke-jr_ | w/o zaptel, even |
| 15:50.56 | Katty | tekati: way to set off my hilight (= |
| 15:50.57 | sevard | jsharp: iirc you only need ztdummy if you're running 2.4.x |
| 15:50.59 | [TK]D-Fender | sevard : Should be in there somewhere |
| 15:51.37 | salviadud | what was ztdummy for? |
| 15:51.48 | salviadud | i heard it was necessary for conferencing |
| 15:51.50 | luke-jr_ | sevard: 2.4 is still used? |
| 15:52.03 | LostFrog | I thought you still needed ztdummy for 2.6, but you didn't need USB or zaptel hardware. |
| 15:52.09 | Vagabond | sevard: here's the fbsd sources http://www.freebsd.org/cgi/pds.cgi?ports/devel/newt |
| 15:52.09 | sevard | luke-jr_: I used 2.4 up until a month or so ago |
| 15:52.16 | luke-jr_ | salviadud: emulates parts of a digium card, or something |
| 15:52.21 | sevard | [TK]D-Fender: seriously guy, they're not in there. |
| 15:52.33 | Vagabond | they might work ;) |
| 15:52.37 | luke-jr_ | sevard: 2.4 doesn't even support my system |
| 15:52.42 | sevard | Vagabond: that's more than I found.. I doubt if that'll work ;\ |
| 15:53.09 | sevard | luke-jr_: 99% of my PCs are dumpster dives and 2.4 was always good to me. |
| 15:53.58 | tekati | ztdummy it is. Thanks for all the answers. |
| 15:55.29 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 15:55.34 | chiardon | Hi jsharp |
| 15:56.47 | Vagabond | sevard: part of the trick is searching for newt rather than libnewt |
| 15:56.51 | chiardon | I'm needing an urgent system expantion . . . then I was thinking to have aroun 50 new VoIP extension and forget the channel bank ans those stuff |
| 15:58.03 | chiardon | then I need some guide in relation with whta kind of gateways ans wich CPU additional resources this expantion could be demanding? |
| 15:58.38 | DoktorGreg | chiardon, you want at least a dual xeon for that load |
| 15:58.48 | jsharp | If you're going with gateways and run with ULAW codec on the gateways, you won't need much more CPU horse power. |
| 15:58.50 | chiardon | I have it! |
| 15:59.14 | DoktorGreg | well, i take that back |
| 15:59.30 | DoktorGreg | you would want the dual xeon with 50 active channel |
| 15:59.30 | *** join/#asterisk cytrak (n=kvirc@adelphi.geofocus.com) |
| 15:59.38 | cytrak | by any chance anyone here happens to have openvpn server running on the same box as asterisk ? |
| 16:00.04 | jsharp | You only need the dual xeon for 50 active channels if you're transcoding codecs. |
| 16:00.50 | gmonxx | is there an easy way to set volume.persists on polycom phones? |
| 16:01.16 | chiardon | DoktorGreg: now with my actual 56 extentions I'm using the dual xeon . . the must I add another one in the case I will expand to 50 VoIP new wxtwnrtions? |
| 16:01.35 | [TK]D-Fender | gmonxx : its in the provisioning files... |
| 16:01.43 | jsharp | Its all a matter of how many concurrent calls you're going to have. |
| 16:01.46 | [TK]D-Fender | gmonxx : Just 3 values to choose fro... pretty quick. |
| 16:02.01 | jsharp | You can have a single machine handle 1000 extensions, provided only a subset of them are active at any time. |
| 16:02.05 | gmonxx | i set up all the phone through the http admin |
| 16:02.13 | gmonxx | now i have to use a boot server? |
| 16:03.02 | chiardon | jsharp . . around 100 - 120 concurrent calls!! |
| 16:03.09 | [TK]D-Fender | gmonxx : For this, yup! |
| 16:03.19 | jsharp | All your extensions would be active at any given time? |
| 16:03.21 | gmonxx | seriously? |
| 16:03.25 | [TK]D-Fender | gmonxx : HTTP setup for Polycom is a total wasste.... not how its meant to be done. |
| 16:03.34 | [TK]D-Fender | gmonxx : Yes seriously. |
| 16:03.37 | chiardon | jshar . . all the time! |
| 16:04.01 | [TK]D-Fender | gmonxx : I do it at work, home, and for many places I consult. a 5 minute job when you know what you're doing. |
| 16:04.02 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 16:05.58 | jsharp | chiardon: All your extensions are active placing calls to the PSTN? How are you connected to the PSTN? |
| 16:06.13 | chiardon | yeppppppppppp!!!!!! 2 E1s |
| 16:06.47 | jsharp | So run all your gateways at alaw so there's no transcoding needed, and you should be good to go. |
| 16:07.15 | chiardon | jsharp . . but at first you told me ULAW!!! |
| 16:07.30 | jsharp | Cause i blatently assumed you were using T1s. |
| 16:07.42 | chiardon | happpppppppppppppppp!!!!brrrrppp! |
| 16:08.31 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 16:08.34 | chiardon | jsharp...but can I put all my traffic with the VoIP gateways without any kind ao T1/E1 card??? |
| 16:09.00 | jsharp | i'm confused now. |
| 16:09.22 | jsharp | I thought you had E1s already. |
| 16:09.43 | chiardon | jsharp . .yepppp |
| 16:09.57 | *** join/#asterisk ToTo (n=ToTo@host157-211.pool872.interbusiness.it) |
| 16:10.04 | jsharp | Connected to Asterisk? |
| 16:10.19 | chiardon | but thinking to put all the PSTN channels running in the offices with VoIP!! |
| 16:10.30 | vader-- | anyone in here use digium cards? |
| 16:10.47 | chiardon | fuiiiiii . . . .fuifuuuuuuuuuuuu |
| 16:11.09 | vader-- | im wondering if you do a modprobe on a driver does that automatically load the driver? |
| 16:11.45 | jsharp | I'm *really* confused now. |
| 16:12.10 | chiardon | jsharp . . . must go out for a minute... can i reconnect with you after? |
| 16:12.14 | jsharp | Sure. |
| 16:12.25 | chiardon | jsharp . . TIA! |
| 16:13.01 | gmonxx | how long to learn it [TK]D-Fender? |
| 16:13.14 | *** part/#asterisk Prival (n=someone@64.235.216.178) |
| 16:17.46 | *** join/#asterisk btm (n=btm@66.213.193.150) |
| 16:18.02 | sevard | Whoohoo! I think I finally have my TDM400P installed! lspci doesn't list it though, unless it's being reported as 01:01.0 Ethernet controller: Intel Corporation 82547GI Gigabit Ethernet Controller |
| 16:18.06 | sevard | erm |
| 16:18.16 | sevard | 02:02.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface |
| 16:19.12 | jsharp | That's it. |
| 16:19.31 | sevard | rad. |
| 16:19.37 | sevard | dmesg is showing Zaptel |
| 16:20.30 | sevard | Is it common for dmesg to show Auto FXO when I have FXS modules installed? |
| 16:21.53 | [TK]D-Fender | sevard : FXS cards use FXO signalling |
| 16:21.58 | sevard | oh yes |
| 16:21.59 | sevard | duh |
| 16:22.02 | jsharp | Sure you've got FXS modules? And you have the power connector connected? |
| 16:22.30 | sevard | I'm pretty sure I have FXS modules and yes the rail is connected to the molex. |
| 16:23.38 | sevard | so if I have some SIP 2002s and I want to plug them into the TDM400P just for testing, the modules on the TDM400P need to be FXS right, because the connecters on the back of the SIP 2002 is also FXS |
| 16:23.57 | jsharp | FXS plugs into FXO. |
| 16:24.01 | jsharp | And vice versa. |
| 16:24.13 | sevard | That' |
| 16:24.15 | [TK]D-Fender | sevard : SPA-2002 = FXS, your TDM = FXS = FAILURE |
| 16:24.25 | sevard | That's what I was saying to the sales guy but he said I was wrong. |
| 16:24.38 | jsharp | Do you have red or green modules on your TDM400? |
| 16:24.46 | sevard | I have red modules |
| 16:25.38 | jsharp | Then you have FXO modules. |
| 16:25.47 | sevard | alriiiiiiiiiiiiiiiiiight. |
| 16:26.26 | sevard | so the SIP 2002 is FXS which uses FXO signaling the TDM400P is FXO and uses FSX signaling, all is well? |
| 16:26.26 | jsharp | So then you can go SPA-2002 to TDM400 without a problem. |
| 16:26.37 | jsharp | Exactly. |
| 16:27.08 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 16:27.23 | *** join/#asterisk RoyKa (n=roy@80.239.107.70) |
| 16:29.03 | redondos | Can I use the official zaptel driver for x100p clones? |
| 16:29.15 | jsharp | You can try. |
| 16:29.16 | redondos | Or should I use a patched one... (I forgot what I did when I bought it) |
| 16:29.24 | redondos | It isn't working, the official one. |
| 16:30.06 | aSaDo | i m workin with a polycom ip301 with asterisk@home 2.7 and it has an awful echo problem, any1 know how can i solve it? |
| 16:32.10 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 16:33.40 | saftsack | is it possible to send the number of the caller if i transfer the call? |
| 16:34.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 16:34.37 | shmaltz | on a debian system what is the apt-get install command to install zaptel 1.2.5? |
| 16:35.09 | Hmmhesays | apt-cache search zaptel |
| 16:35.26 | Hmmhesays | apt-get install <name of package> |
| 16:35.39 | *** join/#asterisk chiardon (n=chiardon@200.71.58.39) |
| 16:36.38 | chiardon | hi jsharp . . . are you busy? |
| 16:37.18 | jsharp | Nope. |
| 16:37.30 | sevard | jsharp: then configure my zaptel.conf for me |
| 16:37.37 | sevard | heh, jk, but seriously. mad confusing |
| 16:37.44 | jake1932 | <PROTECTED> |
| 16:37.54 | shmaltz | Hmmhesays, but that search doesn't show 1.2.x, only the older ones |
| 16:38.16 | sevard | The devil uses emacs. |
| 16:38.24 | jsharp | The devil is my bitch. |
| 16:38.36 | chiardon | jsharp . . . my situation: i nedd to add around 50 extensions to my system . . and I'm planning to do it with VoIP only!! |
| 16:38.38 | sevard | I like pico and vi :) |
| 16:38.46 | jsharp | What is your system now? |
| 16:38.47 | sevard | I've always hated emacs. |
| 16:39.00 | salviadud | pico is nice |
| 16:39.08 | sevard | Pico is effing great. |
| 16:39.09 | Hmmhesays | zaptel.conf isn't confusing |
| 16:39.10 | jake1932 | nano = pico? |
| 16:39.11 | chiardon | jsharp: in tha case wich are the main considerations and the moore apropiate hardware?? |
| 16:39.13 | Hmmhesays | VI ALL THE WAY |
| 16:39.15 | sevard | Hmmhesays: sure sure. |
| 16:39.25 | saftsack | if a person calls me now in this example named P1 he has the number 123456. now i want to transfer him to a second telephone. if i do so i want that his number is showed on the second telephone |
| 16:39.28 | frawd | vi also |
| 16:39.30 | sevard | jake1932: nano is a similar editor, however pico is a part of the pine package. |
| 16:39.35 | jake1932 | ok |
| 16:39.40 | Hmmhesays | nano is ok |
| 16:39.45 | jsharp | chiardon: What do you have running right now? 2 E1s connected to what? |
| 16:39.53 | sevard | nano does syntax highlighting iirc which is much cooler |
| 16:40.03 | salviadud | yeah, maddox would be proud |
| 16:40.04 | sevard | I haven't played with it much but I might be switching |
| 16:40.09 | Hmmhesays | and it makes me coffee n shit |
| 16:40.24 | chiardon | jsharp: 2 E1, TE400p, dialplan with 60 extentions plus 2 Atas! |
| 16:40.31 | jsharp | nano makes you shit? I'd cut back, then. |
| 16:40.37 | sevard | Hmmhesays: I don't know how much more confusing it could be for a guy with no telco background |
| 16:40.40 | chiardon | ER1 s conecte to channel banks |
| 16:40.49 | Hmmhesays | I have not much of a telco background |
| 16:41.00 | sevard | ass monkey. |
| 16:41.08 | chiardon | jsharp: sorry E1s connectes two channels banks |
| 16:41.20 | jsharp | 2 E1s from your telco, 2 E1s to channel banks? |
| 16:41.59 | chiardon | 2 E1s from the telco with 2 channel banks conected and some ATAs |
| 16:42.41 | chiardon | jshap: dial plan 60 extensions |
| 16:42.57 | jsharp | Go with a couple of SIP gateways run in alaw mode. Bang, problem solved. |
| 16:43.26 | jsharp | Minimal CPU overhead needed, since you'll not be transcoding anywhere. |
| 16:43.30 | sevard | Hmmhesays: Oww my knee.... pansy. |
| 16:43.54 | Hmmhesays | lol |
| 16:44.27 | sevard | Hmmhesays: help me with zaptel.conf and I won't sue for knee replacement. |
| 16:46.14 | *** join/#asterisk Creathir (n=Creathir@207.71.17.206) |
| 16:46.36 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net) |
| 16:46.47 | nitam | does anybody tryed to compile zonedata.c alone ? |
| 16:46.50 | Hmmhesays | pay me and i'll help you |
| 16:46.50 | Creathir | hello everyone |
| 16:46.58 | sevard | Hmmhesays: Can I pay you in sexual favors? |
| 16:47.01 | *** join/#asterisk nothinman (i=shakey@adam100.neoplus.adsl.tpnet.pl) |
| 16:47.07 | Hmmhesays | greenbacks, not brokebacks |
| 16:47.14 | jsharp | Only if you clean up the channel afterwards. |
| 16:47.16 | sevard | hahaha |
| 16:47.32 | Hmmhesays | what problems are you having |
| 16:47.43 | nothinman | Masters, how can I set variable with spaces? (from AGI script -- something like SET VARIABLE this is my variable) |
| 16:47.43 | sevard | telco lingo |
| 16:47.48 | sevard | Masters!haha |
| 16:47.58 | nothinman | I've tried qutes but it doesn't work... |
| 16:48.02 | *** join/#asterisk alib80 (n=chatzill@196.211.66.154) |
| 16:48.21 | nothinman | sevard: "people who know more about asterisk than I do" ;) |
| 16:48.42 | Hmmhesays | why would you want to do that |
| 16:48.47 | tzafrir | nitam, what for, exactly? |
| 16:49.04 | chiardon | jsharp . . OK . .but what brand name VoIP gateways you recommend?? and what other considerations related with the hardware? |
| 16:49.20 | tzafrir | nitam, why not use libtonezone? |
| 16:49.28 | nothinman | I want to change callerid from script to something with spaces :/ |
| 16:49.31 | Creathir | I have a question about zaptel cards, specifically, it is not compiling on my FC5 64bit kernel... is this to be expected? |
| 16:49.41 | nitam | tzafrir: coz i need to add tones for my country, argentina. |
| 16:49.45 | jsharp | I don't have any exact brand names to recommend. I've just used Cisco VG224s. They're pricey, though. |
| 16:49.52 | Hmmhesays | set callerid name |
| 16:50.25 | tzafrir | CrashHD, no. What version of zaptel do you use? |
| 16:50.54 | *** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk) |
| 16:50.55 | Hmmhesays | sevard: what are you trying to set up? |
| 16:50.58 | *** join/#asterisk PakiPenguin_ (i=uppal@linuxpakistan/admin/pakipenguin) |
| 16:51.05 | tzafrir | Creathir, no. What version of zaptel do you use? |
| 16:51.09 | chiardon | jsharp . .OK I'll be doing and telling my progress about! |
| 16:51.14 | jsharp | Okee. |
| 16:51.22 | *** join/#asterisk stack_ (n=stack@63.239.190.202) |
| 16:51.22 | Creathir | «tzafrir» I'm using the latest CVS checkout |
| 16:51.32 | alib80 | hi all i was wondering if anyone knew how to send a url with the agent/extension to jabber via the Queue command? |
| 16:51.50 | chiardon | jsharp . . thanks and a 1/4 pound of colombian coffe in your account!!! |
| 16:52.00 | tzafrir | Creathir, Any special reason to use the trunk? try the latest release or the stable branch. |
| 16:52.26 | sevard | Hmmhesays: SIP 2002s to the TDM400P |
| 16:52.45 | Creathir | «tzafrir» okey dokey, was just going by what a guide was telling me on voip-info.org |
| 16:52.48 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 16:53.01 | nitam | tzafrir: do you know how to compile that source ? |
| 16:53.02 | Creathir | so grab the latest stable branch instead of the trunk itself? |
| 16:53.24 | tzafrir | make 'target' on the zaptel dir |
| 16:53.25 | alib80 | sending the url is fine, but to find out who to direct the url too is another story? |
| 16:53.36 | Hmmhesays | sevard ok, whats the problem you're having |
| 16:53.44 | nitam | tzafrir: thats all ? make zonedata.c ? |
| 16:53.50 | *** join/#asterisk razu (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
| 16:53.50 | tzafrir | It is used in both zaptel and libtonezone, IIRC |
| 16:53.52 | Hmmhesays | you just don't know how to configure zaptelc.conf? |
| 16:54.01 | alib80 | ${CHANNEL} just tells one what the incoming channel is |
| 16:54.05 | sevard | Hmmhesays: yeah |
| 16:54.17 | tzafrir | nitam: basically : 'make' . Specifically: make libtonezone.lo |
| 16:54.20 | razu | hi ... can anyone tell me how to test iax connection quality between 2 hosts ? |
| 16:54.29 | stack_ | does roundrobin in a queue always start with the first member, or does it start from the last person dialed? |
| 16:55.06 | nitam | tzafrir: mm do i need to recompile the whole driver again ? |
| 16:55.30 | tzafrir | nitam, if you want it in the driver: you need to rebuild zaptel.ko |
| 16:55.38 | nitam | oh |
| 16:55.38 | froguz | stack_, last person dialed |
| 16:55.52 | sevard | Hmmhesays: there's a lot of lingo in here I don't get. |
| 16:56.06 | Hmmhesays | it's all pretty basic for that, you just have fxo modules in there? |
| 16:56.10 | stack_ | froguz: is there a way to get it to start with the first person every time? |
| 16:56.24 | sevard | Hmmhesays: Yup |
| 16:56.34 | nitam | great. Thank you tzafrir |
| 16:56.52 | Hmmhesays | and what are you going to give me if I fix it for you |
| 16:56.58 | sevard | Hmmhesays: blow job? |
| 16:57.00 | froguz | if you do so, then it wouldn't be round robin anymore |
| 16:57.12 | stack_ | froguz: then how would I do that? |
| 16:57.25 | Creathir | «tzafrir» thanks for your help... should have just thought of doing that.... |
| 16:57.31 | Vagabond | stack_: isn't that what ringall does? |
| 16:57.36 | Creathir | «tzafrir» been a long morning... |
| 16:57.39 | Hmmhesays | um.... |
| 16:57.51 | stack_ | I thought ringall rings every phone all at once |
| 16:57.54 | Hmmhesays | I got hit on enough last night, and i'm still a little bit disturbed |
| 16:57.59 | Hmmhesays | so knock it off |
| 16:57.59 | sevard | Hmmhesays: hahaha |
| 16:58.18 | sevard | Hmmhesays: I'm afraid I probably don't have anything you want |
| 16:58.24 | Vagabond | stack_: oh, hmm |
| 16:58.31 | *** join/#asterisk bmg505 (n=leon@dsl-165-142-253.telkomadsl.co.za) |
| 16:58.32 | sevard | Hmmhesays: I have some empty glass coke bottles |
| 16:58.58 | Hmmhesays | You got 20 bucks and a paypal account? |
| 16:59.00 | froguz | i don't know... i have never worked on queues, i just know the way round robin algoritm works |
| 16:59.51 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 17:00.41 | sevard | Hmmhesays: /msg |
| 17:00.50 | wunderkin | froguz, check queues.conf, roundrobin should start from the top every time i believe, rrmemory is what starts after the last person dialed last time |
| 17:01.03 | brodiem | should busydetect=yes be set on a channelized T1 going to a TE210P? I'm getting reports of some dropped calls, and the latest one shows this in the log: DEBUG[10304] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/14-1, DEBUG[10304] channel.c: Got a FRAME_CONTROL (5) frame on channel Zap/14-1. I'm wondering if I disable busydetect if * will be able to recognize when a line is hungup |
| 17:01.32 | wunderkin | brodiem, i think that is only for analog, so no |
| 17:01.58 | brodiem | wunderkin, yeah that's what I read on voip-info, but I wasn't sure if it applied since it was a channelized T1 |
| 17:02.13 | wunderkin | the signalling will tell it when there is a hangup |
| 17:02.41 | brodiem | wunderkin and would that be only if the telco supports "disconnect supervision"? |
| 17:02.46 | froguz | wunderkin, you're right |
| 17:03.45 | wunderkin | brodiem, um well i'm just learning, but i don't think that applies for digital |
| 17:03.54 | froguz | i though it worked just like round robin channel group |
| 17:04.48 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 17:05.16 | sleepy_one | hi everyone, anyone know why asterisk might lose audio after about 3 min? |
| 17:06.35 | `Sauron | Anyone know why * would think my zap/1-1 wouldn't be hung up properly? |
| 17:06.53 | *** join/#asterisk justinu|laptop (n=Justin@72.18.13.34) |
| 17:07.37 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 17:09.04 | *** join/#asterisk my007ms (n=sao@196.202.70.179) |
| 17:09.05 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 17:09.12 | *** join/#asterisk kshyvy (i=kshyvy@ip-82-177-96-22.nm.e-zet.pl) |
| 17:11.19 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 17:14.31 | *** join/#asterisk TransientPattern (i=mkomitee@B1-66ER.matrix.gs) |
| 17:16.21 | brodiem | wunderkin, I just turned it off and it seems hangups are still being detected so that's good :) |
| 17:17.23 | brodiem | Can someone tell me if changing faxdetect=both to faxdetect=incoming will affect outgoing faxes in any way being sent from a physical fax machine? |
| 17:17.30 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 17:17.46 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 17:17.51 | brodiem | There seems to be some voicemail tones on outside numbers that trigger * into thinking it needs to send a fax, and redirects the call to the fax extension |
| 17:18.10 | brodiem | so I want to know if I set faxdetect=incoming if outbound faxes will still work normally |
| 17:19.47 | pigpen2 | Speaking of faxing: I wonder if anyone has this issue: 1 out of 5 faxes via fxs on TDM2400 (remote * box) via IAX Trunk (ulaw) via Gateway * Box with PRI seems to loose a few inches of fax data... |
| 17:20.05 | pigpen2 | ideas? |
| 17:20.17 | stack_ | so I have three extensions (1, 2, 3). When you dial 1, on unanswered or busy, it dials 2. If 2 is unanswerd or busy it dials three. I'd like to play musiconhold over top of this process... is that possible? |
| 17:20.40 | pigpen2 | stack_, yes. |
| 17:21.37 | stack_ | pigpen2: how? |
| 17:21.37 | pigpen2 | carefully. |
| 17:21.38 | pigpen2 | no..this is easy. |
| 17:21.38 | pigpen2 | Ok..just have the dial plan do: |
| 17:22.33 | [TK]D-Fender | stack_ : Sure its possible, and you'd more often than not implement that as a call Queue.. |
| 17:22.42 | pigpen2 | exten=123,1,Dial(SIP/123,30,m) |
| 17:22.46 | pigpen2 | exten=123,1,Dial(SIP/124,30,m) |
| 17:22.50 | pigpen2 | exten=123,1,Dial(SIP/125,30,m) |
| 17:22.52 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:23.06 | pigpen2 | oops....increment the 123,1 to 123,2, 123,3 |
| 17:23.06 | stack_ | [TK]D-Fender, but I always want the queue to start at 1 and goto 2 and goto 3... but that doesn't seem possibl |
| 17:23.15 | sleepy_one | anyone know why asterisk might lose audio after about 3 min? any ideas or suggestions? tia :-) |
| 17:23.19 | [TK]D-Fender | pigpen : After fixing those priorities and then adding a Goto to loop ;) |
| 17:23.27 | stack_ | pigpen2, is that because of the 'm' option? |
| 17:23.37 | pigpen2 | m is music |
| 17:23.39 | [TK]D-Fender | stack_ : "m" = MoH instyead of ringing |
| 17:23.46 | stack_ | i completely missed taht |
| 17:23.54 | stack_ | should have researched that more |
| 17:23.55 | pigpen2 | yeah...the "30" is the timeout |
| 17:24.04 | pigpen2 | yeah..just google "asterisk dial command" |
| 17:24.05 | *** part/#asterisk my007ms (n=sao@196.202.70.179) |
| 17:24.21 | pigpen2 | sleepy_one, sounds like a nat issue. |
| 17:24.31 | stack_ | yeah, I knew about the options, just missed the 'm' option somehow |
| 17:24.32 | pigpen2 | or a crappy connection |
| 17:25.12 | pigpen2 | Well, I guess no one has ran accross my fax thing...so I will take my laptop out and test ..... |
| 17:25.17 | sleepy_one | pigpen2, thanks :-) there is no NAT involved it happens on Zap channels |
| 17:25.27 | pigpen2 | everything local? |
| 17:26.06 | sleepy_one | pigpen2, PSTN <-> TDM400p <-> * <-> SIP phones on the LAN |
| 17:26.27 | pigpen2 | ok..you just answered my question... |
| 17:26.32 | alib80 | >hi all i was wondering if anyone knew how to send a url with the agent/extension to jabber via the Queue command? |
| 17:26.51 | pigpen2 | sleepy_one, so, after 3 min...voice goes away.... |
| 17:27.11 | sleepy_one | pigpen2, yes about 3min audio drop on Zap |
| 17:27.18 | pigpen2 | one way or both? |
| 17:27.42 | sleepy_one | both "I think" |
| 17:27.51 | sleepy_one | but I'm not 100% sure |
| 17:27.53 | pigpen2 | hmm...test it...find out...that may help... |
| 17:28.05 | pigpen2 | also, try to test the audio, leaving sip out of it.... |
| 17:28.13 | pigpen2 | that also may help figure it out. |
| 17:28.16 | pigpen2 | yes..it is odd. |
| 17:28.34 | sleepy_one | I'm trying to figure out if anyone else is having this issue |
| 17:28.55 | pigpen2 | Well, i am running the same config without the issue. |
| 17:28.58 | pigpen2 | same card too. |
| 17:29.06 | pigpen2 | no issues. |
| 17:29.11 | sleepy_one | in the US or overseas? |
| 17:29.24 | pigpen2 | Us in the US |
| 17:29.31 | sleepy_one | I see |
| 17:29.40 | pigpen2 | you? |
| 17:29.46 | sleepy_one | this card is in the UK, but I'm in the US |
| 17:29.54 | pigpen2 | Lucky you. |
| 17:30.06 | sleepy_one | aye |
| 17:30.09 | pigpen2 | always fun to troubleshoot remote stuff. |
| 17:30.24 | sleepy_one | oh ya, LOTS of fun! |
| 17:30.55 | *** join/#asterisk DoktorGreg (n=Greg@70.91.121.89) |
| 17:31.19 | pigpen2 | I am preparing for a 240 phone and a 130 phone deployment with asterisk.... |
| 17:31.26 | pigpen2 | just a -bit- to plan for.... |
| 17:31.31 | pigpen2 | but atleast it is all local. |
| 17:31.37 | sleepy_one | geez |
| 17:31.48 | sleepy_one | Quad Span TE4xx cards? |
| 17:32.13 | pigpen2 | The 240 will only have a single PRI...so we will use a dual port... |
| 17:32.21 | pigpen2 | in fact the same for the second one. |
| 17:32.30 | pigpen2 | I have a 4 port for my services... |
| 17:33.24 | sleepy_one | 240x48 or 240x46 system then? |
| 17:33.40 | pigpen2 | huh? |
| 17:33.56 | sleepy_one | 240 phones x 48 or 46 phone lines? |
| 17:34.10 | sleepy_one | 2 PRI's worth |
| 17:34.39 | pigpen2 | Actually I have 2 pri's in my quad...which we sell off in trunks... |
| 17:34.49 | pigpen2 | this customer only will have a single pri for their stuff. |
| 17:35.03 | sleepy_one | oh I see |
| 17:35.26 | pigpen2 | but yes...23 usable channels...but if we get another pri, we plan to run nfast to pickup a couple of channels. |
| 17:36.35 | pigpen2 | sleepy_one, I doubt your audio drop is on the zap side... |
| 17:36.42 | pigpen2 | are these "real" pstn lines? |
| 17:37.42 | sleepy_one | yup |
| 17:37.56 | sleepy_one | brb |
| 17:38.03 | *** join/#asterisk SoMeOnEnUlL (n=morris@p1563-adslbkkct1.C.csloxinfo.net) |
| 17:38.05 | pigpen2 | gotta go...later. |
| 17:45.13 | mut | is there any way to tell what revision a digium quad pri card is? |
| 17:45.18 | mut | from the machine |
| 17:45.25 | mut | w/o having to turn it off and look insude |
| 17:45.27 | mut | inside |
| 17:46.31 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:46.35 | generalhan | whats up everyone ! |
| 17:48.10 | generalhan | hmmm not too lively in here this morning, i didnt even get the usuall "the sky" or "gas prices" response today |
| 17:48.25 | sevard | my penis |
| 17:48.53 | Dr-Linux | sevard: grrrrrrrrr |
| 17:48.59 | *** join/#asterisk Damin_PDA (n=pocketir@229.sub-70-217-253.myvzw.com) |
| 17:49.01 | sevard | loffle. |
| 17:49.27 | wunderkin | mut, i thought it showed in dmesg |
| 17:49.38 | Damin_PDA | yo... |
| 17:50.08 | vader-- | are any of oyu guys using a digium pri card and a digium tdm card at the same time? |
| 17:50.18 | generalhan | vader--: I am |
| 17:50.21 | vader-- | im alittle confused as to how to setup the zaptel config |
| 17:51.11 | generalhan | vader--: it took me FOREVER to figure it out ... digium went in and couldnt even figure it out and i guessed my way through it |
| 17:51.13 | *** join/#asterisk falz (i=falz@proxy.supranet.net) |
| 17:51.37 | sleepy_one | vader--, trust in the Wiki |
| 17:51.56 | vader-- | generalhan ya im using a T100P and a TDM2460E |
| 17:52.04 | vader-- | 24 FXS channels |
| 17:52.04 | generalhan | hmm |
| 17:52.15 | vader-- | do you mind sharing your conf files? |
| 17:52.21 | sleepy_one | http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zaptel.conf |
| 17:52.38 | generalhan | that might be a different story from mine then ... i have a dual pri card and a TDM40B |
| 17:52.54 | falz | I've got a few TDM400Ps. I've been using 7 channels, but would like to use all 8. zttool reports "Total/Conf/Act" as "4/4/3". when I try to activate channel 8 (the unused one) I simply get "Unable to reconfigure channel '8'". I've been looking at zapata.conf and zaptel.conf. anywhere else these are configured? |
| 17:53.03 | falz | (or must I reload kernel modules) ? |
| 17:53.17 | sleepy_one | http://pastebin.ca/51237 |
| 17:53.30 | Damin_PDA | falz... yes.. |
| 17:53.47 | falz | yes, reload modules? |
| 17:54.00 | generalhan | Can anyone help me to understand why im starting to get a serious echo on my pri lines ? |
| 17:54.01 | falz | (after modifying zaptel.conf?) |
| 17:54.14 | generalhan | i need to figure out how to fix this before my boss kills me |
| 17:55.12 | mut | Found a Wildcard: Wildcard TE405P (2nd Gen) |
| 17:55.14 | mut | thats all i get |
| 17:55.30 | mut | nothing about revision |
| 17:55.37 | mut | less 2nd gen is revision 2 |
| 17:55.38 | vader-- | sleep_one thats for using a T100P card along |
| 17:55.40 | vader-- | alone |
| 17:55.50 | vader-- | im trying to use a T100P and a TDM2400 card together |
| 17:55.51 | Damin_PDA | mut you got it... |
| 17:56.49 | generalhan | vader--: that was for both i looked at it |
| 17:56.51 | [TK]D-Fender | generalhan : Get ready to start tweaking EC settings all over the place, and maybe recompiling Zaptel with another EC routine, etc, playing with gains... |
| 17:56.59 | [TK]D-Fender | ECHO = Fun! (if you're into masochism) |
| 17:57.03 | generalhan | like rx and txgain ? |
| 17:57.12 | [TK]D-Fender | generalhan : Yes, but thopse settings are last... |
| 17:57.21 | mut | i'm puttin that sangoma in tonite |
| 17:57.21 | generalhan | ok well im looking around the wiki right now |
| 17:57.29 | mut | we'll see how she fares |
| 17:57.29 | [TK]D-Fender | generalhan : How long have you been running on your PRI? |
| 17:57.37 | generalhan | 3 - 4 months now |
| 17:57.42 | *** join/#asterisk JackEstorm (n=thinkthi@ip68-225-72-125.no.no.cox.net) |
| 17:57.44 | generalhan | and its JUST starting |
| 17:57.47 | [TK]D-Fender | mut : I love 'em.... always 100% in my books |
| 17:57.57 | generalhan | it used to only affect my 7960s but now they are all getting nailed |
| 17:58.27 | [TK]D-Fender | generalhan : Maybe load has increased to a point that places a burden on your server... |
| 17:58.41 | [TK]D-Fender | Then again... I've been so long without echo, what should I know? ;) |
| 17:58.48 | generalhan | i seriously doubt its a load on my server |
| 17:59.09 | generalhan | i WAY overkilled the needs for asterisk .. and thats ALL this server does |
| 17:59.11 | [TK]D-Fender | generalhan : All just thoughts... it can come from so many directions.... |
| 17:59.27 | generalhan | well poop; lol |
| 17:59.30 | vader-- | any of you guys have sip firmware for the cisco 7940G's? |
| 17:59.41 | vader-- | im waiting on my smartnet stuff to go through |
| 17:59.54 | [TK]D-Fender | generalhan : Make sure your card is clocking right (ask the telco for an error count), verify CPU load, look at your zapata.conf settings for EC, and go from there |
| 17:59.57 | vader-- | cdwg dumbasses never wrote down the serials to my phones before sending them out |
| 18:00.13 | Damin_PDA | vader:SUUUURREEEE U R... |
| 18:00.17 | vader-- | i had to go through today and retype all 60 phone's serials |
| 18:00.20 | vader-- | and send them to them |
| 18:00.53 | vader-- | hehe im just getting impatient looking at the phones not being able to use them :) |
| 18:00.55 | Damin_PDA | vader get polycom..dump cisco.. |
| 18:01.00 | iCEBrkr | Damin_PDA: where is you? |
| 18:01.04 | vader-- | i just bought 60 cisco 7940G's |
| 18:01.07 | vader-- | too late to switch |
| 18:01.32 | *** join/#asterisk Thock (n=kvirc@216.119.93.253) |
| 18:01.42 | Thock | Hey all |
| 18:01.46 | salviadud | damn, i hate freepbx |
| 18:01.46 | Damin_PDA | . |
| 18:02.03 | salviadud | you see this ¬¬ |
| 18:02.05 | sleepy_one | salviadud, ? |
| 18:02.08 | salviadud | that's my angry face |
| 18:02.10 | Thock | Trying to get my A200 installed, but at the very end of the install for the wanpipe drivers, i keep getting a bunch of "differs in signedness" errors |
| 18:02.13 | Damin_PDA | ice. cal.me..2164104184.. |
| 18:02.17 | Thock | when it tries to compile the wancfg et al |
| 18:02.40 | Thock | bison, gcc, zlib, openssl, all installed with associated --devels |
| 18:03.03 | vader-- | when you modprobe something does that always reload the driver later? |
| 18:03.20 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 18:05.40 | vader-- | when you modprobe something does that always reload the driver later? |
| 18:05.52 | Thock | ? |
| 18:06.06 | sleepy_one | vader--, no |
| 18:06.20 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:06.39 | marcus2 | man, whats the story with nufone |
| 18:06.50 | sleepy_one | vader--, if you want a kernel module to load at startup you have to put it in modprobe.conf usually depending on your distribution |
| 18:06.59 | vader-- | debian |
| 18:07.18 | sleepy_one | vader--, modules.conf then I think |
| 18:09.43 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 18:10.21 | kaz0358 | for testing purposes is it possible to cross connect the two ports on a TE210P? |
| 18:10.25 | *** join/#asterisk brusso (n=BrunoRus@200.204.201.250) |
| 18:10.37 | brusso | hello |
| 18:10.47 | jsharp | Sure. Use a T1 crossover cable. |
| 18:11.27 | brusso | somebody of Brazil??? |
| 18:12.07 | Thock | when it tries to compile the wancfg et al |
| 18:12.07 | Peaceful | Anybody else out there notice 'joinempty' and 'leavewhenempty' not working with queues in asterisk 1.2.7.1? |
| 18:12.11 | Thock | Trying to get my A200 installed, but at the very end of the install for the wanpipe drivers, i keep getting a bunch of "differs in signedness" errors |
| 18:13.45 | kaz0358 | jsharp, okay.. i'm not using a crossover cable. that is probably the issue i'm having. i have span 1 set to "0" for the timing source and span 2 is set to "1" for the timing source... which should be master and slave respectively.. if the timing source is messed up will that always cause a "red" alarm? |
| 18:14.21 | jsharp | No. Red alarm is Loss of Signal/Loss of framing. |
| 18:14.35 | jsharp | Either you don't have cables right, your T1 is connected, or your linecoding and framing don't match your T1. |
| 18:15.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:15.11 | kaz0358 | jsharp, okay.. good. i'll keep that in mind. off the top of your head do you know which pairs need switched? i'll have to make a cable myself |
| 18:15.20 | jsharp | 1-2 and 4-5 |
| 18:15.26 | jsharp | 1 goes to 4, 2 goes to 5 |
| 18:15.35 | DoktorGreg | kaz0358, are you using a bristuff fork of asterisk? |
| 18:15.47 | *** join/#asterisk inv_Arp (i=junya@adsl-10-153-159.mia.bellsouth.net) |
| 18:16.01 | kaz0358 | jsharp, thank. i appreciate the help |
| 18:16.06 | kaz0358 | err thanks |
| 18:16.33 | jsharp | No problem. |
| 18:16.59 | kaz0358 | doktorgreg, no i'm just using Asterisk 1.2.7.1 |
| 18:17.10 | DoktorGreg | kk just checking |
| 18:17.27 | DoktorGreg | I tried same thing with bristuff distro |
| 18:17.37 | DoktorGreg | and it simply doesnt work with pri |
| 18:18.05 | kaz0358 | doktorgreg, well i'll let you know how my testing goes when i get back. i have to hike across campus now.. :) |
| 18:18.29 | *** part/#asterisk brusso (n=BrunoRus@200.204.201.250) |
| 18:20.22 | *** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
| 18:22.51 | vader-- | ok in my dmesg im getting |
| 18:22.51 | vader-- | Zapata Telephony Interface Registered on major 196 |
| 18:22.52 | vader-- | Zaptel Version: 1.2.5 Echo Canceller: KB1 |
| 18:22.52 | vader-- | Registered tone zone 0 (United States / North America) |
| 18:23.12 | vader-- | does that mean that the TE110P and the TDM2400 card are registering? |
| 18:23.28 | vader-- | later down the line im getting |
| 18:23.28 | vader-- | TE110P: Setting up global serial parameters for T1 FALC V1.2 |
| 18:23.28 | vader-- | TE110P: Successfully initialized serial bus for card |
| 18:23.29 | vader-- | Found a Wildcard: Digium Wildcard TE110P T1/E1 |
| 18:23.29 | vader-- | Registered tone zone 0 (United States / North America) |
| 18:23.33 | vader-- | but nothing about the TDM card |
| 18:24.38 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6E6BC.pool.t-online.hu) |
| 18:24.39 | jsharp | Are you modprobing wctdm as well? |
| 18:24.43 | Hali_303 | hi |
| 18:25.08 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 18:25.12 | Hali_303 | after upgrading, I get no sounds from asterisk using a SIP client (xten) what should I check? |
| 18:25.37 | vader-- | jsharp i added into my /etc/modules wctd |
| 18:25.38 | vader-- | m |
| 18:26.32 | jsharp | modprobe it manually and see if the card shows up? |
| 18:26.58 | *** join/#asterisk dasuberdavid (n=david@gateway.digium.com) |
| 18:27.14 | vader-- | when i do modprobe wctdm nothing happens |
| 18:27.42 | jsharp | Sounds like its not seeing the card at all. |
| 18:28.38 | vader-- | would it be modprobe wcfxs? |
| 18:29.45 | jsharp | I don't think so, but you can try it. |
| 18:30.29 | [TK]D-Fender | modprobe wctdm24xxp <---------- |
| 18:30.33 | sleepy_one | vader--, modprobe wctdm24xxp |
| 18:30.42 | jsharp | Or that. |
| 18:30.55 | sleepy_one | [TK]D-Fender, you beat me by a few ms |
| 18:30.59 | websae | anyone do T.38 faxing? |
| 18:31.06 | websae | if so how's that working out? |
| 18:31.17 | vader-- | do i replace the xx's with the correct number or just use it like that? |
| 18:31.27 | sleepy_one | no |
| 18:31.32 | mut | http://www.shortpacked.com/comics/20050309a.gif |
| 18:31.33 | sleepy_one | vader--, modprobe wctdm24xxp |
| 18:32.00 | vader-- | ok cool |
| 18:32.00 | vader-- | thanks |
| 18:32.17 | vader-- | why does the wiki's and stuff seem to leave the wctdm2400 cards out? |
| 18:32.20 | Hali_303 | here is the error message I'm getting on the console: Apr 25 20:36:12 WARNING[18853]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 379608165@81.182.230.188 for seqno 2261 (Non-critical Response) |
| 18:33.00 | websae | does anyone do g.711 faxing? |
| 18:33.04 | websae | how's that working? |
| 18:33.54 | vader-- | when i do a modprobe all that happens is the console says Registered tone zone 0 (United States / North America) |
| 18:34.05 | vader-- | that mean it's found? |
| 18:34.29 | sleepy_one | vader--, you must have forgotten to configure zaptel.conf and zapata.conf accordingly |
| 18:34.30 | jsharp | No. You should see it spit out a bunch of lines for each of the FXO/FXS modules that are on the 2400. |
| 18:35.09 | vader-- | gotcha |
| 18:35.31 | websae | jsharp: how's faxing working out for you? |
| 18:35.45 | sleepy_one | vader--, http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zaptel.conf |
| 18:36.14 | sleepy_one | vader--, look for the TDM400p config, the TDM2400 is the same except it has more ports |
| 18:36.17 | jsharp | 95% or so reliability. I'm using Quintum ASG200s and Grandstream HT286s to talk to our Quintum CMS960. |
| 18:36.50 | vader-- | and the order of the channels is the order i load the modules correct? |
| 18:36.55 | jsharp | Right. |
| 18:37.12 | vader-- | so in my /etc/modules i load zaptel, then TE110P and then the TDM2400 |
| 18:37.14 | *** join/#asterisk lzhang (n=lewiszha@67.95.13.46) |
| 18:37.31 | vader-- | so channels 1-23 are the PRI line and channels 24-48 are the analog TDM card |
| 18:37.35 | lzhang | can anybody extranet.polycom.com is working funny? |
| 18:37.47 | jsharp | vader--: Right. |
| 18:37.50 | generalhan | vader--: NO |
| 18:37.57 | kshyvy | :) |
| 18:38.02 | generalhan | channel 24 is your D Chan for your PRI |
| 18:38.09 | generalhan | you need to start at 25 on your TDM |
| 18:38.15 | jsharp | ohright. |
| 18:40.14 | websae | Has anyone use GRANDSTREAM'S ATA with T.38 support? |
| 18:41.40 | jsharp | yes |
| 18:41.55 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net) |
| 18:42.00 | websae | how do they work out? |
| 18:42.16 | jsharp | Extremely well. |
| 18:43.03 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:43.08 | jsharp | They work well even on some of our bursty, laggy, very latent satellite links. |
| 18:43.10 | qseek | hello all |
| 18:43.53 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 18:44.03 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-249-193.red.bezeqint.net) |
| 18:45.48 | mut | anyone process credit cards online, what merchant do you use? |
| 18:45.56 | Hmmhesays | paypal, lol |
| 18:46.07 | mut | .. |
| 18:46.24 | mut | heh no |
| 18:47.10 | *** join/#asterisk jarrod (i=jarrod@juniperyour.net) |
| 18:47.43 | websae | authorize.net |
| 18:48.09 | stack_ | transaction central |
| 18:48.45 | nahirean | i suggest authorize.net |
| 18:48.49 | jarrod | yoh... im using a cisco as a gateway to pstn and my asterisk cdr are reporting incorrect numbres |
| 18:49.06 | jarrod | or much larger figures than my telco provider is reporting |
| 18:49.34 | brad_mssw | uhh, don't use authorize.net, use a software solution a la monetra |
| 18:50.33 | stack_ | if you're taking orders over the phone or internet, authorize.net is fine... if you are doing face to face, get a terminal |
| 18:51.27 | *** part/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 18:51.27 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:54.58 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:56.39 | brettnem | how do I debug deadlocks.... I am getting so sick of asterisk.. <sigh> |
| 18:56.45 | brettnem | anyone have any pointers? |
| 18:57.05 | brettnem | <no pun intended> |
| 19:00.06 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 19:01.27 | zoa | jsharp, grandstream has t.38 ? |
| 19:01.31 | zoa | like real t.38 ? |
| 19:01.37 | zoa | and not some fake fax passthrouh ? |
| 19:01.51 | brif8 | does the number of modules effect or relate the size of Memory footprint * Uses (top reports Mem: 905016k total, 890660k used, 14356k free) why so little free ? |
| 19:02.55 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 19:02.58 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:02.59 | falz | probably the OS caching memory |
| 19:03.15 | falz | which it does, but will give it to processes if they need it |
| 19:03.24 | jsharp | zoa: Yes, true t.38. |
| 19:03.27 | jsharp | No passthru. |
| 19:03.29 | zoa | amazing |
| 19:03.39 | justinu|laptop | brif8: you need to figure out what processes are using up all your memory |
| 19:03.40 | zoa | what ata supports that now ? |
| 19:04.04 | jsharp | The 286 does, at least. |
| 19:04.33 | brif8 | justinu|laptop: can you suggest something, there are minor backgrounds (like sshd, tftp, etc..) but the big one is asterisk (or it should be) |
| 19:04.38 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 19:04.45 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 19:05.04 | falz | brif8: as I said, it's the OS using the memory for cache, unlikely that it's a process. |
| 19:05.06 | justinu|laptop | brif8: if restarting asterisk causes the memory usage to drop, you've found a memory leak |
| 19:05.22 | justinu|laptop | i've found huge memory leaks, so it's not unheard of |
| 19:05.35 | falz | you can, however, use top and sort by memory. |
| 19:05.56 | techman97_andy | hey all - I'm getting this message in the CLI when someone presses "0" from the voicemail directory - WARNING[15606]: app_directory.c:314 do_directory: Can't find extension 'o' in current context. Not Exiting the Directory! |
| 19:06.01 | zoa | im off to bed |
| 19:06.03 | zoa | ciao |
| 19:06.07 | techman97_andy | any idea which conf file I can edit to allow someone to "zero-out"> |
| 19:06.08 | techman97_andy | ? |
| 19:06.19 | dlynes | techman97_andy: you don't have the operator extension defined ('o') |
| 19:06.29 | dlynes | techman97_andy: for that context |
| 19:06.32 | techman97_andy | OH! The LETTER "o" |
| 19:06.43 | dlynes | techman97_andy: correct |
| 19:06.46 | tasat | Question about asterisk's DTMF detection/supression: when one caller presses a digit, most of the sound is supressed, except for a short blip. Is there something I should change in the code? Do I need a faster machine? Any ideas? |
| 19:06.46 | techman97_andy | DOH! |
| 19:07.09 | falz | it's not a lowercase zero :) |
| 19:07.17 | dlynes | falz: lol |
| 19:07.18 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 19:07.34 | techman97_andy | ugh - I can't believe I didn't realize that was an "o" |
| 19:07.40 | techman97_andy | blibbity blabbity |
| 19:07.46 | dlynes | tasat: what dtmfmode are you using, and what codec are you using? |
| 19:07.50 | sleepy_one | blah |
| 19:08.01 | brif8 | falz: top sorted by memory shows mysqld ,festival and asterisk can't really stop * as it is a production type system |
| 19:08.03 | sleepy_one | j/k |
| 19:08.09 | dlynes | tasat: you cannot using inline dtmfmode for compressed codecs |
| 19:08.16 | tasat | dlynes: codec: ulaw, w/ RFC... |
| 19:08.19 | falz | brif8: yea, but are any of them taking hundreds of MB of RAM? |
| 19:08.36 | *** join/#asterisk IceManRISK (n=kart@201.66.46.17) |
| 19:08.39 | dlynes | tasat: yeah...the dtmf codes aren't long enough for some people |
| 19:08.57 | dlynes | tasat: you'll have to make a change to your c code for that, and then recompile |
| 19:09.09 | tasat | dlynes: ahh, ok. Is that in dsp.c? |
| 19:09.21 | brif8 | falz: each of the mysqld have 2.9% in the MEM column (11 entries) asterisk has 1.2% (20 entries) |
| 19:09.27 | dlynes | tasat: no idea...never encountered the problem myself, so i haven't had to fix it :) |
| 19:10.06 | tasat | dlynes: is it a latency thing? (why my codes need to specified as longer) |
| 19:10.27 | dlynes | tasat: apparently the default length for the tone in asterisk isn't long enough |
| 19:10.38 | dlynes | tasat: but like i said...i've never encountered the problem, personally |
| 19:11.02 | dlynes | tasat: i've seen at least two other people on this channel complain about it though...they modified their asterisk code to fix it |
| 19:11.26 | brif8 | would reducing the number of modules loaded help * function better ? |
| 19:11.43 | tasat | dlynes: ok, thanks -- you've been a big help... I feel better now |
| 19:11.50 | dlynes | tasat: you could also try switching to dtmfmode=info ... asterisk supposedly converts between formats |
| 19:12.38 | tasat | dlynes: you see that making a difference over RFC? |
| 19:12.44 | dlynes | tasat: if you've got an outbound sip provider though, they probably use rfc2833 |
| 19:12.52 | dlynes | tasat: It's fixed some of my problems, yes |
| 19:13.30 | tasat | dlynes: at least one of my providers doesnt support info... |
| 19:13.52 | dlynes | tasat: Yeah...so try setting yoru sip phones to use info, but set your provider to use rfc2833 |
| 19:13.59 | brad_mssw | hah, anyone else get the 'sixtel competitor outage' e-mail ? |
| 19:14.02 | dlynes | asterisk should perform dtmf conversion on it |
| 19:14.13 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 19:14.19 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:14.31 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:15.10 | brif8 | falz: I realize the OS loads memory ready for use, but I'm trying to find factors that will help improve * preformance |
| 19:15.26 | pigpen2 | Hi all. I am wondering if I can do any fax tuning. I have this: |
| 19:15.31 | falz | ok, I missed the first portions of that. I thought you were just wondering why memory wasn't free. |
| 19:16.00 | dlynes | pigpen2: you can adjust your txgain/rxgain |
| 19:16.06 | pigpen2 | Fax - TDM2400E - * - IAX2 Trunk (ulaw) - * - PRI |
| 19:16.11 | pigpen2 | hmm. |
| 19:16.18 | jarrod | we need t38 in asterisk |
| 19:16.23 | jarrod | im having to go cisco->ser |
| 19:16.25 | jarrod | to make it work |
| 19:16.31 | jarrod | and bypass my softswitch |
| 19:16.51 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:16.53 | pigpen2 | Well, this is what I am experiencing...right around the second page... I am getting anywhere .25" to 1" that disappears. |
| 19:17.04 | pigpen2 | otherwise...it is pretty good. |
| 19:17.10 | pigpen2 | so...still the gain? |
| 19:17.15 | brif8 | falz: sorry. would reducing modules loaded help * perform better ? |
| 19:17.16 | pigpen2 | or more a jitterbffer? |
| 19:17.18 | dlynes | pigpen2: oh...no idea |
| 19:17.40 | dlynes | pigpen2: so it's not a transmission error then...just a cosmetic error? |
| 19:17.44 | pigpen2 | yeah..actually, I am pretty happy with it...but the faxes are primarly medical results....quality matter... |
| 19:18.29 | pigpen2 | Well, that brings up a good subject, out of 9 pages of text, is "cosmetic" (ie: a few lines missing) acceptable? |
| 19:18.37 | jsharp | quality fax? |
| 19:18.41 | pigpen2 | it has been so long since I have been around -alot- of faxing... |
| 19:18.52 | *** join/#asterisk aSaDo (n=a@200.68.82.185) |
| 19:18.54 | dlynes | pigpen2: regardless of whether it's acceptable or not, is your fax machine reporting a transmission error? |
| 19:18.59 | pigpen2 | no. |
| 19:19.04 | pigpen2 | no error reported. |
| 19:19.06 | dlynes | pigpen2: That's all I wanted to know |
| 19:19.15 | pigpen2 | dlynes, sorry... |
| 19:19.35 | dlynes | pigpen2: Have you tried rxfax or txfax to see if it's actually receiving and/or transmitting the tiff file appropriately? |
| 19:19.42 | *** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu) |
| 19:20.05 | pigpen2 | dlynes, err...I haven't seen that tool...i take it grabs an image before and after? |
| 19:20.20 | dlynes | pigpen2: yeah..it's a dialplan application |
| 19:20.23 | kaz0358 | jsharp, the cross connect worked. and we now have the wiring issue between asterisk and our avaya switch figured out |
| 19:20.29 | pigpen2 | cool...I will look into it. |
| 19:20.30 | pigpen2 | thanks. |
| 19:20.34 | jsharp | kaz0358: Goodgood. |
| 19:20.44 | dlynes | pigpen2: Yeah...if the resulting tiff file isn't copascetic |
| 19:20.56 | dlynes | pigpen2: then there's probably an error somewhere else |
| 19:21.08 | pigpen2 | yeah...I guess it shows the principle of shit in - shit out. |
| 19:21.28 | dlynes | pigpen2: Yeah...that way you can see what it looks like before it gets transformed by other equipment |
| 19:21.42 | pigpen2 | so do you think .25" is not bad for 9 pages (of missing text that is)? |
| 19:22.01 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 19:22.06 | dlynes | pigpen2: could just be a difference in margin sizes of the receiving/sending fax machines, too |
| 19:22.19 | pigpen2 | yeah, but it happens in the middle of the doc... |
| 19:22.33 | dlynes | pigpen2: ah...that could be a transmission problem then |
| 19:22.51 | dlynes | pigpen2: maybe a network glitch or something |
| 19:22.57 | pigpen2 | yeah...so I may be chasing my tail...but with any new setup, customers are picky. |
| 19:23.22 | dlynes | pigpen2: yeah...as soon as they know it's different, they try to find fault with it...i.e. look for something that may or may not exist |
| 19:23.25 | pigpen2 | true...well the fax is directly connected to the TDM2400....then they have a fiber 45MB line to the gateway * server... |
| 19:23.34 | [TK]D-Fender | pigpen : It happens when the Fax loses part of the data stream and just continues on its merry way. I have PO's come in in LANDSCAPE where the lost segment affected the QTY colum and its absence was amost undistiguishable! |
| 19:24.29 | pigpen2 | so, you would think this is a normal fax occurance? |
| 19:24.42 | dlynes | lol |
| 19:25.01 | *** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) |
| 19:25.10 | dlynes | yeah...don't you wish companies would get rid of those damned fax machines? :) |
| 19:25.49 | generalhan | might as well anyway .. a lot of companies out here use the whole internet faxing software, like winfaxpro so they might as well be sending emails |
| 19:25.52 | lzhang | they should make a scanner style appliance that scans and emails |
| 19:26.24 | pigpen2 | Well, speaking of winfax....that is their "servers". But I also experienced it with my apple notebook...which has proven pretty reliable. |
| 19:27.16 | generalhan | we have winfax here but i havent really had a lot of time to play with it yet so we still use the old fashioned machines |
| 19:29.39 | *** join/#asterisk liran_ (n=liran@80.178.14.98.adsl.012.net.il) |
| 19:29.50 | *** part/#asterisk Hali_303 (n=surfk@dsl51B6E6BC.pool.t-online.hu) |
| 19:29.58 | dlynes | mind you, i'm still sitting here on a 64M pentium II, so I suppose I shouldn't talk about using ancient hardware :) |
| 19:30.51 | pigpen2 | nice.... |
| 19:31.20 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 19:31.36 | dlynes | yeah..it might be old and slow, but it still makes a pretty good X11 terminal :) |
| 19:32.45 | jsharp | LTSP |
| 19:32.55 | dlynes | jsharp: ? |
| 19:33.09 | jsharp | Er, rather...LTSP client. |
| 19:33.20 | dlynes | local telephone service provider? |
| 19:33.38 | jsharp | Linux terminal server project. |
| 19:33.42 | dlynes | ah |
| 19:33.59 | dlynes | heard of it...not familiar with what it is though |
| 19:34.01 | *** join/#asterisk darby_t (i=darby_t@aaov39.neoplus.adsl.tpnet.pl) |
| 19:34.08 | *** join/#asterisk bonfire1 (n=bon@bzq-88-155-15-152.red.bezeqint.net) |
| 19:34.14 | bonfire1 | anybody knows how to use (IF i can use) a PCTel Voice-Modem as a FXO card? |
| 19:34.22 | dlynes | I just run kde on this machine, and then do everything through remote X sessions |
| 19:34.32 | jsharp | One big monster linux machine, thin/diskless clients for workstations. |
| 19:34.40 | jsharp | yeah, me too. |
| 19:35.02 | Vagabond | heh, sounds like the old days |
| 19:35.07 | dlynes | right now though, i'm on vc's because i'm busy burning dvds |
| 19:35.17 | jsharp | Full circle award. |
| 19:35.19 | dlynes | don't want to use up memory on kde |
| 19:35.43 | *** join/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 19:35.49 | dlynes | Vagabond: I still use my 64MB 586, too...works great as a firewall machine |
| 19:37.12 | unmanaged | I have a question. When a call is not answered from a call-file , how do I tie that back into my dial-plan the "failed" exten does not work and the "t" exten does not do it .... |
| 19:37.47 | jsharp | Show off. |
| 19:37.49 | pigpen2 | with gentoo |
| 19:37.52 | dlynes | jsharp: I just got a couple of those...the guy i got them from was using them for a J2EE development environment :) |
| 19:37.57 | jsharp | Uh |
| 19:38.05 | jsharp | Slooooow. |
| 19:38.13 | pigpen2 | Sorry, I saw everyone moaning about their POS...I had to cheer them up. |
| 19:38.25 | tasat | dlynes: still haven't found where the tone length is set. Any ideas? |
| 19:38.38 | dlynes | tasat: one sec, and i'll see if can find it |
| 19:38.52 | jsharp | POS? Nah, mines been running almost a year and a half now. Gotta change it out once I move, though. |
| 19:39.12 | jsharp | It'll fall over at the wirespeed of the new connection. |
| 19:40.49 | pigpen2 | should I have the echo canc turned off on my fax port ? (tdm2400e) |
| 19:41.17 | pigpen2 | mind you I get 1 lost line per 8 pages... |
| 19:41.22 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 19:41.27 | pigpen2 | or leave it the hell alone, it works. |
| 19:43.47 | unmanaged | hmm |
| 19:46.06 | dlynes | tasat: yeah...looks just as confusing to me |
| 19:46.50 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
| 19:46.59 | IceManRISK | Hey |
| 19:47.03 | IceManRISK | anyone here uses a2billing ? |
| 19:47.34 | mog_work | hello marty's friend |
| 19:47.58 | tasat | dlynes: I was look at dsp_detect in dsp.c -- do you make anything of that? |
| 19:48.10 | *** join/#asterisk stoffell_h (n=stoffell@d51A4D12B.access.telenet.be) |
| 19:48.37 | dlynes | tasat: Didn't you say the other end wasn't recognizing the tone you were sending? |
| 19:49.27 | dlynes | pigpen2: Yes...you can't have echocancellation on for faxing |
| 19:49.37 | tasat | dlynes: the DTMF is being recognized -- only the other end hears a blip, rahter than silence |
| 19:49.50 | dlynes | pigpen2: i don't know if autodetect=fax or autodetect=both turns it off, or not |
| 19:49.57 | tasat | lynes: I should say, asterisk in the middle recognizes the DTMF... |
| 19:50.22 | tasat | dlynes: it's just not fully silenced -- same for SIP and IAX2 |
| 19:50.31 | dlynes | tasat: so asterisk isn't generating the dtmf then? |
| 19:50.31 | brif8 | curious I'm reviewing my log files and I find an entry "DEBUG[12295] chan_sip.c: Stopping retransmission on '3495....@ip.address' of Request 102: Match Found" could someone explain ? |
| 19:51.00 | dlynes | tasat: asterisk is only transferring it? |
| 19:51.06 | stoffell_h | dlynes, tzafrir, seems the 1-way audio with the xorcom is solved! (no echo canceller, and activated busy detection) |
| 19:51.16 | stoffell_h | tnx for the help this morning |
| 19:51.28 | dlynes | stoffell_h: ah...col |
| 19:51.30 | dlynes | stoffell_h: ah...cool |
| 19:51.41 | tasat | dlynes: a pots phone to gateway (not sure what they've got) is generating the DTMF |
| 19:51.43 | dlynes | stoffell_h: so you get to play with the door sensor input on it yet? |
| 19:52.16 | stoffell_h | no, not played with that.. not needed at that location, sadly enough.. |
| 19:52.22 | dlynes | ah |
| 19:52.51 | tasat | dlynes: PSTN -> gateway -> asterisk -> softphone |
| 19:52.53 | brodiem | could someone tell me what exactly this means: DEBUG[18241] channel.c: Didn't get a frame from channel: Zap/14-1 |
| 19:53.20 | jsharp | It didn't get a frame from channel Zap/14-1. Probably because the channel was hung up. |
| 19:53.28 | *** join/#asterisk MacDome (n=eseidel@A17-255-105-107.apple.com) |
| 19:54.00 | brodiem | jsharp, i'm trying to figure out why calls are sometimes dropping |
| 19:54.04 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za) |
| 19:54.18 | brodiem | and that was the next thing I saw after seeing that Zap/14 was answered |
| 19:54.30 | stoffell_h | brodiem, what zap device you're using? |
| 19:54.37 | naturalblue | hi. Anyone here using a sipura 3000 |
| 19:55.02 | brodiem | stoffell_h, te210p |
| 19:55.19 | dlynes | naturalblue: yeah...lots of people |
| 19:55.19 | stoffell_h | brodiem, ah, that's a single PRI, right? Using t1 or e1? |
| 19:55.28 | brodiem | stoffell_h, it's a channelized T1 |
| 19:55.35 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:55.48 | brodiem | but te210p is a dual card |
| 19:55.48 | dlynes | tasat: yeah...I don't think that was the other guy's configuration that modified the asterisk code |
| 19:56.00 | stoffell_h | hm, maybe you have to set the timeout for the channel restarts higher? |
| 19:56.07 | naturalblue | dlynes: mine was sending the callerid through to the asterisk box but after some changes it seems to have stopped, i can't work out what i changed. any ideas. |
| 19:56.20 | dlynes | tasat: I don't know if your issue even calls the dsp.c...i woudl assume that's only called if you're generating inline dtmf, not rfc2833 |
| 19:56.28 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:56.31 | dlynes | tasat: try rtp.c |
| 19:56.53 | dlynes | naturalblue: there's a number of settings on the sipura 3000 for the caller id feature |
| 19:57.07 | dlynes | naturalblue: Are you just experimenting with it? |
| 19:57.13 | tasat | dlynes: but its a problem with iax as well |
| 19:57.33 | dlynes | tasat: ah...no idea then |
| 19:57.54 | naturalblue | dlynes: i plan to use it for my pstn house line connection |
| 19:57.56 | brodiem | stoffell_h, which setting is that? |
| 19:58.03 | dlynes | tasat: Try asking Qwell if you see him around...he seems to be pretty knowledgable about that kinda thing |
| 19:58.19 | dlynes | naturalblue: Just do a reset on it then, and start over again |
| 19:58.35 | tasat | dlynes: ok... |
| 19:58.40 | tasat | dlynes: thanks |
| 19:58.53 | dlynes | naturalblue: You know how to do a reset on it, right? |
| 19:59.08 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 19:59.14 | stoffell_h | brodiem: resetinterval (check voip-info.org). but normally it only resets unused channels though. but you can set it to a higher interval |
| 19:59.17 | dlynes | edmonchuck! |
| 19:59.18 | naturalblue | a factory reset |
| 19:59.38 | asterboy | when I type 'show warranty' it says I have none! I want my money back! |
| 19:59.51 | asterboy | oh wait... |
| 20:00.15 | asterboy | lol |
| 20:01.19 | dlynes | naturalblue: Yeah |
| 20:01.37 | justinu|laptop | i'd buy that for a dollar |
| 20:01.42 | dlynes | naturalblue: didn't realize you were asking for confirmation |
| 20:01.46 | *** part/#asterisk unmanaged (n=unmanage@64.89.118.139) |
| 20:01.49 | generalhan | "... and you can too" |
| 20:01.50 | naturalblue | do you really reckon i'll have to to get this working, there isn't just a couple of options i could check |
| 20:02.06 | dlynes | naturalblue: there's about 5 or 6 options for callerid |
| 20:02.16 | dlynes | naturalblue: you can try playing with all of them |
| 20:02.40 | naturalblue | are they in different places or do you mean just the regional page |
| 20:02.42 | dlynes | naturalblue: I don't have a gui up at the moment and a sipura 3000 online to let you know what those are |
| 20:02.58 | dlynes | naturalblue: no...they're on the pstn and line 1 pages if I remember correctly |
| 20:03.19 | naturalblue | ok, cool, i'll have a check around |
| 20:03.24 | dlynes | naturalblue: and they're all down towards the bottom of the page |
| 20:03.48 | dlynes | naturalblue: the spa3000 refers to it as 'CID', not caller id |
| 20:04.05 | *** join/#asterisk groogs_ (n=groogs@d226-27-136.home.cgocable.net) |
| 20:06.04 | naturalblue | whats a CID client ID |
| 20:06.08 | *** join/#asterisk ToTo (n=ToTo@host157-211.pool872.interbusiness.it) |
| 20:06.19 | *** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 20:06.27 | stoffell_h | any good advice on having an extra leading 0 added to the (missed) call lists of hard phones? (in case dialling 0 is needed, same goes for 9..) |
| 20:07.44 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 20:09.01 | dlynes | naturalblue: huh? |
| 20:09.50 | dlynes | stoffell_h: Set(CALLERID(num)=0${CALLERID(num)})? |
| 20:10.09 | dlynes | stoffell_h: before you dial the sip extension, that is? |
| 20:10.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 20:10.47 | stoffell_h | dlynes, hm, ok, only for the incoming calls (from external line, PRI that is) then.. will try something like that! |
| 20:11.11 | dlynes | stoffell_h: exactly...for extension to extension calls you wouldn't do that |
| 20:11.33 | dlynes | stoffell_h: it'll prepend that to all calls though, not just missed ones |
| 20:11.41 | DarKnesS_WolF | when i aster asterisk i got the music on hold working ! every time i start it |
| 20:11.49 | stoffell_h | okay, cool, thanks! (weird, didn't find references on voip-info;) but, i didn't know the right keyword |
| 20:11.53 | dlynes | stoffell_h: to do it for missed ones only, you'd have to write a utility that's tailored for those specific phones |
| 20:12.19 | dlynes | stoffell_h: I didn't look it up on voip-info.org...it's just something i thought up just now |
| 20:12.22 | stoffell_h | that's okay, missed/received/etc... they all need the leading 0 when getting external lines, thanks |
| 20:12.51 | jsharp | Why not just prepend the 0 when you go to dial from *? |
| 20:13.02 | generalhan | jsharp: thats what i was thinking too |
| 20:13.11 | dlynes | jsharp: cause he didn't ask for that? |
| 20:13.41 | generalhan | dlynes: but thats still a good suggestion, why add a 0 to the callerid when you dont have to ... |
| 20:13.49 | dlynes | generalhan: shurg :) |
| 20:13.54 | jsharp | dlynes: True, but I was offering an alternative, perhaps. |
| 20:13.54 | generalhan | lol |
| 20:13.54 | stoffell_h | jsharp, the problem is, the phone (hard phone) has the number in it's memory.. and * expects "0NN.." and not "NN.." (that's how it's setup, to keep the "user experience" the same as before) |
| 20:14.42 | naturalblue | dlynes: found what it was, i had the delay before answer to low so it wasn't getting a chance to get the id |
| 20:14.44 | kaz0358 | in zapata.conf you could have a context per channel if you so wanted? you could just alternate with 'context=c1', 'channel => 1', 'context=c2' and 'channel => 2' ... |
| 20:14.50 | stoffell_h | alternative is to disable "dial 0 for outside line" altogether... but not wanted in this case :) |
| 20:15.04 | dlynes | naturalblue: ah..you had it set to 3? |
| 20:15.12 | dlynes | naturalblue: or 1 or something i mean? |
| 20:16.18 | dlynes | naturalblue: btw...is it getting hoooked into an asterisk box? |
| 20:17.07 | naturalblue | yep |
| 20:17.23 | naturalblue | i had it on 0 and put it back to 3, also works on 1 |
| 20:17.28 | dlynes | naturalblue: and do you use any call forwarding features on your analog line? |
| 20:17.54 | naturalblue | like if im not in forward to my mobile? |
| 20:18.00 | asterboy | are there any softmodems that interface with Outlook to popup records based on CallerID? |
| 20:18.13 | dlynes | naturalblue: correct |
| 20:18.21 | naturalblue | no not at present, i plan to in the future |
| 20:18.25 | asterboy | Identaphone is one by the looks of it |
| 20:18.28 | naturalblue | you having trouble with it |
| 20:18.29 | dlynes | naturalblue: ok, if that's the case |
| 20:18.41 | dlynes | naturalblue: you'll want to set the answer delay a bit higher |
| 20:18.53 | naturalblue | dlynes: you haven an issue with that |
| 20:18.57 | dlynes | naturalblue: the reason being is that it'll ring twice on your line before forwarding |
| 20:19.09 | dlynes | naturalblue: asterisk will pick it up thinking there's a call |
| 20:19.39 | dlynes | naturalblue: but the phone company and the sipura don't reverse the tip/ring when a call forward happens, so asterisk doesn't realize the call's been hung up |
| 20:19.49 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 20:19.59 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 20:20.02 | naturalblue | dlynes: i see. i can get asterisk to do the forwarding for me instead of the sipura or do you mean telco based CF |
| 20:20.08 | dlynes | naturalblue: no, i'm not having an issue with it, but i've run into the problem before |
| 20:20.19 | dlynes | naturalblue: i mean telco based cf |
| 20:20.41 | naturalblue | i wouldn't be using telco based. my telco suck |
| 20:20.47 | dlynes | naturalblue: obviously if you've only got one analog line and no voip lines, you don't want to be getting asterisk to do call forwarding |
| 20:21.08 | dlynes | naturalblue: when asterisk does call forwarding, it ties up two lines, not one |
| 20:21.26 | dlynes | naturalblue: with telco call forwarding, it only ties up one line temporarily (about 5 seconds) |
| 20:21.32 | groogs_ | whats this email i got from sixtel? "A large competitor of ours, who we will not name, has announced that their carrier has terminated their service." .. who? |
| 20:21.41 | groogs_ | anyone know? |
| 20:21.55 | dlynes | groogs_: seems a lot of people on this channel have gotten that email today |
| 20:22.05 | naturalblue | dlynes: i see your point. i'll stick with VM |
| 20:22.15 | *** join/#asterisk guugmember (n=Ignacio@200.30.176.197) |
| 20:22.25 | guugmember | anybody from voipjet here? I am a good customer |
| 20:22.26 | naturalblue | how do they deal with MWI, any issues |
| 20:22.30 | justinu|laptop | nufone |
| 20:22.54 | dlynes | naturalblue: who's 'they'? |
| 20:22.54 | guugmember | nufone seems down, even I have a positive money balance with you |
| 20:23.10 | naturalblue | sipura 3000's |
| 20:23.14 | groogs_ | oh yeah, nufone has a bunch of stuff on their page about it |
| 20:23.15 | asterboy | anyone heard of this company? http://goldcalling.com/techint.html |
| 20:23.23 | groogs_ | " Telesthetic has chosen to terminate our DID services before allowing us to properly migrate the network elements to our new carrier." |
| 20:23.25 | dlynes | naturalblue: they generate their own mwi tone |
| 20:23.29 | vader-- | does anyone know if the Digium TDM2400P card will suppose being in a PCI-X slot? |
| 20:23.35 | asterboy | Looks like an overpriced service |
| 20:23.58 | guugmember | groogs, but is your termination workin properly |
| 20:24.00 | naturalblue | dlynes: grand. thanx for all your help. |
| 20:24.08 | *** join/#asterisk saftsack (n=saftsack@p54A7EE58.dip.t-dialin.net) |
| 20:24.08 | saftsack | hi |
| 20:24.09 | guugmember | groogs, I am about to switch to you |
| 20:24.12 | groogs_ | guugmember: i don't use them |
| 20:24.15 | dlynes | naturalblue: yeah...basically you'll want to disable call waiting on the sipura probably |
| 20:24.23 | saftsack | will there be a betatest from digium for the b410p card? |
| 20:24.40 | guugmember | justinu|laptop, do you work in nufone? |
| 20:24.42 | naturalblue | dlynes: i will had to do the same on my grandstreams |
| 20:24.48 | justinu|laptop | guugmember: no |
| 20:24.51 | dlynes | naturalblue: i haven't gotten it working with asterisk |
| 20:25.07 | dlynes | naturalblue: but then again, i haven't spent a great deal of time trying to get it to work either |
| 20:25.14 | dlynes | naturalblue: I've got better things to do with my time |
| 20:25.17 | jsharp | Why no ISDN card with a U interface on them? |
| 20:25.23 | groogs_ | i was actually just going to setup my system to use sixtel again, i had removed them a long time ago when they were having network problems, and forgotten |
| 20:25.38 | guugmember | anybody from Teliax here? |
| 20:25.52 | groogs_ | forgot i even was a customer of theirs until they sent me that email, still have some money on account |
| 20:28.32 | sleepy_one | vader--, yes it should work fine |
| 20:29.01 | vader-- | hmmm for some reason the bios on this server isn't even seeing the card |
| 20:29.06 | vader-- | ive tried a couple different slots |
| 20:29.37 | sleepy_one | vader--, what do you mean isn't seeing the card?? what kind of board do you have? |
| 20:29.50 | *** part/#asterisk dalfry (n=dalfry@gateway.ishisystems.com) |
| 20:30.02 | vader-- | it's a dell poweredge 2800 server |
| 20:30.03 | sleepy_one | 64bit PCI-X is backward compatible with 32bit PC |
| 20:30.03 | vader-- | brand new |
| 20:30.20 | vader-- | with a digium tdm2400p card |
| 20:30.35 | sleepy_one | does the kernel see the card when you lspci or lspci -v or lspci -vv |
| 20:31.06 | *** join/#asterisk Samoied (n=Samoied@201-25-253-22.fnsce703.dsl.brasiltelecom.net.br) |
| 20:31.17 | generalhan | vader--: i need to ask you right now |
| 20:31.32 | generalhan | does your PE server have hot swap power supplies ? |
| 20:31.43 | vader-- | ya |
| 20:31.46 | tasat | whats the best way to reconstruct incoming RTP packets? |
| 20:31.46 | generalhan | cause i have a PE2850 and i had to do some magic to get it to work |
| 20:31.55 | *** join/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) |
| 20:31.55 | generalhan | ok how are you plugging in your TDM card to power ? |
| 20:31.58 | snitt | hi. |
| 20:31.59 | tzafrir_laptop | could anybody please pm me the output of cat /proc/zaptel/* from a system with a tdm2400p card? (preferebly one with both fxs and fxo modules) |
| 20:32.04 | vader-- | generalhan i had to buy a special adapter to get some molex connectors |
| 20:32.15 | vader-- | from dell |
| 20:32.17 | generalhan | ok just making sure i got that outta the way first |
| 20:32.19 | vader-- | ya |
| 20:32.25 | generalhan | ouch you bought it from dell. |
| 20:32.27 | vader-- | hehe i opened the server up like two weeks ago |
| 20:32.27 | generalhan | ok nevermind |
| 20:32.35 | vader-- | and found out there is no freaking molex connectors |
| 20:32.43 | vader-- | so i had to order this special peice |
| 20:32.49 | generalhan | what does it day when you try and load wct24xxp ? |
| 20:32.55 | sleepy_one | you don't need it actually |
| 20:33.04 | generalhan | haha sleepy !!! |
| 20:33.17 | sleepy_one | an external AC adapter with molex works just fine :-D |
| 20:33.19 | generalhan | he can do what i did ! |
| 20:33.29 | sleepy_one | exactly |
| 20:33.41 | jsharp | That'll get you the Congressional Medal of Ugly. |
| 20:33.57 | generalhan | jsharp ... i made mine look good |
| 20:34.05 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 20:34.08 | sleepy_one | http://www.newegg.com/Product/ShowImage.asp?image=12-156-101-06.jpg,12-156-101-03.jpg,12-156-101-04.jpg,12-156-101-07.jpg&CurImage=12-156-101-04.jpg&Description=BYTECC%20BT-200%20USB2.0%20to%20IDE%20Cable%20With%20Power%20Adapter%20-%20Retail |
| 20:34.17 | generalhan | dremmeled out a pci cover and slid the molex in through there |
| 20:34.25 | sleepy_one | all you need is an AC adapter like this |
| 20:34.35 | LostFrog | A server with no drive power? |
| 20:34.45 | sleepy_one | silly Dell! |
| 20:34.47 | generalhan | LostFrog: its all internal |
| 20:34.59 | jsharp | Drives are 80-pin SCA. Power is on the SCA connector. |
| 20:35.03 | generalhan | the hotswap PS's power the MB in the shoot |
| 20:35.05 | dlynes | LostFrog: anything with fxs ports on it needs power |
| 20:35.28 | guugmember | is there any voipjet support telephone number? |
| 20:35.40 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 20:35.41 | LostFrog | yes, dlynes, I know. |
| 20:35.46 | generalhan | Just so everyone knows .... i havent reported this to digium cause i thought i was the only one that it happened to EVER ... Using a TE210P and a TDM40B i had to skip a channel number to make it work. the TDM should have started on chan 49, but i had to start it on chan 50 to make it work |
| 20:35.50 | *** part/#asterisk falz (i=falz@proxy.supranet.net) |
| 20:35.56 | LostFrog | Ring voltage is expensive, dlynes. |
| 20:36.01 | vader-- | im checking in the bios to see if the card registers |
| 20:36.09 | vader-- | i have a TE110P and a TDM2400P |
| 20:36.16 | vader-- | only the TE110P was registering |
| 20:36.29 | dlynes | LostFrog: exactly...between 30 and 70VDC |
| 20:36.35 | vader-- | i just put the TDM2400P in the pci 32bit slot to see if that registers |
| 20:36.59 | generalhan | i just need to know what it says after you do modprobe zaptel; modprobe wct2xxp; then the modprobe on the 24 card |
| 20:37.13 | generalhan | i think its wct24xxp ? |
| 20:37.14 | sleepy_one | what about lspci tho? was it shown in lspci ? |
| 20:37.15 | vader-- | nothing generalhand |
| 20:37.27 | LostFrog | I was commenting on his lack of molex connectors. |
| 20:37.38 | vader-- | when i do a cat /proc/intterupts it only shows the te110p |
| 20:37.59 | sleepy_one | vader--, please run lspci and procinfo |
| 20:38.31 | sleepy_one | vader--, cd zaptel*; ./zttool # you can also run zttool and see if it's found |
| 20:39.06 | asterboy | I have yet to see a zttool |
| 20:39.14 | asterboy | libnewt is needed for one |
| 20:39.28 | asterboy | <PROTECTED> |
| 20:39.57 | asterboy | zttool would be nice to have without needing libnewt for three |
| 20:40.56 | vader-- | hmmm i tihnk i might have a bad card |
| 20:41.02 | dlynes | guugmember: closest i've been able to find is fastsupport@voipjet.com |
| 20:41.17 | guugmember | dlynes, jeje, mee too |
| 20:41.18 | vader-- | when i go into the bios i see the TE110P card listed and it tells me the slot it's in |
| 20:41.24 | vader-- | but it doesn't list the TDM2400P |
| 20:42.04 | LostFrog | vader--: change slots and see if it show up. |
| 20:42.06 | dlynes | guugmember: did you try https://www.voipjet.com/contact.php? |
| 20:42.07 | sleepy_one | Does Linux see the card tho? |
| 20:42.12 | vader-- | sleepy na |
| 20:42.25 | vader-- | neither the bios or linux sees the card |
| 20:42.33 | vader-- | lostfrog i have put it in every slot |
| 20:42.47 | sleepy_one | that's with the power plugged in? |
| 20:42.51 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 20:42.52 | vader-- | ya |
| 20:43.15 | sleepy_one | vader--, please try the card in other machine and see if Linux sees it |
| 20:43.31 | sleepy_one | preferably something other than a Dell |
| 20:43.58 | vader-- | i don't have another linux machine here |
| 20:44.10 | vader-- | i can those it into another pc and see if the bios recognizes it |
| 20:44.12 | *** join/#asterisk Hali_303 (n=surfk@dsl51B6E6BC.pool.t-online.hu) |
| 20:44.26 | LostFrog | vader--: download knoppix or another liveCD. |
| 20:44.43 | sleepy_one | LostFrog, you beat me to it, I was just about to suggest that |
| 20:44.52 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 20:45.29 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 20:45.53 | sleepy_one | vader--, Knoppix ( 700MB CD or 3.1GB DVD ) or DSL ( 50MB ) would work |
| 20:46.03 | sleepy_one | http://distro.ibiblio.org/pub/linux/distributions/damnsmall/current/ |
| 20:46.18 | sleepy_one | http://www.kernel.org/pub/dist/knoppix/KNOPPIX_V4.0.2CD-2005-09-23-EN.iso |
| 20:46.24 | LostFrog | Does DSL have lspci? |
| 20:46.50 | sleepy_one | vader--, don't forget to check the SHA1 and or MD5 sums and the PGP / GPG sigs |
| 20:47.12 | LostFrog | I think in DSL you have to cat /proc/pci and interpret the vendor:device ids yourself. |
| 20:47.29 | sleepy_one | LostFrog, not sure if DSL has lspci |
| 20:48.06 | sleepy_one | LostFrog, you could probably install it in the RAMDisk tho if it doesn't come with DSL |
| 20:48.19 | dlynes | LostFrog: or copy your pci.ids into /usr/share |
| 20:48.30 | saftsack | hi |
| 20:48.33 | LostFrog | dlynes: that's true. |
| 20:48.43 | saftsack | has someone informations about the b410p card? |
| 20:49.50 | LostFrog | Is that a ISDN card? |
| 20:49.57 | saftsack | yes |
| 20:50.37 | sleepy_one | <PROTECTED> |
| 20:51.00 | *** join/#asterisk darby_t (i=darby_t@aaov12.neoplus.adsl.tpnet.pl) |
| 20:51.21 | dlynes | sleepy_one: is that a fedora-specific location or something? |
| 20:51.38 | sleepy_one | dlynes, that's where it lives on FC4 |
| 20:51.51 | dlynes | sleepy_one: ah |
| 20:52.29 | dlynes | sleepy_one: i thought /usr/share was the standard because slackware puts it there, and it rarely changes any kernel-specific stuff |
| 20:52.42 | sleepy_one | dlynes, same for CentOS and RHEL |
| 20:53.01 | LostFrog | Same on ubuntu/debian. |
| 20:53.16 | sleepy_one | dlynes, that is FC?, CentOS, RHEL, etc stash it in /usr/share/hwdata/pci.ids |
| 20:53.43 | vader-- | ya im having a feeling that the TDM2400P card is bad |
| 20:53.52 | vader-- | that would suck because it's brand new |
| 20:54.07 | dlynes | sleepy_one: Yeah, but Centos, FC, and RHEL are all based on redhat |
| 20:54.18 | dlynes | but LostFrog's saying it's like that on ubuntu/debian, too |
| 20:54.19 | dlynes | shurg |
| 20:54.20 | sleepy_one | dlynes, true |
| 20:55.09 | sleepy_one | vader--, it is time to abandon the dark side and join the rebels! |
| 20:55.31 | sleepy_one | vader--, get an Opteron next time ;-) |
| 20:55.48 | *** join/#asterisk [ProB]CrazyMan (n=crazyman@p549F4875.dip0.t-ipconnect.de) |
| 20:56.23 | vader-- | hehe |
| 20:56.31 | vader-- | the TE110P card works fine |
| 20:56.35 | vader-- | it recognizes |
| 20:57.07 | generalhan | vader--: what is it that you had to buy from DELL ? i dont supose they have a product link for it ? |
| 20:58.13 | vader-- | na |
| 20:58.17 | vader-- | i can get you the part number |
| 20:58.22 | generalhan | no |
| 20:58.41 | LostFrog | Ooops.. ubuntu has pci.ids in /usr/share/hwdata |
| 20:58.42 | generalhan | i dont need it the AC adapter worked great for me ... i was just wondering if they got you the right thing and you card is actually getting power |
| 20:58.50 | LostFrog | Debian has it in /usr/share/misc |
| 20:59.57 | dlynes | hrmn |
| 21:00.07 | dlynes | wonder where the location is stored then |
| 21:00.15 | dlynes | so the kernel knows where to get that info |
| 21:00.29 | dlynes | cd /usr/src/linux-2.4.26/fs/ |
| 21:00.30 | dlynes | ls -al |
| 21:00.35 | dlynes | ack...mistype |
| 21:00.40 | LostFrog | I thought it was compiled in.. but 292k is large. |
| 21:00.41 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 21:01.51 | LostFrog | I just see #include <linux/pci.ids> |
| 21:01.52 | sleepy_one | dlynes, LostFrog https://66.235.243.163/pci.ids.png |
| 21:02.41 | dlynes | ah...nvm |
| 21:02.56 | dlynes | the pci.ids file in an arbitrary spot on the filesystem isn't even read |
| 21:03.06 | LostFrog | I mean #include <linux/pci_ids.h> |
| 21:03.11 | Dr-Linux | hi |
| 21:03.25 | dlynes | the pci.ids info is compiled into the kernel |
| 21:03.25 | LostFrog | lspci might use it. |
| 21:03.25 | sleepy_one | hello Dr-Linux :-D |
| 21:04.01 | sleepy_one | I don't know about Debian but Knoppix is Debian based and so is Ubuntu |
| 21:04.20 | dlynes | and Slackware isn't ;( |
| 21:04.28 | sleepy_one | aye |
| 21:05.28 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:05.32 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 21:05.38 | guugmember | is there any voipjet support telephone number? |
| 21:05.59 | dlynes | LostFrog: devlist.h and classlist.h are generated from pci.ids |
| 21:06.16 | dlynes | LostFrog: devlist.h and classlist.h are then included during kernel compile |
| 21:06.42 | *** join/#asterisk cian (n=cian@g5.cian.ws) |
| 21:08.11 | sleepy_one | DSL is actually a stripped down debian |
| 21:10.16 | *** join/#asterisk montanagbs (n=msummers@12-32-45-196.static.blackfoot.net) |
| 21:10.23 | LostFrog | I want to play with my USB pen drive again.. I wonder which of my computers could boot one. |
| 21:11.16 | montanagbs | I'm a newbie with asterisk looking for some advice - perhaps I can explain what I'm looking to do and some of you experts could point me in the right direction? |
| 21:11.19 | syzygybsd | Does anyone know how on a Sipura SPA-841 to set the external IP? |
| 21:12.00 | *** join/#asterisk trex005 (n=Travis@oh-65-40-131-234.sta.sprint-hsd.net) |
| 21:12.13 | trex005 | has anyone in here worked with EXGN? |
| 21:12.17 | vader-- | i downloaded dsl |
| 21:12.22 | vader-- | but i can't seem to get a terminal window |
| 21:12.24 | syzygybsd | maybe it would be easier if I just went with tcp.... |
| 21:12.25 | vader-- | to do any commands |
| 21:12.27 | brad_mssw | syzygybsd: you use STUN ... |
| 21:12.40 | sleepy_one | https://66.235.243.163/dsl_.png |
| 21:13.14 | montanagbs | an office setting - ADSL w/telephone service - I want to use that line as the incoming/outgoing line - can I do this or do I need a T1 / ISDN with channels |
| 21:13.20 | sleepy_one | vader--, Click on xterm |
| 21:13.28 | trex005 | okay... assuming that someone read my previous question... and they know about them. They seem like a really small company. does anyone know if they are reliable? |
| 21:14.31 | brad_mssw | syzygybsd: or you should use STUN rather ... it's also specifiable if you go to Admin Login -> Advanced -> SIP -> EXT IP |
| 21:14.55 | montanagbs | I'm not really thinking about VoIP service providers and so on...just using asterisk for routing internal calls and using soft-phones - I'm hoping the connections can be handled by a typical modem on the linux server - am I way off? |
| 21:14.56 | *** join/#asterisk gursikh (n=guriskh1@adsl-68-93-83-152.dsl.hstntx.swbell.net) |
| 21:15.28 | vader-- | ok i ran lspci and the card didn't show up |
| 21:15.36 | sleepy_one | vader--, https://66.235.243.163/dsl__.png |
| 21:15.40 | *** join/#asterisk IceManRISK (n=kart@201.66.46.17) |
| 21:15.43 | brad_mssw | montanagbs: 'handled by a typical modem' ... what do you mean, handling what? I hope you're not planning to do voip over 56k dialup |
| 21:15.54 | sleepy_one | vader--, you want the Xterminal but I guess you always found it |
| 21:16.10 | montanagbs | brad_mssw: no, I've got a 3 mbit connection here |
| 21:16.27 | vader-- | so i guess the card is bad? |
| 21:16.27 | *** join/#asterisk papo (n=mathias@adsl-177-161-fixip.tiscali.ch) |
| 21:16.30 | montanagbs | I just don't have any special boards and I don't want to use a VoIP provider, I want to use my traditional line |
| 21:16.47 | sleepy_one | vader--, what kind of machine did you try it in? |
| 21:16.52 | vader-- | dell desktop |
| 21:17.23 | sleepy_one | vader--, do you have any non-Dell PCs? |
| 21:17.27 | brad_mssw | montanagbs: no, you can't use a generic modem ... technically you can use an Intel/Ambient chipset modem, but I think asterisk 1.2 dropped support for those |
| 21:17.31 | vader-- | ehh i would have to hunt for one |
| 21:17.35 | vader-- | let me go see what i can find |
| 21:17.35 | dlynes | montanagbs: if you go onto ebay, there's plenty of cheap x100p cards |
| 21:17.36 | vader-- | brb |
| 21:17.37 | brad_mssw | montanagbs: you should use like a TDM400P or similar device |
| 21:18.15 | montanagbs | what are my other alternatives - go over IP and use a service provider? |
| 21:18.39 | Hali_303 | hi |
| 21:18.51 | dlynes | montanagbs: x100p, x101p, tdm400p, sipura 3000, grandstream ata-486(?), VOIP service provider |
| 21:19.46 | sleepy_one | montanagbs, yes you can use pure VoIP but VoIP isn't as reliable as having hardware to the PSTN |
| 21:19.47 | dlynes | montanagbs: x100p/x101p can be had for as little as $15USD (about $25 if you want one that works for sure) |
| 21:19.55 | *** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com) |
| 21:20.02 | dlynes | montanagbs: sipura 3000 is about $110USD |
| 21:20.19 | dlynes | montanagbs: tdm400p, with 1 fxo port is about $110USD |
| 21:20.22 | brad_mssw | montanagbs: yes, if you've just got standard POTS lines, you need some sort of FXO device to bring the line into ... I wouldn't recommend x100p's, they're not supported or guaranteed ... you'll most likely have echo problems, etc ... |
| 21:20.50 | dlynes | montanagbs: sipura 3000 additionally has an fxs port, however...useful for hooking up analog phones and fax machines |
| 21:20.52 | montanagbs | dlynes: sorry, those are boards that implement different protocols supported by asterisk then? |
| 21:21.09 | Nivex | dlynes: spa-3000 is about US$87 http://store.voxilla.com/customer/product.php?productid=16144&cat=0&page= |
| 21:21.19 | dlynes | montanagbs: all support the zaptel channel, but sipura 3000 supports sip |
| 21:21.32 | papo | Hm, I'm trying to plan a rather special setup. I'm using SIP with software phones. asterisk is running at home on my gateway. I would like to be able to connect my software phone to that asterisk from my LAN and from outside. But when I connect from outside, it would be nice if that call would be sort of redirected, so that the traffic won't go over my gateway |
| 21:22.19 | papo | hm ok, just read the topic |
| 21:22.20 | brad_mssw | papo: SIP should automatically do that as long as no points are NAT'd, and you have REINVITE enabled |
| 21:22.44 | brad_mssw | at least the RTP packets |
| 21:22.50 | *** join/#asterisk websae (n=icechat5@h69-129-251-26.69-129.unk.tds.net) |
| 21:23.40 | *** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl) |
| 21:23.40 | montanagbs | thanks guys, you've given me a starting point of things to look into, much appreciated |
| 21:23.40 | papo | brad_mssw: Ah, I didn't know that, I will test it. Where can I find more information about this REINVITE thing? |
| 21:24.42 | brad_mssw | papo: some info here : http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite ... second paragraph mainly |
| 21:24.50 | papo | ok, thanks a lot |
| 21:25.00 | HotaruT | hm, is it possible to send a "200 OK" as reply to an SIP INVITE without Answer()ing the channel? (and thereby start the charging, or connect-time counter, etc..) .. or is this not possible according to SIP RFC? |
| 21:25.14 | brad_mssw | papo: also read the notes section about reinvite and NAT |
| 21:25.33 | HotaruT | (my problem is a HiPath 4000, which expects a "200 OK" reply within ca. 2 seconds after an invite) |
| 21:25.54 | *** part/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 21:25.57 | papo | brad_mssw: ok, thanks |
| 21:28.26 | dlynes | Nivex: wow...the prices must have dropped in the last little while then |
| 21:28.36 | dlynes | Nivex: that's our usual supplier |
| 21:28.47 | dlynes | Nivex: well...voxilla.ca |
| 21:29.03 | *** join/#asterisk azeteg (n=azeteg@t6o907p36.telia.com) |
| 21:29.28 | azeteg | anyone with experience of the Swissvoice IP-10s ? |
| 21:30.49 | Dr-Linux | i think Comcast is a huge internet provider in USA? :S |
| 21:32.02 | dlynes | Dr-Linux: yep |
| 21:32.06 | LostFrog | It has almost 50% market share for cable internet. |
| 21:32.13 | sleepy_one | Dr-Linux, yes they are |
| 21:32.36 | sleepy_one | used to be part of ATT IIRC |
| 21:32.48 | vader-- | ok |
| 21:32.57 | vader-- | computer number 3 no luck either |
| 21:33.04 | vader-- | this was a compaq desktop |
| 21:33.09 | Dr-Linux | Comcast is in CA or in all US states? :S |
| 21:33.21 | sleepy_one | Dr-Linux, they are all over |
| 21:33.31 | dlynes | Dr-Linux: in order to get 50% market share, they'd have to have more than one state, wouldn't they? |
| 21:33.53 | Dr-Linux | hhm.. |
| 21:34.18 | Dr-Linux | actually one of our Office is in CA and we have hight speed internet from them, that's why i'm asking |
| 21:34.25 | Dr-Linux | actually i'm from Pakistan |
| 21:34.29 | vader-- | any more ideas sleepy_one? |
| 21:34.53 | sleepy_one | vader--, FedEx me the card to test? |
| 21:34.56 | Dr-Linux | some time their modems needs to restart |
| 21:35.12 | vader-- | hehe i think ill call the company |
| 21:35.17 | vader-- | and have them RMA it |
| 21:35.21 | LostFrog | DSL *does* have lspci and /usr/share/misc/pci.ids, if anyone cares. |
| 21:35.21 | dlynes | Dr-Linux: yeah...cable internet kinda sucks for voip...at least that's my experience |
| 21:35.29 | vader-- | 3 computers and no workie |
| 21:35.34 | vader-- | = bad card in my book |
| 21:35.52 | dlynes | LostFrog: does it have ethereal, iptraf, and nmap? |
| 21:35.54 | sleepy_one | LostFrog, ya I posted screenshots of DSL |
| 21:36.04 | Dr-Linux | dlynes: someone told me that vsat is also not good for voip .. brust ... something problems :S |
| 21:36.35 | dlynes | Dr-Linux: no idea, but isn't satellite dem, only? i.e. to upload data you have to use dialup or something? |
| 21:36.44 | azeteg | nope |
| 21:36.47 | azeteg | it used to be |
| 21:36.50 | sleepy_one | dlynes, I'll tell ya in a sec |
| 21:36.56 | dlynes | ah...that's only analog satellites then? |
| 21:37.12 | Dr-Linux | dlynes: i wish we have good internet in Pakistan, but it's very bad |
| 21:37.13 | azeteg | analog satellite |
| 21:37.36 | dlynes | Dr-Linux: yeah...I can imagine |
| 21:37.44 | azeteg | so noone here has used the swissvoice ip-10s? |
| 21:37.45 | dlynes | Dr-Linux: It's pretty good in India, though |
| 21:38.02 | dlynes | Dr-Linux: but I would imagine the internet's pretty good in Lahore, too...no? |
| 21:38.08 | azeteg | I'm just wondering if they'll integrate well with cisco switches, running trunked voip vlan |
| 21:38.27 | Dr-Linux | and for me? i wish i have 512kb/ps internet in my home, but i have a cablenet connection from a provider and that provider total speed is 256kb/ps for 50 clients |
| 21:38.30 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-92-2.cybersurf.com) |
| 21:38.47 | azeteg | I have 56k at home |
| 21:38.53 | Dr-Linux | dlynes: no never, i'm from Lahore |
| 21:39.08 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 21:39.15 | dlynes | Dr-Linux: ah...just figured internet would be pretty good there because lahore has so many call centers |
| 21:39.16 | *** part/#asterisk Hali_303 (n=surfk@dsl51B6E6BC.pool.t-online.hu) |
| 21:39.23 | Dr-Linux | our all servers are in USA, some in office some in Datacenter |
| 21:39.57 | Dr-Linux | dlynes: they are using vsat and blah blah |
| 21:40.03 | dlynes | ah |
| 21:40.03 | Hmmhesays | god i hate vsat |
| 21:40.10 | Dr-Linux | means, internet is very very expensive here |
| 21:40.27 | Hmmhesays | and horribly laggy |
| 21:40.34 | Dr-Linux | telco is very expensive here |
| 21:40.42 | dlynes | Dr-Linux: yeah..same with india |
| 21:40.59 | dlynes | Dr-Linux: I've heard some of the prices for a T1 in India |
| 21:41.04 | dlynes | Dr-Linux: I just about choked |
| 21:41.11 | Dr-Linux | i'm playing with asterisk and stuff, but i can't afford local call to my girl friend |
| 21:41.24 | Dr-Linux | India is much better |
| 21:41.32 | dlynes | Dr-Linux: Well, if you got married, you wouldn't need to call her :) |
| 21:41.34 | sleepy_one | dlynes, NO ethereal, iptraf, nmap on DSL but it does have apt-get so you can install them |
| 21:41.40 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 21:41.53 | Dr-Linux | dlynes: you won't understand my case so leave it :P |
| 21:42.10 | HotaruT | file: some days ago you tould me, in a SIP call, there can be a RTP connection before doing an Answer(). How can I do this with asterisk? (my specific problem is an hipath 4000, which expects an "200 OK" within 2 seconds after sending a SIP INVITE) |
| 21:42.16 | dlynes | lol |
| 21:42.37 | Dr-Linux | dlynes: just imagine, we have only 2 employee in US office and have 5 MB internet from comcast |
| 21:43.06 | qseek | dr-linux sorry to barge in but isnt that only download speed |
| 21:43.25 | Dr-Linux | and in our headoffice in Lahore we have more then 200 employees and we have 1 MB from best pakistan internet provider |
| 21:43.31 | Dr-Linux | and thats sucks always |
| 21:43.47 | dlynes | damn |
| 21:44.12 | dlynes | sleepy_one: ah |
| 21:44.13 | Dr-Linux | qseek: thats down and up is 1MB |
| 21:44.42 | dlynes | sleepy_one: i guess it's pretty easy to modify the distro, and create your own though? |
| 21:44.44 | qseek | dr-linux: so r u in lahore or in IUS |
| 21:44.48 | Dr-Linux | dlynes: i have 2 asterisk servers 1 is in US office and 1 is in Lahore office |
| 21:44.58 | Dr-Linux | qseek: i'm in Lahore |
| 21:44.59 | dlynes | sleepy_one: so that the cd includes ethereal, iptraf and nmap? |
| 21:45.02 | sleepy_one | dlynes, Yes it's Knoppix based and can install on HDD |
| 21:45.12 | qseek | Dr-Linux: I miss the lassi send some over |
| 21:45.26 | dlynes | sleepy_one: Yeah..I don't wantto install on hard drive though...just want a network troubleshooting cd |
| 21:45.29 | Dr-Linux | but our lahore sip users are registered with US server, not with LHR asterisk server bcoz of damn internet |
| 21:45.34 | dlynes | sleepy_one: that doesn't depend on internet access |
| 21:45.39 | *** join/#asterisk ToTo (n=ToTo@host210-136.pool875.interbusiness.it) |
| 21:45.44 | Dr-Linux | qseek: lassi ? |
| 21:45.46 | sleepy_one | dlynes, then get Knoppix |
| 21:45.48 | russellb | qseek: nortel, huh? |
| 21:46.03 | qseek | hey we can learn too:) |
| 21:46.05 | terrapen | lussi? |
| 21:46.10 | sleepy_one | dlynes, DSL is TINY but it doesn't have what Knoppix has, were's talking 50MB vs. 700MB |
| 21:46.13 | russellb | qseek: heh, welcome :) |
| 21:46.20 | dlynes | terrapen: lassi is a beverage |
| 21:46.22 | Dr-Linux | are you talking about drink? |
| 21:46.30 | terrapen | i know |
| 21:46.40 | Dr-Linux | what's beverage? :S |
| 21:46.41 | terrapen | i've seen it called "lussi" in a pakistani restaurant |
| 21:46.45 | qseek | dr-linux aur ki bhaiji...lassi is drink punab di |
| 21:46.57 | Dr-Linux | oo yess |
| 21:46.58 | dlynes | sleepy_one: nod...I don't need all the extra crap though...just network troubleshooting tools |
| 21:47.10 | Dr-Linux | qseek: lolzzzzzzzzzz desi? :P |
| 21:47.13 | sleepy_one | dlynes, grab http://www.kernel.org/pub/dist/knoppix/KNOPPIX_V4.0.2CD-2005-09-23-EN.iso + http://www.kernel.org/pub/dist/knoppix/knoppix-dvd/KNOPPIX_V4.0.2DVD-2005-09-23-EN.iso |
| 21:47.26 | qseek | dr-linux: sure sure |
| 21:47.37 | Dr-Linux | qseek: where from you? |
| 21:47.57 | qseek | dr-linux: karachi |
| 21:48.21 | sleepy_one | dlynes, 700MB and 3.1 respectively worth the huge download |
| 21:48.24 | Dr-Linux | qseek: nice, so what you think about Pakistan internet? |
| 21:48.52 | qseek | dr-linux : dont know never used it...been a long time |
| 21:49.31 | Dr-Linux | qseek: then what you use for your Asterisk? |
| 21:49.53 | Dr-Linux | terrapen: from where? :S |
| 21:50.03 | terrapen | hmmm...probably mexico or guatemala |
| 21:50.10 | terrapen | or possibly nicaragua or el salvador |
| 21:50.23 | Dr-Linux | lolz |
| 21:50.43 | qseek | dr-linux..hobby |
| 21:50.44 | Dr-Linux | we have some clients in Guatemala |
| 21:50.50 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:50.56 | brad_mssw | terrapen: sup |
| 21:51.06 | terrapen | oh yeah? i have a good friend from guatelmala |
| 21:51.11 | terrapen | whattup brad |
| 21:51.17 | terrapen | still waiting on my bike :/ |
| 21:51.22 | brad_mssw | not much, how's utah treating ya ? |
| 21:51.31 | terrapen | its nice. it snowed this morning :) |
| 21:51.38 | brad_mssw | wow, snowing still ? |
| 21:51.39 | brad_mssw | geez |
| 21:51.43 | qseek | terrapen: where did it snow |
| 21:51.49 | qseek | i wished it would snow in texas |
| 21:51.53 | qseek | dang it is hot here :) |
| 21:51.54 | Dr-Linux | qseek: what's Asterisk future in Pakistan? :) |
| 21:51.54 | terrapen | well, it didn't stick and it didn't snow much. but i saw snowflakes |
| 21:51.58 | terrapen | qseek: park city utah |
| 21:52.02 | terrapen | qseek: where in Texas? |
| 21:52.08 | qseek | big D |
| 21:52.16 | terrapen | <--- from San Antonio |
| 21:53.01 | brad_mssw | terrapen: seriously thinking about getting a road bike these days, though nothing nearly as expensive as that mountain bike you're getting |
| 21:53.04 | terrapen | brad, you been riding much? |
| 21:53.11 | terrapen | get a fixed gear! |
| 21:53.50 | terrapen | oh, heh, my dad wants $1500 from me for the bike...i guess my consulting wasn't worth that much to him :P |
| 21:53.55 | qseek | dr-linux: dont know..if the infrastructure improves no charm |
| 21:53.58 | brad_mssw | terrapen: been mainly doing paved riding ... haven't had a chance to hit the trails much this year |
| 21:54.04 | qseek | terrapen kewl..i was there a few weeks ago |
| 21:54.33 | terrapen | i'm going for a music festival...taking a little vacation |
| 21:54.45 | brad_mssw | terrapen: haha, well, still, getting that bike for < 25% of MSRP isn't bad |
| 21:54.46 | terrapen | brad, our trails are still snowed over |
| 21:54.48 | qseek | neat terrapen |
| 21:54.52 | Dr-Linux | our bunch of clients are using our IVR services arround the world, we are moving our IVR system to asterisk using AGI in C, but wondering someone says that's not a good approach :S |
| 21:55.28 | terrapen | well, my foundry switches haven't shown up yet...so there's no work for me to do ont he PBX |
| 21:55.33 | terrapen | vacation time! |
| 21:55.38 | Deep6 | what's the largest Asterisk system someone in here has deployed? |
| 21:55.50 | terrapen | deep6, i'm deploying a 350-seat |
| 21:55.56 | terrapen | but its being done in phases |
| 21:56.18 | *** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net) |
| 21:56.28 | Dr-Linux | qseek: soon most of call centers in lhr will move to asterisk |
| 21:56.41 | Deep6 | terrapen, so it's not deployed then |
| 21:56.55 | terrapen | well, partially |
| 21:57.02 | qseek | dr-linux i think so too |
| 21:57.13 | terrapen | i'm moving sections of office at a time |
| 21:57.29 | terrapen | since I can only deploy so fast and also to save a huge expense occuring at once |
| 21:57.32 | dlynes | Dr-Linux: your government has a mandate to use linux in all public offices too, right? |
| 21:57.56 | Dr-Linux | everyone can use Linux |
| 21:57.58 | Dr-Linux | dlynes |
| 21:58.07 | terrapen | I think Pakistan would be neat to visit...too bad most amaericans (me included) are terrified of the place |
| 21:58.14 | Dr-Linux | Linux is not a problem here, bandwidth tho |
| 21:58.30 | dlynes | Dr-Linux: but doesn't government in pakistan mandate that public offices _must_ use linux? |
| 21:58.33 | dlynes | Dr-Linux: or not? |
| 21:59.01 | qseek | dr-linux: man u are up late..it is almost morning in lahore |
| 21:59.03 | Dr-Linux | dlynes: i'm sorry dude i'm not good with english so i don't understand what's "mandate" :S |
| 21:59.14 | dlynes | Dr-Linux: mandate == require |
| 21:59.23 | dlynes | Dr-Linux: make law....sort of |
| 21:59.42 | Dr-Linux | qseek: no problem i'm at home and i work in nights as US time, our all business is in US |
| 21:59.47 | sleepy_one | everyone should be using Linux as far as I'm concerned :-D :-D |
| 22:00.12 | Dr-Linux | dlynes: most of offices is using Windows here |
| 22:00.18 | terrapen | no thanks, i'll run OpenBSD |
| 22:00.19 | dlynes | sleepy_one: including the blonde secretary snorting too much liquid paper that has immense problems with windows? |
| 22:00.20 | qseek | dr-linux.. good for u....got any thing going on in khi |
| 22:00.21 | websae | *? |
| 22:00.35 | sleepy_one | dlynes, heck yeah! :-D |
| 22:00.37 | Dr-Linux | dlynes: never seen much Linux expert here |
| 22:01.08 | Dr-Linux | dlynes: and hardly few guys knows Asterisk here, |
| 22:01.17 | dlynes | lol |
| 22:01.22 | Dr-Linux | most of pplz do not know even asterisk name |
| 22:01.25 | NewSole | even the makes |
| 22:01.28 | NewSole | makers |
| 22:01.32 | terrapen | i wonder how much money could be made on Asterisk in Pakistan |
| 22:01.32 | dlynes | Dr-Linux: oh...you mean in pakistan |
| 22:01.48 | Dr-Linux | dlynes: yes |
| 22:01.52 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 22:02.06 | qseek | i dont think u could make much terrapen |
| 22:02.10 | Dr-Linux | but i know Pakistan has very good future in pakistan, bcoz of bunch of call centers |
| 22:02.27 | dlynes | Dr-Linux: Yeah...a lot of american companies have call centers in lahore |
| 22:02.51 | dlynes | Dr-Linux: and in mumbai and calcutta, too |
| 22:03.03 | Dr-Linux | dlynes: few days back here was a job fare, and there was Asterisk guys seeking stoll, but they found no one :) |
| 22:03.14 | dlynes | stoll? |
| 22:03.35 | Dr-Linux | stall .. or what i can't spell |
| 22:03.56 | dlynes | Dr-Linux: you mean a booth? |
| 22:04.09 | Dr-Linux | my company do not let me go out, even i'm not good with asterisk yet |
| 22:04.16 | Dr-Linux | dlynes: yes |
| 22:04.51 | terrapen | do you think it's safe for americans to walk the streets in Pakistan right now? |
| 22:04.57 | dlynes | lol |
| 22:05.07 | terrapen | dlynes, i'm serious |
| 22:05.14 | Dr-Linux | terrapen: yes, here are many |
| 22:05.36 | Dr-Linux | our many of employee are from USA |
| 22:05.41 | justinu|laptop | i think it's safe |
| 22:05.42 | Dr-Linux | and they come here often |
| 22:05.54 | Dr-Linux | hey justin |
| 22:05.57 | justinu|laptop | the rest of the world is a lot safer than most americans believe |
| 22:06.11 | Dr-Linux | how are youuuuuuuuu? |
| 22:06.20 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:06.26 | justinu|laptop | doing ok |
| 22:06.30 | terrapen | well, i've walked the streets of mexico at night...but i'm not worried about al Qaeda in Mexico |
| 22:06.34 | justinu|laptop | woooork |
| 22:06.44 | Dr-Linux | justinu|laptop: when you are comingggggg here. as i told my mom about you, she was asking when you will come :) |
| 22:06.55 | justinu|laptop | soon i hope |
| 22:06.59 | justinu|laptop | i have a few people to visit in .pk |
| 22:07.00 | justinu|laptop | :) |
| 22:07.16 | Dr-Linux | Al qaeda lolzzzzzzzzzzzzzzz |
| 22:07.47 | terrapen | well, they kidnapped daniel pearl in karachi |
| 22:07.58 | dlynes | if you just tell them you're muslim, they'll leave you alone, won't they? |
| 22:08.02 | Dr-Linux | fuck .. here is no alqauda, just US damn govt: is playing politics and killing pplz |
| 22:08.02 | terrapen | granted, he was a jewish journalist seeking to meet with militants...but still |
| 22:08.40 | Dr-Linux | terrapen: that was only one case? |
| 22:09.08 | terrapen | well, there is also the incident of the US Embassy in (?) 1978 |
| 22:09.09 | Dr-Linux | scroll the history, you will find thousand same things that americans did |
| 22:09.30 | justinu|laptop | that was in iran |
| 22:09.35 | Dr-Linux | but it's 2006 |
| 22:09.37 | justinu|laptop | 1979 |
| 22:09.38 | terrapen | no, it happened in pakistan, too, justin |
| 22:09.44 | *** join/#asterisk Creathir (n=Creathir@207.71.17.206) |
| 22:10.04 | dlynes | Just wave the canadian flag, instead |
| 22:10.09 | terrapen | http://news.bbc.co.uk/onthisday/hi/dates/stories/november/21/newsid_4187000/4187184.stm |
| 22:10.11 | terrapen | oops |
| 22:10.20 | terrapen | http://tinyurl.com/n9vb8 |
| 22:10.57 | justinu|laptop | provoked by the same guy |
| 22:11.09 | terrapen | yep |
| 22:11.23 | sleepy_one | aye |
| 22:11.27 | kaz0358 | i am making a test call from a sip phone to asterisk through TE210P to our avaya switch to a digital phone on my desk.. as soon as i pick up the phone i get chan_zap.c:7915 zt_pri_error: [Span 0 D-Channel 0] PRI: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX |
| 22:11.34 | kaz0358 | i have googled around and haven't found anything... |
| 22:11.53 | Dr-Linux | justinu|laptop: i dont know why US pplz afraid much :S |
| 22:12.03 | terrapen | dr-linux, because of the beheadings. |
| 22:12.13 | justinu|laptop | Dr-Linux: we live in a country driven by fear |
| 22:12.20 | Dr-Linux | beheadings? :S |
| 22:12.21 | justinu|laptop | everything here is "safety first" |
| 22:12.39 | Dr-Linux | hhm.. i see |
| 22:12.39 | dlynes | Yeah...I haven't heard of any beheadings in pakistan, either |
| 22:12.57 | Dr-Linux | what's beheadlings |
| 22:12.59 | terrapen | dlynes, other than daniel pearl...and a Navy SEAL |
| 22:13.11 | dlynes | Dr-Linux: decapitations |
| 22:13.13 | kaz0358 | i tried setting CALLERIDNUM and CALLERIDNAME in sip.conf under the sip phone entry... but it doesn't make any difference |
| 22:13.20 | terrapen | dr-linux, when al qaeda cuts peoples heads off |
| 22:13.26 | terrapen | or anyone, for that matter |
| 22:13.31 | Dr-Linux | however i don't afraid, if i can, i can go anywhere in the world with no fear |
| 22:13.34 | *** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net) |
| 22:13.35 | justinu|laptop | terrapen: traveled the world much? |
| 22:13.41 | terrapen | but specifically, the videotaped beheadings are what scared americans |
| 22:13.45 | terrapen | justin, yep |
| 22:13.55 | Dr-Linux | terrapen: shit ... huh i don't think that's in pakistan |
| 22:14.06 | terrapen | again, daniel pearl was beheaded in Karachi |
| 22:14.10 | techman97_andy | hey all - in the CLI, I'm sitting here watching usage during our pilot period - do you know of a way that the caller ID number can be included somehow in the ANSWER string I'm seeing? |
| 22:14.14 | gursikh | those, were horrible, and were in afghanistan (the videos you speak of) |
| 22:14.31 | dlynes | gursikh: yeah...that sounds more appropriate |
| 22:14.40 | terrapen | i'm not trying to start a fight...i'm just telling you what many of us americans are afraid of |
| 22:14.57 | terrapen | http://en.wikipedia.org/wiki/Daniel_Pearl |
| 22:15.07 | Dr-Linux | terrapen: you guys still remember Daniel perl, but you can't see what's America did/doing here? |
| 22:15.21 | techman97_andy | it's true terrorism...and humans are panicky in groups. AlJezzara (spelling?) broadcasting those videos accomplished that. |
| 22:15.33 | gursikh | Speak for yourself, the videos are not really what "Scared americans" they just added minutly to it. |
| 22:15.35 | justinu|laptop | you're letting them win |
| 22:16.02 | gursikh | full disclosure: I'm american! |
| 22:16.03 | russellb | this discussion needs to stop *right now* |
| 22:16.09 | terrapen | heh |
| 22:16.12 | techman97_andy | =P |
| 22:16.18 | russellb | if there is another comment, you will be muted |
| 22:16.26 | gursikh | Yeah, for real, not the place or time. |
| 22:16.46 | techman97_andy | ok...*smirk*...here's a question to restart matters... |
| 22:16.48 | techman97_andy | hey all - in the CLI, I'm sitting here watching usage during our pilot period - do you know of a way that the caller ID number can be included somehow in the ANSWER string I'm seeing? |
| 22:16.55 | Dr-Linux | hhmm... |
| 22:17.12 | terrapen | it may have been off-topic, yes. but i think we were all being civil. i'll leave it at that. |
| 22:17.27 | justinu|laptop | agreed |
| 22:17.35 | russellb | techman97_andy: you can insert a NoOp(${CALLERID(num)}) |
| 22:17.46 | techman97_andy | cool. thanks! |
| 22:18.48 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 22:18.57 | terrapen | i need a good minipci wifi card from an opensource-friendly company |
| 22:19.00 | terrapen | anybody know of one? |
| 22:19.18 | justinu|laptop | heh, does such a thing exist? |
| 22:19.39 | papo | some call intel opensource-friendly |
| 22:19.50 | darkskiez | http://rt2x00.serialmonkey.com/wiki/index.php/Main_Page |
| 22:20.05 | terrapen | certainly some manufacturer must have released programming specs... |
| 22:20.05 | gursikh | General Question: What is a decent price to pay someone to do a remote setup of asterisk? (just the installation/setup maybe some walktrhough with the hardware. eg. I will buy the phones and the fxs card, install linux on a computer and get them to SSH in to do the setup. Nothing fancy or extra, just the basics) ? |
| 22:20.19 | mog_work | 100 grand..... |
| 22:20.20 | terrapen | intel is bad about releasing blob-only |
| 22:20.30 | mog_work | but for you gursikh ill do it for 50 |
| 22:20.33 | darkskiez | i'll do it for 10 |
| 22:20.33 | sleepy_one | gursikh, I'll do it for half thank |
| 22:20.34 | techman97_andy | rofl - it all depends on how complex the system is and how involved you want to be in the setup |
| 22:20.50 | darkskiez | 10 with bells&whistles :) |
| 22:21.02 | sleepy_one | gursikh, I'll do it for half rather |
| 22:21.09 | mog_work | gursikh, ill pay you 5 dollars to do it..... |
| 22:21.19 | gursikh | you'll pay me? |
| 22:21.31 | sleepy_one | lol |
| 22:21.33 | gursikh | lol |
| 22:21.33 | tasat | question about DTMF supression/masking: on calls coming from pstn through a gateway provider, is it normal to hear short blips, as if the providers aren't properly masking the tones? |
| 22:22.23 | distortion | depends on if they are sending tones inband from the gateway provider |
| 22:22.41 | gursikh | Just the basics, three POTS lines, 4-5 phones. Need to have voicemail, ability for conference calls. What is a reasonable amount to pay to get someone knowledgable to walk me through the hardware setup. Do the installation. And some support afterwords? 100usd? 200usd? 500usd? |
| 22:23.01 | tasat | distortion: the ones I've tried are either RFC2833 on SIP, or whatever IAX is doing |
| 22:24.01 | distortion | rfc2833 sends the tones as messages- it doesnt really matter how they sound |
| 22:24.11 | distortion | not familiar with iax's method |
| 22:24.31 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 22:24.33 | distortion | so, although a little strange (the blips) when using 2833 if it works, be happy :) |
| 22:24.43 | techman97_andy | if I was indeed knowledgeable enough to ask for money...=)...I would send |
| 22:25.23 | darkskiez | gursikh: well, divide it by how many hours work you/they expect that would entail. |
| 22:25.25 | techman97_andy | maybe go for $200 USD for someone to walk me through the hardware setup via phone or chat, to walk me through the conf files, a few phone setups, etc....and maybe 2-3 support calls |
| 22:25.28 | sleepy_one | gursikh, probably between 200 and 400 |
| 22:25.39 | techman97_andy | support beyond that would be renegotiated. |
| 22:26.12 | darkskiez | gursikh: just be aware, that with POTS interface it will never work as well as an ISDN one. |
| 22:26.35 | sleepy_one | aye that's true :-( |
| 22:26.46 | sleepy_one | analog is not as good as ISDN or PRI |
| 22:26.55 | techman97_andy | your mom is better than analog |
| 22:26.59 | techman97_andy | =/ |
| 22:27.02 | gursikh | maybe i have the terminoly wrong, POTS= plain old telephone system? I just meant to say just regular phone lines, not VOIP. |
| 22:27.09 | darkskiez | gursikh: you wont be able to tell if a call has been answered when you call out, it will assumed to have answered as soon as its finished dialling. Also you wont be able hang up on people and free the lines. |
| 22:27.10 | tasat | distortion: if it was just me I'd deal with it, but I intend to have customers -- and it's fairly annoying... sounds like I need to talk with the providers.... |
| 22:27.27 | terrapen | http://www.netgate.com/product_info.php?cPath=26_34&products_id=279 |
| 22:27.32 | terrapen | these look promising |
| 22:27.33 | tasat | distortion: i.e in a conference setting... |
| 22:28.02 | darkskiez | gursikh: yes, that is correct, but with POTS, you may have echo problems and you will lose features you may be expecting. |
| 22:28.02 | sleepy_one | gursikh, POTS= plain old telephone system = PSTN = public switched telephone network = analog |
| 22:28.09 | dlynes | tasat: what phones are you using? |
| 22:28.40 | techman97_andy | I used two x100p wildcards with a pair of good ol' phone lines for 2 weeks before I went to a voip provider - they worked just fine for basic usage. |
| 22:28.47 | gursikh | AH. Ic. So it would be advisable to get a VOIP provider rather than some extra phone lines? |
| 22:28.48 | Dr-Linux | justinu|laptop: you there? |
| 22:28.58 | justinu|laptop | sorya |
| 22:28.59 | justinu|laptop | sorta |
| 22:29.00 | techman97_andy | yeah, cheaper in the end and easier to work with |
| 22:29.02 | tasat | dylnes: this is with a cell phone, or pots phone -> pstn -> gateway -> my asterisk -> my softphone |
| 22:29.03 | darkskiez | POTS is an _interface_ to the PSTN |
| 22:29.05 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 22:29.07 | thock | hey all |
| 22:29.19 | dlynes | tasat: and it's on the softphone that it sounds like crap? |
| 22:29.28 | sleepy_one | gursikh, Yes VoIP is cheaper but not always as reliable |
| 22:29.30 | distortion | tasat: you can also try sip-info. I've had good reliability with that- it will send the dtmf out of band instead of in the rtp with the voice although not sure if it will help at all with your problem, but worth a shot. |
| 22:29.40 | thock | I am totally stuck here- I've got my sangoma drivers installed for my A200, just a single FXO module in it, and for some reason, if i run wanrouter start, it says it can't find the device |
| 22:29.42 | thock | i'm at my wits end |
| 22:29.47 | Dr-Linux | justinu|laptop: http://www.syednetworks.com/pics.zip |
| 22:29.58 | gursikh | currenlty we have one regular line, and 2 lines through vonage which has been ok so far. We were thinking of getting rid of the vonage and getting two additional landlines. |
| 22:30.00 | techman97_andy | I'm using VoiceEclipse as my Voip provider...a tad crappy in the setup dept, but once you get it up and running, it's stable and good |
| 22:30.02 | tasat | dlynes: I captured the RTP packets coming in to my asterisk box and reassembled the audio -- its audio coming in that contains the blips |
| 22:30.22 | justinu|laptop | you're fucked then |
| 22:30.28 | sleepy_one | gursikh, what country are you in? |
| 22:30.31 | justinu|laptop | btw, nice troubleshooting skillz |
| 22:30.31 | techman97_andy | I'm just on a business-class cable connection and haven't lost registration at all |
| 22:30.36 | gursikh | USA- NYC, NY |
| 22:31.06 | sleepy_one | gursikh, vonage is not THAT bad, it depends on your internet provider tho |
| 22:31.10 | tasat | distortion: yeah, in this case, there's a tone in the regular non-event RTP packets that I don't think should be there |
| 22:31.23 | twisted[asteria] | <redneckish accent> New york city!? </accent> |
| 22:31.47 | tasat | distortion: I get the RTP events with RFC2833, and also a blip in the reg. RTP |
| 22:31.57 | Dr-Linux | justinu|laptop: please see that, upload these for you, that's all tribal area |
| 22:32.14 | terrapen | you should upload those to flickr, dr-linux :) |
| 22:32.24 | darkskiez | what are the pics? |
| 22:32.27 | terrapen | its going to take me forever to d/l over this slow T1 |
| 22:32.42 | terrapen | dark: pics of Pakistani tribal areas, apparently |
| 22:32.49 | Dr-Linux | what's flickr? :) |
| 22:32.56 | terrapen | it's a free service |
| 22:33.00 | terrapen | you can upload you pictures to them |
| 22:33.28 | terrapen | http://flickr.com/photos/Defender90 <-- my flickr page |
| 22:33.59 | Dr-Linux | terrapen: but that's very orignal and big one, i went my home and i took all of them for justin |
| 22:34.08 | dlynes | tasat: did you try my suggestion earlier of switching to sip info? |
| 22:34.34 | Dr-Linux | terrapen: wow that's a great pic :) |
| 22:36.09 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 22:36.13 | *** join/#asterisk mitcheloc (n=mitchelo@204.8.143.106) |
| 22:37.44 | tasat | dlynes: no DTMF detect, and still the blip -- my providers don't support sip-info apparently, and the blip is still coming in from them (it's not me) |
| 22:40.28 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 22:40.39 | darkskiez | tasat: (untested idea) you could make asterisk generate inband on the way to your phone, so u hear the whole tone if you want that |
| 22:41.01 | *** join/#asterisk bartpbx (n=bartpbx@p54B02E93.dip0.t-ipconnect.de) |
| 22:41.07 | bartpbx | hello |
| 22:41.44 | tasat | darkskiez: actually I don't want to hear it -- is there a way to cover it up? |
| 22:41.48 | bartpbx | I have a little question about iax and the expire of iax peers and users |
| 22:42.11 | bartpbx | anyone online knowing details about the expire feature? |
| 22:42.11 | tasat | what about having the providers change -- is there something they can do? |
| 22:42.52 | bartpbx | we are using realtime but somehow it looks like some peers are not expiering |
| 22:43.10 | darkskiez | tasat, is this really such a big problem? |
| 22:43.35 | distortion | tasat: test with another provider |
| 22:43.43 | darkskiez | people tend not to push buttons whilst on the phone |
| 22:43.51 | distortion | they arent hard to find :) |
| 22:43.54 | darkskiez | talkin |
| 22:44.10 | justinu|laptop | tell your provider their DTMF clamper sucks ass |
| 22:44.32 | tasat | distortion: I've tried four so far... 3 bad, 1 good (on this issue), but the good on this issue, is bad otherwise :) |
| 22:45.06 | tasat | darkskiez: it's a conference setting, so everyone hears -- it's not good |
| 22:45.26 | tasat | justinu|laptop: that's a good idea... I'll quote you |
| 22:45.30 | justinu|laptop | heh |
| 22:45.41 | darkskiez | tasat: aaah. |
| 22:46.56 | darkskiez | is there any nice comparions of sip providers (in the uk/globally) |
| 22:47.52 | darkskiez | i use sipgate, but they seem to be down very frequently |
| 22:48.26 | bartpbx | the expire row in the iax2 show peer list shows what? where can i find the definition of what is shown there? |
| 22:48.49 | bartpbx | google shows me everything but not the answer |
| 22:51.22 | bartpbx | why is the expire increasing? |
| 22:51.42 | kaz0358 | quick question.. what is the most likely cause for no audio on a pri trunk under asterisk? |
| 22:53.08 | darkskiez | kaz0358: how did u diagnose it as far as the pri? |
| 22:54.52 | kaz0358 | darkskiez, well i've gotten far enough that i can make a call from the asterisk box to our avaya pbx over the t1 pri.. i was testing from sip on asterisk to digital phone on the avaya switch.. but to eliminate something weird with sip.. i did a native bridge call initiating the call using /var/spool/asterisk/outgoing |
| 22:55.25 | kaz0358 | i pickup both calls, but neither phones can hear each other... |
| 22:56.09 | kaz0358 | i turned on debugging on the pri span.. and i'm not seeing anything too revealing that might point to the problem |
| 22:56.09 | darkskiez | Not experience with that, but how about your timings? |
| 22:56.35 | thock | Anyone configured a Sangoma A200 with just one FXO module? I need a hand |
| 22:56.38 | kaz0358 | the only weird error i'm getting is: Apr 25 17:43:58 WARNING[7347]: chan_zap.c:7915 zt_pri_error: [Span 0 D-Channel 0] PRI: XXX Missing handling for mandatory IE 12 (cs0, Connected Number) XXX ... but i think that has something to do with not setting the calling number |
| 22:56.47 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 22:57.00 | kaz0358 | darkskiez, the timing is set to slave.. |
| 22:57.11 | *** join/#asterisk btm (n=btm@66.213.193.150) |
| 22:57.15 | kaz0358 | and the avaya switch is acting as a master in this case |
| 22:58.02 | justinu|laptop | kaz0358: that sounds like you have the wrong pri profile |
| 22:58.16 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 22:58.33 | justinu|laptop | i recommend NI2 on both sides, if possible. |
| 22:58.36 | kaz0358 | justinullaptop, pri profile? you mean the channel mappings or encoding type or framing? |
| 22:58.40 | *** part/#asterisk thock (n=thock@216.119.93.253) |
| 22:58.47 | justinu|laptop | none of the above |
| 22:58.58 | justinu|laptop | theres different variants of q931 |
| 22:59.09 | *** part/#asterisk bartpbx (n=bartpbx@p54B02E93.dip0.t-ipconnect.de) |
| 22:59.10 | justinu|laptop | recommended standard nowdays is national-2 |
| 22:59.15 | justinu|laptop | i think it |
| 22:59.24 | justinu|laptop | it's switchtype=national in zapata.conf on the asterisk side |
| 22:59.47 | kaz0358 | it is currently set to national... |
| 22:59.58 | justinu|laptop | and on the pbx side? |
| 23:00.08 | kaz0358 | it is also set to national on the avaya pbx |
| 23:00.21 | justinu|laptop | k |
| 23:00.35 | justinu|laptop | you could try fiddling with the others, perhaps dms or 5ess |
| 23:01.07 | kaz0358 | okay.. i'll run through those thanks |
| 23:02.09 | justinu|laptop | since it's avaya, i bet it supports the 5ess profiles |
| 23:02.22 | tasat | can someone recommend some others w/ low cost tollfree dids like asterlink, nufone (before), etc.? |
| 23:02.51 | generalhan | tasat: you can get toll-free #s through VoicePulse |
| 23:04.32 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 23:04.45 | thock | anyone here help me out with some zaptel.conf help? |
| 23:05.01 | Dr-Linux | SJphone rocks |
| 23:05.16 | Dr-Linux | i used all, but SJphone free and better than all for me |
| 23:07.42 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 23:08.19 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 23:09.31 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 23:10.14 | Peaceful | Is there a way to do introspection on a queue from the dialplan? I just want to get how many customers are currently on the queue before sending someone to it. |
| 23:10.29 | redondos | tset |
| 23:10.32 | redondos | s/tset/test/ |
| 23:10.55 | distortion | either- i just took 5-10 |
| 23:11.51 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 23:12.35 | marl | hi folks, i know this has proberly been asked a number of times, but i cant find the information in google or viop-info, so am asking here : in the uk, is there anyway to find out the provider for a mobile number? as i can phone o2 numbers REALY cheep with one line (but its expensive for other mobiles) and other mobiles cheep on another line, anyone herd of a way to lookup numbers to get the provider, as the usual of looking at the first part of hte dia |
| 23:12.36 | marl | ling code doesnt work for ported numbers :( |
| 23:12.56 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 23:15.06 | marl | if this is the wrong place to ask this, let me know, so i can ask elseeware :) |
| 23:15.35 | papo | brad_mssw: Hm, I understand the problem that asterisk would reinvite to my client connecting from the internet (which is what I want), but also to the one connecting from my LAN which I don't want. Can I solve this with asterisk or should I use some sort of hack, for example running asterisk on two ports, configuring two servers in my client and firewall one from inside and one from outside? |
| 23:15.53 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:16.44 | Dr-Linux | what's main difference between version 1.2.0 and 1.2.7.1 ? |
| 23:17.11 | dlynes | marl: I don't know about for the UK, but in North America you can find out whose CO it belongs to, but it doesn't guarantee that the number hasn't been ported to another service provider |
| 23:17.25 | dlynes | Dr-Linux: main difference? you mean besides oodles of bug fixes? |
| 23:17.47 | Dr-Linux | dlynes: yeah, and features as well |
| 23:18.06 | marl | dlynes, thats the problem ive found here, * is the ideal solution for what i need, but i cant find a way to get the provider if the number has been changed :( |
| 23:18.09 | Peaceful | Dr-Linux: besides features and bug fixes, not much |
| 23:18.16 | Peaceful | hehe |
| 23:18.16 | dlynes | Dr-Linux: don't know if there's any new features or not, but i know there's a lot of bugs fixed between those versions |
| 23:19.07 | *** join/#asterisk faberk (n=faberk@host251-210.pool62211.interbusiness.it) |
| 23:19.12 | Dr-Linux | dlynes: i'm using 1.2.0 and my box uptime is 115 days, have no problems |
| 23:20.01 | dlynes | Dr-Linux: ah...i've had tonnes of problems |
| 23:20.09 | dlynes | Dr-Linux: 1.2.1 had a major memory leak |
| 23:20.26 | dlynes | Dr-Linux: 1.2.7 fixed some pretty bad subscription problems in 1.2.6 |
| 23:20.34 | Dr-Linux | dlynes: if the new versions have some new features then i can upgrade my Pakistan server to new version |
| 23:20.48 | justinu|laptop | i found the memory leak in 1.2.0 |
| 23:20.50 | justinu|laptop | mixmonitor |
| 23:21.03 | dlynes | Dr-Linux: Well, sip subscriptions are improved drastically in 1.2.7 |
| 23:21.11 | Peaceful | Is there a way to get the number of members in a queue from the dialplan? |
| 23:21.34 | dlynes | justinu|laptop: yeah, and there was another problem with mixmonitor introduced in latter versions which would cause asterisk to segfault when the call was terminated |
| 23:21.51 | justinu|laptop | joy |
| 23:21.56 | Dr-Linux | justinu|laptop: how can i verify if i'm facing memory leak problem? :S |
| 23:22.03 | dlynes | justinu|laptop: it's apparently fixed now, though |
| 23:22.06 | justinu|laptop | if you're not running out of memory, you're ok |
| 23:22.14 | dlynes | justinu|laptop: i'm going to try it out tonight |
| 23:22.50 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 23:23.02 | justinu|laptop | i haven't segfaulted yet, using mixmonitor on 1.2.6 |
| 23:23.37 | dlynes | justinu|laptop: i think it was 1.2.4 or 1.2.5 |
| 23:24.00 | dlynes | justinu|laptop: or maybe even earlier than that |
| 23:24.08 | Dr-Linux | what's mixmonitor? |
| 23:24.09 | dlynes | justinu|laptop: can't remember which version i was running on freebsd |
| 23:24.22 | dlynes | Dr-Linux: monitor, with automatic call leg mixing |
| 23:24.49 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 23:24.56 | Dr-Linux | sox :S |
| 23:25.08 | dlynes | but who knows...maybe the segfault bug only existed on freebsd, too |
| 23:25.24 | dlynes | Dr-Linux: without the need to spawn an external process |
| 23:25.24 | Dr-Linux | dlynes: yeh maybe |
| 23:25.27 | justinu|laptop | haha |
| 23:25.40 | Dr-Linux | i'm using RHEL |
| 23:25.47 | dlynes | justinu|laptop: i'm guessing you're not a big lover of freebsd? :) |
| 23:25.59 | Dr-Linux | lol |
| 23:26.09 | justinu|laptop | oh, no... i think it's great |
| 23:26.11 | justinu|laptop | i'm OS agnostic |
| 23:26.18 | justinu|laptop | i just don't have much experience on it |
| 23:26.20 | dlynes | ah...so what's so funny then? |
| 23:26.31 | justinu|laptop | that it would segfault on one OS, and not the other |
| 23:26.34 | justinu|laptop | it's a userspace app |
| 23:26.44 | dlynes | justinu|laptop: Yeah, but different system libraries |
| 23:27.02 | justinu|laptop | i guess anythings possible |
| 23:27.26 | dlynes | justinu|laptop: lots of stuff crashes in windows that doesn't crash in linux |
| 23:27.37 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 23:27.57 | Shaun2222 | how can i get the extension a agent is currently assigned to? |
| 23:28.29 | dlynes | Shaun2222: ${EXTEN} ? ... just a guess on my part...never used agents |
| 23:28.48 | Shaun2222 | no |
| 23:29.08 | Shaun2222 | ${EXTEN} is just the extension that was dialed... |
| 23:29.30 | Shaun2222 | but you just made me realize somthing.. :) one sec. |
| 23:29.42 | dlynes | or maybe ${DNID} ? |
| 23:30.13 | dlynes | on my system, the two are the same, but i'm not using queues, either |
| 23:34.50 | dlynes | Dr-Linux: btw...you can always check the changelog |
| 23:36.05 | Shaun2222 | dlynes: for what i'm doing either am i |
| 23:37.58 | Mavvie | oh fsck. |
| 23:38.20 | Mavvie | mental note: check the presence of the "restart_asterisk" script on the command line before pressing enter. |
| 23:38.37 | Mavvie | just lost all my calls when I reloaded the asterisk drivers. |
| 23:40.04 | *** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 23:41.25 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 23:41.48 | Shaun2222 | anybody know why when i dial a extension that doesnt exist and the call gets sent to voicemail why the phoen throughs a XML parse error and then acts all funny |
| 23:41.55 | Shaun2222 | this is a 7960 phone.. |
| 23:42.40 | Druken | uhmm...if your dialing an exten that doesn't exsist, why are you getting voicemail ? |
| 23:44.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 23:45.39 | *** join/#asterisk reallost1 (n=reallost@12-215-208-184.client.mchsi.com) |
| 23:48.26 | *** part/#asterisk mitcheloc (n=mitchelo@204.8.143.106) |
| 23:48.42 | reallost1 | I'm having a weird problem here: I can make IAX->SIP calls just fine, and even IAX client -> IAX client just fine. However, on IAX Server -> my iax server -> iax client it loses the audio on one side. |
| 23:48.54 | reallost1 | Anyone awake to help out? |
| 23:50.03 | reallost1 | So Asterisk Server -> Asterisk Server -> Sip client = OK; Asterisk Server -> Asterisk Server -> IAX client =NO Audio |
| 23:50.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 23:51.23 | Druken | reallost1: sounds like server a is trying to send iax client the rtp directly, and it's prolly not possible right? |
| 23:51.49 | reallost1 | the iax client is behind a nat |
| 23:52.03 | reallost1 | would notransfer=yes fix that problem? |
| 23:52.21 | Druken | probably |
| 23:52.28 | reallost1 | k, I'll try that now. |
| 23:52.51 | reallost1 | any other possible reasong? |
| 23:52.59 | reallost1 | reasons even. |
| 23:53.22 | *** join/#asterisk austinnichols102 (n=austinni@h4608adc4.area4.spcsdns.net) |
| 23:53.49 | Druken | not that i can think of, but i haven't put alof of thought into it |
| 23:54.09 | *** part/#asterisk thock (n=thock@216.119.93.253) |
| 23:56.13 | reallost1 | I think I may have found it. The particular server had Trunk=yes set |
| 23:56.25 | reallost1 | thanks |
| 23:58.33 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.75.171) |
| 23:59.37 | Druken | oh man.... this sucks ass.... |
