| 00:00.06 | gaz00 | just some config issues. |
| 00:00.21 | lesouvage | iciccomputing: You also can configure it using the webinterface. That's the whole idea about A@H. You should use the same number, secret and codec as you configured on the ata. Then it should work by calling the number. |
| 00:00.21 | mog_work | some config uissues..... |
| 00:00.24 | mog_work | err issues |
| 00:00.42 | gaz00 | it was having some difficulty loading certain packages, and I didn't have the patience to figure out why. so i did another clean install with CentOS first |
| 00:00.46 | gaz00 | and it's just stuck :p |
| 00:00.53 | gaz00 | so in other words -- part laziness. |
| 00:00.55 | *** join/#asterisk fifer (n=sirfifer@207.202.227.161) |
| 00:01.08 | litage | gaz00: was package management the biggest reason you decided against debian? |
| 00:01.11 | gaz00 | it wasn't a big thing - i'm sure that a little legwork would've fixed it right up |
| 00:01.26 | gaz00 | just about. |
| 00:01.29 | gaz00 | afk |
| 00:01.33 | gaz00 | (back in 5) |
| 00:01.34 | litage | mog_work: which issues in particular? i'm curious because i'm using debian, but haven't found many others using it |
| 00:01.37 | litage | thanks gaz00 |
| 00:02.02 | mog_work | no i was talking about a@h |
| 00:02.08 | mog_work | debian is great for asterisk |
| 00:02.09 | fifer | I'm running into an issue wiht FOP. I have to admit that this is the first time I have tried to use it (Flash Operator Panel) |
| 00:02.55 | fifer | Everything is installed, the script is running, the variables.txt is being populated, but when trying to hit it, the browser just spinns. |
| 00:02.59 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 00:03.00 | fifer | No errors in the httpd log |
| 00:03.12 | litage | mog_work: what makes you say that? |
| 00:03.22 | mog_work | what? |
| 00:03.22 | *** join/#asterisk edy2 (i=anonymou@klover226.bitnet.nu) |
| 00:03.33 | litage | mog_work: that debian's great for * |
| 00:03.40 | fifer | Running out of things to check, other than continuing to go over the install instructions |
| 00:03.47 | litage | mog_work: i'm not disagreeing, but simply interested in others' opinions |
| 00:04.05 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 00:04.07 | gaz00 | back. |
| 00:04.08 | mog_work | debian package management is second to none |
| 00:04.10 | wunderkin | mog mog mog |
| 00:04.12 | mog_work | it makes life easy |
| 00:04.13 | gaz00 | litage: |
| 00:04.18 | gaz00 | why do you ask? what are you running? |
| 00:04.19 | mog_work | wunderkin!!!! |
| 00:04.28 | postel | litage: if you're using it and you got no problems with it why do you even care what other people use? |
| 00:04.57 | _Sam-- | debian package doesnt have asterisk-addons with it |
| 00:04.59 | chaos1\ | hi |
| 00:05.09 | _Sam-- | at least, when i put it on a box last week it didnt |
| 00:05.31 | litage | gaz00: i'm running debian exclusively |
| 00:05.36 | _Sam-- | and the only way to get it, is to make your own...and the box i was making didnt have a compiler |
| 00:05.43 | gaz00 | uhoh.. did i just start a "which distro is better" flamewar? |
| 00:05.46 | litage | postel: because i'm always interested in other peoples' findings |
| 00:06.04 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 00:06.06 | litage | gaz00: hah not at all |
| 00:06.06 | chaos1\ | mog_work:which debian would you recommend? |
| 00:06.09 | chaos1\ | gentoo :P |
| 00:06.11 | mog_work | testing |
| 00:06.14 | _Sam-- | so whoever is making the debian package: please make an addons package |
| 00:06.15 | mog_work | or stable |
| 00:06.20 | mog_work | dont use packages |
| 00:06.24 | mog_work | for asterisk |
| 00:06.33 | _Sam-- | it was a solid state device with no hard drives |
| 00:06.42 | gaz00 | ahh... honestly, i'm not very used to debian. and although it looks like it's well put together, I just didn't have the patience nor the time. |
| 00:06.42 | _Sam-- | was easier than making my own |
| 00:06.53 | gaz00 | knoppix! |
| 00:06.55 | gaz00 | OSX |
| 00:06.59 | gaz00 | bsd :p |
| 00:07.10 | chaos1\ | mog: i just want to give that linux to my clients that will run asterisk on it |
| 00:07.10 | mog_work | asterisk packages on all platforms are junk right now |
| 00:07.11 | mog_work | you need to make your own |
| 00:07.11 | chaos1\ | which debian do u recommend |
| 00:07.23 | mog_work | debian stable or debian testing aka sarge and etch |
| 00:07.28 | _Sam-- | i run asterisk on 4 or 5 debian boxes |
| 00:07.36 | _Sam-- | only 1 pre-packaged |
| 00:07.48 | gaz00 | why do you say that it's junk? |
| 00:08.20 | mog_work | asterisk has had 200 commits in the past 10 days |
| 00:08.21 | _Sam-- | i have a debian version with a gui for a compact flash card if anyone is interested |
| 00:08.30 | gaz00 | 200?!?? |
| 00:08.32 | mog_work | how do you expect any package mantainer to keep up? |
| 00:08.40 | mog_work | go to asterisk-commits |
| 00:08.48 | mog_work | this is a busy BUSY project |
| 00:08.51 | gaz00 | how do you run regression testing on 200 independent commits? |
| 00:09.08 | litage | _Sam--: is it a mod of DSL? |
| 00:09.13 | mog_work | there is no regression testing for asterisk proper |
| 00:09.16 | _Sam-- | no mine is for a much larger CF card. |
| 00:09.22 | litage | ah :) |
| 00:09.23 | _Sam-- | 1G CF cards are cheap |
| 00:09.23 | mog_work | asterisk be is tested top down |
| 00:09.34 | mog_work | but asterisk trunk is just moving |
| 00:09.34 | _Sam-- | it uses debian / xdm / fluxbox |
| 00:09.47 | _Sam-- | and runs /var from ram |
| 00:09.52 | mog_work | there are some people implementing regression testing |
| 00:09.54 | mog_work | and other things |
| 00:10.02 | gaz00 | heh..now there's someone that knows how to run a tight ship. |
| 00:10.09 | gaz00 | interesting. |
| 00:10.13 | gaz00 | you a dev, mog_work? |
| 00:10.17 | mog_work | yes |
| 00:10.20 | mog_work | and work for digium |
| 00:10.46 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:11.03 | gaz00 | Ahh.... that would do it. |
| 00:11.16 | gaz00 | build me an intercom then! :p |
| 00:11.18 | _Sam-- | when did openpbx split from asterisk? |
| 00:11.24 | gaz00 | (yes, i'm aware that that isn't what you do) |
| 00:11.25 | mog_work | around the 1.2 |
| 00:11.33 | mog_work | err maybe 2 months before it |
| 00:11.41 | mog_work | so sometimee this summer |
| 00:12.01 | _Sam-- | what are the primary differences right now? |
| 00:12.04 | Ariel_ | hello everyone |
| 00:12.06 | _Sam-- | hardware support? |
| 00:12.08 | mog_work | no |
| 00:12.13 | mog_work | they have autoconf |
| 00:12.18 | mog_work | and sqllite stuff |
| 00:12.26 | mog_work | bases arent that vastly different |
| 00:12.38 | mog_work | mostly just a %s/asterisk/openpbx |
| 00:12.52 | mog_work | and they keep up with some of our bugfixes |
| 00:13.17 | _Sam-- | i think there are many guis in the works |
| 00:13.30 | gaz00 | AMP is no good for you? |
| 00:13.31 | Ariel_ | many I just need one good one that works |
| 00:13.40 | *** join/#asterisk locksy (n=locksy@mrtg.sisgroup.com.au) |
| 00:13.49 | fugitivo | in 5 months we'll have 1 million guis |
| 00:14.09 | Ariel_ | I use amp but it could be allot better. Amp does not support multi-company nor different context level settings unless you customizie it. |
| 00:14.11 | fugitivo | all useless of course |
| 00:14.22 | iccomputing | anyone have the Problem with AMP where it overwrites you sip.conf??? |
| 00:14.24 | gaz00 | you could, theoretically, just customize it! |
| 00:14.37 | gaz00 | what does it overwrite it with? |
| 00:14.42 | _Sam-- | i dont like AMP myself....that is why i have made another one in php |
| 00:14.48 | Ariel_ | iccomputing, use the sip_custom.conf for your settings |
| 00:14.57 | _Sam-- | but i dont think that every config option will ever be available from a GUI |
| 00:15.00 | _Sam-- | there are just too many |
| 00:15.15 | fugitivo | right, that's why guis are useless |
| 00:15.16 | gaz00 | i doubt that it would, nor should. |
| 00:15.25 | _Sam-- | not useless |
| 00:15.28 | _Sam-- | they can save time |
| 00:15.36 | fugitivo | ok |
| 00:15.36 | Ariel_ | GUI's are needed for the user end. And some admins |
| 00:15.37 | gaz00 | GUIs are meant for quick changes.... if one thing is changed regularly, slap it into the GUI |
| 00:15.45 | _Sam-- | especially with realtime now |
| 00:15.48 | gaz00 | deep configs, do it with a text editor. |
| 00:15.50 | _Sam-- | a quick gui to sql tables |
| 00:15.51 | fugitivo | Ariel_: a pbx shouldn't administered by a user :) |
| 00:15.53 | gaz00 | imho. |
| 00:16.01 | gaz00 | lol |
| 00:16.04 | iccomputing | Ariel, I have been using sip_additional.conf and it overwrites it.. |
| 00:16.26 | _Sam-- | i use realtime...and hence having a front end to manage sql data is reasonable |
| 00:16.27 | Ariel_ | fugitivo, yes it does.... newer ones need different times changed, add moh and simple changes |
| 00:16.28 | iccomputing | I like the GUI for the detailed call reporting |
| 00:16.34 | _Sam-- | unless you want to add users from an sql command line |
| 00:16.48 | fugitivo | Ariel_: only a few options |
| 00:16.50 | Ariel_ | additional.conf is aways overriden |
| 00:17.15 | fugitivo | also, a pbx shouldn't depend on a database |
| 00:17.15 | iccomputing | so I need to put my custom extensions in sip_custom.conf |
| 00:17.26 | iccomputing | I got it! Thats what I have been missing!! Thanks! |
| 00:17.27 | Ariel_ | fugitivo, I like working the extensions files myself but I know there is a need and I see it every day. |
| 00:17.33 | _Sam-- | fugitivo: arent you the same person who argued that any GUI (ie x-windows) for server management isnt "serious admin"? |
| 00:17.34 | Ariel_ | iccomputing, yes |
| 00:17.47 | fugitivo | _Sam--: yes, that's me |
| 00:17.49 | _Sam-- | you are just a fool. |
| 00:17.55 | _Sam-- | a time wasting fool at that. |
| 00:17.57 | cripito | ok what i am missing :) |
| 00:18.02 | iccomputing | I have another question...I have one ext. that keeps reporting 401 Unauthorized in my SIP Debug. any thoughts? |
| 00:18.03 | gaz00 | lol |
| 00:18.07 | fugitivo | _Sam--: watch your fingers... |
| 00:18.11 | _Sam-- | keep adminning your boxes from your text console and flat text files |
| 00:18.17 | _Sam-- | and keep telling everyone else they are wrong |
| 00:18.19 | _Sam-- | thats smart |
| 00:18.49 | fugitivo | _Sam--: well, YOU are not smart |
| 00:19.01 | _Sam-- | smart enough to know what works for ME |
| 00:19.04 | iccomputing | I know the passwords and such are correct...I think it may be a NAT problem =( |
| 00:19.07 | _Sam-- | and what saves time |
| 00:19.07 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-48-187.cybersurf.com) |
| 00:19.10 | Ariel_ | ~google sip 401 |
| 00:19.27 | _Sam-- | but for you to sit there and say any GUI is useless, and that SQL is stupid and shouldnt be used...you are a jackass my friend. |
| 00:19.36 | fugitivo | _Sam--: telling people that they're fools doesn't make you a smart guy... |
| 00:19.38 | cripito | well u can't put all the settings in a gui |
| 00:19.44 | cripito | but the mayority yes |
| 00:19.44 | iccomputing | I have looked through most of these threads already |
| 00:19.44 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 00:20.10 | Ariel_ | lets not go there people. They all have there needs. Just don't get into the MySql or SQL or prot.... |
| 00:20.24 | fugitivo | _Sam--: and putting words in my mouth... you are not smart at all |
| 00:20.38 | Ariel_ | iccomputing, post the complete error on pastebin lets look at it. |
| 00:20.48 | cripito | i use my own gui.. build for us with the things we usually work and no complain till now |
| 00:20.51 | iccomputing | thanks! i am doing it now. |
| 00:20.52 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 00:20.58 | Ariel_ | Que pasa aque... |
| 00:21.10 | cripito | :) nada ariel_ solo comentando :P |
| 00:21.17 | _Sam-- | i would rather have a debate with a representative fromt he special olympics |
| 00:21.26 | Ariel_ | lol |
| 00:22.02 | iccomputing | http://pastebin.com/510772 |
| 00:22.57 | Netgeeks | GUIs will save the world ;) |
| 00:23.41 | fugitivo | see AMP, you still need *_custom.conf |
| 00:24.03 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:25.06 | Netgeeks | Thats AMP, just one GUI, written by someone who had certain needs, doesn't mean that everyone has the same needs |
| 00:25.18 | fugitivo | all guis have the same problem |
| 00:25.30 | Ariel_ | iccomputing, strange error it's not finding it's context correctly. |
| 00:25.30 | Netgeeks | and that is? |
| 00:25.44 | Ariel_ | Netgeeks, there gui's |
| 00:25.53 | Netgeeks | *sigh* |
| 00:26.07 | Netgeeks | thats not a problem, thats a statement all oranges are oranges |
| 00:26.13 | Netgeeks | all rocks are rocks |
| 00:26.27 | iccomputing | i dotn think the NAT is the problem...i dmz'd the phone basically...i forwarded 5060-5070 (sip) and 2000-2700 (rtp) in and out of my router.. |
| 00:26.40 | iccomputing | is that context something I can change? |
| 00:26.52 | fugitivo | if you have to edit a text file, it makes the gui useless, just edit all files at hand and you don't need to install a gui :) |
| 00:27.02 | _Sam-- | Netgeeks: save your keystrokes...its a wasted cause. |
| 00:27.12 | _Sam-- | some of us know the value of a gui and what it can do for us. |
| 00:27.20 | _Sam-- | like adding 200 users to a flat file...fun |
| 00:27.27 | iccomputing | how can you get dtailed call reporting without a GUI? |
| 00:27.50 | Jameno123 | iccomputing, ncurses ;) |
| 00:27.52 | fugitivo | _Sam--: you need a gui for that? i see you don't have enought experience to know how to use all linux tools |
| 00:27.58 | Ariel_ | iccomputing, 2000-2700.... did you do this change in the rpt.conf? it should be around 10,000 - 11000 at least that is what I use. |
| 00:28.02 | Netgeeks | Sam, no, I'd actually like to get others opinions. I wrote a gui (and am in the process of extending it) and I would really want to know the opinions of folks so if they have something valid I can incorporate I want to do so |
| 00:28.37 | *** join/#asterisk Cool_One (n=bclinton@h84.65.255.206.cable.htsp.cablelynx.com) |
| 00:28.40 | Beirdo | here's my opinion... GUIs are ick... :) |
| 00:28.45 | iccomputing | i did change this, 2000-2700 is what I have designated for RTP..i have some other things (video) that use 10k and up |
| 00:28.57 | Jameno123 | UI = 1 thing |
| 00:29.00 | Jameno123 | GUI = another |
| 00:29.18 | Ariel_ | iccomputing, ok you made the changes in the /etc/asterisk/rpt.conf.. does this phone support those lower numbers? |
| 00:29.20 | fugitivo | Netgeeks: well, if you are capable of doing a gui, that won't need me to edit a file at hand, then your gui won't be useless |
| 00:29.23 | Netgeeks | everyone has thier right to thier opinion. I don't think stomping on people because they see things differently than I is any good. Better to try and see the world as they see it, so you have a different point of view |
| 00:29.32 | Beirdo | meh |
| 00:29.35 | iccomputing | RTP port wouldnt cause my context and sip to be wrong tho correct? |
| 00:29.41 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 00:29.42 | fugitivo | Netgeeks: and it'll have to be faster than editing a file with vi... |
| 00:29.45 | Netgeeks | My gui isn't targeted to people who could find thier way here |
| 00:29.50 | iccomputing | yes. I set the phone to match |
| 00:30.00 | _Sam-- | *yawn* |
| 00:30.32 | Cool_One | Who is the Asterisk God in this channel |
| 00:30.40 | _Sam-- | fugitivo |
| 00:30.48 | d-tech | the TEXTMASTER |
| 00:30.49 | fugitivo | Cool_One: _Sam-- is the smart guy here |
| 00:30.50 | Ariel_ | iccomputing, you do have nat=yes on the extensions settings and also if your asterisk box is behind a nat you are using externip= and localnet= |
| 00:31.07 | fugitivo | i'm the fool that still uses terminals |
| 00:31.13 | Ariel_ | I did not know there was such a thing as asterisk god. |
| 00:31.15 | _Sam-- | i use terminals as well |
| 00:31.16 | _Sam-- | just in x |
| 00:31.23 | Cool_One | ok ok |
| 00:31.33 | iccomputing | asterisk is on public IP no nat...I do have NAT=YES in my sip_additional.conf |
| 00:31.35 | Cool_One | I am a dumb @()*(* newbie |
| 00:31.35 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 00:31.40 | Cool_One | I admit it |
| 00:31.42 | Ariel_ | Cool_One, ask the question someone might help |
| 00:31.46 | Netgeeks | Mine was written to break down the admin of a hosted vPBX/ITSP/Wholesale system into multiple management levels so users could manage thier preferences, and managers could handle thier stuff, and system admins thier stuff.... no need for one person to change configs if joe user wants his cell phone to ring on roll-over from sip phone, etc. |
| 00:31.57 | Netgeeks | so in that case I saw the need for a gui |
| 00:32.10 | _Sam-- | Netgeeks: you dont have to justify yourself to anyone! |
| 00:32.14 | *** join/#asterisk XARiUS (n=bdarcy@70.230.211.183) |
| 00:32.17 | _Sam-- | as long as it does something for YOU |
| 00:32.20 | fugitivo | Netgeeks: i agree with that |
| 00:32.24 | Cool_One | I can not get my Grandstream GXP2000 phone to register on my asterisk server |
| 00:32.27 | fugitivo | Netgeeks: i don't agree with a gui for admins |
| 00:32.32 | *** join/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 00:32.40 | _Sam-- | Cool_One: what does the console say |
| 00:32.56 | Cool_One | Jan 17 18:24:40 NOTICE[1159]: chan_sip.c:7673 handle_request: Registration from '<sip:201@192.168.1.10;user=phone>' failed for '192.168.1.201' |
| 00:32.58 | Ariel_ | GS work with asterisk without much issues. |
| 00:33.17 | XARiUS | Anyone ever had any problems bridging calls coming thru asterisk, out to an IAX terminator to a cellphone? |
| 00:33.25 | Ariel_ | looks like both are on same network from that post |
| 00:33.41 | Ariel_ | XARiUS, yes it happens |
| 00:33.50 | XARiUS | it's just this one phone and it's got me baffled.. |
| 00:33.57 | XARiUS | a sprint phone. |
| 00:34.28 | XARiUS | hrmf, go figure.. |
| 00:34.49 | Ariel_ | you need to add a wait to do the connection |
| 00:34.59 | XARiUS | yeah cause I've got verizon, tmobile and a few other cellphones which work just fine |
| 00:35.09 | XARiUS | but this one guys sprint just won't bridge, * releases both legs of the call |
| 00:35.43 | XARiUS | a wait ... at the iax termination point I would assume you mean, Ariel? |
| 00:36.14 | XARiUS | my * box just hands calls off to my iax provider.. so don't see how me doing a wait would accomplish anything.. lol |
| 00:37.57 | iccomputing | what else could my context be other than "from-internal"?? |
| 00:37.59 | *** join/#asterisk litage_ (n=nick@203.220.55.70) |
| 00:38.34 | Ariel_ | iccomputing, that is the one it needs for amp |
| 00:38.42 | iccomputing | ahh |
| 00:38.44 | iccomputing | ok |
| 00:39.23 | fugitivo | XARiUS: what are you exactly trying to do? |
| 00:39.42 | XARiUS | fugi, folks just have their cellphones ringing off their main extensions + their desk phones |
| 00:39.43 | Ariel_ | XARiUS, I had a strange macro along time ago for that but it was via a pri line. |
| 00:40.00 | XARiUS | ie: SIP/yourphone&IAX2/provider/15555551234 |
| 00:40.18 | XARiUS | so that if htey are out of the office or wherever they get a chance to snag the call |
| 00:40.32 | XARiUS | it's worked flawlessly for ages till this one guy just got a new sprint phone |
| 00:40.35 | iccomputing | thanks for the help all! im outa here! |
| 00:40.51 | XARiUS | wish I had pri's like I used to... |
| 00:40.54 | XARiUS | but I don't work there anymore.. lol |
| 00:41.04 | fugitivo | XARiUS: do you get any error or..? |
| 00:41.12 | *** part/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 00:41.15 | XARiUS | well, there is different behaviour when it works and when it doesn't work.. |
| 00:41.20 | XARiUS | lemme pastebin the console details |
| 00:41.29 | fugitivo | ok |
| 00:42.17 | *** join/#asterisk Berkey (n=Berkey@nv-208-13-143-215.sta.sprint-hsd.net) |
| 00:42.31 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 00:44.17 | *** part/#asterisk Berkey (n=Berkey@nv-208-13-143-215.sta.sprint-hsd.net) |
| 00:44.37 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 00:44.44 | *** join/#asterisk JMcA (n=jmcadams@71.31.33.169) |
| 00:44.58 | XARiUS | k sorry, had to XX out numbers to protect the innocent. |
| 00:44.58 | XARiUS | http://pastebin.ca/37148 |
| 00:45.42 | XARiUS | the failed call is landline OR cellphone --> asterisk --> iax terminator --> destination cell phone.. |
| 00:45.58 | XARiUS | the calls that WORK are: SIP/device --> asterisk --> iax terminator --> destination cell phone |
| 00:46.41 | XARiUS | both destination cell phone's in each case being the offending sprint phone. |
| 00:47.09 | XARiUS | Channel 'IAX2/xtech-jnctn-3' ready to transfer <-- that, I've actually never seen before today. |
| 00:47.59 | JMcA | how is the landline or cellphone in the first instance getting into asterisk? |
| 00:48.16 | XARiUS | thru the same IAX provider. |
| 00:48.25 | JMcA | ah |
| 00:48.34 | XARiUS | inbound thru iax provider, into *, * sends it back out same provider to sprint phone |
| 00:49.16 | *** join/#asterisk jebba (n=jebba@ip-216-17-203-198.rev.frii.com) |
| 00:49.38 | JMcA | do you have notransfer=yes on the peer? |
| 00:49.45 | fugitivo | XARiUS: what is biz-stdexten? |
| 00:50.09 | XARiUS | it's this particular company's standard extension macro logic |
| 00:50.14 | XARiUS | I have about 7 companies virtualized on one * box |
| 00:50.20 | fugitivo | cool |
| 00:50.36 | XARiUS | company1-stdexten, company2-stdexten etc, if they want different behaviours. |
| 00:50.40 | *** join/#asterisk detatch (i=detent@dhcp-100.fresno-dc2.brandxnet.com) |
| 00:50.43 | XARiUS | all in seperate conf files to make like fairly sane. |
| 00:51.09 | XARiUS | yay for #include is all I can say about that. |
| 00:51.10 | XARiUS | :) |
| 00:51.12 | detatch | anyone here familular with the asterisk call manager? |
| 00:51.29 | JMcA | XARiUS: generally a good bet with config files for most systems |
| 00:51.47 | XARiUS | yeah well when I first started with asterisk, you could include other contexts.. but not other files lol |
| 00:51.54 | XARiUS | 0.7.2 I think it was? I forget.. |
| 00:52.09 | JMcA | ya...even just including other contexts is good...but other files is awesome |
| 00:52.14 | Ariel_ | .5 you could include other files |
| 00:52.23 | JMcA | so, when did dundi get added? |
| 00:52.32 | Ariel_ | dundi 1.2 |
| 00:52.45 | XARiUS | so anyway, any more light u can shed on that wait statement ariel? or with sprint and nextel are you just sometimes hosed? |
| 00:52.46 | Ariel_ | actually there was a patch for 1.05 along time ago . |
| 00:52.46 | fugitivo | XARiUS: do a sip debug, that pastebin doesn't give us too much info |
| 00:52.50 | JMcA | ah...and, of course, suse packages 1.0.9 :/ |
| 00:53.01 | XARiUS | well, there won't be any sip debug... |
| 00:53.07 | XARiUS | it's iax --> asterisk --> iax --> sprint cellphone |
| 00:53.08 | XARiUS | :) |
| 00:53.11 | *** join/#asterisk astoria (n=tom@user-7e5a43.user.msu.edu) |
| 00:53.21 | XARiUS | sip device -> asterisk -> iax -> sprint cellphone works just fine lol |
| 00:53.31 | astoria | How long does it typically take to provision an Iaxy? |
| 00:53.44 | *** part/#asterisk kratzers (n=kratzers@65.119.216.4) |
| 00:53.54 | fugitivo | XARiUS: oh, the SIP one works :) |
| 00:53.55 | JMcA | XARiUS: do you have notransfer set? |
| 00:54.03 | XARiUS | lemme check |
| 00:54.59 | XARiUS | nope it's not, I can try setting it on the peer, one sec. |
| 00:55.34 | fugitivo | XARiUS: can you pastebin your iax setup for xtech-jnctn-3? |
| 00:56.17 | XARiUS | sure, it's identical to biz-jnctn, just different usernames. |
| 00:56.29 | fugitivo | did you pastebin that one? |
| 00:56.38 | Ariel_ | astoria, after you setup the files about 1.5 minutes |
| 00:56.48 | Ariel_ | it's it's on the same network |
| 00:56.54 | XARiUS | not yet but I will |
| 00:57.13 | astoria | Ariel_: wow, I didn't expect it to take that long. I'll have to let it sit a while. |
| 00:57.34 | astoria | Ariel_: I did a tethereal to make sure things were going past my firewall OK, and it sent a PROVISION packet at first and nothing since... |
| 00:59.18 | XARiUS | bleh, the offending cellphone is currently in use, I can't test the notransfer just yet. |
| 01:00.18 | XARiUS | fugitivo: http://pastebin.ca/37151 |
| 01:03.31 | XARiUS | sweet, I think notransfer=yes fixed it |
| 01:03.37 | JMcA | I thought it might |
| 01:03.48 | detatch | can someone take a look at this? http://pastebin.com/510823 |
| 01:03.55 | XARiUS | nice dude, appreciate it. :) |
| 01:03.59 | JMcA | glad to help |
| 01:04.21 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 01:04.22 | detatch | im currently in the middle of building a new asterisk pbx with the new 1.2.1 code |
| 01:04.48 | detatch | the pastebin is output from a call manager 'show queue $quename' |
| 01:05.02 | detatch | running on the old 1.0.3 code |
| 01:05.52 | detatch | on the new 1.2.1 the "(brand: "brand-name" <callerid>)" isnt there (we use * to run a branded call-center) |
| 01:06.23 | ManxPower | Well my tires are fixed. |
| 01:06.40 | detatch | any help would be appreciated |
| 01:08.16 | JMcA | ok...nothing purchased yet for my home, so, pros and cons of a sipura 3000 vs a TDM-400p w/ 1 FXS and 1 FXO |
| 01:09.10 | Mark_Halverson | i've been told the 3000 has issues...i am running the digium card with 2 fxo ports - very happy |
| 01:09.36 | JMcA | Mark_Halverson: any issues that you've heard of specifically? |
| 01:10.01 | JMcA | 'cause the digium card is more expensive from what I've seen than the 3000 |
| 01:10.54 | Ariel_ | JMcA, I have used both and they both work fine |
| 01:11.14 | fugitivo | JMcA: i can only talk about the tdm400p |
| 01:11.22 | Ariel_ | I like the sipura do to if you have any power problem it can by pass to the line |
| 01:11.47 | fugitivo | JMcA: i have issues where i need to reboot the server (once in a while) |
| 01:11.49 | Ariel_ | But if your going to get faxes then the tdm11b is the only way to detect them. |
| 01:11.51 | JMcA | Ariel_: yeah...I didn't know if the tdm could do the bypass, guess not, I would assume it would at least require power |
| 01:12.12 | Ariel_ | correct no bypass |
| 01:12.35 | JMcA | faxes would be a feature that I might use once a year...if I need to send or receive a fax right now, I just take it to work |
| 01:12.48 | fugitivo | Ariel_: no faxes with the sipura? |
| 01:12.55 | JMcA | fugitivo: yeah...i have to bounce my wrt-54g occasionally, too *shrug* |
| 01:12.58 | fugitivo | JMcA: you can get a x100p for faxes, it's $20 |
| 01:13.04 | Ariel_ | don't you just love this.. boil nipples for 5 minutes before use... (baby bottles). |
| 01:13.21 | Ariel_ | fugitivo, can't detect it. unless you get nwdetect |
| 01:13.36 | ast_new-B | is there any software to build dialplan such as web based dialplan ? |
| 01:13.51 | Ariel_ | lol no not a gui |
| 01:13.54 | Netgeeks | lol |
| 01:13.57 | Nugget | yay google talk is open now! |
| 01:14.07 | *** join/#asterisk alladin^ (n=alladin@venere.matrixlabs.org) |
| 01:14.23 | Netgeeks | ast_new-B try looking for Asterisk@Home or AMP |
| 01:14.29 | alladin^ | hi ppl |
| 01:14.35 | Netgeeks | Asterisk Management Portal I think is the acronym's full name |
| 01:14.40 | fugitivo | Netgeeks: what's your gui? |
| 01:14.44 | Ariel_ | ok it's boiling... 5 minutes needed.... |
| 01:14.46 | alladin^ | got a strange problem upgrading from asterisk 1.0.9 to 1.2.1 |
| 01:14.47 | JMcA | Nugget: yup...a bit buggy from what I've seen...or perhaps exposing bugs in other server implementations...but open |
| 01:14.49 | Netgeeks | http://www.netgeeks.net/ |
| 01:14.50 | JMcA | makes me happy |
| 01:14.59 | ast_new-B | Netgeeks: yes i've test it ... but it's lake lot of feature |
| 01:15.00 | alladin^ | i've some fritz!box clients who are unable to register |
| 01:15.14 | Ariel_ | lakess |
| 01:15.19 | ast_new-B | no dial call limit, duration limit |
| 01:15.19 | Ariel_ | or lacks |
| 01:15.22 | alladin^ | asterisk complains about Failed to parse contact info |
| 01:15.22 | *** join/#asterisk tengulre11 (n=tengulre@222.90.66.4) |
| 01:15.27 | alladin^ | any hints? Thanks. |
| 01:15.37 | Netgeeks | ast_new-B thats a major issue with gui's if you make them simple enough for beginers, then you run into the missing features |
| 01:15.39 | Nugget | JMcA: yeah, I had to ditch jabberd2 and move to ejabberd to get it talking |
| 01:15.40 | *** join/#asterisk _adrian (n=adrian@user-1758.lns1-c9.dsl.pol.co.uk) |
| 01:15.48 | Nugget | but I'd been meaning to do that for a while now. |
| 01:15.53 | Netgeeks | if you add all the features, then the gui is more complex than building the configs by hand |
| 01:15.55 | Ariel_ | that one is easy to fix |
| 01:15.57 | JMcA | I did a connection from gizmo to gtalk |
| 01:16.08 | mog_work | jabberd2 forever! |
| 01:16.10 | JMcA | every time I started typing on gizmo, I got a blank line in the gtalk window |
| 01:16.14 | mog_work | or till jabber3 |
| 01:16.16 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-42-158.cybersurf.com) |
| 01:16.16 | alladin^ | I think that's because the missing Contact: header in the registration request, but the old version was derivating contact data from other registration info |
| 01:16.37 | ast_new-B | Coz ... i'm a new bie ... can you give a sample dialplan script for duration call limit ? |
| 01:16.38 | alladin^ | any patch out there? :) |
| 01:16.50 | ast_new-B | Netgeeks: Coz ... i'm a new bie ... can you give a sample dialplan script for duration call limit ? |
| 01:16.56 | JMcA | I'm assuming the typing notification is generating the extra line(s) |
| 01:16.58 | wunderkin | mog are you still at work? |
| 01:17.26 | Netgeeks | at the CLI do a show application dial you'll see there is a duration limit argument you can pass to the dial commant |
| 01:17.30 | Netgeeks | command |
| 01:17.48 | Netgeeks | I think it's S(<seconds>) |
| 01:17.55 | Netgeeks | not sure off the top of my head |
| 01:18.17 | ast_new-B | and what the best suite for billing system ? |
| 01:18.41 | Netgeeks | I have no idea on that one... I would say mine, but it's not ready for release yet |
| 01:18.54 | Netgeeks | alot of people mention a2billing |
| 01:19.12 | ast_new-B | a2billing is integrated with AAH .... |
| 01:19.53 | fugitivo | Netgeeks: the user interface looks very usefull |
| 01:20.15 | Ariel_ | Netgeeks, it looks nice. How can I get a version of it?? |
| 01:20.26 | Netgeeks | I'll have an online demo with docs up sometime next week, it will be alot easier to get an idea of the function then |
| 01:20.36 | FuriousGeorge | Which word of four letters can be added to the front of the following words to create other English words? |
| 01:20.36 | FuriousGeorge | CARD BOX CODE BAG HASTE |
| 01:20.41 | Ariel_ | ok 1 minute more for the water |
| 01:20.50 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 01:20.51 | FuriousGeorge | i thought code would give it away for me but it doesnt\ |
| 01:21.10 | Netgeeks | Ariel: not sure yet, its a commercial project, but I'm thinking of releasing a conneciton limited verison that would be fine for small users |
| 01:21.16 | JMcA | FuriousGeorge: "post"? |
| 01:21.21 | ast_new-B | Netgeeks: where's the demo url ? |
| 01:21.27 | ManxPower | I can't believe there isn't a NAMPA VSC for Call Pickup (group) |
| 01:21.28 | FuriousGeorge | posthaste? |
| 01:21.38 | FuriousGeorge | whats a postbag |
| 01:21.48 | JMcA | could work |
| 01:21.48 | FuriousGeorge | lol |
| 01:21.55 | Netgeeks | no demo, just screen shots: http://www.netgeeks.net/ it's not a AMP or A@H gui, it's targeted more towards Service providers |
| 01:22.07 | FuriousGeorge | posthaste is a word? |
| 01:22.23 | ManxPower | FuriousGeorge, Yes. It means "do it quickly" |
| 01:22.23 | JMcA | sure...maybe colloquial...means quickly |
| 01:22.40 | _adrian | and a bag the post man uses to carry mail! |
| 01:22.42 | fugitivo | FuriousGeorge: HOST |
| 01:22.48 | Netgeeks | one would think it's two word hyphenated post-haste |
| 01:22.58 | JMcA | Netgeeks: one would think that, but I don't really know |
| 01:23.07 | ManxPower | http://m-w.com/dictionary/posthaste |
| 01:23.12 | FuriousGeorge | hosthaste, hostgift, hostbag, hostbox? |
| 01:23.26 | JMcA | gift? |
| 01:23.29 | JMcA | where did gift come from? |
| 01:23.31 | *** join/#asterisk chaos1\ (i=anonymou@klover226.bitnet.nu) |
| 01:23.32 | ManxPower | bad hoax? |
| 01:23.33 | FuriousGeorge | isnt a hostbox a tabernacle? |
| 01:23.36 | fugitivo | i was going to ask the same question |
| 01:23.38 | chaos1\ | hi |
| 01:23.48 | FuriousGeorge | i mean HOSTBAG? |
| 01:24.02 | fugitivo | no no |
| 01:24.05 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 01:24.17 | chaos1\ | i would be joining a company for sip deployments |
| 01:24.20 | ManxPower | http://m-w.com/info/05words.htm Interesting |
| 01:24.22 | FuriousGeorge | i mean the one i didnt say yet |
| 01:24.24 | chaos1\ | i will be a testing engineer |
| 01:24.31 | chaos1\ | any advice |
| 01:24.42 | fugitivo | SOFT |
| 01:24.44 | chaos1\ | or things to read for ip telephony |
| 01:24.56 | fugitivo | ~docs |
| 01:24.57 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:25.26 | JMcA | chaos1\: protocol specs and other related docs...don't worry so much about docs for specific products until you've got a good grounding in how the protocols work |
| 01:26.10 | chaos1\ | thanks, i have finished reading asterisk guides usually those comprehesnive pdfs |
| 01:26.39 | JMcA | go read the IETF RFC's for SMTP, HTTP, and SIP |
| 01:26.39 | chaos1\ | configured asterisk for x100p and know how to make basic dial plans and stuff |
| 01:26.45 | chaos1\ | but i definitely want to improve on it |
| 01:26.56 | fugitivo | only sip? |
| 01:27.07 | chaos1\ | yup only sip |
| 01:27.17 | fugitivo | then you don't need asterisk docs |
| 01:27.19 | FuriousGeorge | JMcA: your right postbag is a word |
| 01:27.21 | JMcA | fugitivo: I wouldn't wish reading the H.323 protocols on my worst enemy |
| 01:27.25 | FuriousGeorge | its the postman's bag |
| 01:27.40 | chaos1\ | fugitivo:no, they are using asterisk because it integerated with sip very well |
| 01:27.41 | JMcA | FuriousGeorge: wow...I just kinda tossed that off |
| 01:27.53 | fugitivo | asterisk and sip very well? lol |
| 01:28.16 | FuriousGeorge | JMcA: well i dunno if the test will say its right, but ill let you know |
| 01:28.17 | chaos1\ | JMcA:i am not going to develop those protocols, i am just to install,configure the systems |
| 01:28.33 | ast_new-B | how can i monitor each channel in asterisk and record it ? |
| 01:28.33 | JMcA | chaos1\: yeah...you should know the protocols anyway |
| 01:28.42 | chaos1\ | i just hope they provide me the configurations which i can go and install. life would be difficult if i am expected to configure them too |
| 01:28.49 | JMcA | fugitivo: from what I've seen...asterisk does sip better than Cisco IOS does |
| 01:29.01 | pfn | so when are we gonna get SIP/TCP? |
| 01:29.01 | fugitivo | http://www.voip-info.org/wiki-SIP |
| 01:29.02 | FuriousGeorge | If a circle is one, how many is an octagon? 2 4 6 8 12 |
| 01:29.05 | chaos1\ | ast_new-B:isn't that possible through typing help at asterisk command line console |
| 01:29.21 | JMcA | 8 |
| 01:29.21 | justinu | is it 2? |
| 01:29.23 | _adrian | 8 |
| 01:29.32 | chaos1\ | there are always some great commands, e.g. i had to authenticate a user by passwords and found a very nice tag for that to use in extensions.conf all from asterisk> |
| 01:29.38 | fugitivo | FuriousGeorge: iq test? asking other people isn't valid |
| 01:30.00 | FuriousGeorge | fugitivo: the test im taking isnt valid :) its a fun test, what fun is it if i dont share |
| 01:30.00 | chaos1\ | fugitivo:thanks man! |
| 01:30.12 | chaos1\ | hey,any advice on attitude and chit chat stuff |
| 01:30.13 | fugitivo | FuriousGeorge: send the url! |
| 01:30.31 | chaos1\ | what kind of talkin is expected from a test,install guy? Dos and donts |
| 01:31.04 | FuriousGeorge | is a circle considered to have a side |
| 01:31.06 | ast_new-B | chaos1\: thank's for advice |
| 01:31.10 | FuriousGeorge | ? |
| 01:31.20 | FuriousGeorge | http://www.mensa.org/workout2.php |
| 01:31.25 | fugitivo | mensa! |
| 01:31.27 | JMcA | FuriousGeorge: not really, but sorta |
| 01:31.30 | fugitivo | mensa has some great tests |
| 01:31.31 | _adrian | inside and outside ! |
| 01:31.44 | JMcA | FuriousGeorge: it kinda only has one side, it kinda has an infinite number of sides |
| 01:31.45 | FuriousGeorge | JMcA: i guess were sorta right |
| 01:31.59 | fugitivo | i think it doesn't have sides |
| 01:32.02 | fugitivo | what is a side? |
| 01:32.31 | fugitivo | hmm, maybe it has |
| 01:32.39 | fugitivo | "an extended outer surface of an object" |
| 01:33.21 | chaos1\ | kewl puzzle |
| 01:33.56 | chaos1\ | hey! one big problem |
| 01:34.05 | *** join/#asterisk _DAW-LAPTOP (n=DAW-LAPT@adsl-156-90-253.msy.bellsouth.net) |
| 01:34.06 | chaos1\ | the lingo used in ip telephony fleet meetings |
| 01:34.19 | chaos1\ | t1,e1,dsu,nat block,etc... |
| 01:34.35 | *** join/#asterisk usam (n=usam@203.156.61.41) |
| 01:34.52 | FuriousGeorge | there are only a few i cant get on that test, theres the one about the sides (which you guys seem to agree on) theres the one about posthaste which smarty pants over there got, i cant find what other word i can make from "insatiable" |
| 01:34.55 | JMcA | chaos1\: there is lots of online info about those |
| 01:35.00 | FuriousGeorge | and one more doesnt make sense |
| 01:35.32 | usam | hello, Im trying to get asterisk to register to a sip server which is asking me for the MAC address if im using the sip phone. . if i want asterisk to register to it, which MAC address shoudl I use? the MAC of the NIC? |
| 01:35.59 | JMcA | usam: why is it asking for the MAC? |
| 01:36.25 | JMcA | oh |
| 01:36.33 | usam | JMcA: so that i cant use their service on multiple phones? |
| 01:36.46 | JMcA | usam: is this a remote server? |
| 01:37.03 | usam | 1: Special Your MAC #: (info) |
| 01:37.04 | usam | 2: Choose Your Plan: (info) |
| 01:37.39 | JMcA | if its a remote server, you can probably put in whatever you want 'cause there is no way they can know what it really is |
| 01:37.42 | dogtanian | the mac address normally needs to be specified so that TFTP can check/update phone firmware |
| 01:38.06 | JMcA | TFTP doesn't have anything to do with MAC addresses</nit-pick> |
| 01:38.13 | JMcA | more likely DHCP |
| 01:38.21 | usam | hm... lets try to signup without the MAC address then. |
| 01:38.23 | dogtanian | not for cisco phones |
| 01:38.48 | chaos1\ | how good are snorms |
| 01:38.50 | dogtanian | the cisco phones pick up their firmware by looking for the file xxxmacaddressxxx.cnf |
| 01:38.52 | chaos1\ | any experience |
| 01:38.57 | JMcA | not for anything...TFTP is built on UDP/IP...it doesn't know about MAC's at all |
| 01:39.23 | JMcA | ah...ok...that's just a filename...it doesn't really have anything to do with the MAC address in reality |
| 01:39.54 | JMcA | if its a remote server, though...DHCP/TFTP isn't in play |
| 01:40.35 | usam | they might have the "identify by MAC" to access their network? |
| 01:40.52 | dogtanian | yup |
| 01:40.59 | jerlique | hi - does asterisk 1.2 have a problem with its console dial() command? I get "No such command 'dial' (type 'help' for help) |
| 01:41.02 | jerlique | " |
| 01:41.03 | dogtanian | the mac will still be in the packet headers |
| 01:41.15 | usam | ok.. |
| 01:41.17 | dogtanian | although that may mean you need to use the mack of the router |
| 01:41.21 | jerlique | sorry thats version 1.2.1 |
| 01:41.26 | *** part/#asterisk _adrian (n=adrian@user-1758.lns1-c9.dsl.pol.co.uk) |
| 01:41.38 | JMcA | dogtanian: not unless the server is on the same network |
| 01:42.36 | usam | so i should provide the mac of the router? .. cant be,.. coz i use register the MAC of the ATA, then it shouldnt the MAC of the rotuer. |
| 01:42.36 | dogtanian | well, my assumption is that it should be ok if you leave it blank |
| 01:42.38 | usam | ok |
| 01:42.40 | usam | lets try . |
| 01:42.53 | dogtanian | i have a feeling it just requests mac incase you're connecting a cisco 79xx |
| 01:43.01 | dogtanian | i may be wrong tho |
| 01:43.32 | usam | they are giving away SPA 1001 .. it might be something with it |
| 01:49.39 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 01:50.00 | *** join/#asterisk CoolAcid (n=jason@216.99.98.39) |
| 01:50.55 | riddlebox | so, I couldnt get any AGI stuff working with python, but it works great with c, can someone explain this? |
| 01:51.58 | *** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 01:52.28 | *** part/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com) |
| 01:53.09 | *** join/#asterisk Soul (n=Soul@87-196-32-99.net.novis.pt) |
| 01:55.33 | FuriousGeorge | banalities is the only word you can make from insatiable |
| 01:55.48 | FuriousGeorge | JMcA: we were right but it wasnt so much sides as lines |
| 01:55.57 | FuriousGeorge | circle has one octogon has 8 |
| 01:56.50 | detatch | can someone take a look at this? http://pastebin.com/510823 |
| 01:56.52 | FuriousGeorge | and POST was right on, i didnt know posthaste was a word :) |
| 01:57.01 | detatch | im currently in the middle of building a new asterisk pbx with the new 1.2.1 code |
| 01:57.05 | trixter | octagon doesnt have 8 lines as a line goes on forever |
| 01:57.07 | detatch | the pastebin is output from a call manager 'show queue $quename' |
| 01:57.17 | detatch | running on the old 1.0.3 code |
| 01:57.24 | detatch | on the new 1.2.1 the "(brand: "brand-name" <callerid>)" isnt there (we use * to run a branded call-center) |
| 01:59.14 | *** part/#asterisk ldav15 (n=ldavis@208.2.145.2) |
| 02:01.08 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 02:02.58 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 02:03.45 | *** join/#asterisk co-bdg^-^ (i=EvilInLo@ws1.bratatex.melsa.net.id) |
| 02:04.02 | FuriousGeorge | trixter: for the purposes of this logic test thats the best answer |
| 02:04.17 | FuriousGeorge | options were 2 6 8 12 |
| 02:04.52 | FuriousGeorge | i wanted option 9 (1+number of angles) |
| 02:05.13 | co-bdg^-^ | s |
| 02:05.58 | _Sam-- | does using the jitterbuffer if you dont need it add latency? |
| 02:07.20 | justinu | yeah, probably some |
| 02:07.28 | _Sam-- | like 10ms? or like 30ms? |
| 02:07.44 | justinu | probably 30... i think it depends on the jb implementation too |
| 02:07.53 | _Sam-- | all 1.2 uses the new style? |
| 02:08.04 | co-bdg^-^ | how can i limit only extention 110 and 112 can dialout ? |
| 02:08.11 | co-bdg^-^ | i've set my dialplan |
| 02:08.16 | co-bdg^-^ | exten => _9.,1,GotoIf($[${CallerID} = 200]?2:3) |
| 02:08.33 | co-bdg^-^ | exten => _9.,1,GotoIf($[${CallerID} = 110&112]?2:3)<-the correct one |
| 02:08.45 | _Sam-- | you could just set those two extensions to use a different context? |
| 02:08.52 | FuriousGeorge | heres a real tough question. if i have 4 asterisk servers with dynamic ips logged into each other with register=> and type=friend. one is on a different ISP (with a less dynamic ip) but all the other ones have problems logging into it (timeout) |
| 02:08.58 | FuriousGeorge | why is that |
| 02:09.05 | *** join/#asterisk NoRemorse (n=bah@202.161.68.2) |
| 02:09.27 | NoRemorse | hi all, any ideas why a bt-101 has probs with the B-party eharing the A-party please? |
| 02:10.35 | _Sam-- | NoRemorse: usually one way audio is firewall / nat related..usually. |
| 02:10.46 | wunderkin | NoRemorse, ? you mean when you are on a call, flash and have multiple conversations.. ? or what |
| 02:11.02 | _Sam-- | i think he means the called party cant hear the calling party |
| 02:11.05 | *** join/#asterisk goh (n=goh@60.49.6.190) |
| 02:11.06 | wunderkin | ya probably |
| 02:11.20 | NoRemorse | no just a simple sip call , via an asterisk proxy. bt-101 calls other users, and the others cant hear the bt-101 person but he can hear them |
| 02:12.07 | _Sam-- | FuriousGeorge: what is your theory on why they cant connect to that one box? |
| 02:12.09 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-42-158.cybersurf.com) |
| 02:13.07 | NoRemorse | so no ideas? seems a common prob on google with no fix |
| 02:14.04 | _Sam-- | i think i gave you an idea...if you dont understand what i said then you may never fix it. |
| 02:14.06 | FuriousGeorge | _Sam--: im guessing it has something to do with dynamic ips and different isps |
| 02:14.23 | FuriousGeorge | its not even consistant, right now, 2 arent logged into 4th box but one is |
| 02:14.39 | _Sam-- | how do the boxes get the dynamic ips of eachother? |
| 02:15.19 | goh | any source that can help me receive call from fwd.pulver. A success follow the instruction and manage call out to fwd.pulver. but I can't receive call from pulver. The singal sound like reach the trunks only. |
| 02:15.30 | FuriousGeorge | _Sam--: i use domainname@dynamictodnsservice.org |
| 02:15.47 | FuriousGeorge | i mean name.service.com |
| 02:15.48 | NoRemorse | should I port forward 5060 and 5004 back to the bt-101 ? |
| 02:15.56 | _Sam-- | FuriousGeorge: i would propose something like a wget script from each box every 2 or 3 minutes |
| 02:16.04 | _Sam-- | then you could just log the wget and get the ip |
| 02:16.09 | _Sam-- | and connect to the right ips all the time |
| 02:16.15 | *** join/#asterisk _zigo__ (n=ogiz@m14.net81-64-48.noos.fr) |
| 02:16.16 | _Sam-- | i do that now for a couple hosts |
| 02:16.22 | _Sam-- | i have them wget a file from a webserver |
| 02:16.27 | _Sam-- | then i parse the logs cut -f 1 |
| 02:16.28 | *** join/#asterisk rowter (n=woot@201.145.5.26) |
| 02:16.47 | _Sam-- | that way i always have the correct ip for most boxes without them having to register |
| 02:17.03 | _Sam-- | but you could use the IP to connect to the right box |
| 02:17.36 | rowter | its possible to send a dial tone key when you are in a conversation and react to it? |
| 02:17.49 | NoRemorse | should I port forward 5060 and 5004 back to the bt-101 ? |
| 02:18.06 | FuriousGeorge | that solution is feasable but a bit outside the scope of my knowledge. i get that it resolves dynamic.todns.com name to an ip that changes, and so a restart should fix it. what i dont get is why its timing out to the /correct/ ip when it has it |
| 02:18.33 | FuriousGeorge | the box they cant log into is the one that changes ip LEAST often (once every month or less) |
| 02:18.41 | _Sam-- | you sure its not your named or something that is caching the ips |
| 02:18.58 | _Sam-- | or possibly asterisk is caching the ip based on the name |
| 02:19.09 | _Sam-- | i dont know how asterisk handles name lookups |
| 02:19.16 | FuriousGeorge | im sure * has the right ip, and im sure its timing out for no apparent reason |
| 02:19.22 | FuriousGeorge | i can ping the name from the console |
| 02:19.29 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.28.106.2) |
| 02:19.37 | FuriousGeorge | im logged into all those boxes with eyebeam using the name and that never goes down |
| 02:19.46 | FuriousGeorge | ever |
| 02:20.19 | FuriousGeorge | i use the name of the difficult box (which is my home box) for a bunch of different services remotely, only iax is not working |
| 02:20.20 | *** join/#asterisk klictel (n=klictel@modemcable119.206-200-24.mc.videotron.ca) |
| 02:20.20 | Trazz | exten => 5,2,Dial(SIP/2000,80,rt) ; permit transfer what is the 80 ? |
| 02:20.24 | _Sam-- | and you expect asterisk to switch to the new ip of the dynamic box without restarting it (sorry to sound redundant) |
| 02:20.33 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 02:20.34 | FuriousGeorge | no |
| 02:20.42 | _Sam-- | so even after a restart it still doesnt connect |
| 02:20.47 | FuriousGeorge | _Sam--: the ip is correct |
| 02:20.47 | _Sam-- | to the right host |
| 02:20.55 | FuriousGeorge | yeah |
| 02:20.55 | FuriousGeorge | brb |
| 02:21.14 | Trazz | exten => 5,2,Dial(SIP/2000,80,rt) ; what does the 80 represent? |
| 02:21.20 | _Sam-- | timeout |
| 02:21.25 | _Sam-- | in seconds |
| 02:21.46 | Trazz | so it will go to that extesion for 80 seconds before coming back / |
| 02:21.47 | Trazz | ? |
| 02:22.24 | _Sam-- | it will dial sip/2000 for 80 seconds before timing out |
| 02:22.31 | Trazz | ok |
| 02:22.44 | FuriousGeorge | _Sam--: correect, even after i restart or reload chan_iax2.so it still times out |
| 02:22.45 | Trazz | if i wanted 4 or 5 rings how do i calculate that |
| 02:23.00 | _Sam-- | i dont calculate in rings anymore |
| 02:23.21 | _Sam-- | you could call your asterisk yourself and time your rings |
| 02:24.09 | _Sam-- | furious i dont get it really...it has to be something dns cache related somehow |
| 02:24.16 | _Sam-- | that is my theory anyways |
| 02:24.22 | _Sam-- | the way something is doing the lookups / caching |
| 02:24.33 | _Sam-- | im could be, and usually am, wrong, but that is my theory |
| 02:25.01 | FuriousGeorge | _Sam--: when i iax show registry, it shows the right IP, so * isnt doing the caching |
| 02:26.22 | *** join/#asterisk welles (n=welles@222.90.15.150) |
| 02:27.01 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 02:27.29 | Trazz | Sam,, exten => 5,1,SetMusicOnHold(from-sip) should this be from-sip or from default ? |
| 02:27.50 | *** join/#asterisk shawn (n=welles@222.90.15.150) |
| 02:31.32 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 02:32.35 | *** join/#asterisk rob0 (i=1007@sorry.no-ip-here.net) |
| 02:32.47 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 02:32.52 | *** part/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 02:35.15 | goh | may I know when poeple said ISDN line that mean got how many channel inside. |
| 02:36.34 | *** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net) |
| 02:36.42 | *** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 02:41.00 | *** join/#asterisk wellng (n=welles@222.90.15.150) |
| 02:42.37 | litage | when someone refers to a "softswitch", what are they talking about? |
| 02:42.56 | mog_home | software telephony switch |
| 02:43.08 | mog_home | the biggest most know one is asterisk |
| 02:43.13 | litage | ah |
| 02:44.19 | *** join/#asterisk pengyong (n=lala@222.185.197.28) |
| 02:46.36 | _Sam-- | you are wasting your energy |
| 02:46.38 | _Sam-- | er |
| 02:46.52 | mog_home | ? |
| 02:46.58 | _Sam-- | wrong win! |
| 02:47.59 | mog_home | righttt |
| 02:49.15 | *** join/#asterisk Someone123 (n=m00p@pcp0011543623pcs.mainf01.in.comcast.net) |
| 02:52.02 | Trazz | Sam,, |
| 02:52.03 | Trazz | exten => 6,1,SetMusicOnHold(default) |
| 02:52.03 | Trazz | exten => 6,2,Dial(SIP/2001,80,rt) ; permit transfer |
| 02:52.03 | Trazz | exten => 6,3,Voicemail(u1234) |
| 02:52.12 | Trazz | i never hear a bit of music on hold |
| 02:52.21 | _Sam-- | why would you from that line? |
| 02:52.22 | Trazz | its silence and then finally a ring |
| 02:52.25 | _Sam-- | there is nothing to play it |
| 02:52.29 | _Sam-- | you just set it |
| 02:52.40 | Trazz | ouch..how do i play ? |
| 02:52.53 | _Sam-- | try this: |
| 02:52.53 | _Sam-- | exten => 6000,1,Answer |
| 02:52.53 | _Sam-- | exten => 6000,2,MusicOnHold() |
| 02:52.55 | ronaldl79 | Trazz, you need to add the 'm' option. |
| 02:53.12 | ronaldl79 | Or, as he suggests above...depends on what you want. |
| 02:53.28 | Trazz | which is better ? |
| 02:53.31 | ronaldl79 | (I use the 'm' option to play music in place of ringing) |
| 02:53.45 | ast_New-B | whatis or tag in asterisk variable ? |
| 02:53.52 | ronaldl79 | (But, not always...just depends) |
| 02:54.04 | Trazz | sam, thanks. |
| 02:54.06 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-26-179-224.indy.res.rr.com) |
| 02:54.06 | _Sam-- | sure |
| 02:54.07 | Trazz | i will test now |
| 02:58.35 | ast_New-B | how can i limit only extention 110 and 112 can dialout ? |
| 02:58.43 | ast_New-B | somebody please help me |
| 02:59.30 | Ariel_ | ast_New-B, create the proper context for your users |
| 02:59.53 | *** join/#asterisk welles (n=welles@222.90.15.150) |
| 03:00.06 | ast_New-B | and then ? |
| 03:00.27 | *** join/#asterisk need-book (n=va2003ch@203.119.9.9) |
| 03:00.27 | Trazz | any idea why it takes so long to go from the time i hit the 5 to ring? |
| 03:00.33 | need-book | Hi |
| 03:00.59 | need-book | anyone can help me about ENUM and Asterisk |
| 03:01.04 | Ariel_ | ast_New-B, you create the context and give each context what you want it to use with includes. See the extensions.conf.sample in the /usr/src/asterisk/configs directory |
| 03:01.05 | Trazz | lots of dead silence |
| 03:01.21 | Ariel_ | Trazz, is this a sip phone |
| 03:01.31 | need-book | I upgrade to Asterisk 1.2.1 |
| 03:01.43 | need-book | and we can not lookup ENUM |
| 03:01.48 | Trazz | the phone i am transferring to softphone |
| 03:02.12 | VJ | Hello |
| 03:02.13 | Ariel_ | but your on a sip phone? |
| 03:02.38 | Trazz | you call in and then hit 5 and it takes a good 15 seconds before it starts to ring the softphone |
| 03:02.41 | Trazz | or maybe 10 seconds |
| 03:03.06 | Ariel_ | Trazz, ok is this via your ivr? |
| 03:03.11 | Trazz | yes |
| 03:03.24 | Ariel_ | it's waiting to see if you want any other digits |
| 03:03.33 | Trazz | how can i avoid that? |
| 03:03.41 | *** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com) |
| 03:03.42 | Ariel_ | how are you extension setup |
| 03:03.43 | VJ | I need to dial a number from my asterisk box via VOIP |
| 03:04.00 | Ariel_ | VJ, we all dial via voip all the time it's the same to asterisk |
| 03:04.01 | PMantis | can anyone give me a US phone number that's definitely listed in e164.org ? |
| 03:04.04 | Ariel_ | just use the dialing rules |
| 03:04.16 | Trazz | Ariel, What do you want to see? |
| 03:04.21 | VJ | my carrier wants the call to be authenticated via user name and password |
| 03:04.46 | Ariel_ | Trazz, pastebin.ca your section for the ivr and what your including. |
| 03:04.53 | Ariel_ | vj so setup the account |
| 03:05.04 | VJ | i have the carrier's IP address, port (i.e. 5070) and the user name and password |
| 03:05.05 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 03:05.13 | VJ | i changed the port in sip.conf |
| 03:05.44 | *** join/#asterisk jef_ (i=fischer@p54847227.dip.t-dialin.net) |
| 03:05.52 | VJ | and also created the account in sip.conf |
| 03:06.16 | *** join/#asterisk pengyong (n=lala@222.188.137.148) |
| 03:06.58 | warthawg | i'm running asterisk on ubuntu, and it doesn't find my sound files. they are in /var/lib/asterisk/sounds and /usr/share/asterisk/sounds. should they be somewhere else? |
| 03:07.04 | Ariel_ | VJ, ok then exten => _x.,1,dial(Sip/${EXTEN}@accountname,20,) |
| 03:07.09 | Trazz | Ariel, http://pastebin.com/510948 |
| 03:07.16 | Dandan | hey, is there a sip/iax/h.323 clinet in java? |
| 03:07.26 | VJ | http://pastebin.com/510949 |
| 03:07.36 | Dandan | (private dialpad anyone?) |
| 03:08.37 | Ariel_ | Trazz, you should not do that don't include the outbound dial rules for your inbound. |
| 03:08.41 | Ariel_ | seperate them |
| 03:09.01 | Ariel_ | it's trying to match the full 11 digites or more in the bv section |
| 03:09.18 | Trazz | ok can you paste back how it should look ? |
| 03:09.29 | Trazz | i want to make sure i get it right |
| 03:09.54 | *** join/#asterisk shawn (n=welles@222.90.15.150) |
| 03:09.58 | Ariel_ | sperate the sip extensions and keep them simple. like include=local-ext make that context. Then look at your logic |
| 03:10.54 | Ariel_ | see the extensions.conf.sample it has great includes there it's located at /usr/src/asterisk/configs/ |
| 03:10.55 | VJ | after that i have the following entry in my extensions.conf http://pastebin.com/510950 |
| 03:11.59 | Ariel_ | vj your using an agi to do your dialing |
| 03:11.59 | warthawg | what directory does asterisk look in for sounds? |
| 03:12.07 | VJ | yes |
| 03:12.07 | PMantis | An example e164.org number I can test dial please? |
| 03:12.14 | Qwell | /var/lib/asterisk/sounds/ |
| 03:12.19 | Ariel_ | that is where the issues are then |
| 03:12.23 | warthawg | thanks, Qwell |
| 03:12.37 | warthawg | maybe it's a permissions problem |
| 03:12.51 | Qwell | warthawg: don't use extensions |
| 03:12.58 | Qwell | just Playback(filename) |
| 03:13.07 | VJ | is this the right configuration i have |
| 03:13.20 | Ariel_ | VJ, I don't know how your agi works |
| 03:13.30 | *** join/#asterisk cleyverson (n=cleyvers@201.29.76.85) |
| 03:13.39 | *** join/#asterisk techie (i=gus@antibala.com) |
| 03:13.42 | cleyverson | hello peopple |
| 03:14.15 | cleyverson | what have I to install first, asterisk or zaptel?? |
| 03:14.28 | Ariel_ | zaptel then asterisk |
| 03:14.37 | wunderkin | cleyverson, the installation instructions are at www.asterisk.org under download |
| 03:14.40 | Trazz | Ariel, Don't they all have to be included to be seen / |
| 03:14.50 | Ariel_ | Trazz, no just what you want |
| 03:15.00 | VJ | http://pastebin.com/510955 |
| 03:15.07 | Ariel_ | your ivr should only have the local extensions. The goto take care of the other parts |
| 03:15.09 | cleyverson | thanks |
| 03:15.34 | warthawg | Qwell, right. it's very strange. the console messages add the gsm and say the file doesn't exist |
| 03:15.47 | VJ | http://pastebin.com/510957 |
| 03:15.54 | warthawg | and it plays the .gsm files from voicemail |
| 03:16.02 | VJ | i have pasted the complete sip.conf |
| 03:16.10 | VJ | and extensions.conf |
| 03:16.25 | Ariel_ | ahh your using vicidial |
| 03:16.30 | VJ | http://pastebin.com/510955 & http://pastebin.com/510957 |
| 03:16.32 | VJ | yes |
| 03:16.36 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 03:16.40 | Trazz | Ariel, don't i need the local extensions to use this ? |
| 03:16.40 | Trazz | exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 03:16.40 | Trazz | exten => _1NXXNXXXXXX, 2, congestion() |
| 03:16.49 | ManxPower | my place is apparently a ladybug haven |
| 03:17.01 | Nugget | ladybugs are plusgood. |
| 03:17.10 | Trazz | or only make in the context for the phone definition in the sip.conf ? |
| 03:17.39 | Qwell | Nugget: They're doubleplus good |
| 03:17.53 | Ariel_ | Trazz, first step. context for your users. not extensions like if your users are from-internal that context would include=local-ext and include=ldcontext. |
| 03:18.06 | *** join/#asterisk FastJack_ (i=fastjack@p5091FD9A.dip.t-dialin.net) |
| 03:18.49 | VJ | in the pasted code, i have changed the port to 5070 and then added the username and secret in sip.conf |
| 03:19.22 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:19.23 | Ariel_ | I don't know how vicidial uses it's agi to dial.... I would say the problem is there |
| 03:20.53 | VJ | but right now its only manual call that i am trying to make |
| 03:20.57 | VJ | from asterisk |
| 03:21.27 | Ariel_ | vj your using a agi to do the actual dial |
| 03:21.52 | cleyverson | jef_, brasileiro?? |
| 03:21.54 | VJ | so, what should i paste for you to know how i am using it |
| 03:21.57 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 03:22.33 | Ariel_ | The actual agi would help. But only if I am able to see how they do the dial |
| 03:23.17 | brookshire | i didn't think you could dial out with agi |
| 03:23.27 | brookshire | at least i couldn't 3 years ago, lol |
| 03:24.15 | VJ | what would te location for that in my system? |
| 03:24.22 | VJ | i mean agi? |
| 03:24.36 | VJ | which file will contain its coding |
| 03:25.02 | Ariel_ | ; to dial out with 9 as the prefix |
| 03:25.02 | Ariel_ | exten => _9X.,1,AGI(call_log.agi,${EXTEN:1}) |
| 03:25.02 | Ariel_ | exten => _9X.,2,Dial(${TRUNK}${EXTEN:1},,tTo) |
| 03:25.02 | Ariel_ | exten => _9X.,3,Congestion |
| 03:25.10 | Ariel_ | vj that is your part for the dial out |
| 03:25.30 | Ariel_ | you need to define the TRUNK correctly |
| 03:25.45 | VJ | ok |
| 03:25.49 | Ariel_ | TRUNK=SIP/trunk/ |
| 03:25.54 | Ariel_ | that is what you have it as now. |
| 03:26.00 | VJ | correct |
| 03:26.30 | Nugget | VJ's just junk in his TRUNK |
| 03:26.55 | VJ | what else should be defined here |
| 03:27.39 | Ariel_ | VJ, is the trunk name in the sip.conf called trunk |
| 03:27.44 | VJ | yes |
| 03:28.35 | *** join/#asterisk burton (i=mimx@w201.ljudmila.org) |
| 03:28.47 | VJ | i was able to make calls properly when my PSTN provider was authenticating me via prefix |
| 03:28.55 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 03:29.02 | VJ | which i defined nest to TRUNK=SIP/trunk/ |
| 03:29.34 | VJ | but now i have changed my PSTN provider, he needs to get the authentication done via user name and password |
| 03:29.47 | VJ | and also the port to be changed to 5070 |
| 03:30.00 | VJ | i changed the udp port in sip.conf to 5070 |
| 03:30.10 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 03:30.32 | VJ | then i m defining a username and secret in sip.conf in declaration of trunk itself |
| 03:30.34 | Ariel_ | you can do that in the [trunk] context on the sip and added username= and secret= |
| 03:31.00 | VJ | yes |
| 03:31.18 | VJ | but its not working that way |
| 03:31.20 | Ariel_ | ok what is the error you get on the cli |
| 03:31.26 | *** join/#asterisk aster][sk-newB (n=seymore@203.215.73.192) |
| 03:31.35 | aster][sk-newB | hello |
| 03:31.37 | VJ | i can see the call getting dialled out |
| 03:31.55 | Ariel_ | ok the do sip debug and see what it says about the call |
| 03:32.04 | VJ | ok |
| 03:33.48 | *** join/#asterisk Lee__ (n=lee@dsl254-122-010.nyc1.dsl.speakeasy.net) |
| 03:34.01 | ast_New-B | anyone can tell me to make a group extention in extentions.conf ? |
| 03:34.32 | Lee__ | ast_New-B, use the & to sepperate peers in a Dial application |
| 03:34.35 | Ariel_ | ast_New-B, like dial(sip/234&sip/1234&Sip/3000) |
| 03:35.01 | Lee__ | is it impolite to ask about distro packages here? |
| 03:35.08 | PMantis | Anyone else succesfully using EnumLookup ? |
| 03:35.26 | Ariel_ | Lee__, don't know but go for it... |
| 03:35.38 | brookshire | lee: you'll start a war though.. |
| 03:35.43 | brookshire | ;) |
| 03:36.07 | Lee__ | well, I've found one backport but I'm curious if someone can recommend a Debian backport of 1.2.1 and all deps for sarge. |
| 03:36.48 | *** join/#asterisk StoryFire (n=StoryFir@71.226.252.76) |
| 03:36.49 | Lee__ | but seriously. Asterisk is complicated to package and I'd rather not build my own if someone else has already and is using it in production. |
| 03:36.49 | brookshire | lee: there is one on unstable |
| 03:37.03 | brookshire | but like.. i think it is really really unstable |
| 03:37.09 | Lee__ | brookshire, I know that but I'm not letting unstable on my production servers |
| 03:37.20 | Lee__ | I can backport myself but it takes a while. |
| 03:37.27 | brookshire | but 1.2.1 is stable |
| 03:37.34 | brookshire | they have unstable debs |
| 03:37.35 | Lee__ | no, /Debian/ unstable |
| 03:37.40 | cleyverson | what to do to update linux? |
| 03:37.43 | Ariel_ | stable lllllllllll now really. |
| 03:37.49 | Lee__ | production servers run sarge... |
| 03:37.54 | Lee__ | mantra... |
| 03:37.57 | cleyverson | have I to do something first to update? |
| 03:38.31 | brookshire | http://lists.alioth.debian.org/mailman/listinfo/pkg-voip-maintainers |
| 03:38.35 | brookshire | you can ask them :) |
| 03:38.55 | Lee__ | brookshire, the debs in unstable won't run on sarge IIRC cause they depend on GCC 4 |
| 03:39.33 | ast_New-B | Ariel_ : is this correct exten => _9.,1,GotoIf($[$[${CALLERIDNUM} = 200&202]]?100:2 ) |
| 03:39.53 | ast_New-B | my scenario is just exten 200 and 202 can dialout |
| 03:40.31 | Lee__ | I'll poke around more. Thanks...and one more question: |
| 03:40.44 | Ariel_ | ast_New-B, why not use context includes |
| 03:40.45 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:40.51 | *** join/#asterisk aster][sk-newB (n=seymore@203.215.73.192) |
| 03:41.16 | ast_New-B | Ariel_ : can you give me a sample |
| 03:41.30 | Lee__ | is there a prefered way to set callerID blocking but keep the number and not name for 911? I read that setting it to "Anonymous" <NXXNXXNXXX> works but it was inconsistant on two servers for me. |
| 03:41.54 | PMantis | Does anyone have a number that I can test with DUNDi or e164.org ? |
| 03:42.41 | VJ | can i use H323 instead on SIP in asterisk |
| 03:42.47 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 03:42.52 | Ariel_ | VJ, yes if you install h323 |
| 03:42.54 | Flauto | hi all |
| 03:42.56 | Flauto | how are you doing |
| 03:43.05 | Flauto | anyone here is using voipstunt? |
| 03:43.07 | VJ | is that a reliable kind of thing |
| 03:43.09 | Flauto | i have a question |
| 03:43.12 | *** join/#asterisk StoryFire (n=StoryFir@71.226.252.76) |
| 03:43.13 | Ariel_ | but h323 is worst |
| 03:43.19 | VJ | :( |
| 03:43.32 | VJ | SIP is recommended |
| 03:43.44 | Flauto | how can i setcallerid for voipstunt service? |
| 03:44.57 | Nivex | so, I'm really tempted to buy a SPA-941, but the fact that none of Sipura, Linksys, or Cisco make any mention of the thing on their site kinda worries me. |
| 03:44.58 | Nugget | +1 512 538-0508 is in e164.org if you want to test. |
| 03:45.22 | *** join/#asterisk ACiDV (n=acidv@modemcable247.11-37-24.mc.videotron.ca) |
| 03:45.51 | Ariel_ | Nivex, the 941 is listed on there support /firmware page |
| 03:45.54 | ACiDV | Someone have tested domain=domain,context in sip.conf ? |
| 03:45.54 | Flauto | mine is showing something really odd |
| 03:46.13 | Flauto | sometimes some random number or unknown |
| 03:46.46 | [TK]D-Fender | Nivex : Its a decent phone, but the Polycom IP 501 is a serious competitor at +/- 20$ more |
| 03:47.30 | Nivex | [TK]D-Fender: everybody raves so about the polycom |
| 03:47.40 | Ariel_ | Nivex, they work. |
| 03:48.14 | [TK]D-Fender | Nivex : I run an all-Polycom shop at work. They are VERY solid phones. |
| 03:48.32 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 03:48.37 | [TK]D-Fender | I use an SPA-941 at home because I bought it on faith and for variety (and to save 20$USD ;)) |
| 03:48.38 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 03:48.41 | Mavvie | afternoon. |
| 03:49.02 | Mavvie | way cool, when I execute this AGI script the asterisk server reloads its configuration, while all it was supposed to do is a Dial :-) |
| 03:49.27 | [TK]D-Fender | In retrospect and considering I'm going to go for my Polycom certification I'd have been better to have chosed the IP 501 |
| 03:49.32 | Nugget | Mavvie: it's a Value Add(tm) :) |
| 03:49.39 | Flauto | please people. is there a way that i can setcallerid for voipstunt? |
| 03:50.13 | trixter | no |
| 03:50.18 | Nugget | I don't even know what "voipstunt" is, Flauto, so try not to get snitty that I haven't answered your question. |
| 03:50.21 | Ariel_ | Flauto, have you asked them? I don't know that service or in fact I have never heard about voipstunt |
| 03:50.31 | *** join/#asterisk bmg505 (n=leon@c1-56-4.rndf.isadsl.co.za) |
| 03:50.33 | trixter | voipstunt is the finera (voipbuster) company.. |
| 03:50.41 | Flauto | it is a free phone call service |
| 03:50.45 | Flauto | not bad at all |
| 03:50.49 | trixter | finerea has like 10 companies, 1899.co.uk, voipbuster, sipdiscount, etc |
| 03:50.59 | Ariel_ | free...hummm |
| 03:51.09 | trixter | and you cant set caller id |
| 03:51.15 | Nivex | hmm... I can get a 501 for $177.77 shipped or an SPA-941 for $139.95 shipped |
| 03:51.18 | Ariel_ | Now I know why your getting different caller id's |
| 03:51.24 | Flauto | trixter, not at all? |
| 03:51.35 | [TK]D-Fender | Nivex : Where for 139.95? |
| 03:51.37 | Flauto | ariel |
| 03:51.39 | Ariel_ | Nivex, polycom is worth the extra |
| 03:51.42 | Flauto | yes |
| 03:51.50 | Nivex | [TK]D-Fender: store.voxilla.com w/ coupon code |
| 03:51.50 | Flauto | it is showing unknow |
| 03:51.53 | Flauto | 646 |
| 03:51.59 | Flauto | blah blah blah |
| 03:52.34 | Flauto | depending where you call |
| 03:52.53 | Ariel_ | Flauto, sometimes if you just pre-set the number in the callerID without the name it will work. But it's just with some of there end points that really matter. |
| 03:53.12 | *** join/#asterisk ryansc (n=ryansc@adsl-065-015-206-109.sip.bix.bellsouth.net) |
| 03:53.32 | Flauto | just use setcallerid(xxxxxxxxxx)? |
| 03:53.44 | *** join/#asterisk cleyverson (n=cleyvers@201.29.76.85) |
| 03:53.48 | litage | what are some of your experiences with setting up high-availability for Asterisk servers? |
| 03:53.59 | Math` | Flauto: CALLERID(number)=xxxxxxxxxx |
| 03:54.05 | cleyverson | hello, have I to unstall an existing version to update?? |
| 03:54.38 | Flauto | math, dont need to use setcallerid. just callierid is okay? |
| 03:54.48 | cleyverson | hello, have I to uninstall an existing version to update?? |
| 03:54.48 | [TK]D-Fender | Nivex : Points for Polycom : 2n'd eth port, 10/100, better screen / speakerphone / stand. For the 941 its cheaper, smaller, 2.5mm headset jack, 1 more line if you're going to use all 4 line-keys for 1 reg. |
| 03:54.59 | Qwell | Flauto: with the Set application |
| 03:55.08 | file[laptop] | blargh |
| 03:55.13 | Flauto | thanks qwell |
| 03:55.16 | [TK]D-Fender | brb |
| 03:55.22 | Flauto | i will try |
| 03:55.34 | ast_New-B | thanks all, it's works now ... call duration, ext grouping and limiting |
| 03:56.06 | ast_New-B | what the best brand for ATA hardware |
| 03:56.26 | cleyverson | Have I to uninstall an existiong version to update asterisk? |
| 03:57.00 | joe | has anyone seen the voicmail system not allow you to re record a name when reassigning an extension to someone else? I reset the password to the extension number and all that to re-initialize the voicemail which it does prompt you to set it up, but it doesn't allow the new reccording to take |
| 03:57.02 | ast_New-B | my plan is to use analog phone and connected to asterisk server |
| 03:57.05 | Nivex | so wait, the 941 doesn't have a speakerphone? I thought it did. |
| 03:57.09 | joe | and yes I did a reload |
| 03:58.10 | Nivex | or is he asserting that the polycom speakerphone is simply better? |
| 03:58.33 | usam | ast_New-B: any brand that is not using PA chipset ;) |
| 03:58.52 | Flauto | ariel, qwell, did not work |
| 03:59.04 | *** join/#asterisk ahattar (n=kjsd@ool-43551487.dyn.optonline.net) |
| 03:59.11 | Qwell | Set(CALLERID(num)=5551212) |
| 03:59.41 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 04:00.26 | Flauto | okay, qwell |
| 04:00.28 | Flauto | thanks |
| 04:00.30 | Flauto | let me try |
| 04:01.45 | mog_home | hey how come those linksys phones arent on the linksys site? |
| 04:01.49 | mog_home | whats up with that |
| 04:01.56 | mog_home | the only phone they show is that skype one |
| 04:01.57 | Qwell | shh...they're secret |
| 04:02.05 | Ariel_ | joe if you have made changes in the voicemail.conf did you stop asterisk and restarted it? |
| 04:02.13 | tengulre11 | does the asterisk supported fax function? |
| 04:03.18 | *** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-26-179-224.indy.res.rr.com) |
| 04:03.22 | Ariel_ | mog_home, I think there in linbo between linksys and sipura |
| 04:03.32 | mog_home | so lame |
| 04:03.37 | mog_home | i just wanted a quote |
| 04:03.38 | Ariel_ | tengulre11, it does support some fax functions |
| 04:04.24 | Flauto | qwell, still did nto work |
| 04:04.27 | Flauto | guess it just does not work that way |
| 04:04.30 | *** part/#asterisk jimmy_deanPB (n=jhodapp@cpe-65-26-179-224.indy.res.rr.com) |
| 04:04.40 | need-book | hi |
| 04:04.41 | tengulre11 | Ariel_: we have 20 users in our company, and need sending/receiing fax! |
| 04:04.48 | Ariel_ | Flauto, more like they don't support you setting the callerID |
| 04:04.53 | need-book | which hardware do I need to use SMS with asterisk ? |
| 04:05.43 | Flauto | Jan 17 22:03:48 ERROR[19534]: pbx.c:1408 ast_func_write: Function CallerId not registered it is telling me this |
| 04:05.44 | Ariel_ | tengulre11, ok so set it up that way. asterisk supports analog lines and also faxboards with hylafax. There are many ways to get this done via asterisk. |
| 04:06.25 | tengulre11 | Ariel_: faxboards? which one? |
| 04:06.29 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 04:07.03 | Qwell | Flauto: yes, I do mind... |
| 04:07.07 | ManxPower | If I wanted to create a DVD using my own video what software would you recommend? /msg me is fine. |
| 04:07.16 | ManxPower | i.e. nice looking menus, etc |
| 04:07.22 | Flauto | sorry qwell |
| 04:07.26 | [TK]D-Fender | ManxPower : Roxio |
| 04:07.28 | Math` | ManxPower: Adobe Encore DVD |
| 04:07.45 | Nugget | iMovie :) |
| 04:07.49 | ManxPower | [TK]D-Fender, Why? Math`, Why? |
| 04:08.07 | [TK]D-Fender | ManxPower : Cheap easy works.... |
| 04:08.08 | Math` | ManxPower: cuz its the only one I played with :P |
| 04:08.13 | Flauto | bye ariel |
| 04:08.22 | ManxPower | I'll check out Roxio. |
| 04:08.41 | Nugget | no! buy a mac! |
| 04:08.51 | Math` | lol |
| 04:09.08 | [TK]D-Fender | Nugget : I just installed a nasty OS-X theme for WinXP :) |
| 04:09.19 | Nugget | ew |
| 04:09.46 | Nivex | [TK]D-Fender: welcome back :) |
| 04:09.48 | Trazz | whats the purpose of exten => _9NXXXXXX, 102, busy() |
| 04:09.54 | Trazz | the 9 ? |
| 04:10.07 | Flauto | qwell, it is saying that Jan 17 22:03:48 ERROR[19534]: pbx.c:1408 ast_func_write: Function CallerId not registered |
| 04:10.10 | Flauto | what does it mean |
| 04:10.12 | [TK]D-Fender | Nugget : Works rater well actually. Minimize to dock,(SVG's) desktop search, scaling & fonts, animations... not bad |
| 04:10.13 | Qwell | CALLERID |
| 04:10.19 | Qwell | case matters |
| 04:10.23 | Trazz | becaue i see this in local context ignorepat => 9 |
| 04:10.34 | [TK]D-Fender | Nivex : jsut had to reboot to have it take effect :) |
| 04:11.09 | ManxPower | Egads! Roxio has like a gazillion different products |
| 04:11.10 | Nivex | [TK]D-Fender: so I'm guessing that, having dealt with both phones, you would agree with Ariel and say the polycom is worth the extra dough? |
| 04:11.13 | Flauto | let me try again |
| 04:11.17 | Flauto | thanks qwell |
| 04:11.25 | [TK]D-Fender | Trazz : Ignorepat tells * that for Zap channels to continue dial-tone even after the 9 is received |
| 04:11.39 | Math` | for skinny too afaik |
| 04:11.54 | Math` | (maybe for iax2 too) |
| 04:12.07 | [TK]D-Fender | Nivex : I would. If you're going to do a job do it right. Though with that in mind I'd almost stretch and say go for the IP 601 |
| 04:12.27 | *** join/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 04:12.27 | Flauto | qwell, it is still showing unknow |
| 04:12.29 | [TK]D-Fender | ManxPower : Easy DVD creator I believe its called |
| 04:12.30 | Trazz | ok when i try to dial out i have to use 1+# and can't use 9+1+# |
| 04:12.33 | Trazz | is that normal ? |
| 04:12.36 | Nivex | I don't need to go too overkill... this is for a home system for me to experiment on |
| 04:12.38 | Qwell | Flauto: What are you putting, exactly? |
| 04:12.53 | [TK]D-Fender | Trazz : You can do whatever you want.... its your choice. |
| 04:13.07 | Flauto | Set(CALLERID(num)=3129620228) |
| 04:13.15 | Flauto | qwell |
| 04:13.20 | Qwell | Flauto: What version of asterisk? |
| 04:13.27 | Qwell | Don't even say anything lower than 1.2 |
| 04:13.27 | Flauto | the latest one |
| 04:13.29 | Qwell | just don't |
| 04:13.32 | Math` | lol |
| 04:13.36 | Flauto | hehe |
| 04:13.43 | Flauto | it is the newest one |
| 04:13.44 | Qwell | what "latest one"? |
| 04:13.45 | Trazz | ok so do i need the 9 here ? exten => _9NXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 04:13.51 | Flauto | 1.2.1? |
| 04:13.53 | Math` | Trazz: show version on cli |
| 04:13.55 | mog_home | i run asterik .9 |
| 04:13.57 | Qwell | Flauto: You're 100% sure? |
| 04:13.58 | mog_home | qwell |
| 04:14.00 | Qwell | mog_home: scary |
| 04:14.03 | Math` | mog_home: why? lol |
| 04:14.09 | mog_home | i wonder who has the longest uptime |
| 04:14.10 | Qwell | cvs from 02/2003? |
| 04:14.15 | mog_home | im messing with qwell math |
| 04:14.18 | Math` | lol |
| 04:14.18 | [TK]D-Fender | Nivex : Lets say the 601 would be the last high-end phone you're likely to buy and definately won't disappoint. |
| 04:14.19 | Flauto | yes, qwell |
| 04:14.23 | Trazz | Asterisk 1.2.1 built by root |
| 04:14.27 | Math` | ok |
| 04:14.30 | Flauto | no qwell |
| 04:14.30 | mog_home | the longest asterisk uptime ive had at home was 2 months |
| 04:14.34 | mog_home | but then i had to reboot box |
| 04:14.35 | Qwell | Flauto: Do you have autoload=no in modules.conf? |
| 04:14.37 | [TK]D-Fender | Nivex : but at $250 is a decent chunk of change. |
| 04:14.43 | Flauto | i just rebuilt it about two weeks ago |
| 04:14.44 | remmo | is anyone using vierling gsm gateways here? |
| 04:14.47 | mog_home | as i was upping the ram |
| 04:14.51 | ronaldl79 | I will be interviewing Mark Spencer this week, and just completed the 'landing' page for it (http://www.ronaldlewis.com/coffee). Also, I welcome questions from the community, in addition to those I already have. PM me here, or send an e-mail from the web site. Thanks! |
| 04:14.59 | Flauto | let me check |
| 04:15.25 | Flauto | it says yes |
| 04:15.27 | Trazz | Math ? |
| 04:15.43 | Flauto | autoload=yes |
| 04:15.54 | mog_home | ronaldl79, i want to ask marko why he ignores me in the office ^_^ |
| 04:15.55 | Flauto | qwell |
| 04:16.01 | cleyverson | I'm traying to update the asterisk but the follor error appeard (make: ** [editline/libedit.a] Erro 1 |
| 04:16.01 | cleyverson | )..who knows what is that?? |
| 04:16.12 | alphaque | ronaldl79: i think a new angle, instead of getting mark to talk abt asterisk all the time, is to talk about digium's plans for their line of cards |
| 04:16.14 | mog_home | you need libncurses-dev |
| 04:16.27 | mog_home | lol thats tup secret alphaque |
| 04:16.37 | Qwell | Flauto: from the asterisk cli, do a show version |
| 04:16.41 | Qwell | What does it say? |
| 04:16.57 | ronaldl79 | Sounds good, alpha. |
| 04:17.02 | joe | so changes to voicemail.conf don't take effect on a reload? from the asterisk console? you must restart or relaod the service is that right? |
| 04:17.21 | cleyverson | Need I libncurses-dev to install?? |
| 04:17.22 | Flauto | Asterisk SVN-trunk-r7230 built by root @ DevilNet on a i686 running Linux on 2006-01-04 01:47:18 UTC |
| 04:17.33 | mog_home | yes |
| 04:17.45 | Trazz | what is the 9 for in this line exten => _9NXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 04:17.48 | Trazz | do i need it |
| 04:18.16 | Flauto | qwell, it is correct, right? |
| 04:18.24 | cleyverson | ok but, where can i found it? |
| 04:18.25 | Qwell | That isn't 1.2.1 |
| 04:18.32 | cleyverson | do u know? |
| 04:18.48 | mog_home | what distro |
| 04:18.50 | [TK]D-Fender | Trazz : dialing "9" before the number to dial is jsut a PBX tradition and ensures that you can use 0-8 as the starting digit for internal #'s and features |
| 04:18.53 | cleyverson | suse 9.3 |
| 04:19.02 | mog_home | use yast |
| 04:19.04 | Flauto | qwell, what is it then |
| 04:19.11 | Flauto | would you let me know what shoudl i do? |
| 04:19.14 | Qwell | svn trunk |
| 04:19.15 | cleyverson | i cant find there |
| 04:19.24 | Trazz | Tk, the system is nto requiring me to use 9 |
| 04:19.26 | Trazz | i can dial right out |
| 04:19.26 | mog_home | its ncurses-dev |
| 04:19.30 | Flauto | use cvs to compile again? |
| 04:19.30 | mog_home | its there |
| 04:19.44 | Qwell | Flauto: no |
| 04:19.54 | Qwell | You need to figure out why it isn't loading the CALLERID function |
| 04:20.05 | [TK]D-Fender | Trazz : I never said it was a REQUIREMENT. I just said thats the traditional metheod for doing an outside dial-string |
| 04:20.33 | cleyverson | now i founded |
| 04:20.45 | Flauto | qwell, i think it is loading the function. it did not show any error the last time i tried |
| 04:21.06 | cleyverson | let me try to install again, thanks |
| 04:21.25 | ManxPower | It's snailing outside (snow/hail) |
| 04:21.30 | mog_home | k |
| 04:21.56 | Qwell | What's snow? |
| 04:22.08 | Qwell | I've seen hail once or twice...never heard of snow though |
| 04:22.28 | Flauto | Executing Set("SIP/jianxun-6a78", "CALLERID(num)=3129620228") in new stac |
| 04:22.33 | mog_home | is rain/sleet here manx |
| 04:22.42 | Flauto | but my pther phone was showing unknow |
| 04:22.45 | Qwell | Flauto: okay, and what happens? |
| 04:22.50 | Flauto | qwell |
| 04:22.55 | Qwell | Then your phone sucks. NEXT!!! |
| 04:23.09 | Flauto | <PROTECTED> |
| 04:23.18 | [TK]D-Fender | Flauto : jsut do a NoOp of the callerid right after the SET to prove if it worked and its not the provider.... |
| 04:23.36 | mog_home | qwell dont channel bkw |
| 04:23.46 | Qwell | mog_home: somebody has to |
| 04:23.50 | mog_home | no |
| 04:23.53 | mog_home | no one does |
| 04:23.55 | Qwell | k |
| 04:24.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 04:24.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 04:27.24 | cleyverson | I'm traying to install but a new error appeard... /usr/lib/gcc-lib/i586-suse-linux/3.3.5/../../../../i586-suse-linux/bin/ld: cannot find -lssl |
| 04:27.24 | cleyverson | collect2: ld returned 1 exit status |
| 04:27.24 | cleyverson | make: ** [asterisk] Erro 1 |
| 04:27.24 | cleyverson | <PROTECTED> |
| 04:27.36 | edwin_ | install the ssl package. |
| 04:27.42 | mog_home | now you need libssl-dev |
| 04:27.59 | chaos1\ | ~docs |
| 04:28.00 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:29.50 | cleyverson | do u know the righ package name?? |
| 04:30.09 | mog_home | just search ssl |
| 04:30.12 | cleyverson | because a lot of them was show at yast |
| 04:30.26 | mog_home | or seach ssl and dev |
| 04:31.06 | cleyverson | i have installed the package openssl |
| 04:31.18 | mog_home | right you need the openssl-dev |
| 04:31.22 | mog_home | you need dev librarie |
| 04:31.23 | mog_home | s |
| 04:31.42 | cleyverson | ok |
| 04:31.48 | cleyverson | let me try again :) |
| 04:31.52 | mog_home | k |
| 04:33.31 | cleyverson | let me ask something...what is the right way...why do I need to install zaptel before asterisk? |
| 04:33.51 | mog_home | asterisk doesnt not depend on zaptel |
| 04:33.59 | Flauto | qwell, i used noop and it is showing the callerid i setup but the phone got called still shows unknow |
| 04:34.13 | mog_home | but it wont build parts you need for aterisk to work with zaptel if it doesnt detect zaptel installed |
| 04:34.34 | cleyverson | but i read that is necessary instal zaptel to uso iax |
| 04:34.39 | mog_home | no |
| 04:34.45 | cleyverson | reaaly? |
| 04:34.46 | mog_home | its necessary for iax trunking |
| 04:34.58 | mog_home | and musiconhold |
| 04:35.00 | cleyverson | i need connect to * servers |
| 04:35.16 | mog_home | you dont need to use iax trunking to talk iax between servers |
| 04:35.19 | mog_home | its just usefull |
| 04:35.42 | cleyverson | then..what do I really need to install?? |
| 04:36.06 | Qwell | mog_home: does MoH *really* need a timer? |
| 04:36.09 | mog_home | well do you want music on hold or trunking |
| 04:36.15 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.252) |
| 04:36.21 | Qwell | I mean...will it "work" without it? |
| 04:36.21 | mog_home | it could theoratically be done with out |
| 04:36.22 | cleyverson | not really |
| 04:36.23 | mog_home | but it isnt |
| 04:36.32 | mog_home | well then you really just need asterisk |
| 04:36.42 | mog_home | its nice to have all |
| 04:36.59 | cleyverson | its work just with asterisk? |
| 04:37.19 | Flauto | qwell, thanks anyway though i think now it is the service which does not allow set callerid |
| 04:37.26 | mog_home | but no confrencing, moh , or trunking |
| 04:37.48 | mog_home | i always compile zaptel as those are nice things |
| 04:37.50 | Flauto | mog, what does it mean by trunking? |
| 04:37.51 | mog_home | but you dont neeed them |
| 04:37.55 | mog_home | iax2 trunking |
| 04:37.56 | cleyverson | i dont know y but someone told me that my config files were nice, an d was not working propably because i did not install zaptel |
| 04:39.05 | *** join/#asterisk muzzz (n=chatzill@60.48.153.162) |
| 04:39.13 | *** join/#asterisk suarog (n=suarog@202.127.102.186) |
| 04:39.19 | *** join/#asterisk cbrown (n=cbrown@fw.vpxp.com) |
| 04:39.37 | cleyverson | oh my god...a lot of erros an warning appeard :( i am becoming crazy lol |
| 04:40.15 | Trazz | Areil, still there? |
| 04:40.48 | ast_New-B | Ariel in my country is popular band singer... :D |
| 04:41.12 | Flauto | trazz i think he said that he was going to bed |
| 04:41.16 | Flauto | a while ago |
| 04:41.20 | Trazz | ok |
| 04:41.20 | wasim | here its a washing detergent |
| 04:41.30 | [TK]D-Fender | Ariel in my GF's daughter's movie collection is a MERMAID! Beat that! ;P |
| 04:42.02 | suarog | !list |
| 04:42.12 | Trazz | wtf ? |
| 04:42.18 | xtrvd | Trazz: [20:08] * Ariel_ it's bed time..... see you all later. |
| 04:42.25 | Darkhalf | Jeez. I've seen people ask for lists from some random channels, but this one? |
| 04:42.26 | Trazz | eheeh |
| 04:43.07 | Trazz | whats up with the !list crap |
| 04:43.25 | suarog | calm down guy |
| 04:43.49 | Darkhalf | Everyone's chill, maaan. |
| 04:44.30 | suarog | sorry |
| 04:44.41 | cleyverson | someone here knows how to fix the problem that I sended to pastebin?? I am trayig to install the * |
| 04:44.42 | suarog | i thought i can download the asterisk here |
| 04:45.17 | Trazz | must not be any bots.. usually most channels !list = !kickban |
| 04:45.29 | Trazz | download from main site |
| 04:46.06 | cleyverson | someone could help me plz? :) |
| 04:46.11 | suarog | sorry im a new comer to this channel |
| 04:46.19 | Trazz | heheh ... |
| 04:46.40 | Trazz | go here http://www.asterisk.org/ on right side of page |
| 04:46.44 | Trazz | all teh downloads are ready for ya |
| 04:46.45 | [TK]D-Fender | suarog : Go to the WIKI and look for the page detailing how to download and install *. |
| 04:46.47 | [TK]D-Fender | ~docs |
| 04:46.48 | jbot | somebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:46.48 | Trazz | i just queued them for ya |
| 04:47.39 | [TK]D-Fender | cleyverson : Youa re missing all sorts of essential development libraries. Learn fast how to get them and install them or you'll waste a lot of time. |
| 04:50.03 | Trazz | are these times normal or long ? |
| 04:50.04 | Trazz | exten => s,n,Set(TIMEOUT(response)=10) |
| 04:50.04 | Trazz | exten => s,n,Set(TIMEOUT(digit)=9) |
| 04:50.23 | cleyverson | [TK]D-Fender, what libraries are those?? I really dont know |
| 04:50.38 | Trazz | The Mermaid libraries.. |
| 04:50.58 | [TK]D-Fender | Trazz : Digit shoud be more like 3 |
| 04:51.13 | [TK]D-Fender | cleyverson : whats your latest error? |
| 04:51.24 | cleyverson | i sended to pastebin |
| 04:52.28 | [TK]D-Fender | cleyverson : Link please.... I don't intend to have to go searching for it... |
| 04:52.35 | cbrown | http://pastebin.com/511057 |
| 04:52.52 | cleyverson | i fixed 2 problens installing 2 new libraries but others erros and warnings appeard.. and I have no idea how to fix |
| 04:53.09 | cleyverson | thanks cbrown |
| 04:53.20 | cbrown | chan_zap.c:62:2: #error "You need newer libpri" |
| 04:53.49 | Qwell | cbrown: install a newer libpri |
| 04:54.14 | cbrown | I was just quoting from cleyverson's paste. |
| 04:54.44 | [TK]D-Fender | make sure you got a MATCHING set of libpri, zaptel & asterisk, then try again... |
| 04:57.27 | cleyverson | i have installed the version 1.0.6-4 is it the newst version?? |
| 04:57.59 | [TK]D-Fender | cleyverson : Horribly NOT. |
| 04:58.28 | [TK]D-Fender | you missed 1.0.7 / 8/ 9 / 10, and 1.2.0 and 1.2.1. Where on earth did you get that version from? |
| 04:58.44 | [TK]D-Fender | read the damned topic for crying out loud! |
| 04:59.12 | Math` | 1.0.6? wow |
| 04:59.35 | cleyverson | yast told me that hehe |
| 04:59.53 | mog_home | never use asterisk packages from distros.... |
| 04:59.56 | [TK]D-Fender | Keep the hell away from pacgaged * and jsut download the damn source |
| 05:00.33 | cleyverson | ok, i downloaded libpri from digium cvs...let me try to install it |
| 05:00.42 | drumkilla | and 1.2.2 |
| 05:00.53 | [TK]D-Fender | cleyverson : ALL MATCHING VERSIONS! |
| 05:01.17 | [TK]D-Fender | drumkilla : You mentioned its impending release.. got anything OFFICIAL to add? ;) |
| 05:01.18 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.2 has been released -//- http://www.asterisk.org/ (website will be updated soon, but the tarballs are on the ftp) |
| 05:01.30 | [TK]D-Fender | drumkilla : That'll do :) |
| 05:01.34 | drumkilla | :) |
| 05:01.38 | drumkilla | and on that note, I'm going to bed |
| 05:01.44 | [TK]D-Fender | g'nite! |
| 05:01.49 | drumkilla | you too! |
| 05:01.56 | cleyverson | cant i install just the lest version?? i |
| 05:02.05 | Darkhalf | Is there a changelog posted or do I have to actually grab the tar. |
| 05:02.14 | drumkilla | yes, there is a ChangeLog on the ftp |
| 05:02.23 | drumkilla | or will be shortly ... it may take a little bit for the mirrors to catch up |
| 05:02.31 | drumkilla | you will see a ChangeLog-1.2.2 |
| 05:02.42 | [TK]D-Fender | Darkhalf : I'd bet you'll ahve to wait till tomorrow for the posting or downlaod the Tarballs |
| 05:02.48 | drumkilla | for asterisk, zaptel, and libpri |
| 05:03.05 | Trazz | can you stack multiiple sounds in same command? |
| 05:03.06 | Math` | 1.2.2 is getting released now? |
| 05:03.16 | drumkilla | Math`: just did it in the last couple hours |
| 05:03.22 | Qwell | Trazz: I just tried last night with background...seems to have worked |
| 05:03.22 | Math` | oh nice |
| 05:03.26 | Qwell | Background(hail&north) |
| 05:03.42 | Trazz | so instead of playback ? |
| 05:03.43 | Qwell | (Have I mentioned lately that I love the builtin sounds?) |
| 05:03.48 | Qwell | Trazz: playback may work too...try it |
| 05:04.18 | Trazz | ok ) |
| 05:05.08 | Trazz | i am trying to see if you can do transferring to sales department |
| 05:05.10 | Trazz | with default sounds |
| 05:05.21 | Trazz | or tranferring to sales |
| 05:05.23 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 05:05.35 | *** part/#asterisk suarog (n=suarog@202.127.102.186) |
| 05:05.55 | *** join/#asterisk locoast (n=locovox@OL218-21.fibertel.com.ar) |
| 05:06.05 | locoast | printf('Hello world'); |
| 05:06.21 | Qwell | '? |
| 05:06.24 | locoast | anyone having asterisk 1.2 running with Stanaphone? |
| 05:06.25 | Qwell | You're new, aren't you? |
| 05:06.38 | cleyverson | where the libpri is generally installed? i just to see the version installed :) |
| 05:06.40 | locoast | no, just kidding, 2am here |
| 05:06.40 | *** part/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 05:06.49 | luke-jr_ | locoast: used to |
| 05:06.57 | locoast | 1.2.1? |
| 05:07.02 | luke-jr_ | oh, no |
| 05:07.14 | locoast | i had 1.0.9 without a problem |
| 05:07.16 | luke-jr_ | 1.0.7 |
| 05:07.22 | locoast | but 1.2.1 doesnt seem to work for me |
| 05:07.34 | luke-jr_ | you know Stanaphone recently changed their setup? |
| 05:07.38 | locoast | yes, yes |
| 05:07.45 | locoast | I upgraded to 1.2.1 last week |
| 05:08.02 | locoast | so it was working with the stana-in/forwarded to stana-account number |
| 05:08.31 | locoast | it's very weird, may be anyone can show some light here... |
| 05:08.43 | brookshire | 1.2.2 is out |
| 05:08.47 | locoast | My * box responds to the invite |
| 05:08.55 | locoast | but the other side doesnt get it |
| 05:09.03 | locoast | it retries and retries up to the maximum |
| 05:09.18 | *** join/#asterisk Skumling (n=skumling@fw.sg12.dk) |
| 05:10.00 | locoast | checking out top of the SVN right now |
| 05:10.38 | dos000 | anyone i am using asterisk 1.0.9 and i have issues with asterisk not sending the bye messages. Anyone can tell me pointers to this issue ? i have been googling with no definite answer |
| 05:10.59 | locoast | making |
| 05:11.05 | dos000 | anyone i am using asterisk 1.0.9 and i have issues with asterisk not sending the bye messages. Anyone can tell me pointers to this issue ? i have been googling with no definite answer |
| 05:11.36 | locoast | ... |
| 05:11.38 | dos000 | oops |
| 05:12.05 | locoast | not much help... |
| 05:12.46 | Math` | dos000: upgrade to 1.2.2 |
| 05:13.15 | brookshire | now that the link is fixed on the website, lol |
| 05:13.23 | locoast | <PROTECTED> |
| 05:14.06 | dos000 | Math`, good ... i am assuming this is a known bug fixed in 1.2 ... or is this wishful thinking ! |
| 05:14.37 | Math` | It is wishful thinking |
| 05:15.00 | Math` | there was so many changes between 1.0.9 and 1.2.2 to look back if that was an issue |
| 05:15.11 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 05:15.16 | locoast | iw-asterisk*CLI> show version |
| 05:15.16 | locoast | Asterisk SVN-trunk-r8179 built by root @ iw-asterisk on a i686 running Linux on 2006-01-18 05:04:04 UTC |
| 05:15.18 | locoast | lets test |
| 05:15.37 | Trazz | Q, i tried exten => 5,4,Playback(and|goodbye) and it said and and then hung up |
| 05:15.40 | Trazz | not goodbye |
| 05:15.59 | locoast | same thing. |
| 05:16.19 | locoast | it's not working with stanaphone (for me) |
| 05:16.44 | dos000 | Math`, so this may still be in 1.2. ... i am trying to figure the issue so i know i need to upgrade. Thats why i was looking for some pointers/guidance |
| 05:16.54 | locoast | Jan 18 02:08:40 WARNING[16793]: chan_sip.c:1226 retrans_pkt: Maximum retries exceeded on transmission 60f6fd6c341eaf31163ac3ec2e7f61a7@66.114.240.26 for seqno 200 (Critical Response) |
| 05:17.02 | Flauto | locoast, i use stanaphone and it works great |
| 05:17.28 | locoast | Flauto, lattest version of *? |
| 05:17.33 | Trazz | Math, anyway to play multiple sounds in same line like that ? |
| 05:17.35 | Trazz | exten => 5,4,Playback(and|goodbye) |
| 05:17.38 | Trazz | not working |
| 05:17.39 | Math` | no |
| 05:17.41 | Qwell | & |
| 05:17.44 | Math` | Playback(and&goodbye) |
| 05:17.53 | Trazz | ah |
| 05:17.55 | Trazz | my fault |
| 05:17.58 | Math` | yup |
| 05:18.06 | locoast | Flauto, lattest version of *? 1.2.2? |
| 05:18.10 | locoast | Flauto, lattest version of *? 1.2.x? |
| 05:18.53 | Flauto | i think i am using 1.2.1 |
| 05:18.58 | Flauto | but now can i tell |
| 05:19.02 | Math` | show version |
| 05:19.18 | Flauto | Asterisk SVN-trunk-r7230 |
| 05:19.19 | Qwell | haven't we already been through this? |
| 05:19.23 | Qwell | like...seriously? |
| 05:19.28 | Flauto | what is this then |
| 05:19.34 | Qwell | svn trunk, as I said earlier |
| 05:19.50 | locoast | i'm on SVN-trunk-r8179 |
| 05:20.05 | locoast | Does anybody know what version corresponds to r7230? |
| 05:20.10 | Qwell | trunk |
| 05:20.12 | Qwell | 7230 |
| 05:20.18 | Trazz | is there sounds that i can use to accomplis transferrint to sales department ? |
| 05:20.24 | Trazz | dam cant type |
| 05:20.34 | locoast | what was the first release before that trunk |
| 05:20.34 | Trazz | is there sounds that i can use to accomplish transferring to sales department ? |
| 05:20.49 | Flauto | qwell, that means it is not 1.2.1 or 1.2.2? |
| 05:20.55 | Flauto | i mean, i really dont' get it |
| 05:21.06 | Flauto | i was expecting something showing 1.2.? |
| 05:21.09 | Qwell | no |
| 05:21.12 | Qwell | because it isn't 1.2 |
| 05:21.25 | Flauto | if it is, it would show 1.2? |
| 05:21.31 | Qwell | yes |
| 05:21.33 | Flauto | okay |
| 05:21.42 | Flauto | i will makeupdate from cvs |
| 05:22.19 | *** join/#asterisk De_Mon (n=de_mon@fl-71-0-69-21.dhcp.sprint-hsd.net) |
| 05:22.24 | locoast | Flauto, Qwell seems not to be having a good day... You downloaded a development version from the SVN trunk. |
| 05:22.32 | locoast | Which is not an official release. |
| 05:22.48 | Flauto | oh |
| 05:22.51 | Flauto | thanks |
| 05:22.52 | locoast | However, it's probably its between two official releases. |
| 05:22.54 | Flauto | locoast |
| 05:22.56 | Qwell | good day...bad day... |
| 05:23.04 | Qwell | they're all just days |
| 05:23.30 | Flauto | locoast, i am not using official releases, that is true |
| 05:23.33 | cleyverson | I'm trayng to install zaptel firts..i already installed all required libraries (i guess) but the some erros appeared..link http://pastebin.com/511082 |
| 05:23.46 | cleyverson | someone help me plz |
| 05:23.53 | locoast | as you got it from the SVN, it's probably 1.2.x |
| 05:23.55 | Flauto | locoast, should i compile everything or just asterisk after i make update? |
| 05:24.16 | locoast | everything you updated |
| 05:24.20 | edwin_ | make[1]: *** Sem regra para processar o alvo `modules'. Pare. |
| 05:24.24 | edwin_ | you should translate that one :-) |
| 05:24.34 | Flauto | zaptel, libpri and asterisk? |
| 05:24.43 | locoast | it depends on the one you update |
| 05:24.44 | Flauto | or even sounds and addons? |
| 05:24.54 | locoast | when you update asterisk, you have to recompile asterisk |
| 05:24.56 | locoast | and so on |
| 05:25.03 | locoast | Flauto, you have stana working right now? |
| 05:25.26 | Flauto | yes |
| 05:25.37 | locoast | can you please email me your sip.conf -excluding your passwords- to leandror@gmail.com ? |
| 05:25.59 | locoast | I cant get it to work |
| 05:26.09 | locoast | in fact, i'm going to try with your trunk |
| 05:26.45 | hmodes | hrmm |
| 05:27.17 | cleyverson | edwin => withou rules to process the 'modules'. Stop |
| 05:27.22 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-72-130-156-112.san.res.rr.com) |
| 05:27.48 | Flauto | locoast, can you see me talking to you privately? |
| 05:28.05 | locoast | yes, however i cant respond |
| 05:28.09 | locoast | since im "anonymouse" |
| 05:28.13 | locoast | he |
| 05:28.16 | hmodes | don't suppose anyone has had problems with sox not feeding MoH proper audio frames occasionally when sending raw audio? |
| 05:28.26 | locoast | but you can post it there |
| 05:28.38 | locoast | i can see it |
| 05:28.39 | *** join/#asterisk tetsuzan (n=rider@201.2.206.4) |
| 05:28.41 | cleyverson | edwin: r u there? |
| 05:28.47 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:29.18 | tetsuzan | can anyone tell me |
| 05:29.23 | tetsuzan | what is "VAT CODE" ? |
| 05:29.27 | Flauto | okay, locoast |
| 05:29.29 | Flauto | wait |
| 05:29.31 | franck | Hi |
| 05:29.33 | locoast | tnx |
| 05:29.47 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 05:30.15 | Corydon76-home | VAT=Value Added Tax |
| 05:30.20 | franck | I have been trying the demo of asterisk, when the background intro plays and then I hangup the Zap phone, it does not hangup till the end of the background. Is that normal behavior? |
| 05:30.24 | *** join/#asterisk lunk (n=lunk@66.152.8.184) |
| 05:30.28 | Corydon76-home | Basically it's local sales tax |
| 05:30.34 | tetsuzan | Corydon76-home, this tax, is defined by who? |
| 05:30.43 | tetsuzan | local? |
| 05:30.48 | Corydon76-home | Various European countries |
| 05:30.56 | franck | tetsuzan: local |
| 05:31.13 | tetsuzan | i'm installing MCC 1.2 to control the billing |
| 05:31.15 | tetsuzan | prepaid and postpaid |
| 05:31.24 | tetsuzan | on my local asterisk (sip channels) |
| 05:31.31 | locoast | thanks, checking it right now |
| 05:31.31 | Flauto | locoast, you got it? |
| 05:31.40 | tetsuzan | but, thanks... |
| 05:31.43 | tetsuzan | :) |
| 05:31.45 | franck | tetsuzan: vat is like sales tax, except only the comsumer at the end of the chain pays it |
| 05:31.47 | locoast | yes, checking it |
| 05:31.49 | locoast | 1 sec |
| 05:31.58 | tetsuzan | thanks franck |
| 05:32.13 | tetsuzan | :) |
| 05:32.23 | franck | tetsuzan: all the others (companies) can get it re-imbursed on their purchases, but have to add it on their sales... |
| 05:32.23 | tetsuzan | i'm porting the "mcc" software |
| 05:32.24 | tetsuzan | to freebsd |
| 05:32.36 | tetsuzan | and i'm translating to 4 languages too |
| 05:32.42 | Mavvie | it says value added, it all it adds to is the cost. |
| 05:32.59 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:33.10 | tetsuzan | hmm |
| 05:33.14 | Mavvie | tetsuzan: dumb question: what is mcc? |
| 05:33.30 | tetsuzan | it's a billing solution for asterisk |
| 05:33.39 | Mavvie | aha. |
| 05:33.41 | tetsuzan | Postgresql solution |
| 05:33.56 | tetsuzan | these billing solutions (opensource) |
| 05:34.00 | tetsuzan | that we see on the internet, |
| 05:34.10 | tetsuzan | are only with mysql, |
| 05:34.10 | Mavvie | have to look at that in the near future. |
| 05:34.15 | franck | if you buy something 10$ on the VAT is 10% you can get rimbursed $1 now you sell back this item for $20 with a vat of $2... so basically you have paid $2-$1 to the taxman |
| 05:34.27 | tetsuzan | thanks franck |
| 05:34.31 | franck | the comsumer buys the stuff at $20 and cannot get it re-imbursed |
| 05:34.33 | tetsuzan | copy that |
| 05:35.31 | franck | ok, anybody on my demo problem and hanging up during a background? |
| 05:35.50 | franck | should the system detect the hangup much earlier? |
| 05:35.57 | franck | or is it normal? |
| 05:36.04 | locoast | Flauto, got it, copied (just added the qualify=yes), however same thing.... |
| 05:36.14 | locoast | i'm building your trunk version |
| 05:36.30 | Flauto | is it working? |
| 05:36.33 | locoast | nop |
| 05:36.35 | locoast | same thing |
| 05:36.48 | Flauto | what problem do you have |
| 05:36.51 | Flauto | what do you see |
| 05:36.53 | locoast | retrying the 200 OK after receiving the first invite |
| 05:37.02 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-136-35.dhcp.sprint-hsd.net) |
| 05:37.16 | Flauto | what do you have under general in sip.conf |
| 05:37.17 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 05:39.02 | Flauto | that qualify is not important at all |
| 05:39.21 | locoast | [general] |
| 05:39.21 | locoast | context=incoming |
| 05:39.21 | locoast | allowguest=no |
| 05:39.21 | locoast | bindport=506 |
| 05:39.21 | locoast | bindaddr=0.0.0.0 |
| 05:39.32 | locoast | srvlookup=yes |
| 05:39.32 | locoast | tos=184 |
| 05:39.32 | locoast | tos=lowdelay |
| 05:39.32 | locoast | defaultexpiry=300 |
| 05:39.40 | locoast | canredirect=no |
| 05:39.40 | locoast | canreinvite=no |
| 05:39.40 | locoast | musicclass=default |
| 05:39.41 | locoast | language=en |
| 05:39.48 | locoast | relaxdtmf=yes |
| 05:39.48 | locoast | rtptimeout=60 |
| 05:39.48 | locoast | rtpholdtimeout=300 |
| 05:39.48 | locoast | useragent=Asterisk PBX |
| 05:39.48 | locoast | dtmfmode=rfc2833 |
| 05:39.53 | locoast | compactheaders = yes |
| 05:39.53 | locoast | qualify=1000 |
| 05:39.53 | locoast | registertimeout=60 |
| 05:39.55 | locoast | callevents=no |
| 05:39.57 | locoast | nat=no |
| 05:40.02 | locoast | context=incoming |
| 05:40.07 | locoast | disallow=all |
| 05:40.07 | locoast | allow=ulaw |
| 05:40.29 | locoast | register => xxxxxxx:xxxxxxx@sip.stanaphone.com/xxxxxxx |
| 05:40.32 | locoast | thats it |
| 05:40.51 | Flauto | it should be okay |
| 05:40.52 | *** join/#asterisk msw (n=msw@rdu-nat.rpath.com) |
| 05:41.34 | locoast | WAY TO GO |
| 05:41.34 | Flauto | though, i don't have relaxdtmf=yes, rtptimeout, rtpholdtimeout, and useragent |
| 05:41.39 | locoast | i just changed |
| 05:41.47 | locoast | compactheaders to NO |
| 05:41.49 | locoast | and it worked |
| 05:42.00 | Flauto | good |
| 05:42.05 | Flauto | it is working now? |
| 05:42.06 | locoast | thanks for the help |
| 05:42.07 | locoast | yes |
| 05:42.11 | locoast | PERFECTLY |
| 05:42.14 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 05:42.14 | Flauto | great |
| 05:42.19 | locoast | at least for the inbound calls |
| 05:42.31 | Flauto | whtais that compactheaders? |
| 05:42.35 | Flauto | what is it that for |
| 05:42.44 | locoast | dont have an idea |
| 05:42.51 | locoast | will auto answer my posts in the digium and stana forums |
| 05:43.05 | Flauto | hehe |
| 05:43.07 | Flauto | okay |
| 05:43.45 | *** join/#asterisk MasterKill (n=not@71.226.252.76) |
| 05:43.54 | Flauto | what is relaxdtmf? |
| 05:43.58 | *** join/#asterisk cyber (n=kani@220.247.244.39) |
| 05:44.15 | cyber | good evening guys |
| 05:44.20 | *** join/#asterisk mgoh (n=goh@60.49.6.190) |
| 05:44.22 | pifiu | anyone here a polycom guru? |
| 05:44.35 | Qwell | ~pb |
| 05:44.37 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 05:44.38 | Qwell | locoast: ^ |
| 05:44.56 | *** join/#asterisk Alex1 (n=A@188.90.233.220.exetel.com.au) |
| 05:44.56 | mgoh | any can help can't receive in coming call |
| 05:45.21 | *** join/#asterisk ryansc_ (n=ryansc@h139-055-149-047.adsl.navix.net) |
| 05:45.30 | Corydon76-home | mgoh: do you have an extension s in your default context? |
| 05:46.02 | mgoh | what mean extension s? |
| 05:46.34 | Corydon76-home | exten => s,1,..... |
| 05:46.34 | mgoh | I have create a few extension. |
| 05:46.34 | Qwell | Corydon76-home: you asked for that one |
| 05:46.34 | Qwell | :p |
| 05:46.40 | locoast | yes, sorry about that.. . but noone was saying anything... |
| 05:46.41 | Corydon76-home | Qwell: I did? |
| 05:46.45 | locoast | very quiet |
| 05:46.52 | locoast | tha channel was |
| 05:46.54 | locoast | lol |
| 05:46.57 | Qwell | indeed you did |
| 05:47.29 | Qwell | Corydon76-home: Have you been getting anything from the lists today? |
| 05:47.38 | Corydon76-home | Sure |
| 05:47.56 | Qwell | silly lists |
| 05:47.57 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 05:47.58 | Corydon76-home | Had 45 -dev messages when I got back from my on-site |
| 05:48.02 | Qwell | nice |
| 05:48.19 | Qwell | I wonder if my mail server really was down also, and it bounced them all, so it shut me down, heh |
| 05:48.37 | Corydon76-home | Our T1 is currently down... |
| 05:48.44 | *** join/#asterisk rob0 (n=rob0@sorry.no-ip-here.net) |
| 05:49.35 | cyber | are there any h323 geeks |
| 05:49.53 | cyber | i tried the h323 interface for like 2 weeks and all failed |
| 05:50.55 | mgoh | who familiar with AMP here? |
| 05:51.27 | Corydon76-home | Nope |
| 05:52.08 | *** join/#asterisk muzzz (n=chatzill@60.48.153.162) |
| 05:52.47 | Corydon76-home | Geeks don't use H.323... only nerds... |
| 05:53.19 | Math` | lol |
| 05:53.45 | Corydon76-home | and you know I'm right... |
| 05:53.55 | Math` | yes I do |
| 05:54.01 | *** join/#asterisk diclophis (n=diclophi@comet.cisdata.net) |
| 05:54.10 | diclophis | hello |
| 05:54.23 | diclophis | so i have a problem with realtime voicemail |
| 05:54.49 | Corydon76-home | diclophis: upgrade to 1.2.2, it's fixed in there |
| 05:55.00 | Qwell | diclophis: What problem? |
| 05:55.09 | cbrown | lol |
| 05:55.11 | diclophis | 1.2.2 the problem where the context isnt passed into the SQL retrive statement |
| 05:55.13 | Corydon76-home | Qwell: the problem we fixed in 1.2 two weeks ago |
| 05:55.15 | Qwell | oh, right, no more need for that bug number...awesome |
| 05:55.25 | diclophis | i know i am passing into the voicemail cmd, 1@1 |
| 05:55.33 | Qwell | diclophis: Corydon76-home was right then |
| 05:55.34 | diclophis | yet in the debug log all that shows up is |
| 05:55.37 | Qwell | upgrade, problem solved |
| 05:55.42 | Corydon76-home | diclophis: you think I'm joking? |
| 05:55.46 | diclophis | SELECT * FROM voicemailboxes WHERE mailbox = '1' AND context = '' |
| 05:55.52 | Qwell | shh, listen :p |
| 05:55.53 | diclophis | no.. i am just making sure it is the same bug |
| 05:55.56 | *** join/#asterisk v3ector (i=debian-t@tor/session/x-eeccbfa76db86aec) |
| 05:55.58 | Qwell | yes, it is |
| 05:56.00 | Corydon76-home | I've heard that exact complaint FIVE TIMES on the bugtracker |
| 05:56.02 | diclophis | where do i get 1.2.2 ? |
| 05:56.08 | Qwell | same place you got 1.2.1 |
| 05:56.10 | Corydon76-home | It's on the FTP site |
| 05:56.15 | diclophis | rockin |
| 05:56.20 | diclophis | thanks a bundle |
| 05:56.21 | Qwell | Corydon76-home: five at least |
| 05:56.31 | Corydon76-home | Five that I've had to close |
| 05:56.33 | Qwell | I just closed one today I believe it was |
| 05:56.35 | diclophis | so.. where is the best place to get info like that |
| 05:56.49 | diclophis | .... i thought voip-info wiki was up to date... but it seems to not be |
| 05:56.52 | Qwell | diclophis: the channel topic |
| 05:56.59 | Corydon76-home | Best thing to do is to checkout 1.2 SVN |
| 05:57.06 | Corydon76-home | then upgrade it periodically |
| 05:57.16 | Corydon76-home | We guarantee only fixes go into that branch |
| 05:57.18 | *** join/#asterisk Trazzz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 05:57.23 | Qwell | speaking of 1.2 svn... |
| 05:57.45 | Trazzz | is there a way to accomplish "transferring to sales department" or somethig close with default sounds? |
| 05:57.49 | Qwell | Corydon76-home: not sure you got the email from mantis, I think it was flakey at the time... Your patch worked fine on 1.2 for the realtime queues thing |
| 05:58.13 | diclophis | dang.. and i just upgraded to 1.2.1 today |
| 05:58.28 | diclophis | i was hoping that would be better than the old 1.2.0 version i was using |
| 05:58.29 | Corydon76-home | Qwell: I'll mess with it on Thursday... I have another on-site tomorrow |
| 05:58.36 | diclophis | then i find out all my AGI code is bunked |
| 05:58.38 | Qwell | Corydon76-home: yep, no problem |
| 05:58.42 | diclophis | cause of the voicemail context problem |
| 05:58.49 | diclophis | i can rest easy tonite |
| 05:59.00 | diclophis | i was dreading iw ould have to hop into the C code tomorrow |
| 05:59.07 | Qwell | I actually found another issue with voicemail...umm...I forget what it was, heh |
| 05:59.14 | diclophis | .. |
| 05:59.23 | Qwell | didn't have enough time to look at it today...might just be a config issue |
| 05:59.44 | diclophis | well, if it 1.2.2 works like 1.2.1 then i will be fine |
| 05:59.47 | Trazzz | Qwell, do you think with default sounds i can get that done? |
| 05:59.49 | Qwell | oh, right. option 3, option 5 |
| 05:59.56 | diclophis | ...? |
| 06:00.03 | Qwell | doesn't do the contexts thing |
| 06:00.05 | diclophis | as in send voicemail to another box |
| 06:00.08 | Qwell | yeah |
| 06:00.13 | diclophis | which IIRC is option 5 |
| 06:00.16 | diclophis | what does option 3 do?> |
| 06:00.17 | Corydon76-home | With the exception of searchcontexts in voicemail.conf, yes |
| 06:00.18 | Qwell | should work if they're in default, but...I didn't test much |
| 06:00.29 | Qwell | Corydon76-home: nah, it doesn't pass the context to leave_voicemail, iirc |
| 06:00.38 | diclophis | searchcontexts in voicemail.conf |
| 06:00.39 | diclophis | ? |
| 06:00.55 | Corydon76-home | SOMEbody changed the default behavior between 1.2.0 and 1.2.1 |
| 06:00.58 | Qwell | I didn't test with searchcontexts, or default context |
| 06:01.16 | Qwell | yeah...that bugs me |
| 06:01.37 | Qwell | but...what're you gonna do? |
| 06:02.04 | diclophis | bendover and enjoy it? |
| 06:02.16 | Corydon76-home | I can keep reminding people that it happened, to keep another thing like that from happening again |
| 06:02.55 | Qwell | Corydon76-home: it *was* technically a bug though, one way or another |
| 06:02.56 | Corydon76-home | What makes you think anal sex isn't enjoyable? |
| 06:03.00 | Qwell | but, I digress...I disagree |
| 06:03.06 | Qwell | and, you just had to go there, didn't you? :P |
| 06:03.23 | Trazzz | is there a way to accomplish "transferring to sales department" or something close with default sounds? |
| 06:03.29 | Qwell | Trazzz: sounds.txt |
| 06:03.32 | Qwell | sounds-extra.txt |
| 06:03.38 | Qwell | grep is your friend.. |
| 06:04.11 | Corydon76-home | Qwell: why, I'm a perfect little angel... |
| 06:04.19 | Qwell | I'm sure you are |
| 06:04.54 | *** join/#asterisk argos73 (i=1000@jason.argos.org) |
| 06:06.38 | Corydon76-home | Qwell: and I fixed it initially, by changing the documentation |
| 06:06.45 | Qwell | heh |
| 06:06.49 | diclophis | well, on that note i am heading out |
| 06:06.51 | diclophis | thanks all |
| 06:06.59 | *** part/#asterisk diclophis (n=diclophi@comet.cisdata.net) |
| 06:07.19 | Qwell | here's what really bothers me |
| 06:07.34 | Qwell | how many people actually use realtime? for anything... sip, iax, queues, voicemail...whatever |
| 06:07.48 | Qwell | with all the stuff I'd seen...my guess is ...zero |
| 06:07.54 | cyber | i use real time |
| 06:08.06 | Math` | realtime is useful for scalable stuff |
| 06:08.07 | Qwell | maybe I'm just too picky |
| 06:08.14 | Qwell | Math`: when it works |
| 06:08.21 | Math` | if you have 10 extensions its not worth it, but when you have 3000, it is |
| 06:08.24 | Math` | when it works yeah |
| 06:08.40 | Qwell | I mean...don't get me wrong here. I use realtime, and I love it |
| 06:09.03 | *** join/#asterisk pnlarsson (n=pnlarsso@c83-248-7-150.bredband.comhem.se) |
| 06:09.13 | Qwell | some stuff though...I just don't understand how people haven't seen it |
| 06:09.55 | [hC-] | Id like to be using realtime |
| 06:10.10 | [hC-] | ive just read alot of the time that its not as stable/usable as it sounds |
| 06:10.20 | [hC-] | at least, thats the impression i get |
| 06:10.32 | Alex1 | Hello I'm having this problem I will explain it for you . I first mount the hard drive read/write I make my changes with a text editor modifying things files |
| 06:10.33 | Qwell | it's quite usable |
| 06:12.04 | Alex1 | I save the changes mount the hard drive read only and reboot .... but when I reboot the system has reverted to the default please help me I'm ready to smash it I have just spent 10 hours on it help me |
| 06:12.46 | Math` | you used -o remount? |
| 06:13.09 | Alex1 | that's correct Math |
| 06:13.21 | Trazzz | Qwell, i dont see sounds-extra.txt |
| 06:13.29 | Qwell | Trazzz: asterisk-sounds package |
| 06:13.34 | Alex1 | what's the problem |
| 06:13.44 | Trazzz | i have the sounds package.. |
| 06:13.50 | Trazzz | is there two o fthem ? |
| 06:13.56 | Math` | vtl-gw:~/asterisk-sounds# ls |
| 06:13.56 | Math` | CVS Makefile README.txt sounds sounds-extra.txt |
| 06:14.32 | Trazzz | gots it |
| 06:14.44 | Alex1 | Hello I'm having this problem I will explain it for you . I first mount the hard drive read/write I make my changes with a text editor modifying things files I save the changes mount the hard drive read only and reboot .... but when I reboot the system has reverted to the default please help me I'm ready to smash it I have just spent 10 hours on it help me |
| 06:14.54 | *** join/#asterisk alphaque (n=alphaque@60.48.153.162) |
| 06:14.55 | Math` | Alex1: don't repeat yourself |
| 06:15.11 | Qwell | reverted to the default what? |
| 06:15.14 | Math` | if people don't answer they either don't know or don't want to bother thinking |
| 06:15.23 | Qwell | sounds like a Linux question...not asterisk |
| 06:15.54 | *** join/#asterisk bkw__ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 06:15.55 | Alex1 | Qwell: All the changes I make have vanished next time the system boots |
| 06:15.59 | pifiu | who here is a polycom guru? |
| 06:16.10 | Math` | pifiu: just ask, there's a few of em I'm sure :P |
| 06:16.23 | Qwell | I hear bkw__ knows polycom |
| 06:16.32 | pifiu | for some reano every time i deploy a polycom 501 i try to dial or even pickup it says "url calling disabled" |
| 06:16.42 | pifiu | i always forget what causes that |
| 06:16.48 | Math` | uh I never had that |
| 06:17.09 | Math` | I used the example config for asterisk available on the wiki |
| 06:17.21 | pifiu | i always had, and i always fix it, but i dont kno what causes it, and i think i fix it by changing the outbound number |
| 06:17.22 | Alex1 | hello |
| 06:17.26 | pifiu | even though that should not have anything to do with it |
| 06:18.22 | Math` | http://www.imation.com/products/flash_devices/usb_flash_wristband.html |
| 06:18.33 | Math` | When you need to wear your data heh |
| 06:18.50 | littleball | hello, anyone has experience of using realtime? |
| 06:18.58 | [hC-] | hahaha |
| 06:19.05 | [hC-] | Qwell: care to take this one? |
| 06:19.14 | Math` | lol |
| 06:19.21 | Math` | see? people are using realtime :) |
| 06:19.22 | littleball | if i put dial plan etc into the db, then how about extension.conf file? |
| 06:19.23 | *** join/#asterisk Essobi (i=kstone@75.137.26.216.host.teledvance.com) |
| 06:19.32 | [hC-] | er.... that IS your dial plan. |
| 06:19.44 | Qwell | littleball: You can use both |
| 06:19.47 | Qwell | nothing stopping you |
| 06:19.57 | littleball | does it still take effect? I want to disable extension.conf then |
| 06:20.22 | Qwell | You want the entire config in there, or you want to use realtime switch? |
| 06:20.25 | littleball | i just setup the odbc stuff for postgresql. |
| 06:20.25 | Math` | then don't define any context in extensions.conf |
| 06:20.33 | littleball | realtime switch |
| 06:20.38 | Qwell | Then you have to use both |
| 06:20.38 | littleball | ok |
| 06:20.41 | argos73 | any idea if a 50-foot cable between a Merlin Legend and a TE410 is too long??? (do I need a CSU in there???) |
| 06:20.52 | Qwell | [my-context] |
| 06:20.57 | Qwell | switch => Realtime/@ |
| 06:20.58 | Qwell | or some such |
| 06:21.04 | littleball | does it mean asterisk will merge the configuration items from both db and extension.conf file? |
| 06:21.19 | Qwell | no |
| 06:21.29 | Qwell | It will make a hit to the database every time it needs to do a match |
| 06:21.30 | *** join/#asterisk jerlique (n=jerlique@ppp103-178.static.internode.on.net) |
| 06:21.31 | Math` | a switch is just gonna lookup the db every time |
| 06:22.21 | littleball | so, i can have one context using realtime and others still use extension.conf, right? |
| 06:22.34 | *** join/#asterisk wellng (n=welles@222.90.15.150) |
| 06:22.42 | rowter | on asterisk 1.2 ANSWEREDTIME is diferent? am not getting a value.. |
| 06:22.48 | Qwell | littleball: you can even do both |
| 06:23.11 | littleball | Qwell, switch=> is a new in 1.2, right? |
| 06:23.16 | Math` | no |
| 06:23.24 | Qwell | I think realtime switch is |
| 06:23.36 | Qwell | switch isn't though |
| 06:23.37 | Math` | but the switch statement existed for a long time |
| 06:24.08 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 06:24.24 | littleball | Qwell, i cannot find switch statement in my handbook though :) |
| 06:24.36 | Qwell | it's in there |
| 06:24.49 | Qwell | or...should be |
| 06:25.32 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 06:25.38 | *** join/#asterisk clive- (n=pirch@dsl-165-150-91.telkomadsl.co.za) |
| 06:25.39 | littleball | can you give a link for its document? google cannot find it |
| 06:26.10 | Qwell | doc/README.realtime |
| 06:26.14 | littleball | ok |
| 06:26.35 | Telamon | Hello. I've got a strange problem. Recently (just in the past week or two) IAX calls via my asterisk server have started cutting out. SIP calls still work perfectly, but I get periods of about 1 second of silence every 5-10 seconds of the call. Any ideas what I can do to diagnose this? iax2 debug doesn't seem to show any reason for the cutouts. |
| 06:26.38 | Qwell | actually, no, that's not useful |
| 06:26.40 | Qwell | ~wikis |
| 06:26.44 | jbot | wikis is, like, http://www.voip-info.org |
| 06:26.44 | Qwell | check there |
| 06:26.46 | Qwell | ... |
| 06:26.46 | Qwell | there |
| 06:27.50 | *** join/#asterisk oogle_ (n=jart@ool-435721a3.dyn.optonline.net) |
| 06:28.04 | oogle_ | does anyone know where asterisk keeps the berkley db files? |
| 06:28.39 | oogle_ | for some reason when i'm loading asterisk, it's failing to load my /etc/asterisk configurations and loading stuff long deleted |
| 06:29.11 | Nugget | /var/lib/asterisk/astdb I believe. |
| 06:30.18 | oogle_ | thanks nugget |
| 06:30.50 | oogle_ | phew, things seem to have calmed down now |
| 06:30.58 | littleball | Qwell, i really cannot get doc for switch. what is the main purpose of switch statement? |
| 06:31.02 | oogle_ | must have been an update bug |
| 06:32.07 | Telamon | littleball: Are you talking about switch in realtime configs, or switch as in pointing to an entry on another server? |
| 06:32.11 | Qwell | littleball: to do a lookup of an extension against another server |
| 06:32.29 | littleball | both, Telamon, because i cannot get any useful doc about it |
| 06:33.02 | littleball | because i just heard "switch" from Qwell :) |
| 06:33.10 | littleball | so i try to understand it |
| 06:33.52 | Telamon | http://www.asteriskguru.com/tutorials/extensions_conf.html has an entry on using it to connect servers, for realtime you just put "switch => Realtime" in the extensions.conf context to tell it to try and load from the DB. extnconfig.conf maps the tables for realtime. |
| 06:34.15 | littleball | thanks |
| 06:34.45 | Telamon | http://www.asteriskguru.com/tutorials/realtime_pgsql.html gives a bit of an example of the realtime stuff, although it doesn't mention configuring extensions.conf strangely enough (at least that I notice, maybe it's in the graphics...) |
| 06:35.21 | Telamon | Oh, and it's a little out of date. You don't need CVS head for realtime, it's been built in since 1.2.0 at least. |
| 06:36.12 | clive- | ~changelog |
| 06:36.14 | jbot | i guess changelog is dpkg-deb --fsys-tarfile whatever.deb | tar xOf - '*changelog.Debian.gz' | gzip -d | less |
| 06:36.17 | Telamon | Hey Qwell, does 1.2.2 fix all the voicemail problems, or are there still some issues? |
| 06:36.25 | Qwell | Telamon: it fixes severa |
| 06:36.27 | Qwell | l |
| 06:36.45 | clive- | where is the changelog to see whats new in 1.2.2 |
| 06:36.54 | Qwell | the source |
| 06:36.58 | Telamon | Hmm, might give it a try to see if it makes any difference in the IAX2 problems we're having. |
| 06:37.05 | Alex1 | ok i have mounted the drive reade write i have made my changes saved them thaY are there i am ready to reboot . what should i do now ? |
| 06:37.07 | clive- | qwell...lol |
| 06:37.28 | Telamon | clive-: I think he means that there is a Changelog text file in the tarball. :) |
| 06:37.54 | Telamon | Of course, you could just run diff on the two trees, if you're hardcore. ;-) |
| 06:37.55 | Alex1 | reboot |
| 06:37.56 | clive- | oh, ok....thee is a url somewhere with the changelog,,,thats what I am after |
| 06:38.11 | mog_home | gnite people |
| 06:38.12 | *** part/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net) |
| 06:38.46 | Telamon | http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.2 |
| 06:39.09 | Telamon | Funny, the 1.2.1 Changelog gives a 404, but the 1.2.2 one works fine. |
| 06:41.02 | Mavvie | * channels/chan_agent.c: fix speling errors (issue #6227) |
| 06:41.08 | Qwell | Mavvie: nice |
| 06:41.37 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:42.34 | *** join/#asterisk dasuberdavid (n=david@pcp01534754pcs.huntsv01.al.comcast.net) |
| 06:43.08 | Mavvie | I should have remembered the command to update one SVN tag to another one. |
| 06:43.29 | Alex1 | ok i have mounted the drive reade write i have made my changes saved them thaY are there i am ready to reboot . what should i do now ? |
| 06:44.04 | Qwell | Mavvie: perhaps an svn switch |
| 06:44.11 | Mavvie | switch yes. |
| 06:46.14 | Mavvie | I should have written the command down. |
| 06:46.39 | *** join/#asterisk [hC-] (n=hardcore@209.153.195.139) |
| 06:47.08 | *** join/#asterisk muzzz (n=chatzill@60.48.153.162) |
| 06:48.19 | Corydon76-home | Qwell: Set(one=${SET(two=${SET(three=${SET(four=0)})})}) |
| 06:49.13 | Corydon76-home | Is that enough to make your brain hurt? |
| 06:49.35 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 06:49.59 | *** join/#asterisk [hC-] (n=hardcore@209.153.195.139) |
| 06:51.17 | Alex1 | what the fuck is going on i mount the drive rw i change things reboot the computer and its not there ????? Help |
| 06:51.44 | Telamon | Alex1: Did you add the mount to /etc/fstab? |
| 06:51.47 | Mavvie | >svn switch http://svn.digium.com/svn/zaptel/tags/1.2.2/ <- that was the trick |
| 06:52.00 | Qwell | ...UGH |
| 06:52.00 | *** join/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
| 06:52.04 | Qwell | Corydon76-home: wtf was that? :P |
| 06:52.25 | Corydon76-home | Qwell: an example of how to properly use the SET function |
| 06:52.31 | Qwell | nice...was wondering |
| 06:52.45 | Qwell | still not sure I get why, but okay |
| 06:52.46 | Corydon76-home | Somebody put question marks about that on the Wiki |
| 06:52.55 | Qwell | What does SET return? |
| 06:53.02 | Corydon76-home | The value set |
| 06:53.03 | *** join/#asterisk welles (n=welles@222.90.15.150) |
| 06:53.06 | Qwell | I see |
| 06:53.18 | *** join/#asterisk dos000 (n=dos000@wsp05974758wss.cr.net.cable.rogers.com) |
| 06:53.28 | Qwell | try this on for size |
| 06:53.28 | Corydon76-home | You can only read from that function, not write to it |
| 06:54.01 | Corydon76-home | Oh, and I highlighted the functions in red that are only available in 1.3 |
| 06:54.40 | Qwell | Set(four=$[${SET(three=$[${SET(two=$[${SET(one=1)}+1])}+1])}+1]) |
| 06:54.47 | Qwell | That very likely doesn't parse, but you get the point |
| 06:55.16 | Corydon76-home | Yep |
| 06:55.39 | Qwell | actually, it looks like it does parse |
| 06:56.01 | Corydon76-home | Well, that should parse out correctly |
| 06:56.10 | Qwell | That's just fugly |
| 06:56.29 | Corydon76-home | Nah, it's elegant |
| 06:57.49 | Qwell | ooo, gotta throw an ODBC_GET_NUMBER_BY_CALLERID_NUMBER_AND_CHANNEL_TYPE in the works |
| 06:58.22 | Corydon76-home | That's just abusive |
| 06:58.49 | *** join/#asterisk JamesDotCom (i=jamesdot@sweep.bur.st) |
| 07:03.27 | Qwell | I wonder what the limit is for a function name... |
| 07:03.34 | Qwell | I might need to increase that next :p |
| 07:10.24 | *** join/#asterisk shawn (n=welles@222.90.15.150) |
| 07:10.51 | Corydon76-home | There is no arbitrary limit on a function name |
| 07:11.11 | Corydon76-home | If you wanted to, you could have a 2000-byte function name |
| 07:11.15 | Qwell | nice |
| 07:11.34 | Corydon76-home | Now, there is a limit on the total length of a line, imposed by config.c |
| 07:11.45 | Qwell | Would that matter with realtime switch? |
| 07:11.56 | Corydon76-home | Unsure |
| 07:12.13 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-224-179.dsl.irvnca.pacbell.net) |
| 07:12.16 | Corydon76-home | There's also limits on the size of buffers |
| 07:12.22 | Qwell | of course |
| 07:12.36 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 07:13.27 | hmodes | wait() needs an infinite option, speaking of which |
| 07:14.01 | Corydon76-home | VAR_BUF_SIZE is set to 4096 in pbx.c |
| 07:14.01 | hmodes | unless there's a better way to hold a channel open i'm not aware of |
| 07:14.19 | *** join/#asterisk Pazzo (n=Pazzo@host130-250.pool8172.interbusiness.it) |
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| 07:14.27 | Corydon76-home | hmodes: 2 million seconds isn't long enough? |
| 07:14.28 | *** part/#asterisk Jane_Cooley (n=templev@adsl-71-135-34-253.dsl.pltn13.pacbell.net) |
| 07:14.32 | *** join/#asterisk Jane_Cooley (n=templev@adsl-71-135-34-253.dsl.pltn13.pacbell.net) |
| 07:14.37 | Corydon76-home | excuse me, two billion seconds? |
| 07:14.59 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 07:15.02 | hmodes | 8bil seemed like 'forever' when it came up :) |
| 07:15.19 | hmodes | so i just have it waiting 8mil and looping on a timeout |
| 07:15.21 | *** join/#asterisk CleanerX (n=nix@vpnwww01.rz.uni-karlsruhe.de) |
| 07:15.21 | hmodes | but still |
| 07:15.31 | hmodes | it would have been nice to have a wait forever option |
| 07:15.34 | Corydon76-home | If you have to wait 75 years for a digit, I don't think there's going to be one coming |
| 07:15.42 | hmodes | hehe |
| 07:15.44 | hmodes | true |
| 07:15.55 | hmodes | it's a very nit-picky point :) |
| 07:16.07 | Corydon76-home | There's 31 million seconds in a year |
| 07:16.09 | argos73 | dunno there... i've dealt with some people dumb enough to wait that long before hitting an option... |
| 07:16.27 | Corydon76-home | I don't know that the PSTN supports calls that long... |
| 07:16.32 | hmodes | well in that case i'm waiting a couple months between loops |
| 07:16.41 | hmodes | that seems like a waste of cycles every couple months ;p |
| 07:17.24 | hmodes | also, this is a call to a local channel |
| 07:17.28 | hmodes | just to keep it open |
| 07:17.31 | hmodes | forever. |
| 07:17.40 | hmodes | it's a shady moh hack |
| 07:18.36 | Corydon76-home | It's shady, alright |
| 07:18.55 | *** part/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 07:19.33 | hmodes | yeah, the end situation is multicasts that moh subscribes to as necessary... but I had 6 days for a tech demo, so... shady. |
| 07:20.14 | hmodes | also, 1.2.1 moh doesn't like shady raw streams, it seems :) |
| 07:20.33 | hmodes | i forsee a lot of work in my future |
| 07:20.58 | Corydon76-home | I suspect the telephone would crumble into dust before you reached 75 years |
| 07:21.24 | hmodes | the box will easily loose power this year anyway |
| 07:21.26 | justinu | not if it was a western electric 2500 |
| 07:21.38 | hmodes | like I said, I'm nit-picking ;p |
| 07:21.44 | Corydon76-home | groundstart? |
| 07:22.06 | justinu | did they make groundstart 2500s? |
| 07:22.19 | Corydon76-home | Dunno, but I'm going to bed |
| 07:22.28 | justinu | night |
| 07:23.37 | littleball | <PROTECTED> |
| 07:24.00 | justinu | i'm gonna buy an authentic antique candlestick phone |
| 07:24.06 | justinu | it has to work |
| 07:24.14 | justinu | i'm gonna hook it up to an ATA |
| 07:24.34 | justinu | that'll be my phone |
| 07:27.30 | locoast | printf("Goodbye all"); |
| 07:28.12 | *** join/#asterisk EriSan (n=erisan@151.8.109.88) |
| 07:33.55 | hmodes | iaxy is the only thing i'm aware of that can pulse dial |
| 07:34.07 | hmodes | I bet it could even be made to operator-transfer a crank dial phone :) |
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| 08:19.36 | MadCamel | looking for opinions.. for me voicepulse has been a bit unreliable for incoming service, anyone else have problems? |
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| 08:24.12 | *** join/#asterisk KriS|work (n=KriS@212.202.141.92) |
| 08:24.18 | KriS|work | good morning |
| 08:27.27 | *** join/#asterisk cleyverson (n=cleyvers@201.29.64.235) |
| 08:27.45 | *** part/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 08:27.49 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 08:29.29 | MadCamel | 'mornin |
| 08:30.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:30.47 | cleyverson | hello everbody |
| 08:31.04 | KriS83 | If someone dialed *1# How would this extension look like? exten => _*1#XXX? |
| 08:32.14 | *** join/#asterisk lorinc (n=ang@caracas-0847.adsl.interware.hu) |
| 08:32.34 | MadCamel | KriS83 pretty much (if I remember correctly), may have to fiddle with it a bit though. |
| 08:32.41 | *** join/#asterisk Nugget (i=nugget@dazed.slacker.com) |
| 08:32.54 | Qwell | KriS83: Will they be dialing anything after the #? |
| 08:33.17 | KriS83 | yes |
| 08:33.45 | KriS83 | What I want to do is when someone dials *1#number that the call goes out with no callerid |
| 08:34.16 | KriS83 | so then a Dial(Zap/g7/${EXTEN:3}) should work or not? |
| 08:34.20 | Qwell | sure |
| 08:35.44 | KriS83 | but the exten should be: exten => _*1#XXX. right? as the number to be dialed kann be longer than 3 digits? |
| 08:35.58 | KriS83 | kann = can |
| 08:36.46 | wellng | hi all |
| 08:36.56 | Qwell | KriS83: something like that, yeah |
| 08:37.04 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:37.40 | wellng | tzafrir_laptop, are u there ? i am welles |
| 08:37.58 | tzafrir_laptop | here |
| 08:38.09 | KriS83 | One last question though, where do I find exacly what _ X . etc means? What would I have to search for on voip-info? |
| 08:38.20 | wellng | tzafrir_laptop, i have problems. |
| 08:39.28 | wellng | tzafrir_laptop, now i want to call to pstn using iaxclient through SIP. when answered the callee can hear me ,but i can not hear the callee |
| 08:40.23 | tzafrir_laptop | wellng, is this what you want, or is this what you get? |
| 08:40.44 | wellng | tzafrir_laptop, it is what i get |
| 08:41.34 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:41.39 | tzafrir_laptop | iaxclient through SIP? IAX or SIP? |
| 08:42.03 | tzafrir_laptop | Is there any NAT between you and asterisk? |
| 08:42.10 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:43.04 | puzzled | morning |
| 08:43.36 | *** join/#asterisk welles (n=welles@222.90.199.58) |
| 08:50.57 | af_ | anyone bought something frmo myphonecall (uk)? |
| 08:54.44 | *** join/#asterisk Student (n=Atesliii@85.106.171.225) |
| 08:57.25 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 09:00.13 | nextime | anyone using chan_woomera? |
| 09:00.19 | *** join/#asterisk wellng (n=welles@222.90.199.58) |
| 09:02.11 | *** join/#asterisk strecher (n=123687@calderdale.ac.uk) |
| 09:02.19 | wellng | tzafrir_laptop, i come back |
| 09:03.16 | *** join/#asterisk JMcA (n=jmcadams@pixout.appriss.com) |
| 09:03.31 | tzafrir_laptop | wellng, did you see my questions? |
| 09:03.59 | wellng | tzafrir_laptop, no. i just disconnect |
| 09:04.17 | tzafrir_laptop | Do you connect through IAX or through SIP? |
| 09:04.34 | wellng | tzafrir_laptop, through SIP |
| 09:06.22 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 09:06.57 | wellng | tzafrir_laptop, i found a strange thing. for example i dial through 1234 but the asterisk show dial(IAX2/2345-3,SIP/99999@context). in dial string i think it should be 1234 while not 2345 |
| 09:07.09 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 09:07.39 | wellng | tzafrir_laptop, why it change? |
| 09:07.54 | tzafrir_laptop | wellng, do you connect to asterisk (from the sip client) through NAT? |
| 09:08.56 | tzafrir_laptop | or is there any firewall in the way? It seems that RTP packets in one directions don't get through |
| 09:09.29 | many | dwooop. |
| 09:09.37 | *** join/#asterisk welles (n=welles@222.90.199.58) |
| 09:09.42 | tzafrir_laptop | anyway, I got to go now |
| 09:09.52 | *** join/#asterisk VG (n=vijay@220.227.161.180) |
| 09:10.00 | VG | HEllo guys |
| 09:10.28 | jebba | anyone using lcdial() ? It appears the SQL statement it uses for querying the db is foo. |
| 09:10.40 | VG | i need to change the dialing port to 5070 |
| 09:11.24 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 09:13.10 | *** part/#asterisk wellng (n=welles@222.90.199.58) |
| 09:13.19 | *** join/#asterisk wellng (n=welles@222.90.199.58) |
| 09:13.37 | VG | any idea where all do i need to configure the same |
| 09:15.03 | VG | anyone there |
| 09:15.20 | iDunno | nope. |
| 09:15.22 | wellng | tzafrir_laptop, sorry i network is so bad |
| 09:15.25 | iDunno | no one at all :) |
| 09:15.46 | wellng | tzafrir_laptop, then how to configure the nat? |
| 09:16.50 | tzafrir_laptop | welles, basically, start with nat=yes in the peer settings. Also read about nat on voip-info . There should be a link to that in the page of the sip.conf file. Sorry, got to go now |
| 09:17.20 | VG | can anyone tell me where all do i need to change the configuration of the port number |
| 09:17.23 | wellng | tzafrir_laptop, ok thanks |
| 09:17.37 | VG | like default is 5060, i need to change that to 5070 |
| 09:17.40 | backblue | elvisthedj: are you using realtime? dont use it. |
| 09:17.47 | VG | where all do i need to edit it |
| 09:17.48 | backblue | because it has some nat issues |
| 09:18.17 | VG | hi backblue |
| 09:18.33 | backblue | hi vg |
| 09:18.33 | VG | can you tell me how can i start dialing through 5070 port |
| 09:18.36 | *** join/#asterisk gowmonster (i=gowmonst@ruby.colloquium.co.uk) |
| 09:18.45 | VG | i need to change the port to 5070 from 5060 |
| 09:18.59 | VG | where all do i need to edit it other than sip.comf |
| 09:19.02 | VG | *sip.conf |
| 09:19.10 | backblue | hum i'm not shor, let me check |
| 09:19.13 | tzafrir_laptop | VG, you'd get a better respone if you state the softwares involved |
| 09:19.29 | backblue | [general] |
| 09:19.29 | backblue | bindport=5060 ; UDP Port to bind to (SIP standard port: 5060) |
| 09:19.33 | backblue | this? |
| 09:20.06 | backblue | VG: if you dont use 5060 all the "world" will not call you. |
| 09:20.41 | VG | i need to only for outbound setup |
| 09:20.53 | VG | and my carrier supports only 5070 |
| 09:21.07 | VG | where else do i need to change this |
| 09:21.11 | gowmonster | any1 came across a problem with g.729 codec and bristuff-0.2.0-RC8q? |
| 09:21.11 | iDunno | so set the port in sip.conf |
| 09:21.13 | VG | because even after doing this |
| 09:21.21 | VG | and sip reload |
| 09:21.23 | backblue | VG: firewall? |
| 09:21.36 | VG | i can see the number getting dialled through 5060 in sip debug |
| 09:21.42 | VG | and its not getting connected |
| 09:21.51 | VG | i am dialing using agi |
| 09:22.40 | iDunno | you've got "port=" set in the sip.conf for the 'outgoing' connection, right? |
| 09:22.41 | VG | i simply need it for outbound calling |
| 09:22.48 | VG | yes |
| 09:23.39 | iDunno | hmm |
| 09:26.29 | VG | any idea where all do i need to change this setting |
| 09:27.52 | *** join/#asterisk _SwM_ (n=admin@digitaldatabits.net) |
| 09:31.00 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 09:31.05 | VG | anyone? |
| 09:31.48 | backblue | VG: everyone calling from your * should go to 5070 insted of 5060? |
| 09:32.31 | iDunno | nope, he needs to contact a SIP provider on 5070 |
| 09:32.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:33.04 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 09:33.49 | gowmonster | VG, does your firewall allow traffic both ways on 5070? |
| 09:35.44 | backblue | VG: you can allways do that in firewall. |
| 09:35.47 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 09:36.09 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 09:36.20 | backblue | i dont know if "register => ..." has some port syntax but try :port after hostname. |
| 09:38.46 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
| 09:43.17 | *** join/#asterisk crich1999 (n=crich@p54BFA8CE.dip0.t-ipconnect.de) |
| 09:43.56 | *** join/#asterisk welles (n=welles@61.150.60.159) |
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| 09:50.25 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:50.41 | littleball | hello, i am trying to configure the realtime thing |
| 09:51.16 | frenzy | hey all |
| 09:51.21 | frenzy | what does this mean.. channel.c:2046 ast_indicate: Unable to handle indication 3 - |
| 09:51.26 | clive- | howzit frenzy |
| 09:51.30 | littleball | for the asterisk side, should i only need to change /etc/asterisk/res_odbc.conf and /etc/asterisk/extconfig.conf |
| 09:51.31 | littleball | right |
| 09:51.31 | littleball | ? |
| 09:51.49 | frenzy | clive-: not bad n u |
| 09:52.47 | clive- | good thnks |
| 09:53.31 | frenzy | what does channel.c:2046 ast_indicate: Unable to handle indication 3 mean? |
| 09:53.44 | frenzy | WARNING[8573]: channel.c:2046 ast_indicate: Unable to handle indication 3 |
| 09:55.18 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:56.19 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:59.19 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:04.33 | backblue | littleball: do you will need nat? |
| 10:04.51 | *** join/#asterisk Koenvi (n=kova@tech.ascom.be) |
| 10:06.19 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:07.35 | littleball | backblue |
| 10:07.45 | littleball | Realtime mapping for 'extensions' found to engine 'psqlodbc', but the engine is not available |
| 10:07.52 | littleball | i got this error from cli console |
| 10:08.01 | littleball | s |
| 10:08.35 | littleball | i am confused by the parameters in the asterisk configuration |
| 10:08.52 | littleball | /etc/asterisk/res_...conf and /etc/asterisk/extconfig.conf |
| 10:08.54 | Koenvi | chan_zap.c: Call specified, but not found? |
| 10:09.05 | Koenvi | chan_zap.c: Hangup on bad channel 1/1 on span 1 |
| 10:11.01 | Koenvi | I keep getting these messages... |
| 10:11.21 | nextime | anyone using chan_woomera & g729A? |
| 10:11.40 | Koenvi | found some info in Digium's forum that this is solved some time ago. |
| 10:12.14 | Koenvi | Now I'm using the svn trunk and still the same messages :-( |
| 10:14.56 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
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| 10:26.24 | cypromis | /w 14 |
| 10:26.32 | cypromis | :) |
| 10:27.17 | clive- | cypromosis...hi |
| 10:28.54 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 10:30.49 | RoyK | cypromis: morning |
| 10:31.16 | cypromis | morn |
| 10:31.19 | cypromis | :) |
| 10:31.47 | RoyK | cypromis: any idea why my nice IBM 346 dual xeon64 box uses LOTS of system cpu on only 25 concurrent calls? |
| 10:31.51 | RoyK | echocancel=64 |
| 10:32.02 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:32.04 | RoyK | cpu load ~15% on a dual xeon.... |
| 10:32.37 | gowmonster | is it using mpg123 or inbuilt music on hold? |
| 10:32.43 | cypromis | strange |
| 10:33.07 | RoyK | gowmonster: no, it's a plain zap/iax2 gateway |
| 10:33.11 | RoyK | no transcoding |
| 10:33.13 | RoyK | no nothing |
| 10:33.23 | gowmonster | that is strange |
| 10:33.34 | *** join/#asterisk fulgas (n=fulgas@209.8.233.244) |
| 10:33.57 | RoyK | it is |
| 10:34.03 | RoyK | i've got two A104s in that box |
| 10:34.10 | RoyK | only two PRIs are active atm |
| 10:34.20 | gowmonster | how much memory is it using |
| 10:34.28 | RoyK | memory? lol |
| 10:34.31 | XIN01OZ | sup all Im curious as to what are my options having a T3 as far as wanting to even split usage for voice and bandwidth |
| 10:34.45 | XIN01OZ | found one card |
| 10:34.47 | RoyK | not much |
| 10:34.55 | *** join/#asterisk Medvekoma (i=bear@funyiro.webpress.hu) |
| 10:34.56 | XIN01OZ | oh yeah |
| 10:35.02 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:35.03 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 10:35.18 | XIN01OZ | can it happen at all? |
| 10:35.24 | Koenvi | I'm using a TE110P but getting following error: |
| 10:35.41 | gowmonster | just in case it was using cycles to do swap etc |
| 10:35.50 | RoyK | XIN01OZ: T3? why? |
| 10:36.31 | Koenvi | chan_zap.c: Call specified, but not found? |
| 10:36.45 | XIN01OZ | because I would really like to have one T3 installed in Taiwan and use it for both voice and the net connection for VOIP |
| 10:36.52 | JMcA | at the T3 level, you might be better off splitting it up with an adtran or something |
| 10:37.46 | *** join/#asterisk NoRemorse (n=bah@202.161.68.6) |
| 10:37.47 | XIN01OZ | stuff i cheap over there |
| 10:37.50 | XIN01OZ | is |
| 10:38.09 | *** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net) |
| 10:38.11 | XIN01OZ | im think a t3 might only be like 500 a month |
| 10:38.20 | NoRemorse | hi all, if you ever have a prob where one party cant hear the other but it works the other way, check your firewall has the full rtp port range cleared hehe |
| 10:38.25 | JMcA | XIN01OZ: nice |
| 10:39.01 | XIN01OZ | thanks JMcA im checking int oit |
| 10:39.49 | NoRemorse | does anyone know anything about dtmf payloads? my provider claims I need to set it to 99 but my client hardware wont allow 98 or 99 |
| 10:42.24 | *** join/#asterisk MassiveBlue (n=Massive@80.243.63.82) |
| 10:43.59 | *** join/#asterisk nettie (i=esivieri@85-18-54-38.ip.fastwebnet.it) |
| 10:44.34 | nettie | Hi guys, anyone has problems with asterisk and linksys pap2 call fwd please? |
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| 10:52.33 | tasat | anyone familiar with out the various answering machine detection apps work? |
| 10:52.42 | tasat | familiar with how |
| 10:53.07 | JMcA | I know basically what our own in-house app does...and its not pretty |
| 10:53.19 | tasat | what do you do? |
| 10:54.07 | JMcA | we listen for a brief bit of audio followed by silence (or very low audio, anyway) and assume that a pattern like that is a person answering the phone with a quick greeting...otherwise, we assume answering machine |
| 10:54.39 | tasat | that works? |
| 10:54.49 | JMcA | reasonably well...its hardly perfect |
| 10:55.21 | tasat | I saw a couple of apps -- machinedetect... and backgrounddetect |
| 10:55.25 | tasat | are you familiar with those? |
| 10:55.34 | JMcA | nope |
| 10:56.00 | JMcA | I only know marginally what our own in-house stuff does...I'm not a developer on it, so I haven't explored the knowledge-space much at all |
| 10:56.27 | tasat | FYI |
| 10:56.28 | tasat | http://www.voip-info.org/wiki/index.php?page=Asterisk%20cmd%20MachineDetect%20(addon) |
| 10:59.00 | *** join/#asterisk BlueMassive (n=Massive@80.243.63.82) |
| 10:59.30 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 11:02.45 | *** join/#asterisk blop (n=blop@213-193-176-151.adsl.easynet.be) |
| 11:04.34 | blop | hum :) |
| 11:05.35 | *** join/#asterisk NghtLght (n=ircproxy@202.179.3.171) |
| 11:11.54 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 11:14.55 | *** join/#asterisk uneuronh (n=umesh@61.95.206.107) |
| 11:15.45 | *** part/#asterisk gowmonster (i=gowmonst@ruby.colloquium.co.uk) |
| 11:16.07 | uneuronh | any one from MPA netowrk solutions , melbourne , australia |
| 11:17.31 | JamesDotCom | hahaha uneuronh |
| 11:17.35 | JamesDotCom | i once did some work for them |
| 11:17.42 | JamesDotCom | dodgy bastards |
| 11:20.08 | uneuronh | hmm ok james u remeber justin there |
| 11:21.26 | uneuronh | JamesDotCom, any idea are they still around in asterisk , caz i did a biiling application for asterisk , wondering if the billing system is still in need so i can work on it and make it open source |
| 11:24.31 | kippi | hi |
| 11:25.01 | kippi | I have a HandyTone-486 but it does not want to connect to my asterisk box, any ideas why? |
| 11:27.16 | *** join/#asterisk mcn (n=mcn@ext-gw.newtoncomputing.co.uk) |
| 11:35.51 | *** join/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net) |
| 11:35.55 | Reverend | hi all. |
| 11:36.46 | Reverend | i got an issue with voicepulse i think... calls in and out sound very bad. alot of pops and clicks. |
| 11:37.02 | Reverend | it's been fine for a while... all of a sudden today it's bad |
| 11:38.16 | syle | i don;t think anyone cares about voicepulse, email them |
| 11:39.08 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:40.05 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 11:45.35 | *** join/#asterisk mmmToop (n=chatzill@196.31.11.194) |
| 11:45.45 | Reverend | so what are the differences between 1.2.1 and 1.2.2 ? |
| 11:46.06 | clive- | toop..:) |
| 11:46.17 | mmmToop | hey bud! |
| 11:46.25 | puzzled | Reverend: how about reading a ChangeLog? |
| 11:46.29 | clive- | how u doing |
| 11:46.32 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 11:46.39 | Reverend | puzzled that's what i'm looking for.... |
| 11:46.52 | mmmToop | all good...how is the calling card bus.? |
| 11:47.21 | puzzled | Reverend: if you subscribe to the svn-commits list than you will always be up to date what changes were made |
| 11:47.38 | Reverend | found it http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.2 |
| 11:47.53 | iDunno | must. remove. pub. from. my. highlights. |
| 11:48.07 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 11:55.43 | *** join/#asterisk FireBlade (i=fire@da-box.net) |
| 11:57.12 | *** join/#asterisk Einar__ (n=einar@einsi.nepal.is) |
| 11:57.42 | FireBlade | hi, I have a Wavecom GSM device hooked up to my computer via a serial cable with Astersik deployed, i was wondering if i could accept calls from the GSM device and route them to the Asterisk PBX? i'd really appreciate some pointers |
| 11:58.29 | Einar__ | Anyone know a online-store where I can buy a Billion ISDN card? |
| 12:00.58 | *** join/#asterisk uneuronh (n=umesh@61.95.206.107) |
| 12:01.07 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 12:01.27 | franck | Hi all |
| 12:01.43 | franck | on the asterisk business edition what is this 128 phone limit? |
| 12:02.45 | franck | kram: ^^^ |
| 12:04.57 | *** join/#asterisk graabein (n=gunnar@bkkb-gw.bitcon.no) |
| 12:05.06 | Reverend | yay... upgrade succeeded... |
| 12:08.47 | tzafrir_laptop | franck, ask Digium support? |
| 12:09.32 | franck | tzafrir_laptop: yes good idea, thought somebody knew... |
| 12:10.06 | nextime | how can i do a stress test to verify how many channel an * server can reach without overloading? Is there any h323 call generator that i can use? |
| 12:11.00 | franck | yes, I'm interested too but with SIP |
| 12:12.00 | puzzled | nextime, frankck: there is an app called sipsak and (iirc) sipp that you can use for load testing but afaik it's sip only |
| 12:13.45 | nextime | ok tnx, i can do stress test with sipsak + yate as sip->h323 proxy :) |
| 12:15.08 | puzzled | Reverend: enjoy :) |
| 12:16.31 | puzzled | nextime, franck: you can also search the archives for a stress testing script that someone posted. was quite a long time already so dig deep |
| 12:17.55 | *** part/#asterisk UlbabraB (n=salama@host205-72.pool8174.interbusiness.it) |
| 12:17.57 | *** join/#asterisk redax (n=redax@r6.hu) |
| 12:18.03 | redax | hi,m |
| 12:18.06 | redax | -m |
| 12:19.04 | redax | which parameter controlls the minimal microphone volume, where a asterisk silences the voices ? |
| 12:20.05 | *** join/#asterisk pengyong (n=lala@218.93.145.142) |
| 12:20.13 | redax | like calling from Zap->SIP |
| 12:20.22 | redax | please help. |
| 12:20.51 | RoyK | asterisk doesn't support rfc3389 confort noise/silence suppression |
| 12:20.58 | RoyK | so there are no such parameters |
| 12:21.05 | franck | puzzled: thanks |
| 12:21.16 | redax | hm. |
| 12:21.34 | redax | then Xten-light should do the noise/silence stuff on the SIP channel? |
| 12:21.48 | franck | about the application background, if I hangup on a ZAP interface while background is running, should * hang up or wait till Bacground is completed? |
| 12:22.18 | RoyK | redax: no. it should not |
| 12:22.40 | franck | redax: you must disable this function in xten |
| 12:23.02 | RoyK | redax: since asterisk is built so incredibly smart that it requires an incoming RTP frame before transmitting one out |
| 12:23.32 | RoyK | meaning silence suppression in one direction will block the other |
| 12:23.51 | puzzled | RoyK: unless you use patch #5374 |
| 12:25.13 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:26.08 | *** join/#asterisk mazza[W] (i=mazzanet@unaffiliated/mazzanet) |
| 12:26.14 | mazza[W] | oh herro |
| 12:26.27 | mazza[W] | what's the console command to nuke a sip channel? |
| 12:26.49 | RoyK | puzzled: ah |
| 12:26.59 | RoyK | mazza[W]: soft hangup? |
| 12:26.59 | redax | then my blSy ISDN phoneshould do noise suppression? |
| 12:27.15 | mazza[W] | yes |
| 12:27.20 | RoyK | noise suppression :) |
| 12:27.32 | redax | then silence :) |
| 12:27.38 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:28.04 | redax | but somebody does |
| 12:28.22 | redax | whoa. |
| 12:29.05 | redax | rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.1.11 |
| 12:29.11 | *** join/#asterisk [bofh42] (n=bofh42@p54828A38.dip0.t-ipconnect.de) |
| 12:29.19 | redax | so Xten Light the one who doing noise suppression. |
| 12:33.59 | redax | yeah. you can configure in Advanced settings: |
| 12:34.01 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 12:34.07 | redax | Transmit Silence: default no |
| 12:34.19 | redax | Thanks for your help. |
| 12:36.57 | redax | another weirdness... the ISDN phone puts a leading zero to the callerID when calling from SIP. like "0test_sip" |
| 12:37.23 | RoyK | bristuff? |
| 12:37.25 | RoyK | or zaptel? |
| 12:37.27 | RoyK | or what? |
| 12:37.31 | redax | bristuff |
| 12:37.37 | RoyK | nationalprefix= |
| 12:37.40 | redax | zaphfc in the box |
| 12:37.48 | RoyK | instead of nationalprefix=0 which is default |
| 12:38.17 | redax | to the zapata.conf ? |
| 12:38.23 | RoyK | yeah |
| 12:38.26 | RoyK | i think... |
| 12:40.19 | redax | hm. not working |
| 12:45.15 | XIN01OZ | usaid "T3? Why?" were u suggesting that I use something better such as TDMoIP? |
| 12:45.17 | XIN01OZ | heh |
| 12:45.41 | mazza[W] | RoyK: |
| 12:46.07 | redax | hm. still leading zero |
| 12:47.05 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:48.36 | RoyK | mazza[W]: |
| 12:48.41 | mazza[W] | any ideas? |
| 12:49.04 | RoyK | XIN01OZ: T3 is far more than what you can do with a single server |
| 12:49.07 | RoyK | also |
| 12:49.16 | Reverend | this is great: http://www.research.att.com/projects/tts/demo.html |
| 12:49.43 | Reverend | i <3 U.K. Audrey |
| 12:49.44 | RoyK | if doing a T3 on a single box and that box blows up, it hurts a lot more than if you'd distributed the load on several, say, quad T1 boxes |
| 12:50.31 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 12:53.10 | XIN01OZ | RoyK: have u ever tried http://actor.loquendo.com/actordemo/default.asp?language=en |
| 12:53.20 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 12:56.20 | RoyK | no. but i still wonder why you ask in particular....... |
| 12:56.39 | XIN01OZ | yeah I was referencing that to Reverened .. heh |
| 12:56.52 | redax | sorry found the leading zero. mea culpa that was my bad |
| 12:57.11 | XIN01OZ | thanks though I think the saved cost for doing TDMoIP would be worth setting up in a location that provides it |
| 12:58.16 | *** part/#asterisk uneuronh (n=umesh@61.95.206.107) |
| 12:58.52 | redax | very last question. if I use 2x zaphfc [1 for TE, 1 for NT] is it possible to dial in on the TE adapter using X75 or HDLC or v110 ? |
| 12:58.56 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 12:59.15 | redax | to reach the box somehow remotely ? |
| 12:59.16 | XIN01OZ | its not much to get Loquendo bound into Asterisk |
| 12:59.56 | *** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk) |
| 13:01.16 | Einar__ | <PROTECTED> |
| 13:01.57 | h3x | heh. hi redax |
| 13:02.03 | redax | heyy hex. :) |
| 13:02.29 | Cazper | Einar: www.komplett.no, dont know if they deliver to iceland |
| 13:02.31 | h3x | sup |
| 13:03.30 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:04.48 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 13:06.09 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 13:06.37 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 13:06.40 | Einar__ | Thanks Cazper. Its a bit pricy in Norway (around $44)... Think I'll wait for another billion ISDN auction to show up on ebay. Bought one there for $7 the other day ;) |
| 13:07.20 | fugitivo | morning people |
| 13:07.37 | tuxinator_linux | linville: Hey John, what's TuxDriver? |
| 13:10.37 | *** join/#asterisk coppice (n=chatzill@33.157.17.210.dyn.pacific.net.hk) |
| 13:10.59 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 13:11.53 | RoyK | morning, coppice |
| 13:11.56 | RoyK | or evening |
| 13:11.58 | tparcina | Hi everybody! |
| 13:12.00 | coppice | hi |
| 13:12.07 | coppice | good morrow |
| 13:12.12 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:12.46 | Reverend | XIN01OZ, that's a nice engine too |
| 13:13.11 | caio1982 | coppice: hi there! have you read that email from denis regarding a voip event in brazil? |
| 13:13.21 | coppice | yes |
| 13:14.05 | fulgas | sunshine and beach |
| 13:14.22 | mut | would a busy server lead to qualify replies getting 'lost'? |
| 13:14.36 | coppice | and nuts - but we have enough of those here |
| 13:14.43 | mut | cause the server * is on right now is overloaded and lately more people have been becoming unreachable |
| 13:14.50 | mut | even with a qualify of 3 seconds |
| 13:15.27 | mut | i have a new dual 3ghz system coming for it but i'm wondering if the problem might be elsewhere |
| 13:16.28 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:17.52 | fugitivo | mut: why is it overloaded? |
| 13:18.33 | mut | fugitivo: it's a spam filter gateway |
| 13:18.53 | fugitivo | you have asterisk in a spam filter gateway? :) |
| 13:18.53 | mut | and does lot of stats gathering with cron |
| 13:19.03 | mut | yep |
| 13:19.30 | mut | will have it's own server come friday/monday or so |
| 13:19.42 | mut | depends when it;s shipped, should have already been here |
| 13:21.15 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:22.29 | johnnyb | If I use two TDM cards, should I stick them on the same interrupt or different ones? |
| 13:23.01 | clive- | jonnyb ifferent ones |
| 13:23.07 | Ariel_ | johnnyb, diffferent ones |
| 13:23.30 | Ariel_ | morning folks |
| 13:23.37 | fugitivo | morning Ariel |
| 13:24.11 | Ariel_ | fugitivo, como esta? Espero que esta bien... |
| 13:24.36 | fugitivo | muy bien, y usted? |
| 13:25.20 | Ariel_ | bien.. Trying to a system back up after an upgrade... 1.2.2 has issues.... with a dial plan from a 1.0.9 |
| 13:25.25 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 13:27.46 | [TK]D-Fender | Ariel_ : Enough things changed just getting to 1.2 |
| 13:27.53 | [TK]D-Fender | Ariel_ : How bad? |
| 13:28.17 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-248.claranet.co.uk) |
| 13:28.26 | johnnyb | Is it bad for a TDM card to share an interrupt with eth0 |
| 13:28.27 | johnnyb | ? |
| 13:28.35 | Ariel_ | [TK]D-Fender, don't know yet. I was called in this morning to see if I can fix it for my ex employer after there IS person upgraded. |
| 13:28.47 | johnnyb | I'm trying to find an appropriate slot. |
| 13:28.49 | Ariel_ | johnnyb, yes very bad |
| 13:28.51 | tuxinator_linux | jonathh: yes |
| 13:28.55 | [TK]D-Fender | johnnyb : It shouldn't share with ANYTHING PERIOD |
| 13:29.09 | johnnyb | Is it possible to tell beforehand which PCI slot has which interrupt? |
| 13:29.20 | [TK]D-Fender | Ariel_ : I might take apeek if you want |
| 13:29.31 | Ariel_ | johnnyb, it might be better to sell both cards and get yourself the new TDM2400 series board. |
| 13:29.33 | tuxinator_linux | johnnyb: motherboard manual |
| 13:29.54 | johnnyb | Ariel_: Where do you get those? |
| 13:29.59 | johnnyb | And how much? |
| 13:30.07 | [TK]D-Fender | johnnyb : Just shut it down, go into your BIO and try to move things out of its way. Then start disabling all the non-essentials until they are alone. |
| 13:30.17 | clive- | ariel, lol,,,,depends if he only need a few ports:),...I would try it and see if it works, otherwise use apic which might help |
| 13:30.21 | Ariel_ | [TK]D-Fender, no I would like for them to feel a little pain first. I have most of it fixed already just need to finish some macro's I did along time ago for them. |
| 13:30.32 | Ariel_ | [TK]D-Fender, thanks for the offer. |
| 13:30.35 | [TK]D-Fender | np |
| 13:31.12 | Ariel_ | tdm400p boards don't like apic much either |
| 13:31.29 | johnnyb | [TK]D-Fender: Thanks |
| 13:31.29 | Ariel_ | clive-, he has 2 boards.... |
| 13:32.52 | [TK]D-Fender | johnnyb : If you can hold your breath a little long there is the A200....... |
| 13:33.11 | fugitivo | 1.2.2 is out? |
| 13:33.27 | [TK]D-Fender | fugitivo : in FTP, the rest to follow today |
| 13:33.33 | clive- | a200?? |
| 13:33.39 | Ariel_ | the new sagoma should be shipping in the next weeks or so... |
| 13:33.58 | fugitivo | upgrade or not upgrade? |
| 13:34.04 | fugitivo | does 1.2.2 fix serious issues? |
| 13:34.17 | fugitivo | where is the changelog? |
| 13:34.28 | Ariel_ | hummm the cvs list has lots of fixes they have been doing for some time now. |
| 13:34.31 | clive- | ariel sangoma introducing an analogue card? |
| 13:34.32 | Reverend | http://ftp.digium.com/pub/asterisk/ChangeLog-1.2.2 |
| 13:34.39 | Ariel_ | clive-, yes |
| 13:34.40 | fugitivo | Reverend: thanks |
| 13:34.57 | Reverend | i upgraded asterisk@home with it this morning and i seem to have no issues |
| 13:35.17 | clive- | Ariel, cool the more options the better....:) |
| 13:35.33 | Ariel_ | Reverend, which aah version |
| 13:35.42 | Reverend | Ariel_, the latest ? |
| 13:35.52 | fugitivo | the problem upgrading asterisk, is that there's a big list of patches + addons to apply... |
| 13:35.53 | [TK]D-Fender | clive- : And they play nice with interrupts, pci slots, and with their new DSP's, *0* echo. |
| 13:35.55 | Ariel_ | 2.2 is ok it has the changes already |
| 13:37.09 | clive- | Fender, pity they dont makle a BRI card:) |
| 13:37.14 | fugitivo | i see a lot of mem leak and backport fixes in the changelog |
| 13:37.40 | Ariel_ | there is a clone for the TDM400p board. Seems that people are now starting to use it.. |
| 13:38.00 | fugitivo | Ariel_: i asked the guy that makes them to be a reseller |
| 13:38.07 | [TK]D-Fender | Ariel_ : Considering Digium didn't design the TDM400, just distributed it.... |
| 13:38.17 | fugitivo | it's not really a clone |
| 13:38.23 | fugitivo | it's the tdm400 original |
| 13:38.41 | fugitivo | digium just got that card, changed the name and distributed it |
| 13:38.43 | Ariel_ | I just saw it. It even has the same numbers that digium has on theres. Seems they might be the ones that make it orginally for them. |
| 13:39.09 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 13:39.48 | fugitivo | Ariel_: are you using the tdm2400? |
| 13:40.19 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 13:40.25 | Ariel_ | fugitivo, no but a few of my customers are.. I need it for this office there using 2 tdm400 and from time to time get echo from Bellsouth lines. |
| 13:40.46 | fugitivo | any complains about the tdm2400?. |
| 13:41.08 | johnnyb | Does anyone know what the SMBus controller is, and if it can be disabled? |
| 13:41.40 | johnnyb | /proc/interrupts doesn't list anything as sharing my TDM IRQ, but lspci shows the "SMBus Controller" to be sharing it. |
| 13:41.49 | Ariel_ | fugitivo, yes I have a few. More of not knowning which ports to start at with the brakeout box. |
| 13:42.04 | *** join/#asterisk TK9 (n=Miranda@p54B2AC84.dip0.t-ipconnect.de) |
| 13:42.12 | Ariel_ | johnnyb, worrie more about extra usb's and things like that. |
| 13:42.34 | johnnyb | Yeah, I had to disable USB to get the first TDM card working. |
| 13:43.08 | fugitivo | usb and sata are a pain in the ass when playing with interrupts |
| 13:43.10 | Ariel_ | fugitivo, it's a very large card the TDM2400 so you need a normal case |
| 13:43.51 | fugitivo | Ariel_: any complain about echo, modules hanging or something like that? |
| 13:43.52 | kippi | hey |
| 13:43.56 | kippi | is anyone using Grandstream Handytone-486 ? |
| 13:44.21 | Ariel_ | fugitivo, not so far. But there new cards you have to give them time... |
| 13:44.51 | fugitivo | yes, i know |
| 13:44.54 | Ariel_ | kippi, are you having issue's with it? |
| 13:45.38 | Ariel_ | fugitivo, biggest problem is getting people to find at what ports it starts from the brakeout box like 24 or 1.. for the lines. |
| 13:46.08 | fugitivo | lol |
| 13:46.10 | Ariel_ | or getting people to configure the wires correctly on a 66 block patch. |
| 13:46.11 | fugitivo | that should be an issue |
| 13:46.38 | [TK]D-Fender | Its all amphenol... hardly rocket-science.... |
| 13:47.05 | Ariel_ | [TK]D-Fender, yes if you know them. you should see all the calls I get about them..... |
| 13:48.23 | RoyK | how can i compile zaptel so it can be profiled with oprofile? |
| 13:48.53 | coppice | amphenol make lots of connectors used in rockets :-\ |
| 13:49.08 | RoyK | 41969767 42.6346 zaptel (no symbols) |
| 13:49.12 | Ariel_ | hello coppice |
| 13:49.14 | RoyK | that's what opreport says.... |
| 13:49.31 | *** topic/#asterisk by drumkilla -> Asterisk 1.2.2 has been released! -//- http://www.asterisk.org/ |
| 13:49.53 | coppice | hi |
| 13:50.00 | mut | O_O |
| 13:50.08 | mut | who knows kevin at telnet? |
| 13:50.24 | iCEBrkr | Would you bite! The hand! That feeds you!! |
| 13:51.42 | RoyK | iCEBrkr: ? |
| 13:51.42 | [TK]D-Fender | :F |
| 13:51.52 | [TK]D-Fender | (CHOMP) |
| 13:51.56 | iCEBrkr | RoyK: I dunno why, but I got Nine Inch Nails stuck in my head. |
| 13:52.17 | iCEBrkr | TK9: Change your nick, punk.. :D |
| 13:52.26 | iCEBrkr | [TK]D-Fender: 'Morning |
| 13:52.29 | mut | someone in here... |
| 13:52.36 | iCEBrkr | mut: Nope.. |
| 13:52.38 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 13:52.41 | iCEBrkr | mut: No one here but us chickens. |
| 13:53.22 | fugitivo | and monkeys |
| 13:53.37 | iCEBrkr | AHh yes, I forgot about the monkeys |
| 13:53.45 | fugitivo | and weasels |
| 13:53.59 | iCEBrkr | lets not for get about BADGER BADGER BADGER BADGER!!! |
| 13:54.08 | jontow | muffin? |
| 13:54.53 | mut | ring ring ring banana phone |
| 13:55.46 | *** join/#asterisk basix (n=basix@202.68.154.221) |
| 13:56.03 | gambolputty | We luv da moon!!! |
| 13:56.17 | tuxinator_linux | goofy morning crowd |
| 14:00.21 | mut | ya |
| 14:00.21 | *** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) |
| 14:00.27 | mut | all the business professionals are on |
| 14:00.33 | mut | who else is up this early |
| 14:01.23 | fugitivo | well, that depends |
| 14:01.33 | basix | not everyone is in the same timezone |
| 14:01.48 | fugitivo | right |
| 14:05.25 | JMcA | stupid 4am maintenance windows |
| 14:05.48 | jontow | hehe, the boss was here at 4am testing the generator |
| 14:05.52 | jontow | getting a nasty storm today |
| 14:05.52 | *** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee) |
| 14:06.36 | [TK]D-Fender | ...| ./ ._ (crash) |
| 14:06.42 | fugitivo | nice ascii art! |
| 14:07.09 | JMcA | with way people are driving around here, you'd think this inch or so of snow was a global catastrophe |
| 14:07.19 | [TK]D-Fender | <- m@d l33t sk1llz! |
| 14:09.31 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 14:11.09 | jontow | we had 1/2" of freezing rain last night |
| 14:11.14 | jontow | wild stuff, that was :) |
| 14:11.27 | jontow | i thanked myself repeatedly for opting in on the studded snow tires all the way 'round in the jeep :P |
| 14:12.51 | SERGEUS | grrrrrr... -34 |
| 14:14.57 | basix | JMcA, why dont you guys use GNU/Linux? |
| 14:16.10 | blitzrage | ummm... I do |
| 14:16.30 | JMcA | basix: huh? |
| 14:16.38 | blitzrage | [TK]D-Fender: you like -- you have no skillz |
| 14:16.38 | basix | <JMcA> stupid 4am maintenance windows |
| 14:17.01 | JMcA | being on Linux or not doesn't affect when you take maintenance windows |
| 14:17.07 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 14:17.19 | caio1982 | tzafrir_laptop: should i update my .debs to depends on the new asterisk 1.2.2 or those from the spandsp bug are okay for checkings? |
| 14:17.20 | basix | JMcA, :| my bad. poor reading skillz :P |
| 14:17.57 | *** join/#asterisk Corydon-w (i=yellow@pdpc/supporter/sustaining/Corydon76-home) |
| 14:18.34 | tzafrir_laptop | caio1982, is there any binary compatability issue that requires this change? |
| 14:18.53 | tzafrir_laptop | I haven't built 1.2.2 yet |
| 14:19.23 | caio1982 | tzafrir_laptop: no, just thinking about an future update by pkg-voip |
| 14:20.28 | tzafrir_laptop | caio1982, I see no reason for an automatic update. I'd rather only force an upgrade if it is actually required |
| 14:21.00 | tzafrir_laptop | Can you try to test this? |
| 14:21.34 | caio1982 | to test specifically what? |
| 14:22.05 | tzafrir_laptop | If asterisk 1.2.2 will still load the current packages |
| 14:23.17 | *** join/#asterisk nothinman (i=shakey@adae86.neoplus.adsl.tpnet.pl) |
| 14:23.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.46 | caio1982 | tzafrir_laptop: i'm almost sure it will, i was just going to adjust it so that when new 1.2.2 packages appears in the archive, maybe my spandsp .debs could be uploaded together them matching the latest version |
| 14:24.00 | mut | anyone have problems with sipura 2002's admin config page not loading correctly? |
| 14:24.08 | mut | locally i can get to it |
| 14:24.14 | mut | but when i try to get to it from here |
| 14:24.21 | caio1982 | tzafrir_laptop: then i'd send to you my unicall and dependencies packages |
| 14:24.22 | mut | like half the page loads and then it dies |
| 14:24.34 | *** join/#asterisk MatsK (n=mk@3.80-203-81.nextgentel.com) |
| 14:24.46 | JMcA | mut: sounds suspiciously like an MTU issue |
| 14:24.58 | tzafrir_laptop | caio1982, is there any specific change that got into 1.2.2 that should have broken compatibility? |
| 14:25.18 | caio1982 | tzafrir_laptop: i dont think so |
| 14:25.23 | mut | JMcA: hm |
| 14:26.23 | caio1982 | tzafrir_laptop: actually i'm just waiting your 'ok' for my updated packages be uploaded ;) |
| 14:27.11 | RoyK | how can i compile zaptel so it can be profiled with oprofile? |
| 14:27.21 | Einar__ | Does anyone know how to configure a custom extension to dial an outbound route in AMP portal? |
| 14:29.38 | *** join/#asterisk fild (i=fdello@adae86.neoplus.adsl.tpnet.pl) |
| 14:29.57 | fugitivo | ~amp |
| 14:29.59 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
| 14:29.59 | *** join/#asterisk vladuk (n=vk@ip.82.144.198.127.stat-2.volia.net) |
| 14:30.06 | vladuk | HI All |
| 14:30.18 | mut | you're probly right JMcA |
| 14:30.21 | *** join/#asterisk shakey (i=shakey@adae86.neoplus.adsl.tpnet.pl) |
| 14:30.26 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 14:30.26 | mut | tried on another customer and it's the same |
| 14:30.32 | mut | same dslam |
| 14:31.00 | vladuk | Would like to discuss couple Asterisk DTMF processing bugs |
| 14:31.34 | ^Howler | vladuk: are there some? I've been playing with DTMF (kinda) lately.. |
| 14:31.50 | *** join/#asterisk troyb (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 14:31.52 | vladuk | 5970 and 6027 from mantis on bugs.digium.com |
| 14:35.19 | ^Howler | ah, ok..I'm not running into those. |
| 14:36.13 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:47.11 | blitzrage | vladuk: yah -- have you tested the patch on 5970 ? |
| 14:47.22 | blitzrage | I reported all the new info on those bugs |
| 14:48.26 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 14:48.44 | QbY | anyone have any suggestions for wholesale DID |
| 14:49.43 | fild | <PROTECTED> |
| 14:49.43 | fild | <PROTECTED> |
| 14:49.43 | fild | <PROTECTED> |
| 14:49.51 | brad_mssw | QbY: junctionnetworks is $2.00/mo ... vonworldwide is $.050/mo |
| 14:50.04 | brad_mssw | QbY: that's $0.50/mo for vonworldwide |
| 14:50.19 | mut | i can't find where on this dslam the mtu is even set |
| 14:50.20 | *** part/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 14:51.43 | [TK]D-Fender | mut : just try "ifconfig eth[#]: mtu 1452 |
| 14:53.54 | *** join/#asterisk razu (n=razu@adsl25957.estpak.ee) |
| 14:53.57 | *** join/#asterisk astoria (n=tom@user-7e5a43.user.msu.edu) |
| 14:54.42 | mut | on the dslam? |
| 14:55.08 | [TK]D-Fender | mut on your gateway |
| 14:55.12 | shakey | guys, how many simultaneous connections can asterisk handle, on let's say xeon 3.2 box? |
| 14:55.34 | [TK]D-Fender | shakey : depends on transcoding / codec / hardware, etc... |
| 14:55.44 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 14:55.44 | [TK]D-Fender | shakey : What do yuo NEED it to suupport? |
| 14:56.22 | iCEBrkr | <VOICE TYPE="doctor evil">1 MILLION CALLS</VOICE> |
| 14:56.50 | RoyK | iCEBrkr: ? |
| 14:57.22 | SERGEUS | shakey, http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations |
| 14:57.42 | SERGEUS | shakey, http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 14:58.05 | shakey | [TK]D-Fender: well, codec doesn't change much, as usually codecs are consuming between 15 and 20 MIPS |
| 14:58.29 | shakey | [TK]D-Fender: but what do you mean saying "transcoding"? |
| 14:58.30 | [TK]D-Fender | shakey .... sure it matters... |
| 14:58.57 | [TK]D-Fender | shaky convering one compressed codec into another real-time for multiple channels add up. |
| 14:59.06 | mut | it already is set on the gateway |
| 14:59.14 | shakey | yea, I know that. |
| 14:59.34 | shakey | so doing this on the fly is the most time-consuming thing? |
| 15:00.10 | [TK]D-Fender | Its cumulative, tahts all. So what do you NEED it to support? |
| 15:00.17 | shakey | what about this "transcoding"? is it decoding and then encoding? |
| 15:00.21 | blitzrage | transcoding is probably one of the most, if not *the* most important factor in determining how many simultaneous calls an Asterisk box can handle |
| 15:00.26 | blitzrage | shakey: yes |
| 15:00.32 | shakey | [TK]D-Fender: 1000 calls at the same time |
| 15:00.47 | blitzrage | you will definately need multiple boxes |
| 15:00.47 | zoa | i'd also say the software is important, without software its close to zero |
| 15:00.50 | blitzrage | like... 10 boxes |
| 15:00.51 | *** join/#asterisk pointer (i=pointer@aj.catt.com) |
| 15:01.03 | blitzrage | 100 calls per dual-xeon is a safe number |
| 15:01.08 | shakey | with what inside? |
| 15:01.22 | blitzrage | maybe even 150 |
| 15:01.22 | [TK]D-Fender | shakey : decent RAM I'd say. |
| 15:01.25 | shakey | 500 for 4x xeon 3.2? |
| 15:01.33 | zoa | no no |
| 15:01.34 | blitzrage | no |
| 15:01.36 | zoa | dont go 500 |
| 15:01.39 | shakey | ram is not a problem, this 16GB should be enough ;) |
| 15:01.41 | zoa | what do you want to do anyway |
| 15:01.55 | iCEBrkr | Lovely |
| 15:02.01 | blitzrage | RAM isn't really that big a factor once the code is compiled and loaded... its mostly CPU |
| 15:02.02 | iCEBrkr | Our T1 just took a shit |
| 15:02.04 | iCEBrkr | no phones! |
| 15:02.08 | iCEBrkr | OH NOEZ! |
| 15:02.13 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 15:02.18 | blitzrage | iCEBrkr: then stop chatting on IRC and go fix it :) |
| 15:02.24 | iCEBrkr | Not my deal man |
| 15:02.26 | iCEBrkr | I'm a programmer |
| 15:02.26 | iCEBrkr | :P |
| 15:02.28 | blitzrage | niiiiice |
| 15:02.39 | mut | anyone ever conf'd a coppermountain dslam? |
| 15:02.43 | *** join/#asterisk XGhosT (i=InFeRnO@200.138.89.108) |
| 15:03.06 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:03.12 | *** part/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 15:03.15 | zoa | yo yo |
| 15:03.36 | shakey | hm, I want basically to create some big sip gateway |
| 15:03.41 | shakey | i can go for cisco... |
| 15:03.48 | shakey | or use asterisk |
| 15:03.50 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:03.57 | XGhosT | Hi there.... i need to buy ip phone!!! |
| 15:03.59 | shakey | and I wonder what to do... |
| 15:03.59 | blitzrage | Asterisk scales by using more boxes |
| 15:04.09 | shakey | exactly. |
| 15:04.21 | shakey | cisco by putting more $$$ in their pocket |
| 15:04.29 | blitzrage | thats kind of the point |
| 15:05.05 | *** join/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net) |
| 15:05.14 | blitzrage | although if you have more money than time and support and learning, then Cisco can be a perfectly valid choice |
| 15:05.15 | shakey | and for the price of cisco I can buy a really nice big-boy-server |
| 15:05.47 | zoa | shakey |
| 15:05.52 | zoa | there a other things you can do |
| 15:05.59 | Katty | also, morning. |
| 15:06.01 | zoa | i play big sip gatewau |
| 15:06.01 | mikeyb_work | I am using a few grandstream phones, all their configs are identical, but some are giving me garbled recordings... can latency cause this? can codec choice cause this? any troubleshooting suggestions? |
| 15:06.03 | blitzrage | you won't scale Asterisk using one huge server -- you'll need to use multiple boxes |
| 15:06.08 | zoa | with 1000$ machines |
| 15:06.13 | mikeyb_work | phones are all bt100 |
| 15:07.03 | shakey | blitzrage: well, still not a problem. saying "big" I meant 4x xeon 3.2 or 8x xeon 3.2 |
| 15:07.46 | kippi | Ariel_: yes I am having a problem with my Grandstream box |
| 15:07.55 | shakey | I can calculate how much cpu time codecs will use, but is there anything else which will concume cpu time? |
| 15:08.11 | QbY | So, I want to provide VoIP service to my little city. I need local numbers, and prices that won't break me.. So I can offer Unlimtied.. Anyone know of a good write-up concerning this.. |
| 15:08.12 | blitzrage | well... one Asterisk box, no matter how big seems to be limited to around 250 sim calls |
| 15:08.17 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:08.32 | blitzrage | zoa: would you agree with that? you're the man :) |
| 15:08.40 | shakey | blitzrage: why do you say that? |
| 15:08.43 | zoa | i for sure wouldnt do more than 250 on a machine |
| 15:08.47 | zoa | you could do more |
| 15:08.48 | zoa | for sure |
| 15:08.53 | zoa | but asterisk will not like it |
| 15:09.04 | shakey | because of...? |
| 15:09.05 | [TK]D-Fender | Katty: mew. |
| 15:09.15 | fugitivo | that sucks, we should move to yate :/ |
| 15:09.33 | zoa | it all depends on what you are doing |
| 15:09.45 | zoa | if you are just doing voip passthrough of the rtp packets |
| 15:10.01 | zoa | you could do maximum 400 or so on a dual xeon |
| 15:10.14 | zoa | i might be able to tweak it for you to get it to 2000 |
| 15:10.24 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:10.26 | fugitivo | hold on |
| 15:10.27 | Katty | erro, mister fender. |
| 15:10.30 | fugitivo | from 400 to 2000? |
| 15:10.31 | zoa | moving to yate wouldnt help |
| 15:10.39 | shakey | four xeons 3.2GHz |
| 15:10.47 | zoa | but that would take a few months to do |
| 15:11.07 | fugitivo | is that tweaking or rewriting code? |
| 15:11.11 | zoa | rewriting |
| 15:11.37 | shakey | hm.. |
| 15:11.42 | fugitivo | why isn't that rewrite on asterisk source already? :) |
| 15:11.54 | zoa | shakey, what is cheaper, 4 times a single xeon or a quad xeon ? |
| 15:12.04 | shakey | fugitivo: there must be some way to get some cash ;) |
| 15:12.07 | [TK]D-Fender | QbY : just use SER and use * for billing / VM / services |
| 15:12.08 | zoa | and what is faster, 4 times a single xeon or one times a quad xeon |
| 15:12.19 | zoa | pick the right one :) |
| 15:12.31 | QbY | TK. And for my DIDs? |
| 15:12.41 | blitzrage | 4 boxes! :) |
| 15:12.43 | QbY | in Podunk, SC |
| 15:12.44 | [TK]D-Fender | QbY : thats another matter |
| 15:12.54 | shakey | zoa: well, you think it's 4 times xeon rather that quad xeon |
| 15:13.00 | shakey | but don't really know why |
| 15:13.20 | blitzrage | because of the code |
| 15:13.27 | zoa | because a dual xeon is like 1,5 single xeons |
| 15:13.40 | zoa | while a dual opteron would be like 1,8 single opterons or so |
| 15:13.55 | zoa | and the more processors you add, the worse it gets |
| 15:14.10 | eKo1 | is that so... |
| 15:14.16 | fugitivo | i'm curious about the rewrite to get 2000 calls |
| 15:14.25 | zoa | and when you look at how asterisk works, it will wait for things more often if you have more calls |
| 15:14.28 | zoa | slowing things down |
| 15:14.31 | shakey | zoa: have you tested it? |
| 15:14.34 | coppice | as soon as you go beyond dual, hardware gets mighty pricey even if you can get the performance to scale |
| 15:14.43 | zoa | true coppice |
| 15:14.51 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 15:14.55 | zoa | shakey: read on scalability on those cpu's |
| 15:14.57 | coppice | mainstream == cheapo |
| 15:15.11 | zoa | asterisk is not written for handling 1000 calls on a processor |
| 15:15.11 | shakey | zoa, yeah, that may be the point |
| 15:15.13 | zoa | to many lists |
| 15:15.25 | shakey | so how many then? |
| 15:15.29 | trixter | ABE is for 120 calls, if you pay more 240 |
| 15:15.30 | coppice | zoa: if the CPUs are mostly crunching DSP, and have enough cache, they can scale very well. |
| 15:15.38 | Lathos42 | Good Morning everyone |
| 15:15.39 | trixter | according to glossy flyers anyway |
| 15:15.39 | brad_mssw | well, you could always go dual core opteron, dual processor, and effectively go 4-way for not that much $$ |
| 15:15.41 | shakey | 200 per dual xeon box? |
| 15:15.55 | zoa | coppice: even, just do empty channels, no rtp |
| 15:15.57 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 15:16.01 | zoa | asterisk still doesnt behave all that well on it |
| 15:16.24 | zoa | anyhow, im off |
| 15:16.26 | zoa | too sick |
| 15:16.27 | zoa | ciao |
| 15:16.56 | blitzrage | lates |
| 15:17.03 | shakey | zoa: thanks./ |
| 15:17.31 | shakey | so, still confused, cisco as a sip gateway or asterisk? |
| 15:17.40 | fugitivo | SER |
| 15:18.00 | fugitivo | asterisk is a pbx |
| 15:18.17 | shakey | ser, sip express router? |
| 15:18.25 | shakey | the one from iptel? |
| 15:18.28 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:18.28 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:18.32 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 15:18.34 | fugitivo | yes, or openser |
| 15:18.54 | fugitivo | you use ser as a sip gateway, asterisk for pstn, voicemail, etc |
| 15:19.06 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:19.12 | shakey | that's what I wanted to do with cisco.. |
| 15:19.26 | shakey | but I will read something about it.. |
| 15:19.28 | shakey | ser i mean |
| 15:19.46 | pointer | anyone here have recommendations for analog->voip termination other than digium cards? |
| 15:19.48 | fugitivo | you won't find too much documentation about it |
| 15:19.52 | Mimmus | what is the "peer-to-peer routing protocol" in Asterisk? DUNDI? |
| 15:20.12 | *** part/#asterisk sivana (n=sivana@mixdown.ca) |
| 15:20.16 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 15:20.24 | [TK]D-Fender | pointer : AudioCodes, MutiTech, Medatrix, Sangoma.... |
| 15:20.24 | fugitivo | pointer: sangoma, i don't use them, but it seems they work ok |
| 15:20.51 | [TK]D-Fender | pointer : What kind of interface, how many ports? |
| 15:20.56 | pointer | sangoma does analog (fxo) interface termination? |
| 15:21.02 | *** join/#asterisk sivana (n=sivana@mixdown.ca) |
| 15:21.04 | *** join/#asterisk sulex_ (n=sulex@host70-20.pool8532.interbusiness.it) |
| 15:21.09 | sivana | tzanger: ping |
| 15:21.16 | pointer | most of the analog customers have 1-5 lines |
| 15:21.17 | Mimmus | pointer: no, I think |
| 15:21.23 | [TK]D-Fender | pointer : With their new A200 card, yes |
| 15:21.30 | pointer | we've tried the TDM400ps |
| 15:21.47 | pointer | and I just bought a TDM2401E hoping the onboard echo cans would fix our problem and it didn't |
| 15:21.59 | fugitivo | pointer: really? |
| 15:22.05 | *** part/#asterisk vladuk (n=vk@ip.82.144.198.127.stat-2.volia.net) |
| 15:22.06 | [TK]D-Fender | http://www.telephonyware.com/telephonyware/sangoma_aa.html |
| 15:22.12 | fugitivo | tell us your experiences with the tdm2400e |
| 15:22.26 | pointer | I'm sitting on about $5k in digium hardware that all echos in every box (about 4 different motherboards) that we tried it in |
| 15:22.45 | fugitivo | pointer: are you using software echocan too? |
| 15:22.50 | Mimmus | for who is interested, there is a Mark interview here http://osdir.com/Article8156.phtml |
| 15:23.00 | pointer | the one I haven't tried yet is the 4 port T1 card w/echo cans....I have it in a box ready to go though |
| 15:23.03 | mut | this dslam is pissin me off |
| 15:23.06 | [TK]D-Fender | pointer : I use Sangoma's A104d w/ hardware EC and haven't heard ANY since. Since they're due to be using the same basic DSP for EC on their analog cards I'd put good money on it... |
| 15:23.14 | shakey | fugitivo: heh, I will try to find some info about this SER, if it fails I'll go for cisco ;) |
| 15:23.32 | pointer | fugitivo: I have echocancel=yes in zapata.conf, yes |
| 15:23.36 | fugitivo | shakey: i know a lot of people is using openser+asterisk |
| 15:23.38 | Cinen | Does the local box need to be able to write to the database for realtime or can I have a main box and replicate all data out to the others? |
| 15:23.45 | pointer | fugitivo: should I turn that off? |
| 15:23.51 | zoa | shakey: multiple asterisk servers as sip gateway |
| 15:23.56 | pointer | fugitivo: I really want this digium hardware to work, honest |
| 15:24.08 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 15:24.35 | pointer | the vegastream 10-port fxo <-> sip boxes work really well as far as echo is concerned |
| 15:25.35 | pointer | fugitivo: I just got the tdm2401E on friday andput it in an intel box with 4 analog lines for testing...I got echo |
| 15:25.55 | pointer | fugitivo: I did have an irq conflict, so I fixed that, but it's still there... |
| 15:26.10 | [TK]D-Fender | pointer : Software EC, or the hardware one? |
| 15:26.22 | pointer | fugitivo: asterisk and the polycom phone are on the same ethernet segment (by themselves) |
| 15:26.35 | pointer | [TK]D-Fender: the hardware one (TDM2401E) |
| 15:27.00 | pointer | am I supposed to set echocancel=no in zapata.conf if the card has hardware EC? |
| 15:27.03 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 15:27.40 | pointer | basically, the sidetone (I believe that's what hearing yourself is referred to) is _really_ loud |
| 15:28.14 | pointer | and (guessing) around 100-250ms delayed |
| 15:28.44 | pointer | I have the vega setup on the same network with the asterisk box and the polycom...and when I terminate on the vega to the polycom, I get no echo |
| 15:29.43 | *** join/#asterisk lunk (n=lunk@66.152.8.184) |
| 15:29.48 | pointer | err...that wasn't lucid vega->sip->asterisk->sip->polycom ip500 == no echo; pstn->tdm2401E->asterisk->sip->polycom ip500 == echo |
| 15:30.02 | pointer | with the latest zaptel downloaded/compiled from yesterday |
| 15:30.23 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfimv.dialup.mindspring.com) |
| 15:30.41 | shakey | okay, okay, everyone is telling me something else. so asterisk or openser? which and why? |
| 15:30.48 | pointer | there is never echo for the person on the PSTN side of the call though...just the local sip caller (polycom in this case) |
| 15:31.07 | shakey | zoa: and can you tweak this asterisk to handle more connections? and for how much? ;) |
| 15:31.10 | blitzrage | sivana: do you work with tzanger? |
| 15:31.27 | pointer | I get the same results with the TDM400p (w/FXOs) that I do with the TDM2401E |
| 15:31.29 | sivana | at times |
| 15:31.33 | sivana | why, what's up? |
| 15:31.55 | brad_mssw | pointer: have you used the 'fxotune' utility ? |
| 15:32.00 | pointer | brad_mssw: yup |
| 15:32.07 | pointer | brad_mssw: it didn't help either |
| 15:32.19 | Hmmhesays | anyone use SER with radius before? |
| 15:32.45 | pointer | brad_mssw: we've also (in the past) tried the other software echo cancellers with the TDM400ps (requires zaptel recompile) |
| 15:32.46 | *** join/#asterisk flashnet[BNC] (i=flashnet@Darkstar.AceShells.com) |
| 15:33.14 | [TK]D-Fender | pointer : how many ports do you need and what other gear do you have now? |
| 15:33.18 | brad_mssw | pointer: wow, no clue ... I've got 2 tdm400ps here ... haven't had echo issues |
| 15:33.31 | pointer | brad_mssw: I truly envy you |
| 15:34.02 | blitzrage | sivana: oh, just noticed you were from mixdown.ca |
| 15:34.02 | pointer | [TK]D-Fender: we're using the vegastream 50s right now...they have 10 fixed fxo or fxs ports |
| 15:34.03 | brad_mssw | pointer: though I did have to adjust my gain a bit at the beginning, don't remember why |
| 15:34.10 | blitzrage | sivana: didn't need anything :) |
| 15:34.11 | pointer | brad_mssw: we tried that too |
| 15:34.26 | fugitivo | pointer: is your wiring ok? |
| 15:34.38 | sivana | hehe |
| 15:34.51 | [TK]D-Fender | pointer : And the vegas don't cause echo problems? |
| 15:34.54 | pointer | fugitivo: the wiring here is like 1 year old. We've had this problem at 4 sites in a 20 mile radius |
| 15:35.02 | sivana | blitzrage: np... we collaborate on different projects, etc.. |
| 15:35.30 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 15:35.35 | pointer | [TK]D-Fender: see above...going through the same asterisk box, with calls placed within 30 seconds of each other...a resounding NO (pun intended) |
| 15:35.48 | coppice | pointer: does the hardware EC appear to do anything at all? |
| 15:36.03 | pointer | [TK]D-Fender: the only thing on the ethernet segment was the asterisk box, a vega50 and a soundpoint ip500 |
| 15:36.14 | [TK]D-Fender | pointer : So just use the veg then... |
| 15:36.21 | pointer | coppice: no, I get the same echo with a TDM400p and a TDM2401E |
| 15:36.29 | fugitivo | what's the point of buying hardware with echocan if you still have echo? :) |
| 15:36.48 | pointer | [TK]D-Fender: the current box we have for testing appears to have CID issues |
| 15:36.55 | [TK]D-Fender | fugitivo : Giving more money to your suppliers! |
| 15:37.04 | fugitivo | [TK]D-Fender: right! |
| 15:37.10 | JMcA | pointer: just a shot in the dark, here....are there configuration settings for anything like a playout buffer size for the ZAP interfaces? |
| 15:37.14 | pointer | fugitivo: I was hoping the 2400E's EC's would fix teh problem |
| 15:37.23 | coppice | pointer: sounds like it isn't even being enabled for some reason. you'd expect some effect, if its only to make things worse |
| 15:37.24 | pointer | s/teh/the/ |
| 15:37.28 | fugitivo | pointer: you tried all lines and you have echo on all of them? |
| 15:37.40 | pointer | coppice: asterisk logs the echo enable/disable |
| 15:37.44 | fugitivo | jbot is speaking? |
| 15:37.49 | JMcA | coppice: maybe...unless the round-trip time is too long to trigger the echo cancellation processing |
| 15:37.50 | pointer | fugitivo: so far, yes...again at 4 sites |
| 15:37.52 | [TK]D-Fender | pointer : 2 ideas (1 a little $, 1 slow) : Sangoma A104d + channelbank. Or Wait a little bit for the A200 /w HW EC to come out. |
| 15:38.23 | fugitivo | pointer: did you set echocan in zapata.conf? |
| 15:38.36 | coppice | JMcA: pointer indicated he gets this with all calls |
| 15:38.36 | fugitivo | pointer: did you adjust the gains? |
| 15:38.42 | pointer | fugitivo: yes...I tried yes, 32,64, and 256 |
| 15:38.48 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 15:38.56 | pointer | fugitivo: yes...we've tried values between -10 and 10 |
| 15:38.57 | fugitivo | pointer: echotraining? |
| 15:39.05 | troyb | alright im out for a bit, later |
| 15:39.06 | JMcA | coppice: all calls through the pstn, right? |
| 15:39.06 | *** join/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 15:39.07 | pointer | I tried 400 and 800 |
| 15:39.26 | fugitivo | hmm, you should return those echocan modules to digium :) |
| 15:39.32 | coppice | JMcA: well only IDD calls are going to give hime much delay on the PSTN |
| 15:39.37 | *** part/#asterisk epoch (i=epoch@octane.breakbeats.org) |
| 15:39.38 | pointer | coppice: but if they go throug the vega (asterisk is just handling SIP), there is no echo...on the same line(s) |
| 15:39.46 | mut | upgrading the firmware on a spa-2002 won't reset it's config will it? |
| 15:39.49 | pointer | s/throug/through/ |
| 15:39.58 | JMcA | right...and its delay that is perceived as echo...and if the delay is long enough, the EC circuit won't treat it as echo |
| 15:40.00 | fugitivo | what's happening with jbot today??? |
| 15:40.24 | coppice | JMcA: sounds like you need to read up about echo |
| 15:40.26 | pointer | the echo is the sip caller hearing themselves (in the polycom handset) |
| 15:40.37 | pointer | even with the volume on the handset turned all the way down |
| 15:41.04 | fugitivo | pointer: you get echo with calls to different pstn numbers? |
| 15:41.10 | pointer | yes |
| 15:41.15 | fugitivo | pointer: did you get any call with not echo? |
| 15:41.21 | fugitivo | s/not/no |
| 15:41.23 | pointer | over the vegastream, yes |
| 15:41.23 | [TK]D-Fender | pointer : If you were to have said HEADSET I might have suspected a little, but not nahdset |
| 15:41.31 | Katty | oh! speaking of headsets |
| 15:41.33 | mut | anyone? |
| 15:41.43 | Katty | i ordered those platronics 510s bluetooth ones. |
| 15:41.44 | JMcA | coppice: I know quite a bit about it...if the round-trip time for the audio is too long, it will outstrip the ability of echo cancellation to deal with it |
| 15:41.48 | pointer | [TK]D-Fender: *nod*, this is a polycom soundpoint ip 500/501 with the stock handset |
| 15:41.51 | Katty | they're a /dream/ |
| 15:42.15 | JMcA | that is frequently a culprit of something along the lines of playout buffers being too big/long |
| 15:42.19 | coppice | JMcA: well, then read up and find the bit you don't seem to have worked out |
| 15:42.20 | Katty | we can even use our bluetooth headsets we use with our nextel blackberry phones (= |
| 15:42.38 | Katty | and answer regular phones, and mobiles, all at the same time! |
| 15:42.45 | *** join/#asterisk javar (n=javar@69.79.217.237) |
| 15:42.47 | fugitivo | Katty: that's cool |
| 15:42.49 | JMcA | the part I don't have worked out is if the ZAP interface has adjustable buffers |
| 15:43.06 | Katty | fugitivo: tis :> |
| 15:43.30 | fugitivo | Katty: how does it work? you connect the headset to all the devices?? |
| 15:44.40 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 15:44.41 | JMcA | pointer: how long of a delay are you hearing on the echo? |
| 15:45.22 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 15:45.31 | Katty | fugitivo: you connect the bluetooth transmitter box thingy to the polycom phone (we have ip500s) and as long as they're both on the same channel... |
| 15:46.04 | Katty | fugitivo: it'll work with cellphone and polycom phone. of course you can't have calls at the same exact time... |
| 15:46.13 | websae | I am curious....anyone know of a good place to get VoIP provisioning equipment? |
| 15:46.28 | pointer | JMcA: it has to be less than 250ms |
| 15:46.55 | [TK]D-Fender | websae : explain |
| 15:47.19 | Katty | fugitivo: it works with the /new/ blackberries. |
| 15:47.29 | fugitivo | Katty: i don't like blackberries :/ |
| 15:47.29 | Katty | fugitivo: we had a bit of an issue with the older purply blackberries |
| 15:47.33 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 15:47.40 | JMcA | ok...from what I'm reading (translation: no personal experience with the digium cards), the max delay on the echo cancellation can be set to is 32ms (echocancel=256) |
| 15:47.46 | Katty | fugitivo: well there's plenty of other cellphones that are bluetooth enabled (= |
| 15:48.14 | websae | Say a voip company like vonage....or broadvoice...they can update their customer's ATA adapters and phones remotely and I am sure they have a provisioning server and what not...know of any solutions? |
| 15:48.16 | websae | anyone? |
| 15:48.24 | fugitivo | Katty: :) |
| 15:48.26 | coppice | JMcA: that has nothing to do with his 250ms |
| 15:48.39 | JMcA | coppice: it can |
| 15:48.45 | Katty | fugitivo: only issue, is that you have to manually hit the answer button on the polycom phones. |
| 15:48.53 | Katty | fugitivo: unless there's a way to rig that. |
| 15:49.00 | JMcA | not saying it definitely does, but it can |
| 15:49.02 | Katty | fugitivo: the nextel phones you can answer directly from the headset |
| 15:49.03 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 15:49.15 | fugitivo | Katty: hmmm, that means you have to be at your desk |
| 15:49.19 | Katty | anthm: does your headset let you answer the phone directly from it, or do you have to be at your desk? |
| 15:49.19 | websae | TKD-FENDER.:know of anything |
| 15:49.35 | coppice | pointer: for local calls you almost never need more than a few milliseconds of echo cancellation, unless someone is being tricky and taking your calls through a VoIP path. if the canceller isn't cancelling, something is wrong |
| 15:49.51 | anthm | gotta press the answer button on the phone |
| 15:49.57 | Katty | anthm: k |
| 15:50.28 | pointer | coppice: agreed. I just want to know how to fix it. |
| 15:50.49 | websae | anyone know of any VoIP provisioning equipment to manage ATA adapters? |
| 15:50.49 | pointer | Katty: that's something you send from asterisk |
| 15:50.53 | coppice | pointer: I think you are at the complaining point :-) |
| 15:51.16 | fugitivo | lol |
| 15:51.17 | Katty | pointer: hmm? |
| 15:51.21 | fugitivo | coppice: he should |
| 15:51.27 | Katty | anthm: can you make a certain extension do an auto answer? |
| 15:51.37 | pointer | correction. I'm wrong |
| 15:51.43 | anthm | never tried |
| 15:52.06 | mut | JMcA: what other than an mtu issue might it be? |
| 15:52.10 | fugitivo | Katty: some phones have autoanswer |
| 15:52.22 | Katty | what's Headset Memory Mode mean? |
| 15:52.24 | mut | i have the mtu the same on everything but the dslam |
| 15:52.27 | Katty | on these polycoms |
| 15:52.28 | pointer | Katty: you set it in the sip.conf or ipmid.conf (which was merged into sip.conf in the newer firmwares) |
| 15:52.36 | pointer | Katty: there's a good voip-info article about it |
| 15:52.43 | sivana | websae: just the one we wrote in house |
| 15:52.45 | *** join/#asterisk _-_ (n=nabudoco@206.135.48.98) |
| 15:52.51 | coppice | Katty: it remembers it's a headset |
| 15:52.56 | Katty | coppice: thanks |
| 15:53.12 | JMcA | mut: nothing jumps out at me as another likely problem...can you stick ethereal or tcpdump on it? |
| 15:53.17 | websae | hrm |
| 15:53.29 | websae | sivana: you don't know any companies though? |
| 15:53.31 | pointer | coppice: (on the complaining point) I was just hoping someone here had tried something that worked for them that I could try :-\ |
| 15:53.38 | mut | on what? |
| 15:53.45 | websae | how does your provisioning system work sivana? |
| 15:53.57 | JMcA | mut: somewhere in the path to capture the tcp connection traffic |
| 15:54.03 | *** join/#asterisk VJ (n=vijay@220.227.161.180) |
| 15:54.10 | coppice | pointer: I've tried complaining. occasionally it works :-) |
| 15:54.17 | mut | well |
| 15:54.27 | mut | yea |
| 15:54.29 | mut | i can on this pc |
| 15:54.40 | mut | already have, nothing sticks out to me tho |
| 15:54.41 | sivana | websae: it's part of an entire management system, but it will automatically create/ftp the provisioning for all linksys/sipura/polycom/cisco atas |
| 15:54.49 | mut | it's a different amount of data sent each time |
| 15:54.50 | Katty | pointer: http://www.voip-info.org/wiki-Polycom+auto-answer+config <- is that the url you were refering to? |
| 15:54.58 | pointer | coppice: I tried throwing money ($1k for the 2401E) at the problem too, that doesn't seem to have worked for me either |
| 15:55.00 | fugitivo | pointer: complain with digium about the hardware echocan not working |
| 15:55.21 | websae | yeah I am looking to get an entire management system for provisioning and monitoring... |
| 15:55.24 | websae | any suggestions? |
| 15:55.48 | VJ | hi guys |
| 15:55.49 | pointer | fugitivo: I bought the card from voipsupply, so I think I have to open a ticket with them first (which I left a VM to try to do) |
| 15:55.56 | coppice | pointer: money can't buy you everything. :-) |
| 15:55.58 | VJ | i am trying to load extensions.conf in my system |
| 15:56.08 | pointer | Katty: that looks like what we used |
| 15:56.10 | fugitivo | coppice: are you sure? :) |
| 15:56.14 | VJ | and i am getting the following warning messages |
| 15:56.19 | VJ | any hint why it is coming |
| 15:56.21 | VJ | http://pastebin.com/511534 |
| 15:56.23 | pointer | coppice: heh |
| 15:56.25 | Katty | pointer: sadly, i can't keep up with that doc. |
| 15:56.29 | Katty | pointer: it's going over mine head. |
| 15:56.49 | Mimmus | does anyone use LDAP to store realtime extensions? |
| 15:56.54 | fugitivo | VJ: already in use, means that the extension, is already in use... |
| 15:57.01 | coppice | the best things in life are free.... but if you can't get those, money make the deprivation much more comfortable |
| 15:57.25 | fugitivo | coppice: are you talking about women? |
| 15:57.32 | [TK]D-Fender | pointer : My 2 earlier suggestion will likely solve the problem... |
| 15:57.49 | [TK]D-Fender | websae : Depends on the remote device. |
| 15:57.51 | pointer | Katty: basically, you're telling the polycom (via sip.conf) that it should autoanswer when it receives a specific ring type and you're telling asterisk to send it when you dial *74{extension} instead of {extension} |
| 15:58.00 | coppice | fugitivo: fill in whatever light's your candle :-) |
| 15:58.28 | websae | well i am looking for use with just sipura phones or some ata adapter |
| 15:58.46 | pointer | [TK]D-Fender: I don't think the T1 card and channel bank is a fit for this site with only 4 analog lines, but I'll try to keep an eye out for the other card |
| 15:58.51 | websae | TKD-FENDER: any suggestions |
| 15:58.51 | Katty | pointer: hrm. |
| 15:58.57 | pointer | [TK]D-Fender: I've heard a lot of really good things about sangoma |
| 15:59.00 | Katty | also! |
| 15:59.09 | Katty | new horizon's launch has been canceled this morning |
| 15:59.10 | [TK]D-Fender | Katty : Depends what kind of auto-answer you are looking for. if you're looking for a way so that you can jsut activate your BT jheadset and cause the phone to ansewr no.. if its just to make the Polcom answer period, then yes |
| 15:59.59 | [TK]D-Fender | pointer : I has vexed by 2 TE405P with horrid echo and a host of other problems. They all completely vanished with my A104d. |
| 16:00.09 | pointer | Katty: the important part of the dialplan is this -> SetVar(ALERT_INFO="Ring Answer"); |
| 16:00.19 | [TK]D-Fender | websae : Sipura's can be remotely provisioned. |
| 16:00.21 | pointer | Katty: then you just tell your polycom to autoanswer when it sees that |
| 16:00.36 | Katty | pointer: uhh, right. |
| 16:00.44 | `Sauron | Meow |
| 16:00.47 | Katty | some heavy winds knocked out the power last night. |
| 16:00.51 | [TK]D-Fender | Katty : Its just 2 places in SIP.CFG |
| 16:00.53 | pointer | Katty: are you configuring your polycoms via ftp right now? |
| 16:01.07 | Katty | `Sauron: hi! |
| 16:01.15 | Katty | `Sauron: launch is being pushed off till tomorrow (= |
| 16:01.25 | Katty | pointer: sorta. |
| 16:01.27 | `Sauron | w00t |
| 16:01.48 | websae | im looking for a provisioning/management software to do so TKD-FENDER |
| 16:02.41 | JMcA | ok...I'm going home to catch some shut-eye |
| 16:02.53 | Katty | pointer: i setup an ftp server to do a firmware update for our phones. |
| 16:02.58 | Katty | pointer: but not any config stuff. |
| 16:03.02 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:03.15 | kippi | whats the most debuging you can put on asterisk? and what would the command be? |
| 16:03.41 | coppice | asterisk -v*1.0e38 |
| 16:03.51 | [TK]D-Fender | websae : there is no miracle software for what you want. its a custom app you'd need to make to determine that the ATA is on-line and to contact it. |
| 16:03.51 | pointer | Katty: gotcha |
| 16:04.02 | [TK]D-Fender | websae : read the admin & provisioning guides |
| 16:04.20 | pointer | Katty: we use it for config...if you do that, you can use a skeleton sip.cfg for all the phones...and put the autoanswer stuff there |
| 16:04.41 | [TK]D-Fender | Katty : You really should provision them. thats where all the good stuff is like 1-touch voicemail access, presence, etc.... |
| 16:04.57 | Katty | [TK]D-Fender: probably....but it's a bit over my head. |
| 16:05.17 | websae | how can you make the phone automatically update? |
| 16:05.23 | [TK]D-Fender | Katty : I doubt that, and I'd be willing to preconfigure one for you. |
| 16:05.30 | Katty | [TK]D-Fender: but i don't want to. |
| 16:05.37 | Katty | [TK]D-Fender: i'll tackle it when i feel comfortable with it (= |
| 16:05.48 | [TK]D-Fender | websae : Go get the admin & provisioning guides! Check Voxilla & voipsupply for more info. |
| 16:05.58 | Katty | [TK]D-Fender: we have 8 phones here. |
| 16:06.05 | [TK]D-Fender | Katty : Sure, the offer is "open". |
| 16:06.07 | Katty | [TK]D-Fender: i'm not that worried about doing fancy fancy things here. |
| 16:06.14 | Katty | [TK]D-Fender: yet. |
| 16:06.25 | [TK]D-Fender | Katty : For 8 phones I might jsut do the whole job for you :) |
| 16:06.36 | Katty | [TK]D-Fender: you're not touching it, dear. |
| 16:06.50 | [TK]D-Fender | heh |
| 16:06.51 | Katty | [TK]D-Fender: run along and pester someone else's network. |
| 16:07.22 | [TK]D-Fender | *wah* ... meanie |
| 16:07.32 | Katty | [TK]D-Fender: if it makes you feel anybetter, i wouldn't even let anyway ssh into my box when i was setting asterisk up for the first time (= |
| 16:07.45 | *** join/#asterisk Delvar (n=irc@host-83-146-53-34.bulldogdsl.com) |
| 16:08.15 | [TK]D-Fender | Katty : I never said I'd need access to your server at all for what I was offering. |
| 16:08.34 | Katty | [TK]D-Fender: i'm well aware of that |
| 16:08.47 | Katty | [TK]D-Fender: there you go reading into things again :P |
| 16:08.58 | Katty | [TK]D-Fender: i do not want any help, thanks :) |
| 16:09.03 | iCEBrkr | Lies |
| 16:09.14 | [TK]D-Fender | Actually.. I think of it more as "creative writing" ;0 |
| 16:09.31 | Katty | if i needed help, i'd go ask anthm or Hmmhesays ...they both speak kat well. |
| 16:09.58 | [TK]D-Fender | iCEBrkr : No, like I told you before, women don't want you to SOLVE their problems, just to have you understand them and provide company as things come crashing down :) |
| 16:10.04 | Katty | iCEBrkr: wow, i just don't like putting up with your insults this morning. |
| 16:10.15 | coppice | creative writing == progress reports |
| 16:10.18 | iCEBrkr | Katty: Shut your mouth when you speak to me. |
| 16:10.30 | [TK]D-Fender | iCEBrkr : Nice recycling.... |
| 16:10.33 | Katty | iCEBrkr: You Betta Don't(tm) |
| 16:10.42 | iCEBrkr | :P |
| 16:11.09 | iCEBrkr | [TK]D-Fender: I think that's the best line from the movie. |
| 16:11.21 | [TK]D-Fender | Dammit when are Sangoma's A200s going to hit mass distribution!? |
| 16:11.35 | Katty | i'd ask Moc, but he's not in here |
| 16:12.59 | websae | is it possible to provision sip phone remotely forcing the tftp file into the phone? |
| 16:13.36 | [TK]D-Fender | websae : Not through TFTP. Phones are not tftp SERVERS waiting for requests. Either HTTP of FTP. |
| 16:13.55 | [TK]D-Fender | websae : An d jsut go and read the damn docs for the model you are interested in. |
| 16:14.01 | websae | well i have |
| 16:14.04 | websae | for the 841 |
| 16:14.05 | websae | sipura |
| 16:14.16 | websae | i know it has tftp |
| 16:14.40 | [TK]D-Fender | NO TFTP. PERIOD. FTP or HTTP. |
| 16:14.49 | [TK]D-Fender | TFTP is for local provisioning, not remote. |
| 16:14.50 | websae | it has tftp |
| 16:14.55 | blitzrage | TFTP is the only way to go</sarcasm> |
| 16:15.15 | iCEBrkr | [TK]D-Fender: He's talking about the phones TFTPing in to get the updates |
| 16:15.24 | iCEBrkr | or, I'd think so anyhow |
| 16:15.24 | websae | ahh....so ftp or http |
| 16:15.27 | Mimmus | does anyone know Ian Kinner from Digium? He is working on a support case and it would be great if we could chat here |
| 16:15.40 | [TK]D-Fender | iCEBrkr : No if you followed earlier he was asking about remote provisioning like an ITSP. |
| 16:15.41 | iCEBrkr | But I don't think you can 'provision' a phone via TFTP |
| 16:15.51 | [TK]D-Fender | Mimmus : I've dealt with him. |
| 16:16.16 | Mimmus | [TK]D-Fender: OK |
| 16:16.31 | [TK]D-Fender | Mimmus : but no idea how to contact him here |
| 16:16.41 | Mimmus | [TK]D-Fender: no problem, thanks |
| 16:17.06 | Netgeeks | 8grumble8 i hate mornings |
| 16:17.12 | Mimmus | and 'matt'? He is working at a bug... |
| 16:17.41 | Netgeeks | I also very much dislike it when a computer ignores when I hold down the shift key... *grumble* |
| 16:18.24 | Katty | Netgeeks: sounds like you could use a muffin. |
| 16:18.53 | kippi | how can I see when a sip extension is trying to logon? |
| 16:18.53 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 16:19.16 | Netgeeks | That sounds nice, muffin and a hot cocoa |
| 16:19.29 | *** join/#asterisk flok420 (i=nobody@keetweej.xs4all.nl) |
| 16:20.30 | flok420 | hi, after upgrading 1.0.9 to 1.2.2 "exten => s,7,Queue(callqueue|ntTH|||20)" no longer aborts after 20 seconds, it just hangs up the phone after a minute or so. anyone got a suggestion? |
| 16:20.32 | *** join/#asterisk fulgas (n=fulgas@209.8.233.10) |
| 16:21.49 | Netgeeks | my first thought would be to show application queue and make sure the command argument structure didn't change |
| 16:22.05 | *** join/#asterisk jimmy_deanPB (n=jhodapp@72.244.232.226) |
| 16:23.43 | jimmy_deanPB | I just received a lot of the following message from the asterisk console: NOTICE[2608]: chan_iax2.c:1597 iax2_destroy: Avoiding IAX destroy deadlock I don't understand what would cause this though from looking at the source code it's generated by a request for a mutex lock...google doesn't bring up much about this error...any thoughts? |
| 16:24.05 | [TK]D-Fender | jimmy_deanPB : UPGRADE! |
| 16:24.11 | jimmy_deanPB | I'm using 1.2.1 |
| 16:24.22 | jimmy_deanPB | there's nothing to upgrade to |
| 16:24.33 | fugitivo | i think 1.2.2 will hide all those messages... :/ |
| 16:24.35 | flok420 | netgeeks: it still is the same |
| 16:24.40 | fugitivo | jimmy_deanPB: 1.2.2 is out |
| 16:24.53 | Netgeeks | take heart, it's a notice and not an error... |
| 16:25.09 | jimmy_deanPB | well I don't care if it hides it or not, but the person on a line at the time was cut off by asterisk thanks to that condition |
| 16:25.26 | Netgeeks | hrm, that sounds like it should be an error, then :( |
| 16:26.00 | *** part/#asterisk pointer (i=pointer@aj.catt.com) |
| 16:26.00 | fugitivo | i don't think that should cause a hangup |
| 16:26.40 | trixter | I dont think putting thge phone in the cradle should cause a hangup |
| 16:26.48 | Netgeeks | flok420: dunno what to tell you, I'd start debugging.. |
| 16:27.01 | jimmy_deanPB | well it couldn't find my IAX provider then: |
| 16:27.03 | jimmy_deanPB | Jan 18 11:02:33 NOTICE[2608]: chan_iax2.c:1597 iax2_destroy: Avoiding IAX destroy deadlock |
| 16:27.03 | jimmy_deanPB | Jan 18 11:02:33 WARNING[4553]: chan_iax2.c:2747 create_addr: No such host: gwiaxt01.voicepulse.com |
| 16:27.03 | jimmy_deanPB | Jan 18 11:02:33 WARNING[4553]: chan_iax2.c:2940 iax2_call: No address associated with 'gwiaxt01.voicepulse.com' |
| 16:27.25 | fugitivo | ok, we're going from a notice to a warning :) |
| 16:27.26 | jimmy_deanPB | it seems to have no idea about my provider for a second, but then things started working again |
| 16:27.51 | Lathos42 | show modules |
| 16:27.55 | Lathos42 | whoops :) |
| 16:28.45 | jimmy_deanPB | fugitivo, yeah, I've never seen that before but it was weird and the person was not happy that asterisk hung up the call |
| 16:29.16 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 16:29.22 | fugitivo | maybe you lost your connection? |
| 16:29.23 | jimmy_deanPB | I also experience times when a call can't dial out, but then seconds later it can dial out...this is all over a VOIP provider |
| 16:29.43 | jimmy_deanPB | fugitivo, is that common with VOIP? |
| 16:30.06 | fugitivo | well, voip is not 100% accurate |
| 16:30.15 | jimmy_deanPB | but does it ever just plainly hangup |
| 16:30.16 | jimmy_deanPB | ? |
| 16:30.24 | jimmy_deanPB | or a VOIP connection gets lost |
| 16:30.25 | Netgeeks | have you tried using the ip address instead of hostname in our configurations? Just to isolate a DNS issue? |
| 16:30.30 | wasim | jimmy_deanPB: seems like a buggy dns or resovler |
| 16:30.43 | jimmy_deanPB | that's an interesting idea, I might try that |
| 16:30.54 | *** join/#asterisk sdf (n=lala@222.185.19.54) |
| 16:30.59 | jimmy_deanPB | well, but the DNS server is SBC's DSL service DNS servers |
| 16:31.07 | jimmy_deanPB | it's not even using an internal DNS server |
| 16:31.13 | wasim | jimmy_deanPB: you need to talk to bicster about sbc |
| 16:31.20 | jimmy_deanPB | I hate SBC |
| 16:31.26 | jimmy_deanPB | can't wait to get rid of them |
| 16:31.38 | jimmy_deanPB | we should be going to a Covad DSL line soon |
| 16:31.57 | jimmy_deanPB | wasim, does he hate them too? |
| 16:32.10 | wasim | jimmy_deanPB: with a passion |
| 16:32.44 | jimmy_deanPB | the fact that the first computer that connects to the DSL line the first time is required to get a connection back is just stupid |
| 16:32.48 | jimmy_deanPB | it associates a MAC address |
| 16:32.53 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 16:33.51 | iCEBrkr | Welp the phone line problem isn't on our end.. |
| 16:34.06 | iCEBrkr | I wonder who's digging outside our building or down the street. :-/ |
| 16:34.11 | jimmy_deanPB | so what are some things I can do to improve my network to optimize for VOIP? |
| 16:34.49 | *** join/#asterisk fanguin (n=user@p548F782D.dip.t-dialin.net) |
| 16:37.40 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 16:39.43 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 16:43.21 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:43.21 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:44.47 | *** join/#asterisk crich1999 (n=crich@p54BF9F2E.dip0.t-ipconnect.de) |
| 16:49.30 | *** join/#asterisk fiber0pti (n=John@invinine.com) |
| 16:49.43 | fiber0pti | how do I set the caller id of a calls going out of a specific phone? |
| 16:50.19 | wasim | fiber0pti: setcallerid() or put in iax.conf / sip.conf / zapata.conf |
| 16:50.38 | *** join/#asterisk spunz (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:50.42 | fiber0pti | wasim, so under sip.conf under the specific phone entry? |
| 16:50.46 | Dandan | hey, is there a sip/iax/h.323 clinet in java? |
| 16:50.51 | wasim | fiber0pti: yeah |
| 16:51.00 | wasim | Dandan: no, people aren't masochistic enough yet |
| 16:51.27 | wasim | there might be something for sip, i doubt for iax or h323 though |
| 16:51.39 | Katty | anyone know of this UserData folder in %userprofile% is temp stuff or actually usefull/required? |
| 16:52.00 | Katty | yes, i'm well aware this has nothing to do with linux (= |
| 16:52.15 | *** join/#asterisk VJ (n=vijay@220.227.161.180) |
| 16:52.21 | VJ | has anyone used Asterisk SVN-branch-1.2-r7677. |
| 16:52.28 | VJ | which one is reliable |
| 16:52.41 | VJ | 1.2.1 stable version or Asterisk SVN-branch-1.2-r7677. |
| 16:53.10 | wunderkin | there is no 1.2.1 stable version, that is a release |
| 16:53.17 | wunderkin | ..k |
| 16:53.22 | wunderkin | Katty, yes i would keep it |
| 16:53.23 | Hmmhesays | hrm |
| 16:54.15 | *** join/#asterisk VJ (n=vijay@220.227.161.180) |
| 16:54.26 | Dandan | wasim: dialpad worked really well |
| 16:54.27 | Dandan | afair |
| 16:54.28 | Dandan | :) |
| 16:54.57 | Hmmhesays | i need to roadtrip somewhere |
| 16:55.10 | *** join/#asterisk VJ (n=vijay@220.227.161.180) |
| 16:56.07 | Katty | Hmmhesays: i'm going to the university if IL this weekend. |
| 16:56.16 | Katty | Hmmhesays: it's good place to roadtrip to (= |
| 16:56.24 | *** join/#asterisk rhousand (n=rhousand@rrcs-24-199-246-10.midsouth.biz.rr.com) |
| 16:56.35 | wunderkin | vj: there is no 1.2.1 stable version, that is a release.. stay on 1.2 release branch |
| 16:56.51 | fiber0pti | wsaim: setcallerid in sip.conf doesn't seem to work |
| 16:57.54 | wasim | callerid= in sip.conf setcallerid() in extensions.conf |
| 16:58.01 | badboyz | is there any good asterisk monitoring software out there? for monitoring the health of multiple * boxes |
| 16:58.15 | wasim | Hmmhesays: drive down to pk :) |
| 16:59.43 | fiber0pti | wasim: does setcallerid() work if I'm making calls outside of the asterisk box? |
| 16:59.59 | wasim | outside from, yes |
| 17:00.14 | wasim | as long as your uplink understands it ... i.e. |
| 17:00.24 | fiber0pti | wasim: I set mine and it still uses the base number of the asterisk box. |
| 17:00.32 | fiber0pti | however, works internally |
| 17:00.54 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 17:01.22 | *** join/#asterisk SkalTura (n=none@a84-230-105-166.elisa-laajakaista.fi) |
| 17:01.31 | SkalTura | Hi |
| 17:01.34 | SkalTura | i'm new to asterisk |
| 17:01.37 | Hmmhesays | Katty: what brings you there ? |
| 17:01.42 | eKo1 | SkalTura: good for you |
| 17:02.01 | Hmmhesays | eKo1: or bad |
| 17:02.04 | SkalTura | Is configuring asterisk easy, medium or hard, compared to for example something mediocre for linux sysadmin? |
| 17:02.15 | Hmmhesays | bwhahah |
| 17:02.15 | eKo1 | SkalTura: |
| 17:02.17 | SkalTura | ie. Postfix is hard, while Exim is easy |
| 17:02.34 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 17:02.43 | Hmmhesays | SkalTura: can you read? |
| 17:02.50 | Netgeeks | Asterisk is easy and hard, depending on what you want to do with it |
| 17:03.04 | SkalTura | okies |
| 17:03.05 | Hmmhesays | how are your logical thinking skills? |
| 17:03.11 | SkalTura | that about answers: it's gonna be hard |
| 17:03.35 | SkalTura | (acquiring 100 phone lines etc. bunch of IP Phones, call recording, call centre functionality, automated services etc) |
| 17:03.48 | Hmmhesays | if you're trying to make asterisk the swiss army knife of voip, then you might as well give it up now |
| 17:03.48 | SkalTura | Hmmhesays: lol |
| 17:03.49 | [hC-] | Yeah... Give up now. |
| 17:03.50 | eKo1 | yes, that will be hard |
| 17:04.02 | SkalTura | Ok... |
| 17:04.13 | Netgeeks | I wouldn't say that would be hard, I'd rather say, that will be engaging |
| 17:04.22 | mog_work | Hmmhesays, thats what sterisk is |
| 17:04.34 | hmodes | yeah, i'm with them |
| 17:04.42 | SkalTura | basic IP Phone is easy? adding the phone lines (hardware has been ordered for that, i'm just the sysadmin, i don't even know what kind of hardware yet!) to be used? |
| 17:04.46 | Netgeeks | meaning if you take the time to understand what you are doing and think about your config, it's not hard, just 'engaging' |
| 17:04.51 | Hmmhesays | you want calls from your clients on friday night while you're getting bizzay with your hot date... go for it. |
| 17:04.56 | Flauto | do i need to stop asterisk when i recompile it |
| 17:05.07 | hmodes | if * is anything, it's the swiss army knife of voip |
| 17:05.14 | [hC-] | SkalTura: go spend the next 3 days reading www.voip-info.org, instead of asking us :) |
| 17:05.19 | SkalTura | Netgeeks: i think i can do that, i just spent over 100hours making an e-mail system, and getting to know it inside out X) |
| 17:05.32 | eKo1 | what e-mail system? |
| 17:05.35 | jbroome | 100hrs for an email system? |
| 17:05.36 | Netgeeks | but sometimes the spring on the little scissors in the knife gets stuck... |
| 17:05.44 | Mimmus | SkalTura: for me (8 years of experience in unix system administration) was a not so simple challenge |
| 17:05.48 | hmodes | hehe |
| 17:05.49 | jbroome | you're b0ned |
| 17:05.59 | hmodes | so true |
| 17:06.06 | SkalTura | [hC-]: sorry, ain't got 3 days, i promised to be done with it by friday and i haven't done more than compiled & installed it on my test system x) |
| 17:06.12 | eKo1 | if it was ms exchange, then i feel sorry for you |
| 17:06.33 | Mimmus | SkalTura: it is a different world: at first, you need to acquire confidence with acronyms :-) |
| 17:06.34 | Ahrimanes | hm, any reason why exten => 69127070 would have precedence over exten => _69127XXX when exten => 69127300 doesnt? |
| 17:06.46 | SkalTura | Mimmus: okies, you are way more experienced than me, and you keep it as challenge... How long it took time for you to figure it all out? |
| 17:06.56 | [hC-] | SkalTura: i can promise you this. reading voip info or not, you wont have this down in 3 days. |
| 17:07.01 | Mimmus | SkalTura: 6 months ;-) |
| 17:07.04 | [hC-] | SkalTura: s/days/months/ maybe |
| 17:07.09 | Netgeeks | SkalTura: you will be quite the lucky person if you can achieve what you want by friday (acquiring 100 phone lines etc. bunch of IP Phones, call recording, call centre functionality, automated services etc) |
| 17:07.23 | SkalTura | Mimmus: 6 months, you gotta be kidding me! |
| 17:07.25 | [hC-] | Little over achiever we have here. |
| 17:07.38 | [hC-] | Ive been using asterisk for over 2 years and its still not where i want it to be, in our setup |
| 17:07.40 | *** join/#asterisk FastJack (i=fastjack@p5091FD9A.dip.t-dialin.net) |
| 17:07.47 | mog_work | lol |
| 17:07.48 | [TK]D-Fender | SkalTura : Clarify "automated services" |
| 17:07.50 | SkalTura | Netgeeks: no no no, the basic configuratio netc. needs to be done by friday, phone lines haven't been ordered yet, but all the hardware exists or is being shipped atm |
| 17:08.12 | Hmmhesays | external gateways help quite a bit for pstn interface |
| 17:08.33 | [hC-] | SkalTura: im serious about reading voip-info.org by the way, its probably the definitive learning source, besides ordering a book |
| 17:08.33 | Hmmhesays | rarely do I use cards for such purposes |
| 17:08.38 | SkalTura | [TK]D-Fender: my client wants to be able so that when his client call, and input RMA code, Product code or something like that, they can request information of the product, submit a warranty request or on-site repair etc. |
| 17:08.46 | jbroome | note to self: see how SkalTura is doing with his * setup on Friday. :) |
| 17:08.50 | SkalTura | [hC-]: opened it already |
| 17:08.53 | Netgeeks | 100 phone lines (easy bu time consuming), bunch of IP phones (just money and configuration time), call recording (not too hard either, some gotchas), call center functionality (KABOOM! ouch, there is alot implied in that term), automated services (another KABOOM) |
| 17:09.01 | SkalTura | jbroome: hahah :P |
| 17:09.15 | Mimmus | SkalTura: unless you know what are CID, Zap, PRI, DISA, ADSI, AGI, SIP, IAX, MGCCP, SKinny, trunk, blind transfer, directory, ... |
| 17:09.18 | SkalTura | jbroome: in the meantime, want to buy an hell of a e-mail system, with full service? |
| 17:09.35 | *** join/#asterisk secure75 (n=mic@ppp-82-135-73-230.mnet-online.de) |
| 17:09.40 | Hmmhesays | Mimmus: thats a little overboard, for his app |
| 17:09.46 | SkalTura | Netgeeks: call center: waiting lines and such, nothing more complicated |
| 17:10.03 | SkalTura | Netgeeks: perhaps call forwarding if one number doesn't answer |
| 17:10.31 | SkalTura | Mimmus: uuh... Trunk i know i've heard somewhere... and ADSI, and SIP! Iaxy i've heard of too... uhm... |
| 17:10.37 | [hC-] | Hmmhesays: what external gateways do you use? |
| 17:10.43 | Hmmhesays | call centers usually imply some type of queing systems |
| 17:10.54 | Hmmhesays | audiocodes and quintum |
| 17:10.56 | SkalTura | ok, Hmmhesays |
| 17:11.03 | [TK]D-Fender | SkalTura : Lets just say that the basic setup shouldn't be too hard (IP phones, PRI (lines),VM, ACD, and some Q&A product related IVR. |
| 17:11.20 | Mimmus | SkalTura: queues, agents, musiconhold, dynamci/static agents, login, logoff from queue, ... |
| 17:11.21 | SkalTura | so i should look for queueing system, informing the receivee of queued calls (only 1 number needs to answer to those queued calls at first1) |
| 17:11.30 | SkalTura | 100 lines is overkill for our needs currently |
| 17:11.34 | Hmmhesays | people swear up and down quintum sucks, but its easier when you work with their engineers |
| 17:11.59 | [TK]D-Fender | SkalTura : How many lines, using what tech? How many phones? All IP? |
| 17:12.02 | SkalTura | ok |
| 17:12.32 | Hmmhesays | audiocodes documentation blows the proverbial donkey dong |
| 17:12.39 | Hmmhesays | but the hardware is solid |
| 17:12.54 | coppice | industry standard documentation (TM) |
| 17:12.58 | SkalTura | [TK]D-Fender: some calls would be forwarded to mobile or line phones, mostly IP phones tho. 100 phone lines. IP Phones amount: so far unknown, but to my place is being shipped 2 IP phones for testing up, a minimum initial amount would be 3 i thikn |
| 17:13.30 | *** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com) |
| 17:13.31 | *** join/#asterisk blop (i=blop@2001:6f8:204:33:0:0:0:0) |
| 17:13.37 | Netgeeks | If you have to have somehting running by friday and you don't want to pay a consultant, and you aren't concerned about learning the system by friday, then AMP/A@H might be the initial ticket for you, getting you an operating system by your deadline, and allowing you to pick up asterisk knowledge at a more reasonable pace |
| 17:13.52 | Hmmhesays | yes indeed |
| 17:14.02 | [TK]D-Fender | SkalTura Are you looking to run an IVR driver call center? As in hoping the IVR answers most of your callers questions? |
| 17:14.05 | SkalTura | which codec would be recommended btw? GSM, H.232(or something like that) or what? Can't i use OGG Vorbis or is that not suitable for the job? (OGG Vorbis is afaik the best codec there is for low qual audio) |
| 17:14.15 | Hmmhesays | heh |
| 17:14.18 | eKo1 | ogg vorbis |
| 17:14.25 | eKo1 | ? |
| 17:14.34 | SkalTura | [TK]D-Fender: IVR driver? |
| 17:14.39 | [TK]D-Fender | SkalTura : If VoIP is only internal to your LAN then G.711 |
| 17:14.40 | Hmmhesays | you probably want to go with a real voice codec for 1 |
| 17:14.45 | coppice | the best codec I know for low qual audio is LPC 10 :-) |
| 17:14.56 | Dandan | lol |
| 17:14.57 | Dandan | :) |
| 17:14.59 | Hmmhesays | yeah if you want your audio as low quality as they come |
| 17:15.02 | [TK]D-Fender | SkalTura : using menus to give info to your callers. You are mentioning a lot of LINES, but very few phones... |
| 17:15.07 | SkalTura | [TK]D-Fender: nope, not at the first, but later on we would be having other numbers to do automatic thing like that (requesting information related to our products etc) |
| 17:15.08 | Dandan | two cans and a few feet of string |
| 17:15.22 | SkalTura | [TK]D-Fender: nope, over the net will go some traffic if just possible |
| 17:15.26 | *** join/#asterisk kuku5 (i=kuku@c-67-175-218-223.hsd1.il.comcast.net) |
| 17:15.33 | Hmmhesays | those were they days huh, Dandan |
| 17:15.33 | [TK]D-Fender | coppice : Close... that should read "the lowest quality codec I know is LPC10" :D |
| 17:15.44 | Dandan | Hmmhesays: yeah indeed :) |
| 17:16.00 | [TK]D-Fender | SkalTura : Then do you mean 100+ DID's, but fewer lines? |
| 17:16.01 | Hmmhesays | back with the girls still had cooties |
| 17:16.02 | kuku5 | I have a problem with moh, I need 2 phones to have different moh - I tried everything I can come up with. |
| 17:16.23 | SkalTura | [TK]D-Fender: DID? no, a PRI line or what it is called, 100e for 100 pairs |
| 17:16.29 | Dandan | oh, btw. is it possible to have two zones for overhead paging? |
| 17:16.30 | SkalTura | 100e or so per month here |
| 17:16.31 | kuku5 | ...but it jus plays the moh from that everyone else has |
| 17:16.32 | [hC-] | Hmmhesays: how do you find using voice gateways as opposed to zap? I have a hell of a time dealing with echo and line distortion and such on zap, and its really starting to get on my nerves |
| 17:16.33 | Hmmhesays | set your moh before cmd dial kuku5 |
| 17:16.48 | kuku5 | Hmmhesays: what if it is an incoming call ? |
| 17:16.59 | [TK]D-Fender | [hC-] : Handful of analog or PRI? |
| 17:17.02 | Hmmhesays | incoming is relative to your position kuku5 |
| 17:17.16 | kuku5 | incoming off of a pri |
| 17:17.27 | Hmmhesays | [hC-]: i haven't run into echo problems ever on 2nd gen quintum |
| 17:17.28 | coppice | [TK]D-Fender: i know much worse codecs than LPC10 :-) |
| 17:17.28 | SkalTura | [TK]D-Fender: i wish i could answer to that, my client said he is going to order such a line soon, and didn't even know himself more |
| 17:17.36 | *** join/#asterisk DougRoyer (i=doug@24-119-163-152.cpe.cableone.net) |
| 17:17.38 | [hC-] | [TK]D-Fender: analog. I just did an install in costa rica with 4 lines, and while i realize im talking about the worst combo possible, i used x100p clones, which suck ass, but then i couldnt get the tdm400p working with the dell i had used either |
| 17:17.53 | SkalTura | just a sec guys --> 4th payment today and gotta do verifications and mark it paid |
| 17:18.11 | [TK]D-Fender | SkalTura : Well thats the first question. What do we need. THEN you can ask what will do the job with it. |
| 17:18.18 | Hmmhesays | buy an axm 400, the echo can code is pretty good |
| 17:18.24 | [hC-] | [TK]D-Fender: but just the whole thing has issues so far, occasional entire line distortion |
| 17:18.43 | [hC-] | just looking for the "best" solution |
| 17:18.43 | SkalTura | brb got to handle this order (5mins) |
| 17:18.48 | *** join/#asterisk roulduke_ (i=1189rod5@p508D09FD.dip0.t-ipconnect.de) |
| 17:18.50 | Dandan | anyone using voicetronix stuff? |
| 17:18.54 | [TK]D-Fender | [hC-] : Sangoma's A200 w/ HWEC are coming soon... I'm definately going to give one a try... |
| 17:19.17 | Hmmhesays | [hC-] I do grass hut installs where conditions are less than ideal all the time |
| 17:19.36 | file | Hmmhesays: !!! |
| 17:19.41 | *** join/#asterisk daguz (n=leo@208-1-63-34.celito.net) |
| 17:19.41 | Hmmhesays | file! |
| 17:19.48 | mog_work | FILE!!!! |
| 17:19.52 | [hC-] | Yeah me too. I use sangoma a102us for PRI, and i like them a lot. I do get the occasional call drop, but im not sure if thats the sangoma, or the PRI interface itself, or asterisk, or what. |
| 17:19.53 | file | OMG! HI EVERYONE!!! |
| 17:20.14 | SkalTura | ok done with the order |
| 17:20.22 | Hmmhesays | i've had much much better times using external gateways and keeping asterisk IP only |
| 17:20.33 | [hC-] | This might sound stupid, but can reversed line polarity destroy call quality on analog FXO? Cause with one of the jacks they had fucked it up and i wasnt sure which side was which (and didnt have a volt meter) |
| 17:20.39 | SkalTura | [TK]D-Fender: sorry i don't know :( it's a local telephone company (or started as such, biggest here in Finland, Elisa) |
| 17:20.56 | [hC-] | Hmmhesays: good to know, i hadnt considered voice gateways at all |
| 17:21.33 | blitzrage | Hmmhesays: what voice GWs do you like? |
| 17:21.42 | blitzrage | just curious... I don't do much hardware stuff |
| 17:21.42 | Hmmhesays | don't laugh now... |
| 17:21.51 | Hmmhesays | quintum has done well by me |
| 17:21.56 | Hmmhesays | 2nd gen quintum |
| 17:22.01 | blitzrage | nevermind heard of them... |
| 17:22.07 | Hmmhesays | and audiocodes |
| 17:22.16 | SkalTura | ok, should i get my arms in deep with * immediately and just "do it somehow", or spend time reading docs and a lot of them first? |
| 17:22.21 | Hmmhesays | 2nd gen quintum is waaay different than 1st gen |
| 17:22.28 | mog_work | do it SkalTura |
| 17:22.39 | SkalTura | mog_work: just get neck deep with it? |
| 17:22.45 | blitzrage | I'm curious to test the new AsterBank from Xorcom |
| 17:22.57 | [hC-] | Me too |
| 17:22.59 | blitzrage | SkalTura: don't implement into production right away, but just pick something you want to do, and do it |
| 17:23.12 | [hC-] | It will either be really friggin awesome, or really friggin horrible |
| 17:23.13 | mog_work | yup |
| 17:23.15 | blitzrage | using documentation for reference |
| 17:23.19 | kuku5 | Hmmhesays: thanks - that works! |
| 17:23.22 | mog_work | do it for your house |
| 17:23.26 | Hmmhesays | np kuku5 |
| 17:23.29 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 17:23.31 | SkalTura | blitzrage: okies, should i tell the client that more time is needed and first ship me some friggin' hardware and i get on with it? |
| 17:23.37 | Hmmhesays | send me a postcard |
| 17:23.41 | mog_work | yeah SkalTura |
| 17:23.45 | SkalTura | okies |
| 17:23.46 | SkalTura | i'll do that |
| 17:23.48 | mog_work | dont do it for someone right away |
| 17:23.52 | mog_work | do it for yourself though |
| 17:23.53 | blitzrage | SkalTura: pretty much -- I don't recommend saying you can implement Asterisk until you've atleast done it once :) |
| 17:23.59 | blitzrage | totally agreed with mog_work |
| 17:24.07 | SkalTura | hmmph, first i thikn i use some software SIPs with it tho |
| 17:24.11 | SkalTura | any recommendations? |
| 17:24.19 | blitzrage | I like the X-ten softphone |
| 17:24.21 | mog_work | i mean you getting a dropped call or crashed asterisk is a lot different than someone else |
| 17:24.24 | mog_work | i like linphone |
| 17:24.40 | wasim | moziax! |
| 17:24.43 | SkalTura | blitzrage: i never did say that i know, actually i said i don't know ANYTHING about PBX, VoIP etc. |
| 17:24.50 | blitzrage | I used that too wasim -- its decent :) |
| 17:24.52 | SkalTura | blitzrage: and he insisted that i work on that |
| 17:25.01 | blitzrage | SkalTura: awesome! incentive! :) |
| 17:25.02 | mog_work | ooh moziax |
| 17:25.08 | mog_work | i havent played with that |
| 17:25.13 | wasim | jdg is a neat fellow as well, very helpful |
| 17:25.21 | blitzrage | SkalTura: especially if they'll send you some hardware |
| 17:25.25 | SkalTura | blitzrage: basicly, result can be anything, and it will always be better than expected ;) |
| 17:25.37 | SkalTura | where can i get X-ten? |
| 17:25.42 | SkalTura | googling! |
| 17:25.53 | SkalTura | any free ones? |
| 17:25.59 | _Sam-- | http://www.xten.com/index.php?menu=download |
| 17:26.01 | fugitivo | xlite is free |
| 17:26.06 | blitzrage | www.xten.com <-- x-lite is free |
| 17:26.11 | SkalTura | ok |
| 17:26.16 | [hC-] | wait... moziax.. |
| 17:26.19 | blitzrage | runs on Windows and Linux |
| 17:26.21 | [hC-] | I wonder if that is cross platform |
| 17:26.22 | SkalTura | just thought that was the site when i pressed on a advertisement |
| 17:26.25 | [hC-] | what about OSX! |
| 17:26.34 | [hC-] | oh "should be easy to port to mac" |
| 17:26.37 | _Sam-- | zoa is working on a really nice free softclient |
| 17:26.38 | [hC-] | damn them. |
| 17:26.41 | _Sam-- | called idefisk |
| 17:26.48 | _Sam-- | but its iax not sip, its really nice though |
| 17:27.01 | [hC-] | no mac version of it either |
| 17:27.02 | _Sam-- | i think its better than xlite, for free |
| 17:27.03 | [hC-] | bastard! |
| 17:27.13 | SkalTura | ok, what's this thing with IAX and SIP? |
| 17:27.16 | blitzrage | using clients that are a mix of SIP and IAX is probably a good way to learn anyways :) |
| 17:27.19 | SkalTura | what's the difference, which one i should go for? |
| 17:27.38 | zoa | [hC-]: it has a mac version |
| 17:27.47 | fugitivo | SkalTura: i use iax for remote clients, no problems with nat |
| 17:27.49 | blitzrage | SkalTura: two different protocols -- I like IAX for trunks (to providers if supported -- or between asterisk boxes) and SIP is usually end points (phones) |
| 17:27.50 | [hC-] | zoa: it says its coming soon on the website.. |
| 17:27.53 | fugitivo | SkalTura: sip for internal clients |
| 17:27.55 | [hC-] | er, on asteriskguru |
| 17:27.58 | zoa | want a prerelease ? :) |
| 17:28.06 | zoa | privmsg me your email addy |
| 17:28.08 | [hC-] | Yes please :) |
| 17:28.11 | blitzrage | SkalTura: you're going to wantto learn both |
| 17:28.26 | SkalTura | blitzrage: trunk being? sorry for such a newbie questions btw |
| 17:28.37 | blitzrage | SkalTura: check www.asterisk.org and www.asteriskdocs.org for documentation too |
| 17:28.37 | [hC] | I dont mind testing on OSX for you, Im on a 10.4.4 imac g5 |
| 17:28.37 | jbroome | oh man, you are *so* boned. :) |
| 17:28.58 | zoa | ~docs |
| 17:28.59 | jbot | hmm... docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:29.01 | file | blitzrage: I dare you to yell out in class, "SIP sucks" |
| 17:29.01 | blitzrage | SkalTura: trunk just meaning connection between servers |
| 17:29.09 | blitzrage | file: LOL |
| 17:29.09 | SkalTura | so, i should use IAX + SIP both of them, so i can link asterisk boxes all around the place + SIP for endpoints("clients", IP PHones etc)? |
| 17:29.16 | file | blitzrage: and see what oej does :P |
| 17:29.19 | blitzrage | file: funny enough we're talking about IAX2 right now |
| 17:29.29 | file | blitzrage: haha |
| 17:29.54 | blitzrage | SkalTura: use of IAX and SIP is usually very much a personal opinion -- both work well |
| 17:30.12 | file | SIP it to me. |
| 17:30.15 | SkalTura | is that O'reilly book, Asterisk highly recommended? |
| 17:30.20 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 17:30.22 | SkalTura | because i'll put my client to order that for me |
| 17:30.22 | file | yes |
| 17:30.25 | file | it's a great book |
| 17:30.34 | file | I am in no way biased. Honest. |
| 17:30.43 | blitzrage | I think its great too :) |
| 17:30.48 | zoa | that oreilly book is good yes |
| 17:30.52 | zoa | the pakt book is also good |
| 17:30.54 | file | zoatastic! |
| 17:30.57 | SkalTura | blitzrage: ok, any difference in configuration? (i trust the KISS filosophy: Keep It Simply, Stupid! so less configuration == better) |
| 17:30.58 | zoa | all other books are complete junk iirc |
| 17:31.29 | blitzrage | SkalTura: a bit... but the simplest configs are similar between the two |
| 17:31.46 | SkalTura | blitzrage: ok, what if we are talking highly complicated systems? |
| 17:32.12 | blitzrage | SkalTura: its the dialplan that makes the system complicated |
| 17:32.18 | zoa | id go for iax2 between asterisk servers |
| 17:32.20 | zoa | definately |
| 17:32.25 | SkalTura | blitzrage: like in my case: 100 incoming phone lines, VoIP phones, regular phones, call center functionality(+musiconhold naturally), product data request functionality, call recording |
| 17:33.03 | *** join/#asterisk mjmac (n=mjmac@cpe-69-207-227-173.maine.res.rr.com) |
| 17:33.24 | SkalTura | aah, i love that einstein quote on the sample chapter: "Everything should be made as simple as possible, but not simpler" |
| 17:33.44 | blitzrage | SkalTura: configuring peers doesn't increase in complexity when you have more end points -- just more lines :) |
| 17:34.22 | SkalTura | blitzrage: ok |
| 17:34.43 | *** join/#asterisk oej (n=oej@199.227.185.35) |
| 17:34.49 | zoa | hey ho oej |
| 17:35.08 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 17:35.13 | *** join/#asterisk lahaine (n=lahaine@200.65.119-80.rev.gaoland.net) |
| 17:35.22 | SkalTura | uargh, f*cking warm up already -.- |
| 17:35.43 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:36.56 | Netgeeks | my ink jet printer has a duplexer, but sometimes it's a little slow, especially when it takes coffee and smoke breaks |
| 17:37.26 | [TK]D-Fender | Laser = good <- |
| 17:37.55 | SkalTura | hahah Netgeeks x) |
| 17:38.15 | *** part/#asterisk fild (i=fdello@adae86.neoplus.adsl.tpnet.pl) |
| 17:38.20 | SkalTura | lol, within 2 weeks i've printed now about 300 pages |
| 17:38.28 | SkalTura | gotta go & buy meh some more paper |
| 17:38.45 | blitzrage | SkalTura: sample chapter? |
| 17:39.09 | blitzrage | print the whole book! :) www.asteriskdocs.org <-- read the book online |
| 17:39.13 | blitzrage | or buy it and make me happy :D |
| 17:39.34 | mog_work | hey blitzrage i think we bought a few hundred for ya |
| 17:39.39 | SkalTura | wtf, 2 pages missing oO; (other side of the pages) |
| 17:39.54 | Netgeeks | I just love these companies that tell you, "we'll have a tech out there between 7Am and 6PM on wednesday".... come on, it's not like I"m asking the tech to come fix something in moscow.. can't you get a little more granular than that? |
| 17:39.55 | SkalTura | that book is there? :O |
| 17:41.06 | Netgeeks | So I get my arse into the office at 645AM knowing sure well the tech will be here at 645pm but if I'm not here at 7a sharp, the tech will arrive find I'm not here and depart and cancel the appointment... *sigh* |
| 17:41.28 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:42.12 | SkalTura | lol |
| 17:42.47 | SkalTura | wtf is adobe doing, for some reason, some pages are empty! |
| 17:44.09 | *** join/#asterisk twisted-- (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 17:44.09 | *** mode/#asterisk [+o twisted--] by ChanServ |
| 17:44.14 | Netgeeks | what does the pdpc stand for in pdpc.professional.twisted? |
| 17:44.27 | file | peer directed projects center |
| 17:44.39 | blitzrage | file: you spelt centre wrong |
| 17:44.45 | wasim | :P |
| 17:44.48 | file | I'll spell it however I want |
| 17:44.52 | Netgeeks | ah, thanks, for some reason I kept thinking he was some special version of an old mini-mainframe |
| 17:44.54 | blitzrage | lol |
| 17:45.09 | file | blitzrage: eh Life? :P |
| 17:45.22 | blitzrage | *shakey fist* |
| 17:45.30 | file | muahahahaha |
| 17:45.45 | blitzrage | Qwell: I couldn't send you an email to your email address |
| 17:46.01 | blitzrage | Peers and Users and Registrations, Oh mt! |
| 17:46.06 | blitzrage | s/mt/my |
| 17:46.20 | file | blitzrage: :P |
| 17:47.59 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 17:50.59 | SkalTura | hmmph, this'll be good bed time read :) |
| 17:52.03 | SkalTura | hey, can i somehow link skype with asterisk easily? ie, forward skype incoming calls to asterisk and where ever i want to? |
| 17:52.19 | mog_work | no |
| 17:52.44 | SkalTura | damn |
| 17:53.04 | *** join/#asterisk Entropy23 (n=Entropy2@p5492B28A.dip0.t-ipconnect.de) |
| 17:54.12 | *** join/#asterisk RoyK (n=roy@ti211310a080-9456.bb.online.no) |
| 17:56.30 | *** join/#asterisk JMcA (n=jmcadams@71.31.33.169) |
| 17:56.47 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:58.22 | *** join/#asterisk rkioko (n=rkioko@196.200.26.42) |
| 17:58.54 | Einar__ | Can anyone recommand a ISDN card that works with Zaptel? I've got one Billion that works fine, it's just hard to find another one (need 4 lines). |
| 18:03.15 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 18:04.47 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:05.20 | MadCamel | *giggle* I could probably make a hardware-based hackomatic skype<->* bridge using nothing but the stuff under my desk: spare sipura box, skype compatible USB handset, old phone, ducktape. |
| 18:06.09 | Netgeeks | yikes |
| 18:07.14 | MadCamel | it's the electricly conductive type of ducktape too. haha |
| 18:07.41 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 18:09.48 | *** join/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 18:10.19 | JMcA | you could look at what, for example, kopete is doing with skype integration...personally, however, I'll just drop skype on the ground and not worry about it |
| 18:10.30 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 18:10.41 | *** join/#asterisk ceeto (i=cio@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 18:11.36 | ceeto | Hey guys.. I'm using the fax functionality provided by app_rxfax.so and am having a problem. If I run asterisk from the command line as root, it works fine and drops the file in my fax dir. If I let it start as a service running as user asterisk, it doesn't write the file. Any suggestions? |
| 18:11.46 | *** join/#asterisk Lee__ (n=lee@66.11.200.130) |
| 18:12.11 | JMcA | ceeto: that immediately begs the question of what the permissions on the directory are |
| 18:12.26 | ceeto | asterisk asterisk 777 |
| 18:12.31 | ceeto | (for troubleshooting purposes) |
| 18:12.42 | SkalTura | parent dir perms... *whistles* |
| 18:13.14 | ceeto | Skaltura: 755 asterisk asterisk |
| 18:13.35 | ceeto | No errors in * when I set verbose 6 ... |
| 18:13.39 | Lee__ | is it enough to set the SIP caller ID to "Anonymous" <NXXNXXXXXX> to provide users with "caller ID blocking"? |
| 18:13.41 | ceeto | It acts like its working just fine ... |
| 18:13.48 | ceeto | Just never writes the file. |
| 18:15.11 | JMcA | I don't know enough about rxfax to really be able to troubleshoot it...sounds like directory/file permission, but those permission sets should be good |
| 18:15.17 | JMcA | ceeto: what distro? |
| 18:15.40 | ceeto | debian sarge with the latest zaptel but asterisk 1.0.7 from .deb file .. |
| 18:16.11 | ceeto | Yea, it's been a struggle.. Don't want to run * as root all the time.. |
| 18:16.16 | ceeto | Gotta be permissions. |
| 18:16.18 | JMcA | wouldn't be an selinux thing, then, I wouldn't think, though I'm not up to date on the latest debian mojo |
| 18:16.18 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
| 18:16.35 | ceeto | No, I don't have selinux enabled in my kernel (stock debian 2.6) |
| 18:17.40 | Lee__ | ceeto, I am running asterisk + zaptel on sarge. the packages have some permissions problems. |
| 18:18.09 | Lee__ | they can be fixed manually |
| 18:18.25 | ceeto | Lee - that's getting closer to a possible fix - do you remember what you fixed? |
| 18:20.28 | *** join/#asterisk darby_t (i=darby_t@dkh177.neoplus.adsl.tpnet.pl) |
| 18:20.47 | Lee__ | actually no. I installed 1.2.1 backports last night. But those came from sid so I think they have different errors. |
| 18:20.53 | Lee__ | Can you describe your error again? |
| 18:21.43 | ceeto | No specific error at all. When I let the init script start asterisk as user 'asterisk' my faxes don't write in my fax dir which has wide open permissions (asterisk asterisk 777 for testing purposes) .. When I run asterisk as root just from the command line it works fine... |
| 18:22.24 | Lee__ | your fax dir has a permissions problem. turn on debugging and tail your log. |
| 18:22.33 | ceeto | I'm thinking it's a spandsp permission problem but don't know enough about it to troubleshoot properly... |
| 18:23.16 | Lee__ | does the console output anything about recieving a fax? |
| 18:23.29 | ceeto | Yes, just like it would if it was working. No errors. |
| 18:23.57 | *** part/#asterisk mikeyb_work (n=michael@66-193-82-211.gen.twtelecom.net) |
| 18:24.32 | ceeto | spandsp file permissions look fine ... |
| 18:25.23 | Lee__ | it cannot be anything else if the fax is recieved and it isn't writing the file. It /must/ be permissions. you may even be looking in the wrong place if there isn't an error logged. |
| 18:26.22 | ceeto | Well, when I run as root, the file drops in my /usr/share/asterisk/fax dir normally. When I run from the init script, which uses user 'asterisk', it doesn't write to that file. That's the only thing changing from broken to working. I agree it's a permissions problem, but where is what's killing me :) |
| 18:26.32 | ceeto | Where would the fax libs log to? |
| 18:26.37 | ceeto | Thanks for the help, btw! |
| 18:28.08 | ceeto | Specifically, where would app_rxfax.so log to? |
| 18:29.14 | JMcA | perhaps there's a temp file location that its not able to write to? I'd have no clue where that'd be, though |
| 18:29.28 | ceeto | That makes sense... hrm... |
| 18:29.28 | *** join/#asterisk Coder (n=WisKi@85.106.171.225) |
| 18:29.29 | ravenpi | I bet that that is it -- might there be one in /var/spool or somesuch? |
| 18:30.38 | JMcA | ceeto: worse case, strace should turn it up, if you can sort out all the noise |
| 18:31.05 | ravenpi | God bless strace. Don't forget the "-s 128" or so option, to that you can see *all* of what's going on. |
| 18:31.06 | ceeto | Yea, I was hoping for the lazy fix.. hehe.. |
| 18:32.28 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 18:32.40 | KriS83 | Hi |
| 18:33.50 | JMcA | ravenpi: ya...strace is a life-saver |
| 18:34.10 | ceeto | I'd rather have you guys take turns beating me with a C++ book. |
| 18:34.28 | JMcA | nah...strace isn |
| 18:34.36 | Hmmhesays | can anyone explain to me what exactly overlap dial is in the t1/e1 world? |
| 18:34.43 | JMcA | 't terrible...you just get copious output to sort through |
| 18:34.43 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 18:35.34 | JMcA | Hmmhesays: it means that the next digit(s) is(are) sent before the previous one(s) is(are) acknowledged |
| 18:35.45 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 18:36.00 | Hmmhesays | hmm, k |
| 18:37.07 | ravenpi | I've never quite understood what the difference between "overlapial = yes" and "immediate = no" is... |
| 18:37.42 | ceeto | Hey all - Are my config files for 1.07 compatable with 1.21? i.e., if I was to do an upgrade, would I be chasing down config file problems? |
| 18:37.42 | JMcA | dunno...I'm a relative newb at *, so I haven't yet looked at those settings |
| 18:37.47 | KriS83 | I have just updated Asterisk von 1.2.0 to 1.2.1 (due to a Junghanns qozap update)... I use Playtones(ring) in my Dialplan for the IVR, but after the update the quality of "ring" is really really bad. any ideas? |
| 18:41.27 | *** join/#asterisk wizard545 (n=wizard@tor/session/x-1e49f3f5c1939875) |
| 18:41.47 | ceeto | f*ck it, I'm going to the helicopter field.. later all |
| 18:42.34 | wizard545 | hey guys, quick question... what additional hardware do i need for a asterisk system that will take incoming calls on sip, and re-route them to another sip gateway? |
| 18:42.55 | Lee__ | ceeto, do you have a /var/spool/asterisk directory? |
| 18:43.02 | wizard545 | do i need a card to do that? it's just relaying the calls to the best gateway |
| 18:43.12 | ceeto | Yep, that's where my faxes go /var/spool/asterisk/fax |
| 18:43.13 | oogle | wizard545: if it's pure voip, you don't need hardware |
| 18:43.35 | oogle | wizard545: unless you're sending calls to a sip gateway in to the phone ports |
| 18:43.46 | brookshire | wizard: you don't need any hardware.... |
| 18:43.58 | ceeto | Permissions aren't inherited with *nix are they? My /var/spool is 755, but the fax dir below that is 777 ... |
| 18:44.04 | brookshire | now if you try and do meetme.. you might have some timing issues without hardware for timing |
| 18:44.19 | JMcA | ceeto: no, that should be fine |
| 18:44.27 | iCEBrkr | <PROTECTED> |
| 18:44.54 | oogle | ceeto: permissions for a file are usually based on umask |
| 18:45.00 | Lee__ | ceeto, make sure the user asterisk has permissions to read all along that path and write to the final directory |
| 18:45.46 | JMcA | and for good measure, make sure it has "x" all along the path, too |
| 18:46.03 | wizard545 | oogle, i just want to relay the calls from one sip incoming to another sip outgoing, but drop them from the first line so it can get more calls |
| 18:46.11 | *** join/#asterisk jtodd (n=jtodd@dsl027-191-178.sfo1.dsl.speakeasy.net) |
| 18:46.15 | ceeto | is there a way I can test 'write' a file with a specific user? Almost like a reverse su? |
| 18:46.24 | Lee__ | ceeto, sudo |
| 18:46.29 | wizard545 | a calling card sort of thing |
| 18:46.30 | JMcA | ceeto: su - <user> works just fine |
| 18:46.49 | shmaltz | what motherboard whould you guys recommend if I plan on using a Digium t1 card? |
| 18:46.53 | JMcA | though if asterisk doesn't have a reasonable shell, that may not be terribly useful |
| 18:47.51 | wizard545 | how many calls could a p4 box hold? simultaneously anyone have any idea? i search and all i could find were bandwidth calcs |
| 18:48.02 | *** join/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 18:48.02 | ceeto | <PROTECTED> |
| 18:48.06 | ceeto | seems to work |
| 18:48.31 | ceeto | Yep, I can touch a file all the way to /var/spool/asterisk/fax |
| 18:48.35 | JMcA | just "su <user>" would just change permissions without trying to start that user's shell, too |
| 18:48.35 | ceeto | as user "asterisk" |
| 18:48.47 | JMcA | fwiw |
| 18:49.36 | ceeto | Oh well, I'm going to build a 1.21 box anyway, now is as good a time as any ... |
| 18:49.39 | *** join/#asterisk lilliput (n=lilliput@82-47-147-63.cable.ubr11.brad.blueyonder.co.uk) |
| 18:49.48 | lilliput | hi all |
| 18:51.15 | *** join/#asterisk xp_ (n=wgheath@67-102-228-17.adsl.lbdsl.net) |
| 18:51.29 | Netgeeks | anyone here use junction networks with IAX? |
| 18:52.04 | Lee__ | ceeto, there are good backports if you don't feel like compliing, search google. I found one at peen.net |
| 18:52.04 | lilliput | I am searching some documentation / link / script with asterix to play around with an interractive answering machine or bot |
| 18:52.11 | xp_ | I have a channel bank and if my server fails over to the other server I am told I will loose all my calls, is there any way to avoid that? |
| 18:53.05 | JMcA | xp_: move the gatewaying functions off the server...that just moves the failure point to the device doing the gatewaying...though you might find they're more reliable *shrug* |
| 18:53.55 | JMcA | xp_: ultimately, telecom circuits have to land somewhere, and that somewhere is gonna be a point that will drop the calls on those circuits if it coughs up a hairball |
| 18:57.01 | *** join/#asterisk wiz (i=debian-t@tor/session/x-f74705998e795250) |
| 18:57.57 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:58.04 | *** join/#asterisk slayer192 (n=chrisc@66.138.39.225) |
| 19:02.34 | xp_ | JMcA what hardware would accomplish that? |
| 19:02.38 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 19:02.40 | Netgeeks | Any reason why IAX dtmf wouldn't trigger an ivr option? Got an ivr that works fine on a local tdm400 incoming line, but doesn't work via IAX. Asterisk 1.2.1 |
| 19:04.10 | xheliox | Netgeeks: I've had problems with DTMF with IAX when using Jitterbuffer -- though I think it has more to do with my provider than anything else. |
| 19:04.20 | Netgeeks | IAX2 debug shows one of these every time a key is pressed (Rx-Frame Retry[ No] -- OSeqno: 002 ISeqno: 001 Type: DTMF Subclass: 0) |
| 19:04.32 | *** join/#asterisk Amilcar_ (n=xxxxx@201.14.113.52) |
| 19:04.48 | Netgeeks | hrm, jitterbuffer, I seem to remember hearing about that, who is your provider? |
| 19:06.52 | Amilcar_ | i need help. I've setup a queue in rrmemory. Everything works fine, except asterisk keeps sending incoming calls for members that are already in a call!!!! |
| 19:08.02 | Amilcar_ | Please, anyone could help? I don't know what to do! The members are using eyebeam softphones. So the calls are routed to then even when they are already on a call. |
| 19:08.19 | Katty | is there an equivilent for 'tasklist' in linux? |
| 19:08.28 | *** join/#asterisk cyrax^ (n=cyrax@host161-247.pool870.interbusiness.it) |
| 19:08.31 | Netgeeks | top |
| 19:09.01 | Katty | but will it let me connect to /another/ computer? |
| 19:09.11 | Netgeeks | ah! no |
| 19:09.40 | Katty | sad. |
| 19:10.14 | Katty | tasklist has this neat little thing... tasklist /S netboisname /U domain\username /P password |
| 19:10.46 | Netgeeks | I'm sure there are several different apps out there that will do what you want, however, none are going to be installed and running by default |
| 19:10.48 | *** join/#asterisk ronn (n=ronn@62-249-247-240.no-dns-yet.enta.net) |
| 19:10.55 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:11.02 | Katty | oh ah (= |
| 19:11.09 | Netgeeks | you are besically going to need to run some kind of service on the computers upi want to remotely look at |
| 19:11.13 | rob0 | ssh user@host top |
| 19:11.24 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:11.44 | rob0 | (probably need -t or something for that ssh) |
| 19:12.38 | *** join/#asterisk Paulo (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 19:12.41 | rob0 | (maybe consider a script to grab exactly what you want, run instead of top) |
| 19:13.27 | ronn | is there a way to put an IP range in sip.conf host= some IP range ? |
| 19:13.38 | Netgeeks | um |
| 19:13.49 | Netgeeks | depends what you are trying to achieve |
| 19:14.03 | Amilcar_ | Please, anyone could help me in a queue issue? |
| 19:14.18 | Netgeeks | ronn: what is your goal by supplying a range of IPs? |
| 19:14.29 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 19:15.24 | shmaltz | what motherboard whould you guys recommend if I plan on using a Digium t1 card? |
| 19:15.33 | Amilcar_ | Please, anyone could help me in a queue issue? |
| 19:15.43 | PauloS | Hi all. |
| 19:16.39 | ronn | Netgeeks: I will like allow calls from a company who will be sending me calls from any of its 30 boxes (30 different IPs) |
| 19:17.06 | Katty | ahahahaha. |
| 19:17.09 | Katty | hahahahahah |
| 19:17.13 | Katty | taskkill /S |
| 19:17.18 | ronn | so creating 30 entries for one account is a waste of time |
| 19:17.19 | Katty | </evil> |
| 19:17.31 | shmaltz | oej, have you ever gotten 100% on zttest? |
| 19:17.48 | shmaltz | wrong window |
| 19:18.02 | PauloS | Somebody can point me documentation about Brazilian signalling (mfc/R2) ? |
| 19:18.08 | Netgeeks | ronn, you should be able to do that by breaking your entries into a user and peer entry instead of one friend if you haven't already done that |
| 19:18.39 | *** join/#asterisk razu (n=razu@217-159-181-27-dsl.prn.estpak.ee) |
| 19:18.46 | Amilcar_ | PauloS: www.soft-switch.org |
| 19:18.49 | Netgeeks | then in the user entry, you should be able to not choose a host directive and force asterisk to accept calls based upon authentication, use username and secret to set those |
| 19:18.57 | ronn | Netgeeks: how do u mean by breaking the entries? |
| 19:19.15 | rob0 | Katty, all my mumbling above was to you |
| 19:19.28 | Netgeeks | in sip.conf, you have an entry for your company, right? |
| 19:19.31 | Netgeeks | [company] |
| 19:19.34 | Netgeeks | type=friend |
| 19:19.36 | ronn | <PROTECTED> |
| 19:19.37 | Netgeeks | blah blah blah |
| 19:19.51 | Netgeeks | ewwww |
| 19:20.18 | ronn | i need to use insecure = very |
| 19:20.24 | ronn | and specify IP |
| 19:20.29 | Netgeeks | yep |
| 19:20.46 | Netgeeks | but you cannot specify a range for host |
| 19:20.49 | PauloS | I've tryed the recipe at www.soft-switch.org... But seems that it doesnt work with recent asterisk source tree... |
| 19:20.54 | Katty | rob0: kthx. |
| 19:20.55 | ronn | but I wanted to use host = IP range in stead of creating one for each IP |
| 19:21.09 | Amilcar_ | I have a rrmemory queue. All the members are using eyebeam softphones. But the calls are routed to then even when they are already on a call. show queues shows that the member is in-use, but despite that, it contiunes to route incoming calls to the member. Is that normal behaviour?? |
| 19:21.12 | Netgeeks | host=dynamic allows a permit=a.b.c.d/cidr |
| 19:21.23 | Netgeeks | but I don't think that will work for your needs, you could try it |
| 19:21.46 | *** join/#asterisk mjmac (n=mjmac@cpe-69-207-227-173.maine.res.rr.com) |
| 19:21.58 | ronn | <PROTECTED> |
| 19:22.08 | Netgeeks | so if you wanted to allow calls from 10.0.0.0 255.255.255.0 you would have something like this |
| 19:22.11 | *** join/#asterisk mjmac (n=mjmac@cpe-69-207-227-173.maine.res.rr.com) |
| 19:22.12 | Netgeeks | host=dynamic |
| 19:22.19 | Netgeeks | permit=10.0.0.0/24 |
| 19:22.30 | Netgeeks | can't say that will do what you want, but it would be worth a try |
| 19:23.02 | ronn | Netgeeks: i will definitely give it a try. thanks for your tip. |
| 19:23.09 | cyrax^ | Amilcar_, is your problem similar to http://bugs.digium.com/view.php?id=6111 ? |
| 19:23.15 | Netgeeks | best way to find out if it works, is first test with all the machines, then test with a client outside the range to make sure it's blocked |
| 19:23.36 | Netgeeks | otherwise, you might have to go neat tricks with iptables |
| 19:24.13 | rob0 | Katty: one other nice little tip, ssh-keygen(1) and ssh-agent(1) for managing passwordless SSH logins. Gentoo has a "keychain" script which is a frontend to ssh-agent, very handy (and works in any sh/csh shell BTW.) |
| 19:24.29 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-81-201.tvcablenet.be) |
| 19:24.57 | lilliput | nobody to tell me about the scripts ? |
| 19:25.20 | Amilcar_ | cyrax^: yep! |
| 19:25.26 | cyrax^ | good :) |
| 19:25.34 | cyrax^ | as you can see I sent a patch there |
| 19:25.50 | rob0 | lilliput: the wiki? |
| 19:26.00 | cyrax^ | could you please test it and send feedback on the bugtracker ? :) |
| 19:26.15 | Amilcar_ | cyrax^: but there is no other way to asterisk don't route calls to a member that's already in a queue routed call? |
| 19:26.42 | Katty | rob0: you sure are talking over my head (= |
| 19:26.48 | Katty | Netgeeks: thanks for keeping it simple. |
| 19:27.19 | cyrax^ | Amilcar_, I don't know.. anyway the sip channel has a bug using the "inuse" counter of a user.. |
| 19:27.20 | rob0 | oops :) |
| 19:27.34 | cyrax^ | and the patch should solve that problem.. |
| 19:28.00 | Amilcar_ | cyrax^: ok! thank you! |
| 19:28.15 | *** join/#asterisk XARiUS (n=bdarcy@adsl-69-232-66-200.dsl.sndg02.pacbell.net) |
| 19:28.54 | lilliput | rob0, which one ? |
| 19:29.00 | ronn | Netgeeks: host = hostname and then the permit = IP might do it. |
| 19:30.03 | ceeto | Anyone have an example debian init script to start asterisk? |
| 19:30.31 | rob0 | ~wikis |
| 19:30.33 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 19:31.12 | XARiUS | best recommendation for cheap desk phone, budgetone? |
| 19:31.25 | rob0 | <== once thrown out of a boarding house for spreading roomers |
| 19:32.03 | lilliput | thx going o read a bit |
| 19:32.29 | malverian[work] | Any PRI experts here? I'm trying to find out if there is some kind of hardware device that will allow two computers to have access to the same T1 PRI. |
| 19:32.57 | denon | malverian[work]: sure, use asterisk to route them |
| 19:32.57 | justinu | eh? |
| 19:33.20 | XARiUS | ooo.. anyone tried the Linksys - Sipura SPA-841 with asterisk yet? |
| 19:33.34 | malverian[work] | Basically what i have is a "production" asterisk box, and a "testing" asterisk box.. I usually just wait until after hours to test the "testing" box. |
| 19:33.57 | malverian[work] | But I'd like to be able to use it during the day. And I don't want to have it go through the production box, because that's not a valid test of direct PRI usage.. |
| 19:34.38 | *** join/#asterisk SS||Wing|onsite (n=wing@ssrtr.sharpshooterusa.com) |
| 19:34.44 | justinu | you'd have to run a PRI from your test box to your prod box |
| 19:34.52 | justinu | if you want to test like that |
| 19:35.03 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 19:35.40 | *** join/#asterisk KriS83 (n=KriS@212.202.141.92) |
| 19:35.48 | [av]bani | yay got auto-provisioning working with spa-3000 |
| 19:35.52 | Amilcar_ | cyrax^: i think you're right and the patch works ok!! :-) |
| 19:35.57 | cyrax^ | oky :) |
| 19:36.25 | Amilcar_ | cyrax^: that's not exactly what i want, but it DOES solve a problem! :-) |
| 19:36.34 | cyrax^ | could you please test it a bit and send a feedback on the tracker ? :) |
| 19:37.00 | *** join/#asterisk crich1999 (n=crich@port-212-202-0-102.dynamic.qsc.de) |
| 19:37.01 | malverian[work] | justinu, There is no hardware device that would allow what I'm saying? |
| 19:37.13 | justinu | nothing besides a switch |
| 19:37.16 | malverian[work] | I know there are "failover" devices that let you plug two computers into a PRI, and it uses a keepalive.. |
| 19:37.28 | justinu | hmm, never seen that |
| 19:37.35 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 19:37.37 | SS||Wing|onsite | Guys, please advise! I am running * with Polycom IP500's, everything is working on LAN. I have 1 phone outside the network via IPSec tunnel. I can ring the phone, but when a person picks up I can hear them, BUT they can not hear me. Any ideas? |
| 19:38.19 | Amilcar_ | cyrax^: i will! :-) |
| 19:38.23 | ceeto | What version of mp3123 are you guys using for 1.2.x? |
| 19:38.40 | lilliput | ok now I know the name .. is there any small ivr scripts around (ie dtmf interraction) |
| 19:38.51 | cyrax^ | Amilcar_, thank you :) |
| 19:39.15 | dogtanian | *nick is testing irssi* :) |
| 19:39.16 | lilliput | that i could try and modify |
| 19:39.24 | dogtanian | nick - do Alt+2 again |
| 19:39.27 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 19:40.01 | asterboy | any mac users in here? |
| 19:41.06 | malverian[work] | justinu, http://www.junghanns.net/en/ISDNguard_produkt.html |
| 19:41.19 | wasim | asterboy: benjk but he's sleeping |
| 19:41.31 | asterboy | ya, thats who I was hoping to poll. |
| 19:41.32 | justinu | malverian[work]: interesting |
| 19:42.31 | *** join/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 19:42.42 | *** part/#asterisk Lurr (i=user@adsl-067-034-122-207.sip.mia.bellsouth.net) |
| 19:43.02 | *** join/#asterisk BBRdiguez (n=BBRdigue@p54B0123C.dip0.t-ipconnect.de) |
| 19:43.17 | SS||Wing|onsite | anyone? |
| 19:43.43 | Dandan | how to force asterisk to compile vpb module? |
| 19:43.53 | Amilcar_ | cyrax^: Let me ask you... The wrapuptime setting in queues.conf. That setting will never work if asterisk keeps sending incoming calls to every member (despite inuse or not!).... |
| 19:44.06 | *** join/#asterisk alrs (n=lars@69-160-242-101.vnnyca.adelphia.net) [NETSPLIT VICTIM] |
| 19:44.08 | *** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 19:44.24 | Amilcar_ | cyrax^: so queues have to automatically check inuse status before routing a call to a member. Don't you think? |
| 19:44.29 | asterboy | Just need to know if any standard vga monitor will plug into a mac. |
| 19:44.47 | cyrax^ | hmm |
| 19:44.51 | Dandan | asterboy: only with a converter |
| 19:44.53 | Amilcar_ | cyrax^: i don't need to hardcode call-limit in sip.conf in that case, do i? |
| 19:44.55 | Dandan | adapter that is |
| 19:45.00 | asterboy | ok, thx |
| 19:45.01 | *** join/#asterisk gammacoder (n=chatzill@64-132-192-33.gen.twtelecom.net) |
| 19:45.02 | Dandan | afair macs have dvi output |
| 19:45.09 | Dandan | so you need dvi-to-vga converter |
| 19:45.22 | asterboy | thats what I was thinking. |
| 19:45.32 | asterboy | easy to pickup |
| 19:45.32 | Dandan | :) |
| 19:45.38 | SS||Wing|onsite | Guys, please advise! I am running * with Polycom IP500's, everything is working on LAN. I have 1 phone outside the network via IPSec tunnel. I can ring the phone, but when a person picks up I can hear them, BUT they can not hear me. Any ideas? |
| 19:45.41 | Dandan | now i need help witj vpb's :( |
| 19:45.54 | cyrax^ | Amilcar_, : i'm sorry but I don't know about that.. |
| 19:45.56 | asterboy | want to run ~vpb |
| 19:46.00 | asterboy | ~vpb |
| 19:46.19 | Dandan | SS||Wing|onsite: looks like a firewall issue... |
| 19:46.24 | *** join/#asterisk areski (n=areski@189.Red-83-60-89.dynamicIP.rima-tde.net) |
| 19:46.32 | Dandan | rather: rtp packets do not get to the server... |
| 19:46.32 | Dandan | :) |
| 19:46.32 | Amilcar_ | cyrax^: ok! :-) |
| 19:46.41 | Dandan | try playing with canreinvite |
| 19:46.44 | BBRdiguez | Hi everyone. Does anyone know how to disable FAS on sipura spa-3000 ? I got *->spa3k->PSTN and spa3k picks up before the PSTN call is connected. |
| 19:46.46 | Dandan | maybe that will help |
| 19:46.51 | asterboy | SS, run a diagnostic on the phone to verify the mic equipment is functioning. |
| 19:46.52 | cyrax^ | i just solved the problem with the inuse counter... ;P |
| 19:46.59 | Dandan | if not it is a regular issue with nat/firewall |
| 19:47.18 | cyrax^ | the check is made when a sip_call is started to a user.. |
| 19:47.20 | SS||Wing|onsite | asterboy: Thank you, already verified it, seems ok.. |
| 19:47.33 | Dandan | ~vpb |
| 19:47.36 | Dandan | ~voicetrnix |
| 19:47.38 | *** join/#asterisk dijit0 (n=dijit0@69.110.230.97) |
| 19:47.39 | Dandan | ~voicetronix |
| 19:48.03 | asterboy | SS, you ran the diagnostic from the MENU? |
| 19:48.22 | asterboy | and it spat out whatever you spat in? |
| 19:48.42 | asterboy | jbot, spat is when you spit while talking. |
| 19:48.43 | jbot | asterboy: okay |
| 19:49.03 | SS||Wing|onsite | Basically, when I dial from MY phone located on LAN, my mike works ok for anything inside LAN or outside, BUT when I call the phone located on the other side of IPSec, they can not hear me. |
| 19:49.26 | [av]bani | oh wow, snom supports xml microbrowser now |
| 19:49.48 | asterboy | hmmm, sounds like something is getting dropped...do you have access to a packet sniffer? |
| 19:49.57 | Dandan | ~spit |
| 19:50.01 | Dandan | ~spat |
| 19:50.02 | jbot | spat is, like, when you spit while talking. |
| 19:50.53 | Dandan | jbot, voicetronix is an aussie VoIP hardware manufacturer, go to http://www.voicetronix.com.au |
| 19:50.55 | jbot | okay, Dandan |
| 19:51.01 | Dandan | ~voicetronix |
| 19:51.02 | jbot | methinks voicetronix is an aussie VoIP hardware manufacturer, go to http://www.voicetronix.com.au |
| 19:51.54 | asterboy | Dandan, that voicetronix works with asterisk? |
| 19:52.08 | *** join/#asterisk RoyK (n=roy@ti211310a080-9456.bb.online.no) |
| 19:52.13 | asterboy | They list openPBX |
| 19:52.17 | *** part/#asterisk javar (n=javar@69.79.217.237) |
| 19:52.17 | shmaltz | anybody here using the D-Link 4 port fxo gateway? |
| 19:52.38 | Dandan | asterboy: yup |
| 19:52.48 | asterboy | ah, just found it. |
| 19:52.57 | asterboy | hmmm...how are the prices compared to digium |
| 19:52.58 | Dandan | i just get broken pipe |
| 19:53.51 | Dandan | when trying to initialize asterisk |
| 19:54.40 | xp_ | ok here is my problem, if I have 2 asterisk servers and one fails over to the other one I will loose all the calls that the other one had, is there any way to prevent this? |
| 19:55.31 | Dandan | Ouch ... error while writing audio data: : Broken pipe |
| 19:55.31 | Dandan | Warning, flexibel rate not heavily tested! |
| 19:55.32 | Dandan | Segmentation fault |
| 19:55.36 | Dandan | sh*t |
| 19:55.55 | Dandan | xp_: how do you think that can be prevented? |
| 19:55.56 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 19:56.30 | xp_ | some type of gateway in between the asterisk server and the t1 to ethernet or rhino channel bank? |
| 19:56.38 | ceeto | Yea - I just decided - screw the init scripts... inittab all the way baby! |
| 19:56.52 | asterboy | Dandan, there is something else more serious going on. |
| 19:56.59 | Dandan | xp_: no, I do not know of any such devices... keep asking |
| 19:57.05 | Dandan | asterboy: i figured |
| 19:57.06 | asterboy | Have you tried putting the openpci card in another slot |
| 19:57.08 | asterboy | ? |
| 19:57.14 | Dandan | nope |
| 19:57.18 | Dandan | i only have 2 pci slots |
| 19:57.35 | asterboy | I think your getting an IRQ share problem. |
| 19:57.44 | asterboy | only 2 pci slots? |
| 19:57.49 | asterboy | man that sucks |
| 19:57.50 | Dandan | it shares it with a eth |
| 19:57.54 | Dandan | yeah, all PCI-E |
| 19:58.01 | asterboy | take out the eth for now and try it. |
| 19:58.07 | asterboy | just to see. |
| 19:58.09 | Dandan | can't built in :) |
| 19:58.12 | asterboy | ah |
| 19:58.16 | asterboy | disable via bios? |
| 19:58.17 | Dandan | but will try the other pci |
| 19:58.19 | Dandan | yeah |
| 19:58.23 | Dandan | thanks for pointing me |
| 19:58.24 | Dandan | :) |
| 19:58.43 | *** join/#asterisk PupenoL (n=pupeno@200.123.183.89) |
| 19:59.36 | Dandan | i just work with what i have |
| 20:00.19 | asterboy | Dandan, do you have pricing for those openpci cards? |
| 20:00.27 | *** join/#asterisk ManxPowe (n=ewieling@pcp09535945pcs.gadsdn01.al.comcast.net) |
| 20:00.48 | Dandan | hm... |
| 20:00.48 | Dandan | yeah |
| 20:01.01 | BBRdiguez | Does anyone know how to delay call connection after dial ? |
| 20:01.13 | Dandan | asterboy: http://www.telephonyware.com/telephonyware/search.html?id=tEhqDh4a |
| 20:02.05 | ManxPowe | BBRdiguez, Why? |
| 20:02.23 | ManxPowe | If we know the reason we can give a better and more targeted answer. |
| 20:03.31 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:03.35 | Dandan | shit, now it hares the IP with usb... |
| 20:03.38 | Dandan | argh... |
| 20:03.43 | Dandan | stupid thing, bios here i come |
| 20:03.59 | Dandan | *shares |
| 20:04.06 | asterboy | disable USB and alternatively set the share within BIOS |
| 20:04.26 | *** join/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-225-162.central.biz.rr.com) |
| 20:04.29 | asterboy | Anyone selling Digium FXS Module (S110M) modules? |
| 20:04.31 | Dandan | yeah :) |
| 20:04.36 | Ferrari | Good day all :-) |
| 20:04.57 | Ferrari | anyone familiar with READ_SIZE=160 in chan_zap.c |
| 20:05.03 | asterboy | Or anyone selling broken Polycom phones? |
| 20:05.13 | BBRdiguez | ManxPowe: can you please help me, i have FAS on the 2nd call leg and i want to hide it with asterisk |
| 20:05.23 | Ferrari | I have been told that if i decrease that and add jitterbuffer=40 i can reduce the echo on zap calls |
| 20:05.34 | Ferrari | i have tried but my calls then sound robotic |
| 20:05.39 | ManxPowe | BBRdiguez, Facility Associated Signalling? |
| 20:05.42 | Ferrari | am i barking up the wrong tree |
| 20:06.02 | ManxPowe | Ferrari, jitterbuffer has NOTHING to do with echo |
| 20:06.03 | Dandan | have i mentioned how much i love dell |
| 20:06.09 | BBRdiguez | ManxPowe: False Answer Supervision, i think it's the same |
| 20:06.11 | Dandan | it leaves me no control over irq's |
| 20:06.35 | ManxPowe | BBRdiguez, If you are using analog FXO then the call is considered answered as soon as it's dialed. |
| 20:06.39 | Ferrari | oh yeah the reason i had to increase the jitterbuffer is to allow the gsm recordings to play correctly or so i have been told |
| 20:07.14 | Ferrari | as a result of shortening the READ_SIZE to 16 |
| 20:07.17 | zoa | jitterbuffer might have something to do with echo |
| 20:07.17 | Ferrari | from 160 |
| 20:07.21 | zoa | as the delay increases |
| 20:07.32 | zoa | the echo characteristics might get worse |
| 20:07.33 | rob0 | congrats ManxPowe :) |
| 20:07.53 | Ferrari | http://www.sineapps.com/print.php?rssid=338 |
| 20:08.02 | Ferrari | talks about the concept |
| 20:08.07 | rob0 | get any snow up there on the mountain? We got a little. |
| 20:08.10 | *** join/#asterisk gkchicago (n=gkchicag@66.9.120.18) |
| 20:08.13 | BBRdiguez | ManxPowe: *->spa3000->PSTN , on the leg from spa3000 to PSTN, there's FAS. I want to continue sending trying from asterisk to clients untill the call is picked up |
| 20:08.22 | *** join/#asterisk Pegger (n=peg@pool-68-163-139-134.bos.east.verizon.net) |
| 20:08.29 | zoa | ~seen zx81 |
| 20:08.45 | jbot | zx81 <n=ZX81@81-208-60-204.ip.fastwebnet.it> was last seen on IRC in channel #asterisk, 13d 4h 36m 3s ago, saying: '~ping'. |
| 20:08.45 | Ferrari | anyway when i try it i get robotic sounding conversation and wondered if i was missing something or if this is FALS info |
| 20:08.45 | Pegger | what is the diffrence between isdn and a t1? |
| 20:09.24 | asterboy | http://www.dbstreams.ca/ is selling the OpenPCI 12 ports $1200 CDN. |
| 20:10.06 | asterboy | typical, about $100 per port. |
| 20:10.10 | ManxPowe | BBRdiguez, There is nothing you can do to prevent that. |
| 20:10.16 | asterboy | Dam I sure wish those prices could drop in half. |
| 20:11.00 | Dandan | asterboy: nice |
| 20:11.05 | Dandan | i paid $1.5k us |
| 20:11.16 | *** part/#asterisk BBRdiguez (n=BBRdigue@p54B0123C.dip0.t-ipconnect.de) |
| 20:11.23 | asterboy | Looks like I'm going to be using VOIP termination at my service provider and FXS ports via an unlocked Linksys PAP2 |
| 20:11.33 | asterboy | Dandan, thats is a bit of a difference. |
| 20:11.42 | gkchicago | Hey guys, I'm getting this error when I try to modprobe ztdummy with ubuntu 2.6.12-10 (ztdummy: version magic '2.6.12-10-386 386 gcc-4.0' should be '2.6.12-10-386 386 gcc-3.4') |
| 20:11.42 | asterboy | I should buy and sell on ebay. |
| 20:11.51 | Dandan | asterboy: yeah |
| 20:11.53 | Dandan | company paid |
| 20:11.55 | Dandan | not me though... |
| 20:11.59 | *** join/#asterisk gambolputty (n=gambolpu@64.74.225.131) |
| 20:12.11 | asterboy | company still likes to save a buck usually. |
| 20:12.17 | Dandan | is linksys pap2 unlockable? |
| 20:12.27 | Dandan | asterboy: yeah, they already do, vs. avaya and cisco :) |
| 20:13.07 | asterboy | If I sold for $1300 CDN, thats $100 in my pocket and over $600 savings. |
| 20:13.26 | asterboy | Dandan, yep the pap2 is unlockable. |
| 20:13.39 | *** join/#asterisk ful|work (n=fulgas@209.8.233.10) |
| 20:13.39 | asterboy | makes the cheapest FXS ports you'll ever find. |
| 20:14.04 | wasim | atcom, ag468 |
| 20:14.06 | asterboy | As for FXO, I just terminate with a VOIP provider via SIP. |
| 20:14.08 | wasim | 4 ports for $75 |
| 20:14.18 | asterboy | wasim, thats pretty good. |
| 20:14.43 | Pegger | what is the best way to do isdn failover if I have two machines, could I simply splice the isdn cable two two so if one machine does the secone will take over |
| 20:14.58 | rajiv | where do you find a pap2 for $75 ? |
| 20:15.07 | asterboy | pap2's go for about $10 per port |
| 20:15.26 | asterboy | rajiv, goto ebay. |
| 20:15.38 | asterboy | http://cgi.ebay.com/NEW-Linksys-Vonage-PAP2-VOIP-Phone-Adapter-UNLOCKED_W0QQitemZ5855081413QQcategoryZ99269QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 20:16.27 | asterboy | Pegger, I have a feeling that would be a bad idea. |
| 20:16.44 | xp_ | can I pay someone to train me on asterisk? |
| 20:16.48 | Dandan | i unlocked the callvantage d-link |
| 20:16.50 | Dandan | $30 |
| 20:16.51 | gambolputty | What is the best way to find out if a SIP provider is up automatically in * without doing a dial command first? |
| 20:16.55 | brad_mssw | how do you unlock the pap2 ? |
| 20:17.10 | Dandan | gammacoder: poke it |
| 20:17.24 | gambolputty | ? |
| 20:17.48 | Dandan | heh, :) yeah, somewhere in sip.conf |
| 20:17.49 | Dandan | hold on |
| 20:18.08 | ManxPowe | gambolputty, what's wrong with using a dial command? |
| 20:18.11 | asterboy | you could burn out equipment with over voltage. |
| 20:18.11 | asterboy | but I'm just thinking cautiously. |
| 20:18.11 | asterboy | hook it up to a multimeter before you try it. |
| 20:18.12 | Pegger | asterboy you understand what I am trying to do right |
| 20:18.30 | asterboy | Pegger, yeah, you want a fail safe incase one of the ISDN lines drops. |
| 20:18.31 | rajiv | the pap2 has 2 fxs? hmm. i am looking for 2 or 4 fxo ports |
| 20:18.35 | ManxPowe | brad_mssw, you cannot unlock a PAP2 if it has ever connected to the provider it's loclked to. |
| 20:18.36 | rob0 | xp_: if you don't get an answer, Google and the wiki list consultants |
| 20:18.59 | ManxPowe | Don't use the PAP, use the Linksys |
| 20:19.00 | Pegger | asterboy, no actully i want failsafe incase one of the machines dies |
| 20:19.02 | ManxPowe | ..er...use the SIPURA |
| 20:19.12 | asterboy | brad_mssw, goto broadbandreports.com and search for pap2 |
| 20:19.17 | Pegger | asterboy one isdn two machines |
| 20:19.21 | Ariel_ | Damm I have been trying to fix an error for the pass hour. I They had a O1 instead of 01 that is "Oh" one instead of Zero one... argh they look the same via nano |
| 20:19.43 | ravenpi | So: what's wrong with the $400 D-Link 24-port PoE switch? Everyone else's goes for ~$1K and up... |
| 20:19.51 | brad_mssw | well, anyone know if any other linksys routers tied to vonage can be unlocked? I've got 2 bricks here that I'm trashing |
| 20:20.07 | asterboy | Pegger, ok got ya now. |
| 20:20.09 | gambolputty | The dial command would take the timeout value as the max time to make a call, I need to find out in one or two seconds if the provider is up or not. |
| 20:20.13 | Pegger | brad_mssw, what model number are the bricks? |
| 20:20.29 | Pegger | asterboy would it be ok to just splice the line in that setup? |
| 20:20.33 | ManxPowe | gambolputty, um, qualify=yes, then dial will return immediatly if the provider becomes unreadhable |
| 20:20.42 | asterboy | Pegger, in that case I'm thinking you need a gateway. |
| 20:20.57 | *** part/#asterisk Utah_Dave (n=boucha@c-24-10-151-252.hsd1.ut.comcast.net) |
| 20:21.02 | asterboy | Pegger, I'm certain you cannot splice that line. |
| 20:21.05 | Pegger | asterboy what do you mean GATEWAY? |
| 20:21.16 | brad_mssw | Pegger: err, let me see ... rtsp31 or something, need to go look |
| 20:21.23 | Pegger | asterboy then how are you supos to do failover fo ra idsdn |
| 20:21.23 | asterboy | like a another box or equipment that properly divvies out the ISDN. |
| 20:21.31 | ManxPowe | You cyou cannot "just splice" an ISDN line |
| 20:21.32 | Ariel_ | brad_mssw, I saw a post on voxilla on how to get them unlocked. |
| 20:21.36 | Dandan | gambolputty: qualify=yes? |
| 20:21.48 | gambolputty | qualify=yes is already on in sip.conf, but the provider authenticates off an IP address |
| 20:21.58 | brad_mssw | hah, oops, too late, already trashed them |
| 20:22.00 | gambolputty | so there is no definition for the provider in sip.conf at all necessary |
| 20:22.02 | ManxPowe | gambolputty, one has nothing to do with the other |
| 20:22.15 | gambolputty | then what good will qualify do? |
| 20:22.21 | ManxPowe | gambolputty, MAKE an entry in sip.conf for the provider |
| 20:22.30 | Dandan | it shows you if the sip server is avail? |
| 20:22.51 | Dandan | anywa: i am still looking for VPB users |
| 20:23.07 | ManxPowe | gambolputty, Qualify willdo a "sip ping" once per second |
| 20:23.13 | gkchicago | I'm getting this error when I `modprobe ztdummy` with ubuntu 2.6.12-10 (ztdummy: version magic '2.6.12-10-386 386 gcc-4.0' should be '2.6.12-10-386 386 gcc-3.4') does this mean I can't use gcc-4.0 with ztdummy? |
| 20:23.18 | Pegger | asterboy, then how would you suggest doing failover for the line |
| 20:23.18 | gambolputty | what happens then? |
| 20:23.38 | ManxPowe | gambolputty, then dial will return immediately when you you try to dial |
| 20:23.38 | asterboy | brad_mssw, http://www.dslreports.com/forum/remark,14450684 Unlocking Guide for PAP2 Linksys |
| 20:23.50 | ManxPowe | Dude, you should know all this already. |
| 20:23.54 | brad_mssw | asterboy: yep, found that one |
| 20:23.58 | Dandan | gkchicago: recompile your ztdummy module |
| 20:24.11 | Dandan | it has not been compiled for your kernel |
| 20:24.27 | ManxPowe | When you dial by IP address or hostname you have NO control over codecs, etc |
| 20:24.28 | brad_mssw | asterboy: interesting, oh well, it wasn't a pap2 i had, but the one before it that vonage sent out (2 of them) ... just realized I already trashed them though |
| 20:24.35 | asterboy | Pegger, not sure how to do failover, dollars to donuts someone in here knows...tug on ManxPowe |
| 20:24.58 | ManxPowe | I cannot help with failover for any type of ISDN |
| 20:25.04 | asterboy | trashed how? |
| 20:25.19 | asterboy | ManxPowe...thought you might. doh. |
| 20:25.21 | Pegger | ManxPowe, or t1 |
| 20:25.27 | Dandan | http://www.geocities.com/sigmaz_1/ |
| 20:25.27 | ManxPowe | or T-1 |
| 20:25.28 | Dandan | and |
| 20:25.32 | Dandan | http://www.bestbuy.com/site/olspage.jsp?skuId=6689871&type=product&id=1083712208590 |
| 20:25.33 | JMcA | you can't really "failover" an ISDN line like that |
| 20:25.33 | Dandan | this |
| 20:25.33 | Dandan | :) |
| 20:25.38 | Pegger | err it must be possible some how |
| 20:25.38 | gkchicago | dandan, i'm compiling from cvs, with `make linux26` my symlinks are in place for /usr/src/linux .. |
| 20:25.41 | ManxPowe | At least for INCOMING CALLS. for outgoing calls it's dead simple. |
| 20:25.42 | gambolputty | for the provider definition I have type and host defined, that should be enough |
| 20:25.47 | asterboy | Pegger, have you searched internet yet? Try altavista |
| 20:25.54 | xp_ | ok here is my problem, if I have 2 asterisk servers and one fails over to the other one I will loose all the calls that the other one had, is there any way to prevent this? |
| 20:26.01 | ManxPowe | for INCOMING calls you would need two PRIs and help from your carrier |
| 20:26.01 | gambolputty | there is no username/password authentication at all for them |
| 20:26.02 | JMcA | a telco line like that has to land somewhere...and that somewhere will be a single point of failure for that line...only way to deal with that is multiple lines |
| 20:26.05 | Dandan | gkchicago: hm, was your kernel compiled with 3.4 then? |
| 20:26.23 | ManxPowe | PEOPLE, read the wiki about this stuff. |
| 20:26.59 | xp_ | JMcA I don't understand, can you elaborate? |
| 20:27.14 | Dandan | ~vpb |
| 20:27.35 | gkchicago | dandan, well the kernel comes precompiled from ubuntu, i have the kernel source packages installed.. is there a way to tell? |
| 20:27.55 | Dandan | modinfo afair |
| 20:28.15 | *** join/#asterisk RoyK (n=roy@ti211310a080-9456.bb.online.no) |
| 20:28.28 | JMcA | xp_: telco lines like that are inherently point-to-point...you have a device on each side and they are "connected" to each other in that they "bring up" the line and maintain some sort of state...if the device on one side fails, whatever connections are established on that line are going to be shut down, there is no two ways about it...the best you can do is have another line to another device where you can re-establish your connection quickly |
| 20:28.58 | *** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee) |
| 20:29.04 | ManxPowe | Yoiu peoYou people realize that failover issues have been discussed many times on the asterisk mailing lists, and there are a couple of page son the Wiki about it as well. I'll bet the new Asterisk book talks about it too. |
| 20:29.30 | gkchicago | dandan, http://pastebin.ca/37281 |
| 20:29.30 | klasstek | gkchicago, apt-get gcc-3.4 and set your CC properly |
| 20:29.39 | ManxPowe | There is very little we can say here that has not been said better, and in more detail on these resources. |
| 20:29.42 | ManxPowe | ~mailinglist |
| 20:29.43 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 20:29.45 | ManxPowe | ~docs |
| 20:29.46 | jbot | methinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:29.46 | JMcA | xp_: this is true even for tradition POTS service...telco's have put a *LOT* of money into making their equipment very reliable rather than creating failover schemes |
| 20:29.54 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 20:29.56 | gambolputty | I try to dial [name} but it doesn't work |
| 20:30.01 | gambolputty | [name] |
| 20:30.11 | JMcA | yeah, what manx said...but moreso beyond even just asterisk-specific sources |
| 20:30.29 | Dandan | 3.4 |
| 20:30.31 | Dandan | gcc |
| 20:30.41 | gambolputty | does this "sip ping" mean having to register with a sip provider first? |
| 20:30.48 | Dandan | that's what you need to use to compile the new module |
| 20:31.23 | gkchicago | klasstek, what does "set your CC properly" mean? |
| 20:31.27 | gkchicago | C Compiler? |
| 20:32.10 | klasstek | yes the CC environment variable |
| 20:32.11 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 20:34.21 | Pegger | asterboy, i have does some searching on the internet, but so far it looks like I need some hardware which I am hoping to avoide, I would rather so it simpler the make it more compicated |
| 20:34.23 | gkchicago | `echo $CC` doesn't show anything in there right now, set that to the path to gcc-3.4? |
| 20:34.42 | klasstek | ~google ubuntu compile kernel module gcc-3.4 |
| 20:35.43 | gkchicago | good call |
| 20:35.45 | gkchicago | thanks |
| 20:35.59 | klasstek | export CC=gcc-3.4 |
| 20:36.46 | ManxPowe | gambolputty, no, you do not have to register. |
| 20:37.02 | ManxPowe | gambolputty, You've been using Asterisk for a long time and this is the first time you've done SIP? |
| 20:37.18 | ManxPowe | Registration ONLY notifies the remote server what your IP address is. It does nothing else. |
| 20:37.27 | ManxPowe | Let me repeat that for everyone. It does NOTHING else. |
| 20:41.35 | [TK]D-Fender | ManxPowe : So full auth info is passed again for every call? |
| 20:42.09 | ManxPowe | [TK]D-Fender, correct. |
| 20:42.29 | [TK]D-Fender | ManxPowe : and I presume as not to send further packets to fradulent attempts to identify as an account. |
| 20:42.44 | ManxPowe | [TK]D-Fender, I would assume so. |
| 20:43.07 | ManxPowe | I don't think auth is sent for every RTP packet, just every SIP packet. |
| 20:43.09 | [TK]D-Fender | ManxPowe : I suspected as much. The "using IAX to connect 2 servers" wiki page pretty much gave that away. |
| 20:43.41 | [TK]D-Fender | ManxPowe : Yes, I was implying call control, not RTP, but then again.. thatrs exacty what happens before RTP anyways |
| 20:44.40 | ManxPowe | Much like being handcuffed to a zebra, dialing by hostname/ip address may seem like a good idea at first, but you soon realize it's not. |
| 20:45.48 | [TK]D-Fender | ManxPowe : Actually the way I keep adding typed out sip URI's to my dial-plan for testing with people I'm going to need a web-interface to set it up in ASTDB or something. Manually editig them into the dial-plan is a PITA and worse on phones. |
| 20:46.08 | gambolputty | Why is dialing by hostname not a good idea? Free calls using SIP are fun. |
| 20:46.13 | *** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net) |
| 20:46.26 | ManxPowe | gambolputty, because you have NO control on a per destination basis. |
| 20:46.40 | gambolputty | what do you mean? |
| 20:47.18 | ManxPowe | There is no different in dialing by host .vs. dialing by sip.conf entry except that if you dial by sip.conf entry you can customize the settings for that destination. |
| 20:47.20 | FuriousGeorge | hi all |
| 20:47.58 | ManxPowe | Usually you want disallow all codecs and the specify only one codec to use, or you want qualify, or (in the case of incoming calls) you want to specify what context the call lands in. |
| 20:48.05 | [TK]D-Fender | gambolputty : not to say its not a good idea (I like it personally), its just inconvenient without a web interface to type them into or something |
| 20:48.26 | gambolputty | that's why IP phones should have built in keyboards or USB ports |
| 20:48.35 | [TK]D-Fender | ManxPowe : and can share your dial-plan without having to blot our secrets all the time :) |
| 20:48.44 | gambolputty | to allow for the plugging in of a garden variety USB keyboard |
| 20:48.47 | ManxPowe | [TK]D-Fender, that too. |
| 20:50.40 | *** join/#asterisk jerlique (n=jerlique@lnk59.adl3.adsl.esc.net.au) |
| 20:50.41 | FuriousGeorge | for some reason which i cant for the life of me figure out, since i upgraded to 1.2.1 one of my zap channels is misbehaving. can someone take a look at this and help me speculate http://pastebin.ca/37288 |
| 20:50.54 | FuriousGeorge | maybe i have to specify immediate=no for channel 1 |
| 20:51.21 | ManxPowe | FuriousGeorge, you need immediate=no before that channel |
| 20:51.21 | FuriousGeorge | ? |
| 20:51.40 | ManxPowe | in fact unless you are a kick ass asterisk guru you ALWAYS want immediate=no |
| 20:51.53 | ManxPowe | same applies for "r" option to dial. |
| 20:51.58 | [TK]D-Fender | ManxPowe : Or your name is Bruce Wayne or something..... |
| 20:52.08 | Dandan | ~vbp |
| 20:52.10 | Dandan | argh |
| 20:52.13 | Dandan | voicetronix :/ |
| 20:52.17 | ManxPowe | [TK]D-Fender, Even bruce wayne should have immediate=no |
| 20:52.22 | FuriousGeorge | why is that |
| 20:52.48 | ManxPowe | FuriousGeorge, because immediate=yes is for PLAR lines, i.e. "bat phone" or "hotline phone". |
| 20:52.55 | Dandan | Jan 18 15:52:39 NOTICE[1837]: chan_vpb.c:1705 vpb_pvt* mkif(int, int, int, int, float, float, float, float, int, in: Voicetronix V12PCI channel vpb/1-2 initialized (rxsg=0.000000/txsg=0.000000/rxhg=0.000000/txhg=0.000000)(0xffffffff/0xffffffff/0xffffffff) |
| 20:52.55 | Dandan | Segmentation fault |
| 20:52.58 | Dandan | i get this :/ |
| 20:52.58 | ManxPowe | You pick up the phone and the system dials a predefined number for you |
| 20:53.06 | Pegger | humm i found this, it is exactly what I need but the guy is quite vague abou who is supplying the failover and who makes http://www.voip-info.org/wiki/index.php?page=Asterisk+failover+case1 |
| 20:53.15 | FuriousGeorge | yeah, i use that, i just thought if the default is no, then all the other channels would act like default |
| 20:53.16 | ManxPowe | Now, Commisioner Gordon might have immediate=yes |
| 20:53.25 | FuriousGeorge | lol |
| 20:53.27 | iCEBrkr | So if you SetGroup(), will that call fall off the 'stack' when it's hung up-- Cuz I can't find any UnSetGroup() |
| 20:53.38 | FuriousGeorge | or the doorphone made by viking electronics |
| 20:53.38 | ManxPowe | FuriousGeorge, all settings in zapata.conf apply to all following channel => lines. |
| 20:54.09 | FuriousGeorge | ManxPowe: yeah, check out my pastebin post, i had it set up right sans immedate=no for all the other channels besides doorphone |
| 20:54.20 | FuriousGeorge | which came first |
| 20:54.21 | ManxPowe | FuriousGeorge, you also do NOT want busydetect or callprogress |
| 20:54.52 | FuriousGeorge | sounded useful at the time |
| 20:55.22 | ManxPowe | Trust me, the fewer options in zapata.conf the better. |
| 20:55.41 | ManxPowe | busydetect= is just an alias for randomlydisconnectcalls= |
| 20:56.03 | FuriousGeorge | so far so good but last time i fixed it it unfixed after a few seconds |
| 20:56.30 | *** join/#asterisk backblue (n=moo@87-196-34-198.net.novis.pt) |
| 20:58.04 | FuriousGeorge | still working ;) |
| 20:58.09 | FuriousGeorge | thanks ManxPowe |
| 20:58.21 | FuriousGeorge | p.s. you spelled your nick wrong didnt you? |
| 20:58.34 | ManxPowe | No, I was ghosted. |
| 20:58.55 | FuriousGeorge | i see you've exorcised the deamons |
| 20:59.10 | ManxPower | You can thank me by helping to stomp out misuse of immediate=yes, busydetect=yes, callprogress=yes, and the "r", "t", and "T" options to Dial. |
| 20:59.45 | FuriousGeorge | speaking of t,T |
| 21:00.04 | ManxPower | or you can send me something off my thinkgeek.com wishlist, or send me all your money. |
| 21:00.10 | ManxPower | Any of those will do just fine. |
| 21:00.40 | FuriousGeorge | ill see what i can do |
| 21:01.25 | FuriousGeorge | i still think that falls under a bug. if the default for immediate=no than it should be no. not no for a while then yes a few mintes later :) |
| 21:01.39 | *** part/#asterisk Ferrari (n=Ferrari_@rrcs-24-123-225-162.central.biz.rr.com) |
| 21:02.49 | FuriousGeorge | exten => s,3,dial(${ARG2}/${ARG1},,T) is not working on one of my boxes and i cant figure out why (the t,T part specifically) |
| 21:04.37 | [TK]D-Fender | FuriousGeorge : what's not working? |
| 21:04.50 | [TK]D-Fender | looks like one too many ","'s |
| 21:04.55 | FuriousGeorge | when i hit # the chick doest ask me to x-fer |
| 21:04.58 | *** join/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 21:05.13 | FuriousGeorge | i thought i needed the extra , to escape TIMEOUT |
| 21:05.52 | blitzrage | FuriousGeorge: check features.conf -- is it set to use # for transfers? |
| 21:06.02 | blitzrage | FuriousGeorge: yes.. .you need the second comma to skip over the timeout |
| 21:06.05 | KriS83 | Anybody (from germany) ever worked with Partial Rerouting? |
| 21:06.20 | Tall-guy | I ask this question every 6 months or so: does anyone have a working config using BRI ISDN with national (ie: N.American) NI-1 signalling with asterisk? |
| 21:06.34 | asterboy | Does the Polycom IP 300 Power Adapter plug into the phone or does it have a Data/Power Cable like the IP 500? |
| 21:06.34 | *** join/#asterisk _Roey (n=Roey@h-69-3-4-130.mclnva23.covad.net) |
| 21:06.36 | FuriousGeorge | blitzrage: i thought that was hard coded and couldnt be changed. i'd love to change it to 33 |
| 21:06.36 | _Roey | HI |
| 21:06.37 | FuriousGeorge | ## |
| 21:06.48 | _Roey | SIP is an ISO spec yet? |
| 21:06.51 | blitzrage | FuriousGeorge: check features.conf |
| 21:06.52 | _Roey | since XMPP is not. |
| 21:07.05 | _Roey | Is there a looming battle between XMPP+voice extensions and SIP? |
| 21:07.11 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 21:07.13 | asterboy | Anyone here have an IP 300 from Polycom? |
| 21:08.00 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 21:09.14 | FuriousGeorge | blitzrage: are you sure thats not just for parking, the box that it does work on doesnt have anything special in features.conf |
| 21:10.12 | Amilcar_ | anyone knows how to make asterisk delivers one call at a time to members of a queue? For example, if a member is on the phone, with a queue call, asterisk can't route more calls to that member until it frees. |
| 21:10.30 | blitzrage | FuriousGeorge: |
| 21:10.33 | asterboy | stop hiding...I know someone here has an IP 300, just don't want to admit it do ya! |
| 21:10.34 | blitzrage | [featuremap] |
| 21:10.34 | blitzrage | ;blindxfer => #1 ; Blind transfer |
| 21:10.34 | blitzrage | ;disconnect => *0 ; Disconnect |
| 21:10.34 | blitzrage | ;automon => *1 ; One Touch Record |
| 21:10.35 | blitzrage | ;atxfer => *2 ; Attended transfer |
| 21:10.45 | blitzrage | in 1.2.x at least |
| 21:11.08 | FuriousGeorge | hmm |
| 21:11.17 | FuriousGeorge | i think i gotta upgrade my features.conf :) |
| 21:11.27 | asterboy | you thought you were going to have a great speaker phone and *poof*...no speaker phone. |
| 21:11.31 | [TK]D-Fender | asterboy : What do you want to know about it? |
| 21:11.55 | [TK]D-Fender | asterboy : Speakerphone on the 50x & 60x. SPEAKER on 30x. |
| 21:11.56 | asterboy | Just want to know if the power plugs into the phone or a data/power cable like the IP 500. |
| 21:12.10 | *** join/#asterisk mina (n=mina@modemcable175.94-70-69.mc.videotron.ca) |
| 21:12.22 | asterboy | Got an IP 500 without the data/power cable. |
| 21:12.23 | [TK]D-Fender | asterboy : You need an adapter cable for PoE jsut like the 50x. |
| 21:12.32 | dippo_ | man we are getting distortion and dropped calls like crazy via teliax lately |
| 21:12.37 | dippo_ | anyone else experiencing this? |
| 21:12.40 | asterboy | no PoE |
| 21:12.56 | mina | Anyone successful in getting the latest asterisk in FreeBSD ports (1.2.1) to compile and install ? |
| 21:12.57 | asterboy | Got an IP 500 without the power/data cable. |
| 21:13.28 | asterboy | I'm thinking if the IP 300 had the same type of connection, I'll get a used one on ebay and save the $50 replacement cost. |
| 21:13.57 | asterboy | Mind you they do come in a 5 pack at the price...so I could resell the others on ebay in case anyone else is in the same boat. |
| 21:14.04 | blitzrage | [TK]D-Fender: got any good resources for getting Polycom phones? |
| 21:14.21 | blitzrage | [TK]D-Fender: can you tell jsmith in #asterisk-doc? |
| 21:15.35 | *** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 21:16.29 | warthawg | is there any setting i can set so that asterisk tells me why a sip phone registrations fails? i have verbose/debug == 10 now |
| 21:16.50 | mina | warthawg: sip debug |
| 21:16.57 | warthawg | tks mina |
| 21:17.19 | FuriousGeorge | heres another tough asterisk logic question: FuriousGeorge has 4 asterisk servers (1.2.1) with dynamic ips: Be, Claudia, Filomena, and Juanita. they use dynamic ip to dns services to log into eachother. the domain names are all pointing to the correct ips. i can never get all four logged into eachother at the same time |
| 21:17.50 | FuriousGeorge | right now filomena is registered with all, but cluadia only 2 outta 4 |
| 21:18.33 | rob0 | FuriousGeorge: openvpn might simplify things, is any site relatively static? |
| 21:19.21 | FuriousGeorge | rob0: ironically the most relatively static one is on a different ISP and most difficult for the other pretty dynamic ones to reach |
| 21:19.22 | *** join/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 21:19.50 | QbY | Affinity VoIP Telecom (http://www.affinityvoiptelecom.com) anyone have any horror stories? Compliments? |
| 21:19.54 | FuriousGeorge | it times out even though iax2 show registry shows its looking for the right ip |
| 21:21.00 | rob0 | I'd rig them all up in several openvpn connections. Encryption is optional if latency proves to be a problem. |
| 21:21.09 | FuriousGeorge | as of right now i got all but one. filomena cant register with be (the relatively static one) |
| 21:21.23 | FuriousGeorge | filomena seems to have the most register problems |
| 21:21.30 | dippo_ | can anyone recommend a good IAX trunking service other than teliax? |
| 21:21.32 | dippo_ | i am not happy with them |
| 21:21.42 | rob0 | complex routing tables would enable you to fall back on other VPN links |
| 21:22.11 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 21:22.16 | FuriousGeorge | rob0: how much overhead is vpm w/o encryption? |
| 21:22.31 | [TK]D-Fender | blitzrage : I only got mine through CCP ( http://www.ccpin.com ) with an agent. nothing really special. |
| 21:22.47 | blitzrage | coo |
| 21:23.07 | rob0 | probably very little, I don't notice it even using blowfish. |
| 21:23.19 | rob0 | obviously stay away from 3DES :) |
| 21:23.38 | warthawg | is SIP/2.0 403 Forbidden from SIP debug meaningful? |
| 21:24.13 | _Sam-- | dippo_: ive been trying www.iax.cc |
| 21:24.24 | _Sam-- | they seem ok from my location...im a teliax user also, looking at alternatives. |
| 21:25.44 | _Sam-- | i should know more about how that new place is later today, have a meeting with a client who i have testing it |
| 21:26.06 | _Roey | (hi?) |
| 21:27.33 | FuriousGeorge | rob0: i was just tyhinking: ive never set up a vpn before, wouldnt i need to change the ip ranges on the 4 networks so they arent all 10.0.0.X |
| 21:28.20 | rob0 | not necessarily, but I do keep all my sites on distinct subnets, yes. |
| 21:28.34 | rob0 | If the VPN is only to connect the various * boxen, no. |
| 21:28.53 | rob0 | Each VPN link will have its own IP. |
| 21:29.07 | rob0 | s/will/can/ |
| 21:29.19 | *** part/#asterisk Tall-guy (i=tall-guy@207-195-103-110.regn.hssx.sasknet.sk.ca) |
| 21:29.48 | dippo_ | i am just starting to wonder if there's any IAX trunking service that is viable for mission critical stuff |
| 21:29.52 | Jameno123 | dippo_: junctionnetworks.com |
| 21:29.55 | dippo_ | we have had lots of quality problems and dropped calls |
| 21:30.02 | *** join/#asterisk darby_t (i=darby_t@djw145.neoplus.adsl.tpnet.pl) |
| 21:30.04 | Jameno123 | business class iax |
| 21:30.09 | dippo_ | and i don't need to tell you that it only takes one or two dropped calls by sales people before heads start getting bitten off |
| 21:30.16 | _Sam-- | i use teliax for mission critical stuff at my business...we have 15 people answering incoming sales calls over teliax. |
| 21:30.22 | _Sam-- | but i have a really good route to them |
| 21:30.59 | dippo_ | i'm about 16 hops |
| 21:31.03 | rob0 | s/jbot/#asterisk/ |
| 21:31.05 | _Sam-- | im about 10 / 45ms |
| 21:31.06 | _Sam-- | to teliax |
| 21:31.21 | _Sam-- | my stuff stays on the same network for 3000 miles |
| 21:31.45 | _Sam-- | i get on cogent 2 hops away, and they are on cogent also. |
| 21:31.57 | _Roey | hey |
| 21:31.59 | _Roey | what about my hops |
| 21:32.12 | _Roey | how many chans do I have to hop through to get a response? |
| 21:32.24 | badboyz | is there any good asterisk monitoring software out there? for monitoring the health of multiple * boxes |
| 21:32.53 | *** join/#asterisk w14 (n=mike@84-51-158-179.mikero153.adsl.metronet.co.uk) |
| 21:34.30 | dippo_ | Jameno123: have you been using junctionnetworks.com? |
| 21:34.30 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 21:34.46 | *** join/#asterisk edwin__ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 21:35.30 | _Sam-- | dippo_: someone also recommended asterlink, which i think is run by someone on this channel |
| 21:35.36 | _Sam-- | i think they do IAX trunking |
| 21:36.00 | *** join/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 21:36.40 | dippo_ | i just hope i don't keep running into these same problems and am forced to hop from one to another before I give up and just buy a PRI |
| 21:37.53 | dippo_ | it seems like there might be a need for a site specifically for reviews of this stuff |
| 21:38.09 | dippo_ | sorta like in the voip-info wiki but more specific |
| 21:38.46 | [av]bani | _Sam--: how is teliax reliability and quality? they have outages? |
| 21:38.47 | *** join/#asterisk Naturalblue (n=Kay@195.26.12.229) |
| 21:38.48 | Ayano | Does anyone in here know how to track down the user of an IP? |
| 21:39.10 | dippo_ | Ayano: whois.arin.net will give you info about who owns the IP |
| 21:39.15 | warthawg | track down and kill, Ayano? |
| 21:39.18 | dippo_ | but tracking it down to a specific person is usually up to them |
| 21:39.23 | Ayano | I'm going to. |
| 21:39.51 | warthawg | Ayano, you have to be careful, because their pc may be being used without their knowledge |
| 21:40.09 | _Sam-- | ravenpi: sorry was away for a sec...from my location teliax is mostly reliable....the reliability problems, from my asterisk...arent internet related. |
| 21:40.15 | warthawg | samspade.net does a pretty good job of getting you to their isp |
| 21:40.17 | _Sam-- | er that was [av]bani |
| 21:40.33 | *** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 21:40.37 | _Sam-- | i have had some network outtages with teliax, but the guy is on my instant messenger, and is really responsive |
| 21:40.41 | [av]bani | mostly? |
| 21:40.46 | _Sam-- | yes mostly. |
| 21:40.51 | _Sam-- | they are by no means 100% |
| 21:40.52 | [av]bani | doesnt sound good :/ |
| 21:41.01 | [av]bani | well, compared to PSTN |
| 21:41.07 | Katty | twisted[asteria]: abandon work! |
| 21:41.09 | _Sam-- | since december 14th we've logged at least an hour of downtime because of teliax problems, not my problems. |
| 21:41.11 | [av]bani | nobody is 100%, but how many 9's are there |
| 21:41.19 | [av]bani | bleh |
| 21:41.45 | _Sam-- | luckily i own the business...so if someone wants to come crying "these phone suck"...they are not going to get very far. |
| 21:41.54 | [av]bani | what kind of problems? |
| 21:42.01 | _Sam-- | no phone service |
| 21:42.07 | _Sam-- | no incoming or outgoing calls until they reset something |
| 21:42.16 | _Sam-- | that has happened twice since november 16th |
| 21:42.23 | _Sam-- | the last one was only for about 8 minutes |
| 21:42.23 | [av]bani | :/ |
| 21:42.26 | Katty | [TK]D-Fender: you should abandon work too. |
| 21:42.47 | [av]bani | so youre using teliax for both in and out trunks? |
| 21:42.58 | joe | has anyone seen the voicemail system not allow you to re record a name when reassigning an extension to someone else? I reset the password to the extension number and all that to re-initialize the voicemail which it does prompt you to set it up, but it doesn't allow the new reccording to take effect and I reloaded asterisk service and at the console |
| 21:42.59 | _Sam-- | i have 2 backup accounts for outgoing just in case. |
| 21:44.00 | [av]bani | who else have you used? |
| 21:44.00 | _Sam-- | but yes teliax is my primary incoming and outgoing service for 20 employees at my business |
| 21:44.00 | joe | Ariel_: you mentioned restarting asterisk yesterday which I tried but still no go |
| 21:44.00 | _Sam-- | broadvoice and iax.cc are my backups |
| 21:44.00 | _Sam-- | i havent used either extensively |
| 21:44.00 | [av]bani | heh, i hear lots of bad things about broadvoice |
| 21:44.00 | _Sam-- | they have a server 15ms away from me |
| 21:44.00 | _Sam-- | broadvoice, that is. |
| 21:44.00 | _Sam-- | that is why i wanted to try it |
| 21:44.22 | _Sam-- | iax.cc's server is about 2 hops / 15ms closer than teliax's, to me. |
| 21:45.21 | [av]bani | hmm 32ms to teliax for me |
| 21:45.38 | dippo_ | i have had one outage with teliax, but I could live with that |
| 21:45.45 | dippo_ | what I can't handle is the quality and dropped calls |
| 21:45.50 | [av]bani | :/ |
| 21:45.53 | dippo_ | i am not 100% sure that's teliax, but things are pointing that direction |
| 21:46.02 | _Sam-- | i would check for packet loss and check your route |
| 21:46.08 | _Sam-- | because we have good call quality from teliax |
| 21:46.09 | dippo_ | we have no bandwidth congestion or packet loss that I can identify |
| 21:46.10 | _Sam-- | but we have a good route too |
| 21:46.17 | dippo_ | and our Zap trunk calls are fine |
| 21:46.32 | _Sam-- | there has been some problems ive read from people getting to teliax over one of their backbones |
| 21:46.36 | _Sam-- | in the past week its been acting out |
| 21:46.41 | twisted[asteria] | Katty, whyfor? |
| 21:46.50 | _Sam-- | so maybe you going over that problematic route |
| 21:46.54 | [av]bani | wonder if teliax will do PVC |
| 21:46.56 | _Sam-- | the guy at teliax is really guy |
| 21:46.57 | dippo_ | i hate to leave a company because of something like that |
| 21:47.01 | _Sam-- | er is really cool |
| 21:47.02 | Katty | twisted[asteria]: because i have crayons and coloring books. |
| 21:47.03 | dippo_ | but i can't hang with this for too much longer |
| 21:47.04 | Katty | twisted[asteria]: obviously. |
| 21:47.09 | _Sam-- | i gave him my ip and had him change the way my route goes |
| 21:47.16 | _Sam-- | cause they have bgp |
| 21:47.17 | dippo_ | how did you get in touch with him? |
| 21:47.23 | dippo_ | the last time we had an outage I called and got no answer |
| 21:47.27 | dippo_ | filed a ticket that no one ever updated |
| 21:47.27 | _Sam-- | i have email, instant messanger, direct number, etc. |
| 21:47.30 | _Sam-- | i know how to get him |
| 21:47.31 | brad_mssw | teliax is killing me these days |
| 21:47.35 | brad_mssw | 70+ms latency |
| 21:47.36 | dippo_ | but closed without comment when i noted that it was back up |
| 21:47.39 | brad_mssw | dropped calls ... |
| 21:47.45 | [av]bani | :/ |
| 21:47.46 | _Sam-- | i would call the guy |
| 21:47.54 | *** join/#asterisk lilliput (n=lilliput@82-47-147-63.cable.ubr11.brad.blueyonder.co.uk) |
| 21:47.56 | brad_mssw | most traffic goes over pnap or cogent ... high latency |
| 21:47.59 | [av]bani | i get 32ms to teliax, seems ok from here |
| 21:48.05 | _Sam-- | mines over cogent, no problems here. |
| 21:48.09 | [av]bani | i see savvis and pnap |
| 21:48.17 | dippo_ | i go over AT&T |
| 21:48.21 | brad_mssw | yeah, depends on your peering |
| 21:48.27 | _Sam-- | my client who goes over ATT is complaining about teliax |
| 21:48.29 | [av]bani | we're multihomed |
| 21:48.33 | _Sam-- | i switched them to iax.cc |
| 21:48.40 | brad_mssw | UUNET -> cogent isn't working well |
| 21:48.47 | [av]bani | cogent is doggie poo |
| 21:48.53 | _Sam-- | my cogent is perfect |
| 21:49.08 | brad_mssw | yeah, unless you're _on_ the cogent network, it sucks |
| 21:49.15 | dippo_ | and most of the latency seems to be from AT&T |
| 21:49.16 | brad_mssw | they're peering arrangements just aren't good |
| 21:49.16 | _Sam-- | yeah im on it, i guess luckily. |
| 21:49.34 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au) |
| 21:49.35 | brad_mssw | most of the latency is from cogent from where I'm standing |
| 21:49.42 | brad_mssw | I'm at 17ms leaving the UUNET backbone |
| 21:49.43 | _Sam-- | the guy from teliax told me they had 8 backbone providers |
| 21:49.51 | _Sam-- | ive only seen 2-3 myself |
| 21:49.52 | brad_mssw | then it shoots up to 70ms travelling over cogent the rest of the way |
| 21:50.05 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 21:50.22 | tasat | anyone out there have experience with answering machine detection? |
| 21:50.25 | dippo_ | same here with AT&T |
| 21:50.27 | brad_mssw | junctionnetworks gives me around 43ms ... vonworldwide gives me around 35ms |
| 21:50.40 | [av]bani | from what i can tell teliax is singlehomed with rockynet, who is multihomed |
| 21:50.44 | dippo_ | do you have an IP that junctionnetworks uses for IAX? |
| 21:50.49 | dippo_ | i'm curious how it looks from here to there |
| 21:50.54 | brad_mssw | dippo_: iax.jnctn.net |
| 21:50.56 | dippo_ | i was gonna call them and ask |
| 21:50.58 | _Sam-- | teliax is co-located at rockynet, who is multihomed, from what i can tell. |
| 21:51.17 | [av]bani | wow, i just did a trace to teliax from WV |
| 21:51.23 | [av]bani | went over cogent, and cogent is compltely shit |
| 21:51.24 | dippo_ | hm, not a whole lot better. 20ms |
| 21:51.31 | [av]bani | 200ms and 15% packet loss |
| 21:51.34 | _Sam-- | that junctionnetworks is the closest one to me ive seen so far |
| 21:51.37 | _Sam-- | 20 ms |
| 21:51.39 | [av]bani | major cogent problems heh |
| 21:52.03 | brad_mssw | _Sam--: wow, 20ms is awesome |
| 21:52.05 | [av]bani | you use JN? |
| 21:52.25 | _Sam-- | ive never used JN yet |
| 21:52.36 | brad_mssw | dippo_: 20ms total, or 20ms better ? |
| 21:52.37 | tasat | anyone out there have experience with answering machine detection? |
| 21:52.41 | dippo_ | 20ms better |
| 21:52.46 | dippo_ | 60ms average versus 80ms |
| 21:52.58 | brad_mssw | got JN right now for outgoing, testing them out |
| 21:53.00 | *** join/#asterisk Darwin35 (n=Darwin@sta-208-139-193-162.rockynet.com) |
| 21:53.05 | brad_mssw | dippo_: try iax.vonworldwide.com |
| 21:53.06 | Darwin35 | ok whats going on |
| 21:53.09 | _Sam-- | lol |
| 21:53.20 | _Sam-- | we are debating the quality services provided by teliax |
| 21:53.21 | dippo_ | about the same. 60ms |
| 21:53.31 | _Sam-- | you guys are having problems with teliax lately ? |
| 21:53.35 | [av]bani | wow iax.vonworldwide.com is all over theplace with me |
| 21:53.38 | *** join/#asterisk dca[laptop] (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 21:53.40 | _Sam-- | i think darwin may have some ideas on how to help |
| 21:53.40 | [av]bani | 90ms-200ms |
| 21:53.42 | brad_mssw | dippo_: sounds like the rest of that latency is your network |
| 21:53.47 | Darwin35 | whats the issue with teliax ? |
| 21:53.52 | _Sam-- | these guys are saying that getting to teliax has been teerrible lately |
| 21:53.57 | _Sam-- | and that the calls are dropped, the network sucks... |
| 21:54.03 | _Sam-- | and the call quality is bad...bad latency, you name it! |
| 21:54.20 | [av]bani | :/ |
| 21:54.33 | [av]bani | my latency to teliax is fine |
| 21:54.36 | _Sam-- | these guys are saying ping times 200ms to teliax |
| 21:54.39 | _Sam-- | 20% packet loss |
| 21:54.42 | *** part/#asterisk tholo (n=tholo@nat.sigmasoft.com) |
| 21:54.42 | _Sam-- | lost / dropped calls |
| 21:54.45 | _Sam-- | bad call quality |
| 21:54.50 | _Sam-- | looking for alternatives |
| 21:54.51 | [av]bani | and teliax kicks your dog too |
| 21:54.56 | brad_mssw | no packet loss for me ... just higher latency than 3 weeks ago |
| 21:55.05 | *** join/#asterisk lesouvage (n=lesouvag@82.74.11.143) |
| 21:55.24 | _Sam-- | they are finding that other providers have routes and boxes that are infinitely closer in MS than teliax is |
| 21:55.25 | Darwin35 | please do traceroutes and send them to support@teliax.com and cc them to support@rockynet.com so we can trace this |
| 21:55.38 | dca[laptop] | last i heard teliax pissed off the internet :P |
| 21:55.45 | brad_mssw | Darwin35: did that already |
| 21:55.57 | brad_mssw | Darwin35: haven't responded to me yet :/ |
| 21:56.16 | Darwin35 | I am the tech and have not seen any come across |
| 21:56.32 | _Sam-- | there is no denying that something has been amiss lately with the routes to teliax. |
| 21:56.34 | brad_mssw | Darwin35: I talked to you a few days ago, and pasted the traceroute on your live support on teliax |
| 21:56.39 | _Sam-- | and its affecting people only on certain networks |
| 21:56.41 | brad_mssw | Darwin35: you are Richard on there, no ? |
| 21:56.52 | Darwin35 | yes |
| 21:56.53 | dippo_ | i've got a ticket filed, only for choppy quality and dropped calls, though |
| 21:56.55 | festr_ | hello, is it possible in application Dial playback some file to the caller but as soon as the called channel asnwer? |
| 21:57.05 | Darwin35 | ok let me look |
| 21:57.17 | brad_mssw | Darwin35: seems like the pnap and cogent routes suck ... |
| 21:57.23 | festr_ | it seems that i can play file to called but not to caller |
| 21:57.47 | dippo_ | is 80ms latency considered unacceptable? |
| 21:57.54 | Darwin35 | ok I will send a note to rockeynet on it |
| 21:57.55 | festr_ | this is for annoucing "call can be monitored" |
| 21:57.59 | festr_ | dippo_: relatively ok |
| 21:58.04 | brad_mssw | http://www.teliax.com/forum/viewtopic.php?t=190&postdays=0&postorder=asc&start=15 |
| 21:58.06 | dippo_ | if so I may be screwed, because I don't stand a chance of getting much of anywhere lower than that :) |
| 21:58.09 | dippo_ | (good ol' comcast) |
| 21:58.10 | brad_mssw | some traceroutes on there |
| 21:58.20 | brad_mssw | dippo_: no, 80 isn't good at all |
| 21:58.39 | Darwin35 | we are doing our best but with the backbones screwing up it makes us look bad and its not really us |
| 21:58.42 | brad_mssw | dippo_: not good enough to even attempt to fax over, or anything else that requires quality |
| 21:58.46 | Darwin35 | this is the hard thing |
| 21:58.51 | dippo_ | if 80 is bad enough that an IAX trunk would be subject to quality or dropped calls i might be in a tough spot |
| 21:58.53 | Darwin35 | gives us the bad rap |
| 21:58.57 | brad_mssw | Darwin35: right, it's not your fault, it's rocknet's |
| 21:59.07 | dippo_ | because i don't think comcast's network gets much of anywhere under 60ms |
| 21:59.09 | _Sam-- | Darwin35: you COULD make it better on your customers by having boxes at more than one place! |
| 21:59.10 | dippo_ | in my experience |
| 21:59.15 | _Sam-- | i would be glad to help :) |
| 21:59.19 | Darwin35 | we are |
| 21:59.28 | Darwin35 | we have 3 new boxes going up soon |
| 21:59.38 | Darwin35 | 1 chicago 1 la 1 sf |
| 21:59.52 | brad_mssw | Darwin35: hmm, chicago may not be bad |
| 21:59.53 | Darwin35 | then looking to put 3 in cnada |
| 21:59.55 | _Sam-- | whats the point of 2 in cali and none on the ast coast? :) |
| 22:00.00 | brad_mssw | Darwin35: any eta ? |
| 22:00.00 | _Sam-- | east coast too |
| 22:00.22 | Darwin35 | its hard right now just trying to update hardware and clean up the update mess |
| 22:00.43 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 22:00.44 | Darwin35 | atlanta is on the list and ohio is on the list |
| 22:00.59 | Darwin35 | we plan to add servers where we can |
| 22:01.07 | brad_mssw | atlanta would be kick ass for me ... traffic gets routed from Jax to Atlanta ... basically 2-3 hops for me :) |
| 22:01.10 | Darwin35 | but it takes money and support time |
| 22:01.28 | _Sam-- | i have money but no time! |
| 22:01.44 | brad_mssw | now, right now you all do voip-coX ... and numbers are specific to a certain server, will that still be the case when those new servers come online? |
| 22:01.45 | _Sam-- | id be glad to invest a few bucks in your infrastructure :) |
| 22:01.46 | JMcA | brad_mssw: keep in mind that network paths don't always make sense geographically |
| 22:02.08 | brad_mssw | and you have to get an account moved to the new servers? |
| 22:02.13 | asterboy | time to disconnect...gotta try to unlock a pap2 with firmware 3.1.9(LSc)...Linksys has been locking the admin account to block downgrading the firmware. |
| 22:02.34 | festr_ | dippo_: voip quality is about changing latency. if you have constant 80ms, it is ok |
| 22:02.36 | _Sam-- | if they setup servers and they all use realtime for IAX /SIP users/peers...then you would be able to login to any box |
| 22:02.38 | brad_mssw | JMcA: right, that's true ... but UUNET is big in atlanta, and there's lots of peering arrangements there ... so I'd hope it wouldn't take another route |
| 22:02.45 | tdonahue | hi all, i'm having some problems with my asterisk install |
| 22:02.49 | festr_ | dippo_: 30ms treshold is still relatively ok |
| 22:03.02 | JMcA | brad_mssw: yeah, and in this day and age, in a major city like that, you have a pretty decent chance of that being true |
| 22:03.41 | tdonahue | when I use an application like Playback, Background or Voicemail, the sound files never get played back to the endpoint (a Snom 190) |
| 22:03.42 | asterboy | Pegger, I was putting some thought to the failover...you might look into setting up a second ISDN line with your provider...just don't activate the line so your don't have to pay monthly...anything goes down and your a phone call away to reactivate. |
| 22:03.58 | tdonahue | I have tried this with both Asterisk 1.2 and -HEAD |
| 22:04.00 | Darwin35 | ok I have reported the issue to rockynet |
| 22:04.28 | Darwin35 | they are looking into it |
| 22:04.31 | asterboy | anyway, bbl. |
| 22:04.32 | TedC | is there a way for asterisk to ask the far end of a SIP session to put a call on hold? |
| 22:05.02 | TedC | I know asterisk can hold the call itself, but I have an interop test that wants me to direct their end to hold. |
| 22:05.15 | tdonahue | on both, the RTP stream is set up from the phone to the server, but not the other way around |
| 22:05.33 | brad_mssw | _Sam--: what is iax.cc's server? |
| 22:05.47 | _Sam-- | iax2.sixtel.net |
| 22:06.16 | tdonahue | this problem is happening on a flat network when both the server and the 190 are plugged into the same switch. |
| 22:06.19 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:06.42 | brad_mssw | _Sam--: wow, avg latency for me is only around 38ms to them |
| 22:06.59 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
| 22:07.01 | tdonahue | anyone have any thoughts on what could be causing this problem? |
| 22:07.14 | brad_mssw | _Sam--: Jax -> DallasFW -> theplanet -> them ... |
| 22:07.36 | *** join/#asterisk sapo_original (n=Fenix@200.138.76.58) |
| 22:07.45 | brad_mssw | 6ms faster than JN for me |
| 22:07.53 | Darwin35 | ok I have a update pls stand by |
| 22:07.54 | sapo_original | hi.. anybody can help me with a digium te110p ??? |
| 22:08.48 | *** join/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 22:08.54 | darwin_35 | A DDoS event just before 2pm MST today caused an outage of approximately |
| 22:08.54 | darwin_35 | ten minutes as extremely high packet rates caused routing protocol |
| 22:08.54 | darwin_35 | instability. The attack subsided as we identified the victim but before |
| 22:08.54 | darwin_35 | we could have it null routed upstream. We apologize for any inconvience |
| 22:08.54 | darwin_35 | this caused. Because of the way it transits the network it impacted both |
| 22:08.55 | darwin_35 | Voice and Data services. |
| 22:10.04 | tdonahue | darwin_35: who was that? |
| 22:10.29 | syle | so they firewalled off an ip on the router, or switched bgp routes for that person hehe |
| 22:10.33 | SERGEUS | is there any way to get current value of "billsec" in the middle of diaplan? |
| 22:10.45 | brad_mssw | darwin_35: well ... that doesn't explain why latency has been aweful for the past 3 weeks ... compared to prior |
| 22:10.49 | mog_work | whats wrong sapo_original |
| 22:11.22 | SERGEUS | i'm trying ast_cdr_getvar(chan->cdr, "billsec",&billsectmp, tmpbs,sizeof(tmpbs), 0); but i got 0... |
| 22:11.28 | syle | I have also seen companies use that excuse as a way to upgrade their router as it couldn;t support enough traffic who knows |
| 22:11.32 | *** part/#asterisk darwin_35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 22:12.35 | *** part/#asterisk QbY (n=QbY@adsl-068-209-210-253.sip.cha.bellsouth.net) |
| 22:12.47 | Darwin35 | i am looking at a few now |
| 22:12.51 | Darwin35 | let me seee |
| 22:13.01 | SERGEUS | as far as understand, * calculate "billsec" value upon cdr saving... but i think there shoulf be way to get this value... |
| 22:13.06 | brad_mssw | any way to turn off routing via cogent to those not on cogent's network ? |
| 22:13.42 | brad_mssw | basically assume unless your ip address terminates at cogent, that you're not wanting cogent |
| 22:15.30 | dca[laptop] | i heard Teliax is looking at Viawest as an alternate Datacenter |
| 22:15.44 | *** join/#asterisk bweschke_ (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 22:17.58 | syle | SERGEUS i think you need to understand how modules work |
| 22:19.07 | syle | your not gonna get billsec in the middle of a call, you have to call your hooks into the cdr functions |
| 22:19.08 | SERGEUS | syle, would you so kind to explain me it? :-))) |
| 22:19.47 | syle | noone spoon fed me dude, read the source code like everyone else :) |
| 22:20.13 | SERGEUS | yes i'm looking for it right none :_ |
| 22:20.24 | SERGEUS | right now :) |
| 22:21.33 | syle | what is your goal btw? |
| 22:22.33 | syle | if your talking about billing software, that crap is not easy |
| 22:22.44 | SERGEUS | to calculate cost of call |
| 22:22.46 | syle | i;m already 3 months into c coding one of those |
| 22:23.20 | syle | if thats all you want to do why not just use database triggers |
| 22:23.39 | [av]bani | anyone using grandstream phone with * ? |
| 22:23.40 | syle | insert comes in, calculate cost with the trigger |
| 22:23.55 | SERGEUS | and i also want to add custom fields to cdr so i'll need to edit it anyway |
| 22:24.18 | syle | that should take you 5 min to add new fields |
| 22:24.20 | SERGEUS | syle, that is "astbill" style :) |
| 22:24.43 | syle | lol astbill sucks |
| 22:24.57 | syle | prob why i started coding one |
| 22:25.01 | SERGEUS | i don't want to use triggers, i want to use mysql 4.x |
| 22:25.12 | syle | mysql 5.x is stable |
| 22:25.33 | *** join/#asterisk Los415 (n=los415@c-24-126-63-65.hsd1.ca.comcast.net) |
| 22:25.43 | SERGEUS | i'm a little bit conservative :) |
| 22:25.48 | [av]bani | what's optimal voice frames per tx to use with * and g711u ? |
| 22:25.53 | syle | lol |
| 22:26.03 | justinu | 20ms |
| 22:26.09 | SERGEUS | we are still runing 3.23 at my work :) |
| 22:26.13 | dippo_ | is it normal for most VOIP trunking providers to bill for calls <1min as a full min? |
| 22:26.16 | syle | oww god |
| 22:26.23 | syle | fire the admin :) |
| 22:27.12 | syle | thats like not going to 5.x and letting your php programmers use the full potential of new features to build great sites |
| 22:27.18 | brad_mssw | dippo: don't know about 'most' |
| 22:27.19 | syle | php 5.x i mean |
| 22:27.25 | [av]bani | k, i should use 1 for gxp2000 |
| 22:27.26 | brad_mssw | dippo: teliax doesn't seem to, but junction networks does |
| 22:27.31 | dippo | i can't remember if teliax did, but jnctn does |
| 22:27.32 | dippo | yeah |
| 22:27.42 | dippo | i wonder what sort of difference that'll make for our bottom line |
| 22:27.50 | dippo | i'd imagine we have quite a few calls <1min |
| 22:27.53 | SERGEUS | syle, we don't use php :) c && perl :) |
| 22:27.55 | dippo | no answer/voicemail hangups |
| 22:28.07 | syle | embedded perl i hope |
| 22:28.32 | syle | perl is slower than php for webstuff |
| 22:28.34 | SERGEUS | syle, well our software doesn't updated too oftenly :) |
| 22:28.37 | syle | perl good for everything else |
| 22:28.56 | SERGEUS | holywars are fun :) |
| 22:29.05 | syle | url? |
| 22:29.22 | syle | i promised not to hack your mysql 3.x lol |
| 22:29.30 | *** join/#asterisk craigb_ (n=craig@12-220-217-231.client.insightBB.com) |
| 22:29.31 | SERGEUS | hehehe :) |
| 22:29.59 | SERGEUS | syle, http://www.maxnet.ru - only site uses php :) |
| 22:30.16 | syle | oww wow |
| 22:30.20 | syle | what language is that? |
| 22:30.30 | SERGEUS | syle, russian :) |
| 22:30.48 | craigb_ | I've got users who want to use the forward button on their Polycom 501's to their cellular phones, but let their cell phone voicemail pick up instead of the asterisk extension voicemail if no answer. is this possible? |
| 22:30.48 | syle | i love russian c programmers |
| 22:30.50 | _Roey | yennnnnternet |
| 22:30.51 | _Roey | heh |
| 22:30.57 | _Roey | yentyernyet |
| 22:31.10 | tdonahue | Does anyone have any idea what might cause one way audio from Playback, but not when I use an application like Milliwatt? |
| 22:31.19 | SERGEUS | syle, why? :) |
| 22:31.32 | syle | i had a good friend that was a russian c program, but the maffia forced him to work for them |
| 22:31.40 | syle | s/program/programmer |
| 22:31.58 | syle | now he works at a security firm |
| 22:32.52 | syle | guy who me some of the most advanced trojans, hell he could ping a box and open a root shell on you, hack anything, guy was amazing, great in c and assembler |
| 22:33.06 | SERGEUS | syle, btw, do you have a list of features of your billing? |
| 22:33.27 | syle | yeah lets talk in PM |
| 22:36.05 | syle | so i never heard from that dude again, so nice to meet another russian, hopefulyl mafia don;t take you to :) |
| 22:36.06 | syle | PM |
| 22:38.04 | *** join/#asterisk F41L4M3 (n=SCOTT@209.246.17.218) |
| 22:38.21 | F41L4M3 | i need some asterisk help plz |
| 22:39.44 | F41L4M3 | anyone? |
| 22:40.17 | lesouvage | F41L4m3: wazzup? |
| 22:40.23 | iDunno | (so, narrow it down and the people that are alive might help) |
| 22:40.55 | F41L4M3 | i need to know if asterisk can act as a phone line simulator |
| 22:41.51 | *** join/#asterisk Lots (n=Aces1Up@ip70-189-157-31.lv.lv.cox.net) |
| 22:41.53 | justinu | yes |
| 22:41.55 | justinu | it can |
| 22:41.58 | *** join/#asterisk jonathh (n=asd@host86-142-221-157.range86-142.btcentralplus.com) |
| 22:42.04 | *** join/#asterisk maxis (n=maxis@c-f3c272d5.013-146-73746f29.cust.bredbandsbolaget.se) |
| 22:42.18 | F41L4M3 | if its not too much trouble i would like some info on how to do that |
| 22:42.42 | Lots | hey all didn't know where else to ask this, I have a client that needs a 40-line autodialer that can dial a database of phone numbers with a auto-message and choice of option to leave a voice mail message. what is the best option to do this via voip? |
| 22:45.42 | F41L4M3 | if any one wants to pm me with the info that would be great :) |
| 22:46.03 | *** join/#asterisk zu (n=raz@30-pool1.ras14.floca.alerondial.net) |
| 22:46.05 | zu | hy all |
| 22:46.21 | zu | is cut function broken when used in ael in 1.2.1 |
| 22:46.29 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
| 22:47.57 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 22:48.38 | gaz00 | wow...i wonder what that's for. |
| 22:48.50 | *** join/#asterisk FastJack (i=fastjack@p5091FD9A.dip.t-dialin.net) |
| 22:49.45 | Katty | can someone tell me where recorded audio conversations are located? |
| 22:50.13 | F41L4M3 | can anyone tell me how to use asterisk as a phone line simulator in gentoo |
| 22:50.43 | *** join/#asterisk Soul (n=Soul@87-196-9-93.net.novis.pt) |
| 22:50.45 | Katty | anthm: where are recorded .wav calls stored? |
| 22:51.03 | anthm | recorded how? |
| 22:51.15 | Katty | through my *9 features shortcut |
| 22:51.41 | zu | found the proper ael syntax for substrings |
| 22:51.45 | zu | back to codin |
| 22:51.46 | Katty | i thought it was /var/lib/asterisk/sounds but i don't see it |
| 22:52.03 | justinu | check /var/lib/asterisk/recordings |
| 22:52.35 | Katty | nada. |
| 22:52.41 | justinu | :( |
| 22:52.53 | Katty | where conf file specifies where to dump it? |
| 22:53.44 | w14 | Can anyone help with a SIP bridging problem I have? |
| 22:53.49 | w14 | I am trying to forwward a call received on one SIP trunk to a mobile phone via second SIP channel. |
| 22:53.59 | Katty | anthm: which conf file specifies where to dump recorded conversations? |
| 22:54.48 | anthm | is it /var/spool/asterisk/monitor ? |
| 22:54.49 | w14 | the mobile rings, but when I answer, there's only silence. Both calls are held open by Asterisk, but Asterisk seems to fall out of the loop, leaving the calls orphaned. |
| 22:54.56 | Katty | anthm: oh ah. |
| 22:55.31 | Katty | hrmm. |
| 22:55.39 | Katty | those dates don't corespond to today. |
| 22:55.51 | w14 | the console shows Attempting native bridge of SIP/<incoming call> and SIP/<outgoing channel> |
| 22:55.58 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 22:56.02 | w14 | and then just sits there |
| 22:56.10 | Katty | hmm, not good |
| 22:56.19 | anthm | how are you recording ? |
| 22:56.31 | Katty | anthm: *9 |
| 22:56.53 | Katty | anthm: that's the right folder, but only old stuff is in there. |
| 22:57.06 | *** join/#asterisk RoyK (n=roy@ti211310a080-9456.bb.online.no) |
| 22:57.20 | F41L4M3 | Asterisk as a phone line simulator? HELP ME PLEASE! |
| 22:57.45 | *** join/#asterisk ToTo (n=ToTo@host56-162.pool875.interbusiness.it) |
| 22:57.53 | anthm | doh |
| 22:58.24 | Katty | anthm: which conf file is the directory specified in? |
| 22:58.27 | Katty | anthm: or is it? |
| 22:59.12 | anthm | for one touch i think there is no choice |
| 22:59.17 | Lots | hey syle i would like to hear your ideas although i'm not registered with this channel so can't send pm's. |
| 22:59.19 | Katty | anthm: kk |
| 22:59.52 | anthm | is it an older version? |
| 23:01.36 | Ariel_ | w14, do you have on both sip accounts canreinvite=no |
| 23:01.56 | Katty | wow, show version is completely useless. |
| 23:02.12 | w14 | Ariell_: yes, checked that |
| 23:02.23 | anthm | back when i added that code it used to be auto generated unless you set the variable TOUCH_MONITOR |
| 23:02.40 | anthm | which let you pick your own base name |
| 23:02.52 | Katty | well there are other files in that directory |
| 23:02.57 | Katty | and i know i haven't set that variable |
| 23:03.01 | anthm | you may want to turn on the console and watch if it notices when you dial the code to record |
| 23:03.20 | anthm | set verbose like 4 |
| 23:03.20 | Katty | so...i can only therefore assume, this guy did not record the conversation |
| 23:03.34 | anthm | and try it while you are watching the cli |
| 23:03.50 | anthm | you may need to make such an assumption yes |
| 23:04.13 | Katty | yeah i just recorded one myself...that's definately the right directory |
| 23:04.16 | Katty | anthm: thanks (= |
| 23:04.30 | Zodiacal | anyone know off hand where i set the security code for the flash operator panel? |
| 23:05.15 | F41L4M3 | anyone anyone |
| 23:07.52 | *** join/#asterisk kart_179 (n=kart@201-40-207-109.mganm702.dsl.brasiltelecom.net.br) |
| 23:09.10 | dippo | hm, my trunk to junctionnetworks appears to be ignoring allow=gsm |
| 23:09.16 | dippo | or asterisk seems to be ignoring it, not sure which |
| 23:09.18 | F41L4M3 | Asterisk as a phone line simulator? HELP ME PLEASE! |
| 23:11.28 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 23:12.05 | *** join/#asterisk warthawg_ (n=warthawg@cpe-66-68-176-215.austin.res.rr.com) |
| 23:12.45 | warthawg_ | can anyone point me to a sample sip.conf for a zultys ZIP 2x2? |
| 23:14.08 | shido6 | subway is tasty |
| 23:14.53 | shido6 | sea food & bacon with southwest sauce, black olives, pickles on Parmesan and oregano bread |
| 23:14.53 | Ariel_ | Zodiacal, look in the file on /var/www/html/panel/op_server.cfg |
| 23:15.36 | Zodiacal | ariel_ Thank You! |
| 23:15.40 | F41L4M3 | some one help F41L4M3 he is getting impacient |
| 23:16.01 | Ariel_ | F41L4M3, why? what is it you want to do????? |
| 23:16.06 | *** join/#asterisk RaYmAn-B1 (i=rayman@x1-6-00-40-63-da-39-3f.k191.webspeed.dk) |
| 23:16.19 | w14 | Zodiacal: I think its op_server.cfg, in /var/www/panel, or wherever you have it |
| 23:16.23 | warthawg_ | F41L4M3, patience is a virtue achieved by few men, and never by women |
| 23:16.43 | F41L4M3 | i want to use asterisk as a phone line simulator |
| 23:17.12 | Ariel_ | phone line simulator.... so put a fxo card and setup the pbx.... |
| 23:17.20 | *** part/#asterisk Ayano (n=Ayano_@adsl-70-245-190-90.dsl.spfdmo.swbell.net) |
| 23:17.24 | linlin | http://i11.photobucket.com/albums/a168/linmaster/parabolic/parabolic001.jpg |
| 23:17.28 | linlin | http://i11.photobucket.com/albums/a168/linmaster/parabolic/parabolic002.jpg |
| 23:17.33 | [TK]D-Fender | Ariel_: that'd be FXS card.... |
| 23:17.37 | linlin | look what i have and you dont |
| 23:18.07 | Ariel_ | [TK]D-Fender, yes if there going to plug in a normal phone yes. Correct |
| 23:18.44 | [TK]D-Fender | Ariel_ : He did say a phone LINE simulator. |
| 23:18.58 | [TK]D-Fender | F41L4M3 : Can you tell us a bit more about what that implies to you? |
| 23:18.59 | Ariel_ | [TK]D-Fender, yes I did not read correctly |
| 23:19.20 | Ariel_ | linlin, hehe so what are you going to do with it.... |
| 23:19.40 | F41L4M3 | I work at a store with no phone lines and we need to do repairs on machines involving testing the modem |
| 23:19.59 | F41L4M3 | so i just need a dial tone |
| 23:20.01 | Lots | syle can you go and join #blah |
| 23:20.21 | JMcA | F41L4M3: yes, it should handle that pretty well |
| 23:20.30 | F41L4M3 | and a # so i can use windows hyperterminal to dial in to my desk machine |
| 23:20.32 | Ariel_ | F41L4M3, just get an sipura |
| 23:20.36 | Lots | syle ok join umm #4355 |
| 23:20.37 | [TK]D-Fender | F41L4M3 : What kind of test? Just having a tone on the line at all? |
| 23:21.07 | [TK]D-Fender | F41L4M3 : nothing related to # will give you a DATA grade signal for modems.... |
| 23:21.13 | [TK]D-Fender | * |
| 23:21.53 | F41L4M3 | i don't need to surf the internet with them connected i just need to be able to dial in |
| 23:22.32 | JMcA | you may not get a top speed connection from two modems connected through an asterisk box, but they should at least negotiate a connection |
| 23:22.41 | Ariel_ | modems over voip does not work very well. |
| 23:23.03 | JMcA | Ariel_: yeah, so you keep it local to the box, should work reasonably well |
| 23:23.27 | JMcA | F41L4M3 you do konw that you can just plug a phone line between two modems and get them to connect |
| 23:23.40 | Ariel_ | JMcA, only if you use a tdm400 type board or a channel bank calling between each other. |
| 23:23.43 | w14 | anyone any suggestions for my bridging problem? |
| 23:24.01 | JMcA | Ariel_: sure...which is what I was implying...if not saying explicitly |
| 23:24.12 | Ariel_ | w14, we need more info. do sip debug and then make the call and see what it's doing |
| 23:24.16 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:25.05 | Ariel_ | well it's time to drive home... I should be back on line in about an hour....see you all later. |
| 23:25.16 | shmaltz | anybody here getting zttest results of 100% with a Digium TE110P? |
| 23:25.27 | F41L4M3 | i do need a dial tone |
| 23:25.41 | F41L4M3 | to be sure the modems are completely functional |
| 23:25.58 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 23:26.03 | JMcA | F41L4M3: then an * box with two FXS ports (on, for example, a TDM400p) should do what you need |
| 23:26.10 | wunderkin | shmaltz, nearly impossible.. thats nothing to worry about though.. i think its 99.987 or something you need to be above or something like that |
| 23:26.25 | JMcA | as has been mentioned...I wouldn't count on perfect modem connections, but they should get a reasonable connection |
| 23:26.48 | shmaltz | wunderkin, I know that, however since Digium told me that they did get that, I'm trying to figure out if anybody else did as well |
| 23:27.15 | wunderkin | as an average? i doubt it |
| 23:28.15 | F41L4M3 | any ideas where to find a detailed howto including the configs |
| 23:28.49 | mog_work | wunderkin, |
| 23:28.50 | JMcA | F41L4M3: configuring asterisk is not a trivial task...your config will be relatively simple, but it will take some figuring out on how to do it... |
| 23:28.50 | mog_work | !!! |
| 23:28.55 | mog_work | when are we gonna fix that bug |
| 23:29.23 | JMcA | F41L4M3: asterisk is a PBX, so you're setting up a PBX with two extensions and you're going to dial from one to the other, so most of the intro docs are going to show you how to do that |
| 23:29.44 | lesouvage | i'm trying to format ${CALLERIDNUM} so it ends up in a standard format. exten => s/0Z.,3,SetVar(BELLEND_NUMMER=0031${CALLERIDNUM:1}) What am I doing wrong? |
| 23:29.59 | wunderkin | mog mog mog.. now? |
| 23:30.15 | mog_work | let me finish this one bug |
| 23:30.18 | mog_work | and then we are go |
| 23:33.19 | F41L4M3 | thanks to all that helped i appreciate it |
| 23:34.26 | lesouvage | Or isn't numbermatching supposed to work on the s extension? |
| 23:34.44 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 23:36.10 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 23:37.38 | shido6 | what is todays date? |
| 23:37.44 | Igbothom_III | today |
| 23:38.29 | jbroome | 18th |
| 23:38.40 | warthawg | january, 2006 |
| 23:41.30 | hugo-v6 | hiho |
| 23:41.37 | hugo-v6 | huh 1.2.2? |
| 23:43.52 | *** join/#asterisk Luhiwu (n=marsosa@200.63.89.217) |
| 23:44.39 | Luhiwu | hello all, i'm having problems with Pickup(), i get "No originating channel found" in the debug, anyone can help me with that? |
| 23:45.08 | warthawg | anyone have a working config for a zultys 2x2? |
| 23:45.21 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 23:46.31 | hugo-v6 | lots of bugifxes |
| 23:47.01 | hugo-v6 | someone with snom phones awake? |
| 23:48.55 | *** join/#asterisk Snooker (n=klayton@ras-41.expert.com.br) |
| 23:50.43 | gaz00 | noone's really awake. You might as well ask and hope that they wake up :p |
| 23:51.30 | Luhiwu | second try: i'm having problems with Pickup(), i get "No originating channel found" in the debug, anyone can help me with that? |
| 23:51.36 | warthawg | as cheap as these zultys phones are there has to be someone here using them |
| 23:52.59 | nettie | Hi guys, I'm wondering if anyone experienced problems sending fax through asterisk using alaw codec and the Linksys PAP2 as ata? thanx in advance. |
| 23:55.24 | hugo-v6 | gaz00: i cant belive thyll wake up ;) i get "forbidden" in the snom-display if the called party is busy. now ive the Busy() cmd in my extension (or PRI_CAUSE=17) and now im getting service unavail. which should be busy instead. i dunno why |
| 23:56.35 | *** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0) |
| 23:56.50 | hugo-v6 | nettie: im sending faxes through a analog <> isdn-converter which is connected to asterisk and * via isdn to pstn. works fine. dunno if that helps ;) |
| 23:59.21 | nettie | you're not using SIP then.. |
| 23:59.54 | nettie | and you're not using a voip codec I suppose.. |
| 23:59.59 | hugo-v6 | nettie: nope. also never looked on t.30 (or was it 38?) |
