| 00:01.13 | CrazyTux | Math`, :) |
| 00:01.41 | _VoiceMeUp_COM | and add to modules.conf |
| 00:02.10 | CrazyTux | Math`, MySQL RealTime: Failed to query database. Check debug for more info. where would the debug/log be? |
| 00:02.54 | *** join/#asterisk Defraz (n=t0tal@67.42.167.242) |
| 00:03.29 | Math` | set verbose 10 |
| 00:03.33 | Math` | core set debug 10 |
| 00:03.36 | Math` | and try again |
| 00:03.55 | Math` | and also... try |
| 00:03.55 | Math` | realtime mysql status |
| 00:05.15 | CrazyTux | Math`, says failed to query database, check debug for more info still |
| 00:05.19 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 00:05.22 | Paavum | Does anybody have any experience with SIP BLFs and a Grandstream GXP2000? I think I missed something but I cant seem to figure out exactly what I mean... when I do a core show hints I see all phones idle |
| 00:05.33 | CrazyTux | Math`, shows I'm connected |
| 00:05.43 | Math` | are your tables created? |
| 00:06.08 | Math` | pastebin your ext_config.conf |
| 00:06.13 | CrazyTux | Math`, yea, but may not be proper? |
| 00:06.19 | CrazyTux | Math`, I dont want all of the columns |
| 00:06.58 | Math` | just put all the columns and let them to their default values |
| 00:07.11 | CrazyTux | Math`, the only wiki I find is really old |
| 00:07.17 | CrazyTux | Math`, thats why I dont know whats what |
| 00:07.27 | CrazyTux | Math`, so what exactly can/should the struct be? |
| 00:07.45 | Math` | http://voip-info.org/wiki/view/Asterisk+RealTime+Voicemail |
| 00:07.58 | Paavum | Does anybody have any experience with SIP BLFs and a Grandstream GXP2000? I think I missed something but I cant seem to figure out exactly what I mean... when I do a core show hints I see all phones idle |
| 00:08.25 | Math` | Paavum: if noone answers, noone probably has |
| 00:08.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:09.47 | Paavum | Ok delete the GXP part from it |
| 00:09.50 | Paavum | I think its myc onfigs |
| 00:13.21 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 00:13.32 | *** join/#asterisk voiper1 (n=luke@ozvoip.dsl.onthenet.net) |
| 00:17.17 | CrazyTux | Math`, ok, I did that, no errors, but I setup mailbox # and password, and dosent seem to work? :( |
| 00:18.47 | CrazyTux | Math`, I had to basically get rid of the voicemail.conf file :) |
| 00:18.49 | CrazyTux | Math`, that fixed it |
| 00:20.03 | CrazyTux | Math`, thanks for the help :) |
| 00:22.41 | voiper1 | anyone experience half duplex sound with tdm02b card? |
| 00:33.42 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 00:34.36 | philippel | anyone have any insight into getting many message such as: |
| 00:34.44 | philippel | channel.c: Dropping voice to exceptionally long queue on ... |
| 00:34.58 | philippel | and: sched.c: Request to schedule in the past?!?! |
| 00:35.27 | philippel | just before the whole system goes south, and then needs rebooting to bring back alive |
| 00:35.49 | philippel | there is a fair amount of activty going on with call queues in this environment - and probably 20-30 simultaneous calls at once |
| 00:36.36 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 00:39.58 | ManxPower | philippel: those messages are a side effect of the crash. |
| 00:40.18 | ManxPower | They basically mean "The system is WAY too slow to do what you are trying to do." |
| 00:40.24 | philippel | that is what I was guessing at |
| 00:40.46 | *** join/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 00:41.10 | philippel | I was just asked to take a look - I noticed the system has 512MB of ram and 768MB swap - I wanted to do a little digging around in parallel to telling them to put a lot more memory and up the swap |
| 00:43.14 | ManxPower | see how much swap is being used. |
| 00:43.29 | pabs3 | is there any database of per-country numbering plans in extensions.conf format (specifically Australia)? |
| 00:44.30 | philippel | When it is active, it is middle of the night there (UK) where this system is |
| 00:44.35 | ManxPower | philippel: I'm from the USA where there is a very well defined and simple dialplan. As far as I can tell most countries just randomly decide to change their numbering plans, have variable lengthnumbers, etc. |
| 00:46.03 | philippel | ManxPower this is a fairly busy system, with a very fat feature rich dialplan doing a lot of stuff - it is no surprise at all, for what they are doing, it should have way more memory - but thought I would still look around for other clues in the log |
| 00:46.47 | *** join/#asterisk wotcha (n=jim@cust4716.qld01.aanet.com.au) |
| 00:47.04 | *** join/#asterisk killfill (n=killfill@201.238.233.3) |
| 00:47.06 | killfill | hi. |
| 00:47.26 | killfill | should cat /dev/zap/ctl output something? |
| 00:50.08 | JT | pabs3: how much of the australian dialplan do you need? |
| 00:51.23 | killfill | would anyoue please cat the dev for me? thanks.. |
| 00:51.25 | pabs3 | JT: not really sure, probably just differentiated on charging rate |
| 00:51.38 | pabs3 | (for an office) |
| 00:51.39 | JT | pabs3: ? |
| 00:51.59 | JT | pabs3: the full dialplan is very big and probably has stuff you don't need |
| 00:52.42 | pabs3 | yeah |
| 00:52.48 | JT | pabs3: what state? |
| 00:53.19 | pabs3 | NSW |
| 00:53.35 | JT | oh, easy then, i probably already have what you want |
| 00:54.17 | pabs3 | (by charging rate I mean like free, local, $50/min, etc) |
| 00:54.37 | aptura | 50 is peanuts here in vancouver |
| 00:55.20 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:58.28 | *** join/#asterisk Dantenix (n=Dante@61-163-126-200.fibertel.com.ar) |
| 00:59.09 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 00:59.10 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:00.08 | Dantenix | hi all, someone has experience connecting an axis camera with asterisk? |
| 01:00.27 | JT | pabs3: still there? |
| 01:00.29 | aptura | I used to be a axis print server support tech :) |
| 01:00.42 | sevard | Dantenix: aren't those axis cameras the one with the built in ftp client? |
| 01:01.03 | pabs3 | JT: yup |
| 01:01.24 | aptura | thay are very sweet cameras just a bit on high side. |
| 01:01.32 | JT | ~sydney |
| 01:02.19 | Dantenix | sevard, yes I've read something like that... but I want to use it as a door comm |
| 01:02.26 | JT | ~syddialplan |
| 01:02.28 | jbot | syddialplan is, like, http://www.pastebin.ca/481857 |
| 01:02.38 | pabs3 | thanks :) |
| 01:02.40 | JT | pabs3: look at that, i just uploaded that |
| 01:03.45 | JT | pabs3: it doesn't handle things like callerid barring/unbarring prefixes and premium rate services etc |
| 01:05.10 | *** join/#asterisk Fieldy (i=pgRqAson@gentoo/contributor/Fieldy) |
| 01:05.49 | *** join/#asterisk Blackthorn (n=support@w-l4.smyth.net) |
| 01:06.44 | Blackthorn | Does anyone know if thers any problem in sending a iax call from asterisk 1.2x to a 1.4 server? |
| 01:11.19 | *** join/#asterisk dS_mEmX (n=saber@c74-195-156-49.amrlcmta01.tx.dh.suddenlink.net) |
| 01:11.41 | sevard | Does anyone remember which file does allison talk about the spam and viagra |
| 01:13.33 | *** part/#asterisk dS_mEmX (n=saber@c74-195-156-49.amrlcmta01.tx.dh.suddenlink.net) |
| 01:15.32 | *** join/#asterisk _anand_ (n=anand@59.94.98.157) |
| 01:16.52 | sevard | ah, spam! |
| 01:17.22 | *** join/#asterisk Defraz (n=t0tal@67.42.167.242) |
| 01:20.30 | Math` | sevard: funny one lol |
| 01:23.14 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 01:24.22 | *** join/#asterisk andrewc (n=andrewc@67.50.65.228) |
| 01:24.51 | *** join/#asterisk Tond (n=tond@CPE0014bf30c190-CM0010954a5ffa.cpe.net.cable.rogers.com) |
| 01:26.39 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 01:27.00 | Tond | Hi I have aterisk 1.4.4 but haveing a wierd MOH problem. When I force MOH to work using an extention like exten => 6000,2,MusicOnHold() it works fine. But when I call my mobile phone and put the call on Hold I don't hear the MOH.. On console it says: Started music on hold, class 'default' followed by Stopped music on hold on SIP/..... Any ideas why? |
| 01:27.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:28.59 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 01:29.36 | DrukenLPY | Tond: if you start talking into the cellphone while on hold do you hear the music? |
| 01:29.58 | LeddyHM | Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.0.50 |
| 01:30.05 | LeddyHM | any ideas? |
| 01:30.45 | Tond | hrm.. let me try that |
| 01:30.45 | Tond | :) |
| 01:30.45 | Tond | u think it is VAD ? |
| 01:31.18 | Tond | DrukenLPY> i tried it and no i don't hear anything |
| 01:31.47 | Tond | DrukenLPY> the issue is that right after MOH gives the start message on console it follows it by a stop message |
| 01:31.54 | Tond | so it starts and then immedietly stops |
| 01:32.29 | Tond | but if i dial the extention 6000 which plays MOH, it will work fine... It is so wierd |
| 01:33.15 | _Sam-- | does anyone know if this is a valid SIP_HEADER function: SIP_HEADER(Alert-Info) |
| 01:33.26 | Math` | _Sam--: you need to assign a value... |
| 01:33.56 | Math` | and the new way of doing it is.. SIPAddHeader(Alert-Info: bellcore-r1) |
| 01:33.59 | _Sam-- | im trying to GET a value from the SIP_HEADER to distinguish which number was dialed |
| 01:34.10 | Math` | ah you're getting it |
| 01:34.22 | Math` | thats the correct syntax then |
| 01:34.25 | _Sam-- | ive checked my sip debug, and i cant seem to see anything coming |
| 01:34.40 | Math` | well that doesnt mean anything is wrong with your dialplan |
| 01:34.48 | _Sam-- | i know that -- my dialplan works fine... |
| 01:36.00 | _Sam-- | but, for exampkle, a line like this doesnt echo anything back: |
| 01:36.07 | _Sam-- | exten => 212202XXXX,1,Noop(Alert-Info -> '${SIP_HEADER(Alert-Info)}') |
| 01:36.20 | _Sam-- | in the console i see the Alert-Info -> |
| 01:36.25 | _Sam-- | but nothing for the value |
| 01:36.32 | [TK]D-Fender | LeddyHM, Just spam messages thrown out by Polycoms, nothing to worry about |
| 01:36.35 | Math` | and sip debug was showing a value for that header? |
| 01:36.50 | _Sam-- | i didnt see anything in SIP debug for that header, but was expecting to |
| 01:36.59 | LeddyHM | tk: was hoping not to see them :) |
| 01:37.01 | _Sam-- | based on this: http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice+distinctive+ring+support |
| 01:38.11 | Math` | distinctive ring is sent in Alert-Info thats correct |
| 01:38.21 | Math` | but if you're not seeing the header in sip debug it means its not sent |
| 01:40.22 | *** join/#asterisk Zipper_32 (n=None@d154-5-87-181.bchsia.telus.net) |
| 01:40.26 | _Sam-- | thanks for the help, i will try to find out why i am not receiving that header. |
| 01:41.48 | _Sam-- | THANK |
| 01:42.14 | Zipper_32 | I'm having a bit of trouble with my TDM400P, I have 3 FXO modules installed which are working perfectly, however, when I went to install a FXS module today, I can't seem to get the configuration right so that the "Channel 4: No such device or address" doesn't show up each time I try to start asterisk. |
| 01:42.41 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 01:42.44 | Zipper_32 | I've looked at the configuration examples for Zapata.conf, but I can't seem to figure out what I may be doing wrong. |
| 01:43.02 | [TK]D-Fender | Zipper_32, pastebin your zaptel & zapata |
| 01:43.04 | [TK]D-Fender | ~pb |
| 01:43.17 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:43.31 | Zipper_32 | Will do, one moment, thanks |
| 01:43.42 | *** join/#asterisk netcrusher88 (n=keke@unaffiliated/netcrusher88) |
| 01:44.07 | Math` | is there a linux app to "pipe to pastebin" |
| 01:44.25 | netcrusher88 | so, asterisk says IAX doesn't know how to authenticate me onto FWD for outgoing calls - incoming work fine though |
| 01:44.30 | Math` | like cat extensions.com | pastebin |
| 01:44.30 | Math` | http://pastebin.com/blah |
| 01:46.58 | netcrusher88 | any ideas? |
| 01:47.58 | Zipper_32 | My Zap problem is now here: http://pastebin.ca/481906 FXO channels seem to work, but the 4th module FXS does not. |
| 01:49.26 | Math` | netcrusher88: yeah your peer settings are probably not set properl |
| 01:49.26 | Math` | properly* |
| 01:50.36 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 01:51.08 | [TK]D-Fender | Zipper_32, ok, that looks fine. No did you remember to plug int he MOLEX CONNECTOR to your card? |
| 01:51.58 | Zipper_32 | Yes, my ztcfg -vvv shows "4 channels configured." |
| 01:51.58 | Zipper_32 | FXS, FXS, FXS, FXO |
| 01:52.21 | Zipper_32 | The Molex is plugged in. |
| 01:52.44 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 01:54.18 | [TK]D-Fender | Zipper_32, pastebin "dmesg" |
| 01:55.52 | Zipper_32 | dmesg: http://pastebin.ca/481914 |
| 01:56.31 | Zipper_32 | Oddly enough, I'm getting this error now: http://pastebin.ca/481915 |
| 01:57.30 | _anand_ | Any experiences on interfacing an Asterisk system with Avaya Call Manager 2.0 based systems? |
| 01:58.17 | aptura | x100p installed |
| 01:58.22 | Math` | last time I heard someone trying to interface with avaya hardware the avaya system fried a cisco gateway |
| 01:58.39 | [TK]D-Fender | yeah I saw that |
| 01:58.41 | Math` | the cisco wasnt plugged in and the leds were turning on and off then bang, no more |
| 01:58.51 | [TK]D-Fender | Zipper_32, looks like you have an X100P + TDM400P |
| 01:59.12 | [TK]D-Fender | Zipper_32, Your X100P seems to initialize first, so you should have 1-4 = FXO, 5 = FXS |
| 01:59.24 | aptura | Math, that sounds like a serios Impedence mismatch causing the tx line finles to go. |
| 01:59.56 | Math` | aptura: I just heard the story, didnt play with it |
| 02:00.15 | aptura | I wonder if that is a common issue among competing pbx equipment or not. |
| 02:00.35 | Zipper_32 | I'll get rid of that x100p, |
| 02:00.35 | Math` | kinda hard competition... |
| 02:00.35 | Zipper_32 | I didn't even realize it would initialize first. Thanks. |
| 02:00.53 | Math` | a buddy of mine bought an x100p and sometimes he has to change pci slot for it to be recognized lol |
| 02:01.30 | Tond | Hi is there a way that i can pull a value from MySQL db right in the dialplan? Or must I write a PHP and call it using AGI? |
| 02:01.31 | _anand_ | E1 trunking remains a possibility. We tried with interfacing through FXO/FXS, but traffic was one way |
| 02:01.51 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 02:02.02 | aptura | Math I know it sounds really odd but its very possible both equipments fxo/fxs channls impedence bridges are not following standards. To much of a load imbalance and the heat generated can fry the front end electronics. |
| 02:02.03 | Zipper_32 | .So it's dmesg that shows me that the x100p initializes first? |
| 02:02.37 | *** join/#asterisk andrewc (n=andrewc@67.50.65.228) |
| 02:03.44 | Zipper_32 | Wow... it's all working now. Thanks [TK]D-Fender and aptura, =) |
| 02:03.57 | Zipper_32 | I can't believe that oversight |
| 02:03.58 | aptura | I have seen a extreem case of final overheating on a ICOM tranciver once when something caused a ballanced line to become unbalanced and all the discreate smt's around the final slid off the circuit board when the finals overheated. |
| 02:04.02 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 02:04.29 | pabs3 | is there a shorter way of writing Goto(${MACRO_CONTEXT},${MACRO_EXTENSION},${MACRO_PRIORITY}) ? |
| 02:04.38 | pabs3 | Pop() or something? |
| 02:05.47 | JT | pabs3: how did you find those patterns i sent? |
| 02:06.49 | pabs3 | very handy, thanks :) |
| 02:08.53 | netcrusher88 | how does one use the ${EXTEN} var to get the entire extension? |
| 02:09.12 | Tond | I have a MySQL table with a list of allowed caller IDs. Now what I need to know if that how I can connect to that table and retreive the info from the Dial Plan? What is the best way of approaching this. (I have Asterisk 1.4.4 and I can do this manualy in the dial plan, just need to make it dynamic by connecting it to a DB) |
| 02:09.28 | JT | i derived some of the less obvious patterns (like exactly what all the regional number ranges were) by reading the federal dialplan from ComLaw |
| 02:09.32 | JT | :) |
| 02:10.24 | [TK]D-Fender | Tond, "core show function ODBC" |
| 02:11.03 | Tond | Thanks.. Is there also a MySQL one? Just like the Asterisk Realtime where I get a choice between ODBC and MySQL ? |
| 02:11.31 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 02:12.43 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 02:13.03 | Tond | [TK]D-Fender> "core show function ODBC" doesn't return anything.. Do I need to load the module or something? and I remember reading somweher that using both ODBC and MySQL for realtime can cause issues... is that true? |
| 02:13.53 | [TK]D-Fender | Tond, I'm betting you didn't have UnxiODBC installed when you compiled *. Go set that up now and rebuild |
| 02:14.33 | Nugget | You should use ODBC so that when you later discover how crappy MySQL is it will be less painful to migrate to something better. :) |
| 02:15.17 | Tond | [TK]D-Fender> probably not, because i wan't planning on using ODBC. But is it reliable to use with Asterisk? |
| 02:15.41 | [TK]D-Fender | largely |
| 02:16.04 | Tond | Nugget> What do u recommend then? If I need to retrieve my sip-peers and CDRs from a Databse, what should i use? |
| 02:17.10 | Nugget | I was mostly just taking a cheap shot at mysql, which I think is a pretty dismal database. I'd encourage you to look into postgresql as an alternative, but certainly don't distract yourself from asterisk over my database dogma. |
| 02:17.14 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 02:17.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 02:17.33 | Tond | :) |
| 02:17.49 | *** join/#asterisk Blackthorn (n=support@w-l4.smyth.net) |
| 02:18.12 | Tond | so having asterisk realtime with MySQL and using ODBC in dialplan wn't cause any problems then? |
| 02:18.21 | Nugget | I have no idea. |
| 02:18.29 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 02:18.50 | Nugget | well, I mean I have no idea about mixing odbc and mysql drivers. |
| 02:18.58 | Nugget | accessing mysql via the odbc driver should be just fine |
| 02:19.11 | Blackthorn | I have a pri -- Asterisk 1.2 box --stetup iax -- Asterisk 1.4 box. And my authentication is now working and i'm getting request '@default' does not exist. I have a working default profile in extensions.conf.... |
| 02:19.11 | Nugget | (as fine as mysql can be, at least) |
| 02:19.19 | Tond | k, tnx :) |
| 02:22.36 | Math` | Blackthorn: check your contexts |
| 02:22.49 | Math` | uhm what exactly is the other box trying to call |
| 02:22.54 | Math` | how do you Dial() |
| 02:25.54 | Blackthorn | the recieve iax.conf file has "context=default". On the console when the call comes in it says there is no @default... but in extensions.conf it's there.. and inuse... |
| 02:26.56 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 02:27.18 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 02:30.06 | Tond | [TK]D-Fender> after installing unixODBC do i need to make any config changes to asterisk for it to see / connect to the ODBC DB (other than recompiling)? |
| 02:31.58 | [TK]D-Fender | Tond, you WILL have to set up your DSN to connect to your MySQL DB |
| 02:32.35 | Tond | [TK]D-Fender> ya, but that is a unixODBC config i need to make, correct? |
| 02:33.01 | Blackthorn | the incoming call in the iax with context=default does refer to the default context in extensions.conf correct? |
| 02:33.02 | [TK]D-Fender | Tond, Correct |
| 02:33.28 | Tond | as far as asterisk goes, where do i tell it to use that DSN to connect to the DB and get info? Is there any online doc on this that i can read? |
| 02:34.11 | [TK]D-Fender | Tond, Check out the WIKI & the BOOK. |
| 02:34.19 | [TK]D-Fender | Tond, I don't know the particulars. |
| 02:34.41 | Tond | Ok thanks.. The WIKI hardly has anything about asterisk 1.4.x |
| 02:35.12 | Math` | thats not true |
| 02:35.13 | CCFL_Man2 | i need to make a wav file of my voicemail greeting |
| 02:35.23 | Math` | but most of the doc is in the cli :) |
| 02:35.43 | Strom_M | man, this is a weird problem. I have telephone sets on channel banks connected to a TE406P; whenever users three-way call, there are weird issues with asterisk recognizing double digits |
| 02:35.52 | Tond | oh ok.. that should be fine.. but examples are always great.. :) |
| 02:36.16 | Strom_M | i.e. user dials 91323 but asterisk recognizes 913323 |
| 02:37.22 | [TK]D-Fender | Strom_M, "relaxdtmf=true" <- |
| 02:37.24 | Math` | TE406P thats analog or digital |
| 02:37.34 | [TK]D-Fender | Math`, Digital. |
| 02:37.39 | Strom_M | [TK]D-Fender: Is it "true"? I have it to "yes" |
| 02:37.40 | Math` | yeah you need to tell asterisk to take it easy on the dtmf |
| 02:37.51 | [TK]D-Fender | Strom_M, Flip it and try |
| 02:38.08 | Math` | is this related to var. length dtmf in any way? |
| 02:38.16 | [TK]D-Fender | Strom_M, I have a super shit analog set for power outages whose DTMF royally sucks. it causes that sometimes |
| 02:38.32 | [TK]D-Fender | Math`, No, you are barking up the wrong tree :) |
| 02:38.44 | aptura | I wonder what the S&N of the typical TDM ratio of the cards reciver is :) |
| 02:39.02 | Strom_M | [TK]D-Fender: this happens across the board here, so i dont think its the telephone sets |
| 02:39.07 | Math` | [TK]D-Fender: damnit :P |
| 02:40.04 | [TK]D-Fender | Strom_M, have you done an audio quality test? Echo could reflect some of the DTMF, as well as high-gain cross-talk |
| 02:40.15 | Strom_M | audio quality seems fine |
| 02:40.19 | Strom_M | echo cancellation is on |
| 02:43.47 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:44.40 | Strom_M | testing now with true instead of yes |
| 02:48.32 | Strom_M | well, it seems less finnicky now |
| 02:51.33 | [TK]D-Fender | Strom_M, Check your gains as well |
| 02:52.52 | *** join/#asterisk [hC] (n=hardcore@74.221.128.35) |
| 02:53.08 | Blackthorn | does an incoming call in the iax.conf file with context=default does refer to the default context in extensions.conf? |
| 02:53.48 | Blackthorn | does an incoming call in the iax.conf file with context=default refer to the default context in extensions.conf? |
| 02:54.03 | Strom_M | [TK]D-Fender: i'm not doing any gain adjustment |
| 02:54.14 | Strom_M | Blackthorn: yes |
| 02:56.55 | *** join/#asterisk Ard0gx (n=adiaz0@190.128.163.163) |
| 02:59.38 | *** join/#asterisk hfb (n=hfb@75.80.37.175) |
| 03:01.43 | *** join/#asterisk dracosilv (n=draco@CPE-65-29-47-173.wi.res.rr.com) |
| 03:02.34 | Blackthorn | thanks strom |
| 03:03.28 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:03.36 | Blackthorn | does anyone know why I would get a message "request '@default' does not exist", when there is a working dialplan default in extensions? |
| 03:04.30 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 03:07.21 | Math` | Blackthorn: whats ur dial command you're dooing on the other side |
| 03:08.08 | Math` | if you see only @default I guess you didnt dial an extension, you just dialed IAX2/machine2 |
| 03:08.40 | Math` | so its looking for the "default" context, which you have... and it will probably try to execute whatever's in the "s' extension |
| 03:08.53 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 03:08.59 | Math` | if you want to call a particular extenion you need to Dial(IAX2/server2/extension) and then it will grab extension@default |
| 03:09.24 | aptura | or what ever context you send it to |
| 03:09.36 | Blackthorn | exten => 555,1,Dial(IAX2/remote/${exten}); |
| 03:09.45 | Blackthorn | the far side dial string |
| 03:09.56 | Blackthorn | and in iax the remote context has the user/pass |
| 03:10.06 | Blackthorn | and ip of the remote server |
| 03:11.38 | Blackthorn | do i need to replaxe the exten with the actual extenion number perhaps so the far side would get something like 555@default ? |
| 03:12.11 | Blackthorn | sorry for the mis-spells. i'm pretty tired and been working on this issue for several days now. |
| 03:14.30 | [TK]D-Fender | exten => 555,1,Dial(IAX2/remote/${exten}@default) |
| 03:14.42 | [TK]D-Fender | you need to specify the target context. |
| 03:15.08 | *** part/#asterisk pabs3 (i=daemon@60-242-186-48.tpgi.com.au) |
| 03:21.02 | Blackthorn | ok changed to what you stated, and reloaxed the iax. but it gives exact same message. calling * says call rejected no such extension |
| 03:21.14 | Blackthorn | and recive says no such extention @default |
| 03:21.30 | Blackthorn | and there is a context default in extensions.conf |
| 03:21.42 | Zipper_32 | Is the given extension in that context? |
| 03:21.56 | Blackthorn | yes |
| 03:22.02 | Zipper_32 | Perhaps try a complete wildcard extension of: exten => _. |
| 03:23.15 | Zipper_32 | Maybe something like exten => _.,1,Background(welcome) That would work for all extensions, and you could try to figure out where the problem is. |
| 03:23.28 | *** join/#asterisk Paavum (n=Dorphals@pcsp163-73.supercabletv.net.co) |
| 03:23.31 | [TK]D-Fender | Blackthorn, in the OTHER server? |
| 03:23.56 | [TK]D-Fender | Blackthorn, does default,555,1 exist THERE? |
| 03:23.56 | Zipper_32 | Another possibility could be that you have two [default] contextes in the extensions.conf, perhaps you didn't remove the original in the sample configuration. |
| 03:24.31 | [TK]D-Fender | Blackthorn, And I seriously suggest you minimise your description of things and pastebin the WHOLE mess. |
| 03:25.51 | Blackthorn | the remote server has an active voice incoming/outgoing pri with exteions setup for all of the sip phones. |
| 03:26.20 | Blackthorn | i'm moving all the sip phones to a local server, and just want to send/recive local costs through the pri at the remote server |
| 03:26.55 | Blackthorn | the extenions.conf are pretty much identical.. the remote dial was changed for one phone so i could figure out how to do this |
| 03:27.10 | Blackthorn | the sip phone for test is registered on the new server |
| 03:27.23 | Blackthorn | the remote server extion was changed to push the call through iax to the new server |
| 03:27.57 | Blackthorn | the autentication goes thorugh, but then errors out with thers no default... the sip phone is registered in default and can place calls to like voice pulse etc etc |
| 03:28.39 | Blackthorn | the instructions to do this on the wiki and in the asterisk book make this look like it should be a simple proccess.. hehe |
| 03:29.29 | Blackthorn | the remote is a 1.2 server and the new one i built is a 1.4 |
| 03:31.39 | Blackthorn | iax debug shows "No such context/extension" unforntly i've already said this so i'm not really providing any more info unforntly.. |
| 03:35.14 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
| 03:36.39 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 03:36.51 | [TK]D-Fender | Blackthorn, Perhaps your hearing is impaired. Let me attempt to be more clear. PASTEBIN YOUR CONFIGS. You are turning yourself around in circles and those trying to help will tire sharing your confusion. |
| 03:39.11 | Zipper_32 | I have a question regarding T1/PRI's. If I am to use a T1/PRI card for asterisk, can I somehow take two dial-tone lines off of the PRI through the PBX to support fax machines instead of having analog channels delivered into a building? |
| 03:40.07 | JT | Zipper_32: easy with a channel bank and channelised T1 |
| 03:40.13 | JT | but not so easy in PRI mode |
| 03:40.29 | JT | PRI mode is superior though |
| 03:40.32 | Blackthorn | ok thanks for the info fender.. but i can't pastbin private numbers... |
| 03:42.08 | JT | Blackthorn: ok... just a little paranoid |
| 03:42.14 | JT | no-one said paste your passwords |
| 03:42.21 | JT | you remove/substitute them first |
| 03:42.33 | *** join/#asterisk phobus (n=phobus@crlspr-69.65.75.232.myacc.net) |
| 03:44.34 | Zipper_32 | JT: Would something like a T1 card and 2FXS modules work for me? |
| 03:45.12 | JT | a T1 card and an analogue card with 2FXS ports could work |
| 03:46.50 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 03:47.24 | Blackthorn | http://www.pastebin.ca/482052 |
| 03:48.14 | Zipper_32 | Could you point me in the direction of a place to read the difference between a channelised T1 and a PRI? |
| 03:48.22 | Zipper_32 | Or if you feel like typing... =) |
| 03:50.47 | JT | a channelised T1 has up to 24 channels using robbed bit signalling (RBS / CAS - channel associated signalling), which is sort of like analogue over a digital bearer |
| 03:51.04 | JT | every few timeslots a bit is robbed for rudimentary signalling |
| 03:51.49 | JT | a PRI T1 has 23 B channels for voice, and 1 D channel for out of band signalling (CCS - common channel signalling), that has digital control messages utilising ITU Q.931 / ISUP |
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| 03:55.43 | Zipper_32 | JT: What's the difference when buying Asterisk hardware? |
| 03:55.51 | JT | Zipper_32: none |
| 03:56.16 | Zipper_32 | Alright then, so it all depends on my provider? |
| 03:56.28 | JT | and what you order |
| 03:56.32 | [TK]D-Fender | Zipper_32, Asterisk is software. There is no Asterisk hardware. |
| 03:56.33 | JT | but a PRI is much nicer |
| 03:56.55 | Zipper_32 | Excuse me, Digium hardware, compatible with Asterisk. |
| 03:57.17 | [TK]D-Fender | Zipper_32, and what "difference" are you talking about? |
| 03:57.40 | JT | channelised vs pri |
| 03:57.42 | Zipper_32 | In respect of the T1 PRI and channelised T1 |
| 03:57.45 | JT | hardware is the same |
| 03:58.04 | Zipper_32 | I wasn't sure if there was specific hardware for the specific service. |
| 03:59.52 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:00.33 | Zipper_32 | And with a single port T1 card like the TE120P, can I also route data through unused channels of the T1 for an internet connection to the rest of my LAN? |
| 04:00.36 | Blackthorn | anyt thoughts on that pastbin fender? if not thanks for your time tonight. i gota get to bed. |
| 04:00.57 | JT | Zipper_32: not sure, i know the sangomas definitely can |
| 04:01.28 | Zipper_32 | JT: So it can be done with some cards? |
| 04:01.32 | JT | Zipper_32: yes |
| 04:01.59 | JT | Blackthorn: out of curiosity, why do you have pointless semicolons at the end of your exten directives? |
| 04:02.33 | *** join/#asterisk joe-f (i=joe-f@c-71-201-188-239.hsd1.il.comcast.net) |
| 04:02.48 | Tond | I can't use ODBC_SQL to insert into my database can i? |
| 04:02.55 | Tond | <PROTECTED> |
| 04:03.51 | Blackthorn | jt: thers notes after the ; stating the persons phone exten |
| 04:04.14 | JT | i see |
| 04:04.29 | JT | it's best not to modify what you pastebin too much as you could miss something |
| 04:04.44 | *** join/#asterisk joe-f (i=joe-f@c-71-201-188-239.hsd1.il.comcast.net) |
| 04:05.00 | Zipper_32 | Blackthorn: Is this a new setup? I had a problem helping somebody today where they couldn't dial an extension that was CLEARLY in their default context. The reason was that they had 2 default contextes... |
| 04:05.13 | JT | Blackthorn: so do you see errors in the asterisk cli on the callED end? |
| 04:05.25 | Zipper_32 | "default" is created in the sample configs. You may want to try another obscure context name. |
| 04:06.43 | Blackthorn | zip: i have an existing * server at a remote location with a pri and all of our sip phones registered to it. I'm setting up a local server and moving all sip registrations to it. |
| 04:07.03 | Blackthorn | then any local incoming/outgoing calls on the pri will be sent to the remote server |
| 04:07.56 | Blackthorn | so the extions are identitical except for where i moved the single sip phone registration to the new server and modified the remote server to dial the new server |
| 04:08.43 | JT | Blackthorn: so about those errors i asked for |
| 04:08.57 | Blackthorn | the new server has a default context and contains the one registered sip phone and can place ld calls. but when the remote server sends an incoming call.. i get an error thers no default context |
| 04:10.03 | Blackthorn | the sending server says reject no context. the reciving server says no "@default" context.. beats me :P |
| 04:10.40 | JT | remove the @default bit |
| 04:11.17 | Blackthorn | fromt he dial string or from the reciving iax context? |
| 04:11.47 | shido6 | ZZzZ |
| 04:11.48 | JT | i only see it in the dial string |
| 04:12.23 | JT | hrm iax |
| 04:12.37 | JT | maybe you can leave it then |
| 04:12.58 | shido6 | ok 1 more b4 bed |
| 04:13.46 | Blackthorn | i can change the dial @default to @anything. and it shows up on the reciving server with the same messaage "no context @anything". so i feel the dial strings are cororect. |
| 04:13.59 | shido6 | what do you have Blackthorn? |
| 04:14.09 | shido6 | on server A and what do you have on server B ? |
| 04:15.50 | Zipper_32 | shido6: [20:47] <Blackthorn> http://www.pastebin.ca/482052 |
| 04:17.54 | shido6 | we talked about this |
| 04:17.56 | shido6 | :) |
| 04:18.08 | Blackthorn | thanks for everyone's help. I have to give up and try again tommorw. i just fell asleep for a few minutes. |
| 04:18.39 | Blackthorn | shido we talked about authentication.. which i got that working.. one side i said password = should have been secret = (silly me) |
| 04:19.34 | shido6 | here it is again.http://www.pastebin.ca/482129 |
| 04:19.39 | shido6 | wow |
| 04:19.46 | shido6 | oh well. |
| 04:20.31 | Zipper_32 | Has anybody had experience with a GSM device interoperating with their Asterisk system? |
| 04:23.06 | JT | damn blackthorn was super frustrating |
| 04:23.23 | JT | one side said password=.... yeah it'd be nice if he PASTEBINNED it |
| 04:23.31 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583012.dsl.bell.ca) |
| 04:24.06 | Paavum | Hi, is it true that if I have more than 100 sip extensions I need a SIP gateway? |
| 04:24.20 | JT | no |
| 04:25.00 | Paavum | Whats the use of one? |
| 04:25.11 | Paavum | is it the equivalent of asterisk sip handling? |
| 04:25.28 | JT | i assume you mean one of those hardware boxes? |
| 04:25.39 | JT | that connect to either FXO/FXS ports or PRIs or GSM? |
| 04:25.41 | Paavum | Actually I was kinda wondering of SER |
| 04:25.52 | Paavum | and such kinds of software |
| 04:26.14 | JT | that's not really a gateway device |
| 04:26.18 | JT | it's a proxy |
| 04:26.32 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:27.16 | *** part/#asterisk Ard0gx (n=adiaz0@190.128.163.163) |
| 04:30.11 | Paavum | and its use is... other than allow SIP over NAT (which generally works with a port forward) |
| 04:30.47 | JT | afaik it doesn't provide any more NAT punching power than asterisk does usually |
| 04:30.52 | JT | it can handle registrations |
| 04:30.54 | JT | or do lcr |
| 04:31.02 | JT | and proxy multiple asterisk boxes |
| 04:31.21 | *** join/#asterisk wunderkin (n=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 04:31.53 | Paavum | can you elaborate a bit more on the last item? |
| 04:32.26 | JT | pass off requests to 1 box from a pool of boxes |
| 04:33.42 | Paavum | So if I'm going to have 200 SIP devices (ATAs)... I dont need anything else but asterisk... right? |
| 04:35.11 | JT | depends on the specs |
| 04:35.18 | JT | and what the usage patterns are like |
| 04:36.46 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
| 04:36.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:37.09 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 04:37.28 | nowork | hello, i am still using 1.2.14,how can I unreigster a sip friend |
| 04:39.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:54.27 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
| 04:56.38 | Zipper_32 | I'm not sure if anyone's around, but I have a simple question. When I have an analog phone connected to a FXS, I'm only able to dial 1 number before * hangs the channel up on me, Where am I going wrong? |
| 04:56.51 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:58.04 | [TK]D-Fender | Zipper_32, Check the context its in |
| 05:00.47 | Zipper_32 | Well that was the perfect response. All sorted out now. Thanks [TK]D-Fender. |
| 05:01.08 | [TK]D-Fender | Zipper_32, well that sorted out fasst... ok |
| 05:02.07 | Zipper_32 | I'm setting up a new box with my first FXS card... It's something new to me, but I realized that it had to do with my single digit extensions. I didn't have a failthrough. |
| 05:09.59 | CCFL_Man2 | my cisco 1721 is too old |
| 05:18.09 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
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| 05:47.32 | nowork | hi, anyone know where is the SIP user secret stored. I changed it under /etc/asterisk , and reload asterisk even i did service asterisk restart, but when i show sip user, i still see old secret. not the one i changed at /etc/asterisk/..conf |
| 05:48.43 | netcrusher88 | /etc/asterisk/sip.conf ? |
| 05:49.07 | netcrusher88 | idk... maybe reconnect with the SIP client, i don't know how persistent sessions are with asterisk |
| 05:49.30 | netcrusher88 | is FWD's IAX2 bad, or am I just unlucky? |
| 05:53.56 | sbingner | yes? |
| 05:56.46 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:57.11 | nowork | netcrusher: will show sip users also show iax user? |
| 06:01.26 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:05.37 | *** join/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 06:06.55 | SoloFlyer | is viewcvs down? |
| 06:07.05 | JT | cvs... what? |
| 06:07.13 | SoloFlyer | http://svn.digium.com/view |
| 06:07.20 | JT | oh svn |
| 06:08.34 | SoloFlyer | does it work for you? |
| 06:09.02 | JT | yes |
| 06:09.52 | Juggie | its up for me too |
| 06:10.02 | SoloFlyer | strange... |
| 06:10.55 | SoloFlyer | works now... |
| 06:11.02 | SoloFlyer | thanks guys |
| 06:19.44 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:19.51 | *** join/#asterisk Linx (n=linx@debian.geek.nz) |
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| 06:37.09 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 06:38.52 | xpot | can anyone tell me if it is possible to use variables in voicemail.conf? I am trying to specify extensions from SQL. EX: ${EXT} => ${PASS},${FNAME} ${LNAME},${EMAIL} |
| 06:40.10 | xpot | I am using func_odbc to establish a connection... variables are pulled successfully into dialplan... I just can't figure out how to get te same variable to work in voicemail.conf |
| 06:41.16 | *** join/#asterisk [hC] (n=hardcore@110.59.19.209.transedge.com) |
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| 06:42.32 | xpot | is anyone here still awake? |
| 06:42.59 | *** join/#asterisk oej (n=olle@213.115.215.55) |
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| 06:47.09 | jacq | i am, but i have no idea :) |
| 06:49.12 | *** part/#asterisk Linx (n=linx@debian.geek.nz) |
| 06:58.56 | *** join/#asterisk AsteriskGuy99 (n=Asterisk@adsl-75-32-115-194.dsl.renocs.sbcglobal.net) |
| 06:59.17 | AsteriskGuy99 | Hello. I have a quick question for some kind soul |
| 06:59.27 | AsteriskGuy99 | I'm new to Asterisk, but just configured the basics.... I think |
| 06:59.40 | AsteriskGuy99 | It says that it's reaching my VoiP provider |
| 06:59.53 | AsteriskGuy99 | and the UN/PWs are in there correctly |
| 07:00.07 | AsteriskGuy99 | so my question is: What is the easiest way to make a test phone call with Asterisk? |
| 07:00.24 | AsteriskGuy99 | I'm logged into asterisk right now on the server through SSH |
| 07:00.29 | AsteriskGuy99 | so I can type in commands |
| 07:01.37 | kaldemar | use a soft phone. idefisk is nice. |
| 07:03.12 | AsteriskGuy99 | ok cool |
| 07:03.22 | AsteriskGuy99 | I'll look up idesisk now |
| 07:03.35 | AsteriskGuy99 | but is there a way to make a call straight from the asterisk menu easily? |
| 07:03.38 | AsteriskGuy99 | That's what I was hoping |
| 07:03.58 | AsteriskGuy99 | I know that I won't be able to speak, but at least I could know that server is set up correctly if I can hear a phone ring. |
| 07:06.21 | kaldemar | if you have a sound card in the machine and chan_oss or chan_alsa loaded, you can use the cli command dial to make a call. |
| 07:06.57 | AsteriskGuy99 | cool |
| 07:07.40 | kaldemar | but i'd use a soft phone because it's nice to know that audio is working too. :) |
| 07:09.18 | AsteriskGuy99 | Thanks Kaldemar - I appreciate your help |
| 07:09.24 | AsteriskGuy99 | I'm installing the softphone as we speak :) |
| 07:12.39 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:15.28 | *** join/#asterisk mathai (n=root@dvere.psg.sk) |
| 07:20.08 | AsteriskGuy99 | That softphone does not seem to be working right now. |
| 07:20.14 | AsteriskGuy99 | It's not letting me register an account |
| 07:20.53 | kaldemar | did you define the client in your asterisk box? |
| 07:24.26 | xpot | can anyone tell me if it is possible to use variables in voicemail.conf? I am trying to specify extensions from SQL. EX: ${EXT} => ${PASS},${FNAME} ${LNAME},${EMAIL} |
| 07:24.26 | xpot | I am using func_odbc to establish a connection... variables are pulled successfully into dialplan... I just can't figure out how to get the same variable to work in voicemail.conf |
| 07:29.27 | *** join/#asterisk senski (n=samllewe@60.234.20.178) |
| 07:30.44 | AsteriskGuy99 | kaldemar: I'm not even that far yet. It wants me to create an Asteriskguru account and |
| 07:30.47 | AsteriskGuy99 | it's not letting me |
| 07:31.09 | senski | hi, do you know why i have no audio when using queues but get the audio just fine when not using queues? |
| 07:31.39 | AsteriskGuy99 | As far as Asterisk goes, do I need to define a client before I can make my first asterisk call? |
| 07:31.58 | kaldemar | AsteriskGuy99: oh, just skip to manual configuration and configure your asterisk server in there. |
| 07:32.30 | AsteriskGuy99 | ok cool |
| 07:32.39 | AsteriskGuy99 | What is the easiest way to define a client in Asterisk? |
| 07:32.59 | kaldemar | and, yes, define the client in your asterisk box. take a look at the examples in iax.conf. |
| 07:33.10 | AsteriskGuy99 | ok |
| 07:33.19 | AsteriskGuy99 | my iax.conf should be set up correctly |
| 07:33.35 | AsteriskGuy99 | as my VoiP provider (VoicePulse) had a good config file |
| 07:33.46 | AsteriskGuy99 | but it's time for me to check it carefully I think :) |
| 07:36.06 | kaldemar | AsteriskGuy99: it may be set up correctly to dial your provider, but there is surely no client definition for your softphone if you haven't made one. |
| 07:37.05 | AsteriskGuy99 | ok, I need to look in the manual then |
| 07:37.10 | AsteriskGuy99 | for some example |
| 07:39.00 | AsteriskGuy99 | How can I make sure that the server is using IAX2 instead of SIP? |
| 07:39.53 | mvanbaak | show channels |
| 07:40.02 | mvanbaak | if it starts with SIP it's sip |
| 07:40.10 | mvanbaak | if it starts with IAX2/ it's iax2 |
| 07:41.28 | carrar | What if it starts with ZAP!! |
| 07:41.42 | kaldemar | AsteriskGuy99: see if asterisk is listening to port 4569. and it is able to use both, so there is no harm in having SIP enabled. |
| 07:41.44 | hads | It's electric! |
| 07:41.49 | carrar | hahah |
| 07:41.54 | kaldemar | carrar: then your in a deep mess! |
| 07:41.55 | JT | if it starts with misdn/, run |
| 07:44.24 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 07:45.04 | senski | can i use sip channels instead of agent in queues? would that break the audio? |
| 07:49.02 | *** join/#asterisk Dibbler_ (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 07:49.18 | senski | ok - canreinvite=no |
| 07:49.22 | senski | that old chestnut |
| 07:57.09 | AsteriskGuy99 | Any suggestions on the best documentation for setting up a client? |
| 08:00.09 | AsteriskGuy99 | [iaxuser] |
| 08:00.10 | AsteriskGuy99 | type=friend |
| 08:00.10 | AsteriskGuy99 | context=outgoing |
| 08:00.10 | AsteriskGuy99 | auth=md5 |
| 08:00.10 | AsteriskGuy99 | secret=iaxpassword |
| 08:00.11 | AsteriskGuy99 | notransfer=1 |
| 08:00.13 | AsteriskGuy99 | host=dynamic |
| 08:00.15 | AsteriskGuy99 | allow=all |
| 08:00.33 | AsteriskGuy99 | (I changed the secret before posting it here) |
| 08:00.37 | AsteriskGuy99 | but in any case |
| 08:00.48 | AsteriskGuy99 | does that qualify as a correctly set up client? |
| 08:04.01 | adorah | is there any one with indepth knowledge of audio trunking and paging? |
| 08:04.53 | kaldemar | AsteriskGuy99: yes. in the future, don't paste configurations here, use for example pastebin.ca. |
| 08:05.01 | sergee | can anybody help me with Cisco 5300 ? :) i can't find a reference for an old IOS, |
| 08:05.13 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 08:05.29 | adorah | I need for a project mass paging from multi-ysers to one point but none of them should be able to hear the other-only the one at the end-point. any suggestions? |
| 08:05.41 | adorah | =multi-users.. |
| 08:06.57 | adorah | join #asterisk-dev |
| 08:07.28 | *** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 08:07.33 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:08.52 | AsteriskGuy99 | ok |
| 08:08.55 | AsteriskGuy99 | Thanks for your help |
| 08:09.00 | AsteriskGuy99 | I'll keep trying tomorrow |
| 08:10.07 | *** part/#asterisk senski (n=samllewe@60.234.20.178) |
| 08:11.26 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 08:11.44 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:13.50 | SoloFlyer | anyone heared of a bug relating to the hangup on polarity detection code? |
| 08:14.46 | SoloFlyer | where the provider sends evenly spaced pairs of polarity reversals during the rining phase |
| 08:18.54 | *** join/#asterisk mathai (n=root@dvere.psg.sk) |
| 08:30.29 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 08:39.28 | demlak | how to echo a ";" in the dialplan? im using h,n,system(/bin/echo test ; test) |
| 08:39.40 | demlak | you understand the problem? =) |
| 08:44.10 | *** join/#asterisk friedrich| (n=friedric@e177249042.adsl.alicedsl.de) |
| 08:47.42 | kaldemar | demlak: escape the ; |
| 08:48.11 | demlak | what does this mean? im not realy into this |
| 08:48.39 | kaldemar | echo test \; test |
| 08:48.46 | demlak | ok |
| 08:48.57 | kaldemar | but what are you trying to do? |
| 08:49.38 | demlak | my asterisk is on a minimal embeded linux..w ithout any mail sending programm.. so im writing a little dialplan to send mail with netcat =) |
| 08:52.24 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 08:53.10 | demlak | did not work |
| 08:53.51 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 08:53.55 | skirmisha | guys |
| 08:54.09 | skirmisha | any idea how can i get libosptk for debian |
| 08:56.50 | *** join/#asterisk lorinc (n=ang@pool-4344.adsl.interware.hu) |
| 08:57.20 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:58.48 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 08:59.21 | kippi | can you run the g711 codec on asterisk |
| 09:00.18 | skirmisha | me? |
| 09:03.21 | *** join/#asterisk sumasuma (n=kurukko@61.14.86.23) |
| 09:03.42 | SoloFlyer | delmak: that sounds cool |
| 09:06.09 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 09:11.34 | demlak | well.. asterisk sends the ; correct to echo.. but echo doesn´t echo it |
| 09:11.39 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 09:12.25 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-7-131.w81-251.abo.wanadoo.fr) |
| 09:13.00 | kaldemar | demlak: does system interpret it and assume that the second test is a next command? |
| 09:13.22 | demlak | echo needs also "escaping" |
| 09:15.45 | demlak | \\\; works |
| 09:16.33 | JT_ | kippi: yes, of course |
| 09:23.40 | kippi | how can I find out why some of my calls have jitter and are not very crisp, its going calls that are coming though the IDSN30 |
| 09:25.05 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 09:27.18 | SoloFlyer | my isp sends polarity winks during the ring phase of a call, astersisk assuming that this means that the call has hung up and then that a new call has started... im writing a patch for it, but should this be patched in zaptel or in asterisk? |
| 09:31.00 | Ifaistos | demlak : Use esmtp is very small I am using it also on our embedded devices for emails |
| 09:31.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:32.37 | *** join/#asterisk Giraya (n=giraya@milo.giraya.net) |
| 09:35.00 | Giraya | hi, i have 2 tdm400p, each one on a different server. |
| 09:35.04 | Giraya | the first one has 2 fxo (configured to receive faxes with spandsp and app_rxfax) and work well |
| 09:35.32 | Giraya | the second one has 2 fxo (configured to receive faxes with spandsp and app_rxfax) and 2 fxs (not used) |
| 09:35.58 | Giraya | every server running asterisk 1.2.17 on freebsd |
| 09:36.19 | Giraya | with zaptel 1.4.1 |
| 09:36.44 | SoloFlyer | sounds good so far... whats the problem |
| 09:36.48 | Giraya | the two configurations are the same |
| 09:36.55 | Giraya | the first server can receive fax |
| 09:37.00 | Giraya | the second can't |
| 09:37.24 | Giraya | i've got this message in debug : Fax receive not successful - result (3) Timed out waiting for the first message. |
| 09:37.29 | SoloFlyer | you tried swapping the cards |
| 09:37.33 | Giraya | yep |
| 09:37.50 | SoloFlyer | does it stay witht he card or with the machine |
| 09:38.03 | Giraya | it's always the card with 2 fxo only that can worked |
| 09:38.15 | Giraya | it stays with the card |
| 09:38.27 | SoloFlyer | try taking the 2 fxs off and see what happens |
| 09:38.47 | Giraya | ok |
| 09:39.10 | SoloFlyer | then try swapping the fxo modules between the working card and the not working card |
| 09:39.33 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 09:39.35 | Giraya | ok |
| 09:39.36 | kink0 | hi |
| 09:39.41 | SoloFlyer | sounds like a peice of broken hardware... |
| 09:39.46 | *** join/#asterisk shinao1 (n=shinao1@dial-pool1.lagos.starcomms.net) |
| 09:39.54 | Giraya | yeah i was thinkin about this :( |
| 09:40.14 | kink0 | what fields from SIP leg are passed or affects the Zap leg ? |
| 09:42.00 | SoloFlyer | why? |
| 09:42.12 | Giraya | i didn't mentioned that when i try to call a sip extension on the server through the tdm400p, it works |
| 09:44.09 | SoloFlyer | Giraya: that makes it more interesting... |
| 09:45.11 | Giraya | i know, but i expect an hardware conflict with the 2 fxs modules |
| 09:45.47 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:46.22 | Giraya | so the fxo can't worked correcty with fax handling and spandsp ? |
| 09:47.00 | Giraya | or |
| 09:47.04 | SoloFlyer | huh? |
| 09:47.31 | Giraya | maybe it's a pb with rxgain |
| 09:47.52 | SoloFlyer | check lines with ztmonitor... |
| 09:48.04 | SoloFlyer | when its receiving a fax... |
| 09:48.08 | Giraya | did it already |
| 09:48.42 | Giraya | it's a bit too high but it's the same on the other card |
| 09:48.53 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 09:48.54 | Giraya | is the rxgain can differs between two card ? |
| 09:49.27 | SoloFlyer | i wouldnt have though it would... but it could... |
| 09:49.28 | Putzz | anyone using spandsp with asterisk 1.4.4? when I call the extension waiting for fax it crashes asterisk |
| 09:50.22 | Giraya | ok i'll try that too |
| 09:50.22 | SoloFlyer | Giraya: anyway id try the hardware route, ive got to get back to this |
| 09:50.46 | SoloFlyer | code doesnt write itself.... :( |
| 09:51.25 | Giraya | ok |
| 09:53.49 | Giraya | thk you btw :) |
| 10:04.19 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 10:04.21 | zeeesh | hi |
| 10:05.25 | zeeesh | i have been installed new version of asterisk-1.4.14 .. trying to unzip with this command xvzf but its not workind |
| 10:05.42 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 10:07.48 | Black-Kakugane-1 | what's the full name of the file you downloaded? |
| 10:16.35 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:16.37 | Chris-NB | hi |
| 10:16.50 | Chris-NB | anyone doing conferencing with asterisk 1.2 ? |
| 10:26.27 | SoloFlyer | yep |
| 10:26.27 | SoloFlyer | zeeesh: use tar -zxvf |
| 10:28.26 | zeeesh | ok |
| 10:36.27 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 10:42.49 | puzzled | hi |
| 10:45.34 | SoloFlyer | hi |
| 10:45.47 | SoloFlyer | Chris-NB im doing conference on 1.2 |
| 10:45.55 | Chris-NB | SoloFlyer, how do you do conferencing? |
| 10:46.22 | Chris-NB | if I'm in a call with another person I want to call a 3. one and make a conference with these two |
| 10:46.25 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-efee1f95722f89fd) |
| 10:46.28 | Chris-NB | are u using meetme? |
| 10:47.02 | SoloFlyer | im using meetme |
| 10:47.20 | Chris-NB | how do you do that? |
| 10:47.51 | SoloFlyer | i usually call the third person from another phone then transfer them to the conference room |
| 10:48.33 | Chris-NB | SoloFlyer, ok, I could do that as well, but from one phone? is this possible from your solution? |
| 10:49.04 | SoloFlyer | i usually do the equivilent using the phone i am on, by putting my call to the conference room on hold |
| 10:49.21 | SoloFlyer | ie put my call to conference room on hold |
| 10:49.25 | SoloFlyer | call other person |
| 10:49.36 | SoloFlyer | call conference room |
| 10:49.41 | SoloFlyer | transfer other person to conference room |
| 10:49.49 | SoloFlyer | resume the call i put on hold |
| 10:50.51 | SoloFlyer | im pretty sure you can do it using the meetme admin interface though... |
| 10:52.12 | Chris-NB | hmmm, I'll try |
| 11:02.14 | puzzled | for all you ITSP's out there: http://www.voipfraud.net/ |
| 11:06.00 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 11:06.23 | *** join/#asterisk fujin (i=aj@unaffiliated/fujin) |
| 11:06.55 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 11:09.38 | DarKnesS_WolF | anyone had asterisk up and running on RAID-1 software ?? is there any problems with performance ? |
| 11:11.10 | *** join/#asterisk Fieldy (i=zCEgXA3w@gentoo/contributor/Fieldy) |
| 11:12.40 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:32.27 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 11:32.55 | kippi | hey |
| 11:33.14 | kippi | has anyone loaded SIP on to the cisco handset? |
| 11:37.05 | SoloFlyer | DarKnesS_WolF: Im running asterisk on software raid-1 ive had no problems... |
| 11:37.55 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 11:39.40 | DarKnesS_WolF | SoloFlyer: great :-D |
| 11:42.06 | puzzled | DarKnesS_WolF: afaik the performance of software raid-1 on linux is quite good. iirc there are some reports that google has |
| 11:43.48 | *** join/#asterisk coppice (n=chatzill@94.143.17.210.dyn.pacific.net.hk) |
| 11:47.36 | SoloFlyer | anyone know anything about zaptel |
| 11:49.52 | e-ddie | oh |
| 11:50.02 | e-ddie | wrong window :) |
| 11:50.04 | SoloFlyer | i need to patch in AC detection during the ringing phase in wctdm |
| 11:50.25 | SoloFlyer | wrong window? |
| 11:50.40 | e-ddie | yeah |
| 11:50.46 | e-ddie | would you please go close it? |
| 11:50.52 | e-ddie | it's getting cold in here |
| 11:50.55 | SoloFlyer | lol |
| 12:00.28 | StyleWarz | Anyone can give me a hint what this could be? [May 11 13:55:31] WARNING[27418]: rtp.c:885 ast_rtcp_read: RTCP Read too short |
| 12:03.02 | *** join/#asterisk apardo (n=apardo@87.217.144.161) |
| 12:03.43 | JT_ | the advantage of raid 1 is primarly meant to be reliability, not speed :) |
| 12:03.56 | JT_ | which is why i still prefer a hardware solution |
| 12:04.39 | LeddyHM | raid 1 has faster reads though :) |
| 12:04.45 | LeddyHM | (than single drive) |
| 12:05.07 | LeddyHM | assuming you're using a hardware solution |
| 12:05.19 | LeddyHM | does software raid exist *cough* |
| 12:05.34 | SoloFlyer | yes |
| 12:05.50 | LeddyHM | I was being fecitious |
| 12:06.00 | SoloFlyer | atleast if one of my hard drives dies i dont lose anything |
| 12:06.18 | LeddyHM | software raid is an oxy moron |
| 12:06.20 | JT_ | i'd prefer to do raid1 in hardware |
| 12:06.21 | LeddyHM | IMHO |
| 12:06.31 | JT_ | it has its advantages, this is not one of them |
| 12:06.38 | JT_ | (software raid) |
| 12:06.51 | SoloFlyer | but redundant array of inexpensive disks... its still an array of inepensive disks so how is it an oxymoron |
| 12:07.06 | LeddyHM | you forgot the word software |
| 12:07.06 | JT_ | s/inexpensive/independant/ |
| 12:07.13 | JT_ | these days anyway |
| 12:07.44 | JT_ | with hardware raid 1, no stuffing around to make sure OS will boot properly |
| 12:07.55 | JT_ | and you can easily split the raid by pulling out a drive |
| 12:08.43 | SoloFlyer | at least with software raid i dont have to worry about loseing a controller card |
| 12:09.17 | JT_ | that's a red herring |
| 12:09.18 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 12:09.19 | JT_ | seriously |
| 12:09.21 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 12:09.41 | JT_ | i always hear that excuse, it's a very occurance in reality |
| 12:09.49 | LeddyHM | + rare |
| 12:09.52 | JT_ | also, hardware raid is battery and RAM backed |
| 12:09.54 | LeddyHM | ;) |
| 12:10.14 | JT_ | to ensure the drives are in a contiguous state when there's an unexpected power failure |
| 12:10.15 | SoloFlyer | heh give me hardware raid any day |
| 12:10.22 | hads | If you get a high end card |
| 12:10.37 | JT_ | well i din't count fake raid as hardware raid |
| 12:10.54 | JT_ | pretty much all non-fake raid cards have battery and ram cache backup |
| 12:11.01 | hads | Na |
| 12:11.04 | *** join/#asterisk eltech (n=eltech@ool-457c9ece.dyn.optonline.net) |
| 12:11.08 | zeeesh | normally we can check CLI by using these commands .. (cd /etc/asterisk -r ) or (sudo /usr/sbin/asterisk -r) ... is there any other way to check CLI .... ????????????? |
| 12:11.34 | JT_ | hads: maybe absolute junk doesn't have it |
| 12:11.49 | hads | Sure |
| 12:12.51 | SoloFlyer | in the end if you are running raid software or hardware you have an reliability advantage |
| 12:13.14 | SoloFlyer | but with hardware raid you gain even more of an advantage |
| 12:13.29 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 12:13.41 | hads | No argument there |
| 12:13.51 | SoloFlyer | espically when it comes to performance |
| 12:14.24 | JT_ | SoloFlyer: actually that's debatable for RAID5 |
| 12:14.51 | JT_ | a lot of hw raid cards don't have as much grunt as linux software raid to compute RAID5 parity |
| 12:14.53 | StyleWarz | JT: Software Raid is faster and more reliable in most cases than cheap (< 250 euro) hardware raid :) |
| 12:15.18 | JT_ | StyleWarz: fake raid isn't hardware raid ;) |
| 12:15.19 | *** join/#asterisk Obergandhi83 (n=Obergand@P43d0.p.pppool.de) |
| 12:15.27 | StyleWarz | JT_: ^5 :) |
| 12:15.35 | Obergandhi83 | hi @ all |
| 12:15.43 | *** part/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 12:18.03 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-7-131.w81-251.abo.wanadoo.fr) |
| 12:19.06 | Obergandhi83 | i've got Asterisk 1.4.2, mISDN 1.1.2, mISDNuser 1.1.2 and an echo problem :-) .... echocancel is on .. but sometimes i can hear myself or the other person can hear himself ... |
| 12:28.06 | *** join/#asterisk _shad_ (n=shad@mail.topan.ca) |
| 12:29.04 | _shad_ | Hope you guys can help. I upgraded from 1.2 to 1.4 and now when I receive an incoming sip call, it works the first time but does nothing after that. Doing a sip reload fixes the problem for only one more incoming call. Any ideas? |
| 12:30.21 | LeddyHM | go back to 1.2 |
| 12:31.46 | _shad_ | It's a home system so it is not critical to get it up right away, I would rather try to diagnose the problem. |
| 12:33.27 | zeeesh | i can't c any call which is coming at my asterisk server ... but when i give " show channels " ... i can c it the call are terminating .. but at run time i can't c ... y is so that ... ? |
| 12:33.33 | kippi | can someone help me with my cisco handset? its now just sitting there saying upgrading and seems stuck in that loop, anyideas? |
| 12:36.25 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:42.13 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 12:43.37 | coolbeans | Hi all. I'm building zaptel 1.4.2.1 and it's screaming about not being able to find this file: /include/linux/autoconf.h The file exists, but under /usr/include/linux. I ran the configure script and it worked ok but these errors stil prevent it from building. Any ideas? |
| 12:47.26 | SoloFlyer | symlink |
| 12:48.01 | SoloFlyer | are you building from withing /usr/src/something |
| 12:48.07 | coolbeans | I just removed the zaptel src and exploded the tarball again, it works. Apparently, make clean doesn't work completely. |
| 12:52.32 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:53.15 | *** join/#asterisk Fieldy (i=q5yGPj1a@gentoo/contributor/Fieldy) |
| 12:53.39 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 12:53.42 | LeddyHM | TK! |
| 12:56.16 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 13:00.23 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 13:00.49 | coolbeans | Anyone compilex zaptel 1.4.1 on ubuntu 6.06 LTS successfully? If so, how did you link to your kernel headers? When I try to compile it tells me: You do not appear to have the sources for the 2.6.15-26-server kernel installed. But I in fact do have them installed, but when you install the kernel-headers package it isn't listed as '-server'. Any help would be appreciated. |
| 13:01.31 | coolbeans | Maybe it's overkill since I don't use any TDM cards. In 1.4, is ztdummy still required as a timing source for MeetMe? |
| 13:02.01 | [TK]D-Fender | coolbeans: Yes |
| 13:02.13 | [TK]D-Fender | LeddyHM: Y0 |
| 13:02.26 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 13:07.38 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
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| 13:14.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 13:17.16 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 13:18.42 | Putzz | anyone wanna help me out? I want to send a fax using txfax what should my dial plan look like for that? |
| 13:19.00 | kuku5 | did txfax compile ? |
| 13:19.21 | Putzz | yep |
| 13:19.25 | Putzz | I got it all running |
| 13:19.29 | Putzz | just not sure on dialplan to send fax |
| 13:20.00 | *** join/#asterisk jaike (n=jaike@125.5.144.90) |
| 13:20.55 | [TK]D-Fender | Putzz: How are you dialing out? |
| 13:21.19 | Putzz | well I was thinking something stupid like I call a ext and the extension dials out and sends fax |
| 13:21.24 | Putzz | or something like that |
| 13:22.02 | [TK]D-Fender | Putzz: You need to remove yourself from the equation. You can't be talking on that channel and sending the fax now can you? |
| 13:22.17 | [TK]D-Fender | Putzz: So you needt to use a call-file or AMI originate |
| 13:22.26 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 13:22.33 | Putzz | should have thought of that |
| 13:22.39 | Putzz | lol |
| 13:22.50 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 13:22.56 | Putzz | thats what no sleep does to u |
| 13:23.08 | Putzz | but how would I go on dialing and sending fax with txfax |
| 13:23.10 | Putzz | ? |
| 13:23.13 | Bladerunner05 | Hi all, I use AVM B1 card and I get sometimes problem with echo, who use that card can tell me how to set echo cancellation correctly ? |
| 13:23.20 | *** join/#asterisk Fieldy (i=I97dxZUV@gentoo/contributor/Fieldy) |
| 13:27.22 | [TK]D-Fender | Putzz: I just told you the 2 tools you can use to do this. If you can't figure that out, go to sleep.... you're gonna need it. |
| 13:31.06 | coolbeans | Hey TK - Is 1.4.4 w/static realtime ready for production yet? |
| 13:31.46 | *** join/#asterisk AndrewGearhart (n=chatzill@h1.39.213.151.ip.alltel.net) |
| 13:31.48 | [TK]D-Fender | coolbeans: I don't sue realtime, and most would say to continue waiting a bit |
| 13:32.03 | coolbeans | Ok, thanks ;) |
| 13:32.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:32.18 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 13:41.53 | coolbeans | When I compile mysql support from the addons package in 1.4, do I have to explicitely load the module in modules.conf? |
| 13:42.17 | *** join/#asterisk PioneerVM (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 13:42.21 | PioneerVM | I create a sip section for a phone to register too called "sipuser" -- i tried to register 2 people to the same section and it seems that the second one takes priority, is that true? |
| 13:42.37 | PioneerVM | meaning you can only register one phone to a context? |
| 13:43.23 | *** join/#asterisk afrosheen (n=cj@207.71.49.164) |
| 13:43.42 | afrosheen | hi, can someone point me to a step-by-step install for an iaxy device |
| 13:44.47 | ManxPower | PioneerVM: sip.conf does not have contexts. extensions.conf has contexts. |
| 13:44.57 | ManxPower | you cannot register more than 1 device to the same sip.conf account. |
| 13:45.40 | PioneerVM | ok |
| 13:45.58 | ManxPower | PioneerVM: What we do is set the SIP user id to be the same as the MAC of the device. This works fine for hardphones, not so well for softphones. But we don't use softphones. |
| 13:46.15 | PioneerVM | next question, i have a linksys box hooked to an asterisk sip account thru my router -- it works fine but "sometimes" it wont get an incoming cal |
| 13:46.20 | ManxPower | PioneerVM: You must remember that a SIP device is NOT an extensions. |
| 13:46.44 | *** join/#asterisk Sweeper (i=sweeper@scriggleit.com) |
| 13:46.45 | ManxPower | sip devices are devices. an extension just maps a number to an account |
| 13:46.46 | PioneerVM | yea i made two sip user IDs and it worked i just had to do & in the DIal to ring both |
| 13:46.58 | ManxPower | PioneerVM: set up a port forward for 5060 on your router |
| 13:47.16 | PioneerVM | well i dont need it for my x-lite |
| 13:47.28 | PioneerVM | and i am trying to use x-lite on a PC and linksys box on my same network |
| 13:47.35 | PioneerVM | i got them both working taking incoming calls fine |
| 13:47.38 | PioneerVM | actually even at same time |
| 13:47.46 | PioneerVM | but then the linksys box stops taking incoming calls sometimes |
| 13:47.47 | ManxPower | PioneerVM: when a device sends a packet the router will remember that for X amount of time and you will get incoming calls for that amount of time. X varies depending on the router. |
| 13:48.01 | ManxPower | PioneerVM: better than a port forward is to set the NAT keepalive on the linksys |
| 13:48.10 | PioneerVM | yea that was what i was going to ask |
| 13:48.17 | PioneerVM | i hate port forward someone has to work on that :) |
| 13:48.33 | ManxPower | don't do any OTHER nat setings on the device, just the nat keepalive setting |
| 13:48.54 | ManxPower | PioneerVM: you can use a port forward, a qualify= or a nat keepalive |
| 13:49.07 | ManxPower | oh, or have the device register every 60 seconds. |
| 13:49.10 | PioneerVM | i had to turn on Nat Mapping Enable YES |
| 13:49.16 | ManxPower | Any one of these will keep the port open. |
| 13:49.18 | PioneerVM | before to get it to work |
| 13:49.23 | PioneerVM | ur saying not to use it? |
| 13:49.43 | PioneerVM | whats the qualify= do |
| 13:49.46 | ManxPower | If bith ends are trying to do nat migic it can cause isues. |
| 13:49.51 | PioneerVM | not familiar with that one |
| 13:50.01 | ManxPower | PioneerVM: I do not recommend qualify= as it does not work well |
| 13:50.10 | PioneerVM | im connecting from behind a linksys router at home to behind cisco pix at office |
| 13:50.23 | PioneerVM | had to turn on NAT=YES for the sipuser on that end |
| 13:50.31 | ManxPower | quality= tries to measure response time to a SIP OPTIONS packet. |
| 13:50.42 | PioneerVM | how do i turn that on |
| 13:50.46 | ManxPower | The problem is that if even 1 response is missed, asterisk will consider the device offline. |
| 13:51.30 | PioneerVM | can i use multiple of these, like register every 10 mins, nat keep alive=on, quality= |
| 13:52.06 | DarKnesS_WolF | can mixmonitor record in mp3 ? |
| 13:52.07 | ManxPower | PioneerVM: in theory, but it is totally useles. |
| 13:52.31 | afrosheen | DarKnesS_WolF, I doubt it but you can have a shell script run SOX on it and convert it later |
| 13:52.32 | PioneerVM | ok phone is ringing again ill see what hapens (with nat keep alive) thx |
| 13:52.39 | PioneerVM | how does vonage handle all this crap |
| 13:52.52 | PioneerVM | my vonage box always works, do they use proxies, or stun |
| 13:53.17 | ManxPower | PioneerVM: they do the equiv of NAT keepalive. |
| 13:53.21 | ManxPower | in their box |
| 13:53.32 | PioneerVM | i meant how do they make the connection behind routers |
| 13:53.35 | PioneerVM | do they use the STUN stuff |
| 13:53.40 | PioneerVM | or some proprietary thing |
| 13:54.01 | ManxPower | Um, they do the equiv of nat=yes on their server and nat keepalive on their customer box |
| 13:54.09 | PioneerVM | hmm |
| 13:54.17 | PioneerVM | i have to use stun to get this and x-lite to work |
| 13:54.17 | ManxPower | whiich is exactly what I'm recommending to you. |
| 13:54.24 | ManxPower | PioneerVM: then you did it wrong. |
| 13:54.49 | ManxPower | STUN is almost never needed. |
| 13:54.59 | PioneerVM | hmm strange thats not what counterpath keeps saying |
| 13:55.02 | PioneerVM | they keep praising stun |
| 13:55.18 | ManxPower | PioneerVM: they prolly don't know about Asterisk's nat=yes option. |
| 13:55.24 | PioneerVM | if u can help me fix this then great |
| 13:55.28 | PioneerVM | i have NAT=YES on my sip user set |
| 13:55.31 | ManxPower | All STUN is for is to help with NAT issues before any devices supported nat |
| 13:55.58 | coppice | ManxPower: I don't think you know what STUN does |
| 13:56.35 | ManxPower | coppice: Simple Traversal for UDP NAT ? |
| 13:56.42 | ManxPower | It assists in determining the public IP of a device behind NAT. |
| 13:56.47 | coppice | its a very important protocol today |
| 13:56.54 | ManxPower | It also assists in getting around NAT issues. |
| 13:57.29 | ManxPower | coppice: what can STUN do for a standard situation of Asterisk(publicIP) <-> SIP client (NAT'd) that nat=yes and NAT Keepalive cannot do |
| 13:58.01 | coppice | it can make box<->nat<->nat<->box work |
| 13:58.22 | ManxPower | coppice: So can turning off reinvites in Asterisk. |
| 13:58.40 | DarKnesS_WolF | afrosheen: yes i have this script .. i just thought it might be supported |
| 13:58.41 | ManxPower | But is not what most home users care about. |
| 13:58.46 | coppice | but that is an awful solution. the audio goes the wrong way |
| 13:59.16 | coppice | two boxes behind NAT can find each other. I use it all the time |
| 13:59.34 | Katty | morning lovables. |
| 13:59.36 | afrosheen | DarKnesS_WolF, not yet :) |
| 13:59.38 | ManxPower | coppice: People have two boxes find each other all the time without STUN |
| 13:59.53 | coppice | how? |
| 14:00.04 | ManxPower | coppice: DNS |
| 14:00.07 | Bladerunner05 | Hi all, I use AVM B1 card and I get sometimes problem with echo, who use that card can tell me how to set echo cancellation correctly ? |
| 14:00.31 | coppice | ManxPower: are you trying for the idiot of the month award? |
| 14:00.34 | ManxPower | But you are correct. If the client and the server are on dynamic IPs and NAT I can see how STUN would help in a way nothing else can. |
| 14:01.00 | ManxPower | coppice: "DNS" was a somewhat sarcastic answer. 8-) |
| 14:01.36 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 14:01.47 | [TK]D-Fender | ManxPower: You forgot to tell him to add the all-important "canreinvite=no" :) |
| 14:01.48 | ManxPower | What *I* am saying is that if STUN is so important why is it not even supported in Asterisk (does 1.4 support STUN?) |
| 14:01.54 | [TK]D-Fender | PioneerVM: See above |
| 14:02.17 | ManxPower | 1.2 and before didn't. |
| 14:02.19 | coppice | ManxPower: a *lot* of important stuff is not in 1.4 :-) |
| 14:02.22 | [TK]D-Fender | PioneerVM: and I personally DO recommend "qualify=yes" most of the time |
| 14:02.28 | ManxPower | coppice: you go that right! |
| 14:02.41 | ManxPower | coppice: my pet peeve is that qualify smoothing for SIP is not in 1.4 |
| 14:02.58 | ManxPower | [TK]D-Fender: and I recommend against qualify=yes most of the time 8-) |
| 14:03.00 | [TK]D-Fender | Katty: Mew. |
| 14:03.06 | [TK]D-Fender | Katty: *hugz* |
| 14:03.16 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 14:03.19 | Katty | [TK]D-Fender: How be? |
| 14:03.35 | PioneerVM2 | strange i had to reboot router and when i came back it kept telling me my pw was wrong on nickserv |
| 14:03.36 | [TK]D-Fender | ManxPower: We both get the job done... so we're BOTH right :) |
| 14:03.58 | [TK]D-Fender | PioneerVM : more likely considered "inuse" |
| 14:04.01 | PioneerVM2 | i tried turning off STUN and i still registered but i could not hear any audio coming in -- apparently i need to use STUN |
| 14:04.03 | ManxPower | [TK]D-Fender: Yeah, but your way will make devices randomly unreachable. *tease* |
| 14:04.04 | [TK]D-Fender | Katty: TGIF! |
| 14:04.07 | PioneerVM2 | no it actually said invalid pw |
| 14:04.13 | Katty | [TK]D-Fender: oh, yeah. it is friday. hrmm. |
| 14:04.20 | afrosheen | hi, can someone point me to a step-by-step install for an iaxy device |
| 14:04.20 | ManxPower | PioneerVM: No you don't. You have something ELSE set wrong. |
| 14:04.29 | PioneerVM2 | ok, if you could enlighten me then |
| 14:04.32 | [TK]D-Fender | ManxPower: No... my way would make YOUR devices randomly unreachable. I live in HappyLand! |
| 14:04.33 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:04.36 | ManxPower | PioneerVM: Is the Asterisk server behind a firewall or NAT? |
| 14:04.37 | Katty | [TK]D-Fender: i went bowling last night..with some friend. i bowled a 47 and a 57, and now my forearms hurt >.< |
| 14:04.44 | PioneerVM2 | its behind a cisco pix fw |
| 14:04.55 | [TK]D-Fender | PioneerVM2 : PIX?!?! |
| 14:04.56 | ManxPower | PioneerVM: Yes, that coiuld cause issues. |
| 14:05.06 | [TK]D-Fender | PioneerVM : Dear God... the WORST thing you could have said |
| 14:05.11 | PioneerVM2 | lol |
| 14:05.14 | ManxPower | PioneerVM: get SIP fixup turned off on the PIX to start with. |
| 14:05.20 | Katty | stop being a bunch of bitter geeks. |
| 14:05.29 | ManxPower | PioneerVM: But no NAT?: Firewall only? |
| 14:05.35 | [TK]D-Fender | PioneerVM2 : Go to the WIKI and read up on what you're going to have to do.... Cisco PIX NAT is friggen muder on SIP/RTP |
| 14:05.42 | PioneerVM2 | yea i really dont want to get into a linux/mac/pc, perl/php type debate |
| 14:05.54 | PioneerVM2 | all machines have real world addresses |
| 14:06.12 | PioneerVM2 | so the internal address is same as external but they all route thru one IP |
| 14:06.26 | [TK]D-Fender | PioneerVM2 : I'm just speaking from experience.... it is SPECIFICALLY problematic. Almost anything else would have left you better off |
| 14:06.28 | PioneerVM2 | oh i have been having problems connecting from asterisk to voicepulse with SIP, i have to use IAX2 |
| 14:06.37 | PioneerVM2 | yea well i dont have much choice right now |
| 14:07.03 | PioneerVM2 | tk -- thought u were just getting into those political debates over which type of fw is better in general :) |
| 14:07.05 | ManxPower | PioneerVM: Perhaps you are one of the small number of non-ITSPs where STUN is a good idea. |
| 14:07.06 | PioneerVM2 | didnt realize u meant for asterisk |
| 14:07.18 | ManxPower | It is certinally simplier than setting it up the Asterisk Way |
| 14:07.23 | PioneerVM2 | so the SIP setting is a bad idea on pix? |
| 14:07.37 | PioneerVM2 | i would prefer not to need a stun server or other |
| 14:07.48 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:07.48 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.58 | Katty | tis mog. |
| 14:08.07 | PioneerVM2 | any idea why i cant have my asterisk connect to voicepulse using sip? they said they see my requests but apparently there returns are not getting to me |
| 14:08.41 | ManxPower | PioneerVM: Asterisk is set to fixup packets, The PIX is trying to fixup the already fixed up packets, STUN is trying to fixup the doublefixed up nat packets, and your phone is trying to fixup the tripple fixed up NAT packets. |
| 14:08.44 | *** join/#asterisk Feral_Kid (n=FeralKid@red-corp-201.170.80.61.telnor.net) |
| 14:08.57 | ManxPower | I'm surpized you did not tear a hole on space/time. |
| 14:08.58 | PioneerVM2 | lol interesting let me try that |
| 14:09.04 | *** join/#asterisk kombi (n=kombi@195.158.185.196) |
| 14:09.05 | Feral_Kid | Does anyone use Wengo? |
| 14:09.18 | Katty | Feral_Kid: how about Wendy's instead? |
| 14:09.24 | kombi | yum! |
| 14:09.53 | Katty | [TK]D-Fender: i guess register sip clients is next, eh? |
| 14:10.13 | Katty | or maybe backups. |
| 14:10.20 | *** join/#asterisk federicoco (n=federico@212.34.251.205) |
| 14:10.25 | PioneerVM2 | yea i still needed stun |
| 14:10.28 | PioneerVM2 | even with fixup off |
| 14:10.33 | kombi | can someone point me to some information about powering phones over ethernet? |
| 14:11.05 | kombi | ..got a cisco one here, no lights on it..;( |
| 14:11.10 | ManxPower | kombi: the standard is called 802.3af, IIRC. |
| 14:11.11 | afrosheen | kombi: it's not too complicated, either your phones support PoE or not |
| 14:11.12 | Katty | i think the switch has to support it, kombi... |
| 14:11.20 | Katty | and then the phone too |
| 14:11.23 | ManxPower | Cisco uses their own power over either net protocol in their older phones |
| 14:11.27 | PioneerVM2 | hmm it might have fixed the problem with voicepulse though |
| 14:11.29 | Katty | then, if both is the case, you just plug it in...and chaching, powery |
| 14:11.42 | PioneerVM2 | im not getting the error anymore about not connecting |
| 14:11.58 | kombi | thanks ManxPower + afrosheen! Does it involve soldering or can the NIC do it? |
| 14:12.20 | PioneerVM2 | manx: cool that fixed my voicepulse issue |
| 14:12.25 | Katty | gosh, messy. |
| 14:12.33 | PioneerVM2 | the Cisco SIP fixup turned off |
| 14:12.36 | [TK]D-Fender | Katty: Just copy over your old configs and tweak to current spec... |
| 14:12.43 | PioneerVM2 | the Cisco SIP fixup turned off |
| 14:12.44 | Katty | but that's no fun! |
| 14:12.48 | Katty | i want something shiny and new! |
| 14:12.58 | afrosheen | kombi, there is no soldering involved, and I don't know what NIC you're talking about |
| 14:13.15 | ManxPower | kombi: what PoE standard does your phone support? |
| 14:13.21 | afrosheen | kombi, basically you start with a PoE-supported device, like a Polycom 601 |
| 14:13.29 | afrosheen | kombi, then you match it with a PoE switch |
| 14:13.33 | kombi | afrosheen: the network card I meant.. |
| 14:13.42 | afrosheen | kombi, what network card? |
| 14:13.47 | ManxPower | You either need an ethernet switch that supports PoE or you must have a device to inject PoE into the line. |
| 14:13.50 | kombi | ManxPower: I'll check.. |
| 14:13.53 | Katty | [TK]D-Fender: is iaxcoom and SJphone still the most awesome software clients? |
| 14:13.58 | PioneerVM2 | Ok new question - my employee is trying to connect using X-Lite behind his router in the same was I do at home -- we both have same settings, and same cable modem company, but he cant connect -- I think it may be that his cable modem has built in VOiP Phone -- could it be blocking the sip packets? |
| 14:13.59 | Katty | [TK]D-Fender: also, iaxxcom |
| 14:14.03 | ManxPower | kombi: network cards do not do PoE |
| 14:14.06 | shido6 | then match the switch to a ups that can handle your phones at full load for x time |
| 14:14.07 | Katty | [TK]D-Fender: or however i'm not spelling it. |
| 14:14.08 | afrosheen | man I'm glad kombi asked in here first |
| 14:14.30 | xpot | anyone know if voicemail.conf supports var's such as this example: ${EXT} => ${PASS},${FNAME} ${LNAME},${EMAIL} |
| 14:14.31 | kombi | afrosheen: sorry people.. |
| 14:14.32 | afrosheen | I could just see him soldering all night then ending up with a heap of smoking phones scratching his head |
| 14:14.46 | *** part/#asterisk Feral_Kid (n=FeralKid@red-corp-201.170.80.61.telnor.net) |
| 14:14.46 | ManxPower | xpot: it does not |
| 14:14.46 | Katty | xpot: i know it will do extension info.. |
| 14:14.47 | kombi | like it happened to me many times.. |
| 14:14.49 | [TK]D-Fender | kombi: No, you NIC will NOT power a phone. You need either a PoE Switch or an in-line injector |
| 14:14.58 | Katty | xpot: like read the ext the call is coming in on and not ask you for the mailbox number. |
| 14:15.01 | afrosheen | kombi, what kind of phones are these |
| 14:15.10 | *** join/#asterisk SwK (n=SwK@65.192.110.34) |
| 14:15.13 | Katty | SwK: ! |
| 14:15.14 | kombi | afrosheen: shiny cisco 7941 |
| 14:15.15 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:15.25 | [TK]D-Fender | kombi: Older Cisco's only support Cisco 48v PoE. Newer ones support 802.3af |
| 14:15.51 | [TK]D-Fender | kombi: 7941G supports 802.3 so you can use a standard PoE swithc/injector |
| 14:15.58 | aydiosmio | ~phones |
| 14:16.12 | jbot | somebody said phones was http://bani.anime.net/phones/. SIP Hardphones in order of quality/auggestibility: Polycom (any), SNOM, Aastra 480i, Linksys SPA-9XX, Grandstream, Cisco. |
| 14:16.25 | xpot | Katty: is there documentation somewhere to read ext? |
| 14:16.27 | kombi | [TK]D-Fender: got Jazz and Precision btw;) great! PoE switch is a separate box I take it? (like a big muff) |
| 14:16.40 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 14:16.48 | afrosheen | lol cisco is below grandstream |
| 14:16.53 | ManxPower | kombi: it CAN be a seperate box, it can also be built into the switch |
| 14:17.21 | ManxPower | afrosheen: someone changed that recently. GS was not even on the list before. |
| 14:17.24 | afrosheen | kombi, like Manx said, there are actual PoE switches that sit in place of a standard switch and supply power |
| 14:17.27 | afrosheen | ManxPower, hahahah |
| 14:17.30 | PioneerVM2 | hey manx: that sip fixup thing tunred off just solved our other problem too, thanks |
| 14:17.37 | PioneerVM2 | fixed my employees problem |
| 14:17.40 | kombi | I was thinking of something from within the linux box that runs our beloved pbx.. |
| 14:17.44 | ManxPower | PioneerVM2: I'm smarter than I look |
| 14:17.46 | afrosheen | PioneerVM2, guess they should rename that to Sip breakup |
| 14:18.14 | afrosheen | kombi, doesn't exist and I don't get how that would work anyway |
| 14:18.48 | kombi | ok, understood! Thanks so far, I'm afraid I might ask more silly stuff some time soon.. |
| 14:19.23 | afrosheen | kombi, I'm picturing you sitting at the server with 2 network cards and a single phone...am I right? |
| 14:19.35 | kombi | completely right! |
| 14:19.44 | afrosheen | kombi, it's all clear to me now |
| 14:19.54 | kombi | shiny phone won't light up..;( |
| 14:20.05 | afrosheen | kombi, it should have a power injector cable |
| 14:20.05 | Katty | i hate when shiny phone doesn't light up :< |
| 14:20.20 | aydiosmio | so what's with the cisco phones? |
| 14:20.21 | PioneerVM2 | afro yea, fcrazy |
| 14:20.26 | aydiosmio | compatibility problems? |
| 14:20.29 | PioneerVM2 | here i was thinking i needed that stupid option |
| 14:20.32 | ManxPower | OK, this is just bizarre. Someone has gotten their hands on some LSD. http://www.forbes.com/business/feeds/afx/2007/05/10/afx3708595.html |
| 14:20.46 | Katty | xpot: i'm not sure. |
| 14:20.48 | kombi | afrosheen: didn't come with it unfortunately, it's optional.. |
| 14:20.51 | Katty | xpot: anthm helped me with mine. |
| 14:21.07 | ManxPower | kombi: you will need to buy one. |
| 14:21.31 | Katty | xpot: [TK]D-Fender and Hmmhesays also helped me. |
| 14:21.37 | afrosheen | PioneerVM2, I've found that in general, Sonicwall and Cisco hardware really breaks SIP if you tick a logical option on them |
| 14:21.42 | ManxPower | My main issues with Cisco phones are 1) SIP firmware costs extra. 2) Power supply costs extra. 3) Not all phone support SIP 4) many features are only avaialble with CCM |
| 14:21.45 | kombi | point taken, or rather buy a PoE Switch to supply all phones at once, right? |
| 14:22.05 | afrosheen | kombi, how many phones do you plan on having there, and why Cisco? |
| 14:22.26 | kombi | afrosheen: some 4 and because of Jack Bauer |
| 14:22.44 | afrosheen | kombi, hahaha..bad reasons, I have a 24 ringtone on my polycom here |
| 14:23.05 | kombi | same here.. they look so cool too, to me at least.. |
| 14:23.55 | xpot | Katty: thanks, I will try to catch one of them. thank you |
| 14:23.59 | SwK | Katty! |
| 14:25.13 | afrosheen | kombi, yes, they look good, but so does the polycom ip650 but it's easier to use, comes with power, and sounds better :) |
| 14:25.47 | aydiosmio | ManxPower: what features require CCm for example? |
| 14:26.06 | ManxPower | aydiosmio: softbuttons, I believe |
| 14:26.09 | Katty | SwK: I setup a shiny new server!! 1.4! |
| 14:26.10 | PioneerVM2 | do they make any IAX boxes? |
| 14:26.21 | PioneerVM2 | like analog/digital linksys boxes using IAX? |
| 14:26.21 | ManxPower | aydiosmio: see the mailing list archives |
| 14:26.28 | ManxPower | PioneerVM2: no |
| 14:26.31 | kombi | afrosheen: point taken, too late though.. What kind of PoE switch would you recommend for 4 thingies? |
| 14:26.36 | afrosheen | PioneerVM2, nobody makes iax stuff but digium :( |
| 14:26.42 | PioneerVM2 | from what i read it seems like it would solve a ton of problems |
| 14:26.52 | Katty | iax stuff? |
| 14:26.55 | afrosheen | kombi, a tiny 8 port Linksys or something comparable |
| 14:26.56 | anonymouz666 | hi Katty |
| 14:27.00 | Katty | what sort of iax stuff? |
| 14:27.01 | PioneerVM2 | would be interesting to see a SIP->IAX converter box for your network |
| 14:27.10 | Katty | i thought iax only worked with software stuffs, like iaxcomm |
| 14:27.13 | ManxPower | PioneerVM2: "It's only kinky the first time" "NAT is only hard the first time" |
| 14:27.19 | Katty | anonymouz666: hello |
| 14:27.21 | PioneerVM2 | like a Linksys Pap2t that used IAX instead |
| 14:27.22 | Katty | anonymouz666: (= |
| 14:27.39 | PioneerVM2 | it's just a pain to have to do all these hacks to make things work |
| 14:27.51 | afrosheen | PioneerVM2, personally I wish everything was IAX, because dealing with double nat is a bitch every time |
| 14:27.54 | PioneerVM2 | the protocal for sip just seems overly complicated |
| 14:28.11 | PioneerVM2 | i shouldnt have to use STUN or proxies to get my phone to work, or port forwarding or whatever |
| 14:28.51 | afrosheen | yeah IAX usually just does the Gaijin Smash through firewalls, layers of NAT, etc. |
| 14:28.56 | PioneerVM2 | man the # of options in these linksys converter boxes is mind boggling |
| 14:29.21 | ManxPower | PioneerVM2: The Linksys PAP2s that I worked with only need 4 options different from the default |
| 14:29.29 | PioneerVM2 | yea me too |
| 14:29.33 | PioneerVM2 | just had to set NAT and STUN |
| 14:29.33 | ManxPower | user/password, dialplan, nat keepalive |
| 14:29.47 | [TK]D-Fender | afrosheen: I run double-NAT scenarios all the time. Never an issue |
| 14:29.51 | PioneerVM2 | im just curious what all this crap does |
| 14:29.56 | Katty | there isn't, by chance, a really uber nice iax software phone, is there? better than iaxcomm |
| 14:30.03 | PioneerVM2 | any good page that explains the dialplan |
| 14:30.13 | PioneerVM2 | so i can figure out how to set my area code for example |
| 14:30.18 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:30.26 | [TK]D-Fender | Katty: Idefisk I believe is probably better : www.asteriskguru.com |
| 14:30.42 | Katty | mkay. |
| 14:31.06 | aydiosmio | I haven't been impressed overall with Idefisk |
| 14:31.18 | aydiosmio | I've had some compatibility problems SIP-wise |
| 14:31.32 | aydiosmio | whereas SJPhone worked fine |
| 14:31.36 | Katty | [TK]D-Fender: Ooo, it's shiny blue! |
| 14:31.55 | Katty | iax is so much nicer going through a firewall tho. |
| 14:31.56 | afrosheen | Katty, are you a crow by chance |
| 14:32.03 | *** join/#asterisk phillipk (n=pkey@fw.datafax.net) |
| 14:32.07 | Katty | afrosheen: a ..crow? |
| 14:32.12 | Katty | afrosheen: i have a wow druid. |
| 14:32.19 | Katty | afrosheen: i think that's as close to a crow as i can get. |
| 14:32.26 | afrosheen | yeah crows love shiny things and I count at least 7 from you so far today :) |
| 14:32.32 | Katty | oh. |
| 14:32.33 | Katty | a kinder. |
| 14:32.45 | Katty | maybe a smidgen :> |
| 14:32.52 | Katty | mostly, i just like it when uber neat things happen ... |
| 14:33.05 | Katty | like dial $extension and the cdrom drive ejects. |
| 14:33.19 | Katty | or the machine sshes over to another one, and connects to xmms-shell, and changes my song for me. |
| 14:33.24 | afrosheen | haha |
| 14:33.40 | afrosheen | jacob is in your machine |
| 14:33.57 | Katty | what?! idefisk is only windows? |
| 14:34.10 | Katty | but, but...butbut...but :< |
| 14:34.21 | [TK]D-Fender | Katty: How about we mhave * make coffee for you like mine used to? ;) |
| 14:34.33 | Katty | [TK]D-Fender: meh, coffee. |
| 14:34.41 | Katty | [TK]D-Fender: i'll take some redbull (= |
| 14:35.10 | [TK]D-Fender | Katty: No palpatations, kplzthxbibi ;) |
| 14:35.12 | Katty | [TK]D-Fender: oh, while i'm thinking about it. |
| 14:35.23 | Katty | [TK]D-Fender: lolzomgwtfkthxbi?! |
| 14:35.35 | [TK]D-Fender | Katty: EXACTLY! |
| 14:35.42 | DarKnesS_WolF | the cdr sql table not included somewhere in asterisk source or in asterisk-addon ? |
| 14:35.51 | DarKnesS_WolF | [TK]D-Fender: long time not seen ;-) |
| 14:35.55 | Katty | file: i never got your sms :< |
| 14:36.00 | Katty | file: it hates me :< |
| 14:36.08 | file | Katty: I never got yours :( |
| 14:36.13 | Katty | *sob* |
| 14:36.28 | Katty | [TK]D-Fender: darn you and your lack of magic wand!! |
| 14:36.31 | [TK]D-Fender | file: Take a number! |
| 14:37.20 | kombi | Someone, name any brand of PoE switches |
| 14:37.44 | *** join/#asterisk bbryant (i=brett@nat/digium/x-c7b6c0efe8266ef8) |
| 14:37.55 | kombi | nevermind.. |
| 14:38.27 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 14:38.43 | skirmisha | guys what is the command to see if asterisk is registered on remote side |
| 14:39.23 | aydiosmio | oh no way! the snom phones have built in packet capture!?? |
| 14:39.28 | LeddyHM | iax2 show peers? |
| 14:40.17 | Katty | i had a snom phone once.. |
| 14:40.25 | Katty | it was my first. sniffle. |
| 14:40.49 | Katty | now it just sits on my desk and takes up space. |
| 14:42.06 | [TK]D-Fender | skirmisha: "sip show registry" / "iax2 show registry" |
| 14:42.16 | skirmisha | let me check |
| 14:42.39 | [TK]D-Fender | aydiosmio: Once they grab a packet.... it never comes back! |
| 14:43.12 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 14:43.30 | skirmisha | yes thanks a lot |
| 14:45.19 | aydiosmio | I'm impressed with the features of the snom phones |
| 14:51.43 | *** join/#asterisk marc\cba (n=marc@cpc1-whit2-0-0-cust972.cdif.cable.ntl.com) |
| 14:54.48 | JerJer | is there some trick to making G.722 work (hd polycom fone) |
| 14:55.27 | [TK]D-Fender | JerJer: Should only have to specify the codecs on the phone and call a compatable endpoint |
| 14:55.31 | ManxPower | JerJer: A goat |
| 14:55.36 | [TK]D-Fender | JerJer: Mind you HD is a total waste |
| 14:56.14 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 14:56.17 | Uatec | Greetings |
| 14:56.52 | Uatec | hey |
| 14:57.02 | Uatec | i'm trying to setup group pickup on my phone system |
| 14:57.26 | Uatec | is there something really basic that i have to turn on? becuase when i dial *8# from my sip phone i just get "Not Found" |
| 14:58.13 | ManxPower | Uatec: Why the #? |
| 14:58.25 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 14:58.48 | Uatec | umm |
| 14:58.53 | Uatec | becuase i read somewhere that it should |
| 14:59.08 | Uatec | so *8? |
| 14:59.13 | Uatec | with *8 i get Unavaiable |
| 14:59.14 | Uatec | hmm |
| 14:59.19 | Uatec | AHAH |
| 14:59.21 | Uatec | that's better |
| 14:59.24 | Uatec | nothing to pickup |
| 14:59.27 | Uatec | GREAT :D |
| 14:59.34 | Uatec | ty for correcting my stupid mistake |
| 14:59.43 | ManxPower | Uatec: SOME phones will eat the # before they send the digits to Asterisk |
| 14:59.57 | Uatec | ok |
| 14:59.59 | Strom_M | ah, *8...which conflicts with ten vertical service codes |
| 15:00.10 | Strom_M | like...*82 to unblock caller ID |
| 15:00.42 | Uatec | oh? |
| 15:00.53 | *** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:01.14 | Strom_M | ~vsc |
| 15:01.26 | jbot | methinks vsc is Vertical Service Codes such as *67, *69, *72, and *82. These codes are generally reserved for specific uses, and it's a bad idea to conflict with the official assignments. A list of assigned VSCs for North America is at http://nanpa.com/number_resource_info/vsc_assignments.html and http://www.nanpa.com/number_resource_info/vsc_definitions.html |
| 15:01.34 | JerJer | [TK]D-Fender: customer wants it |
| 15:02.01 | [TK]D-Fender | JerJer: SAD |
| 15:02.14 | JerJer | yep |
| 15:02.59 | SoloFlyer | does asterisk ignore polarity reversals (for the hangup) that happen during the cid spill |
| 15:03.02 | Katty | uh. where do you put primary and secondary dns info in debian? |
| 15:03.19 | Katty | i /thought/ it was in /etc/network/interfaces, but it looks like debian is special. |
| 15:03.22 | [TK]D-Fender | Katty: /etc/resolv.conf |
| 15:03.27 | Katty | Ooo, thanks. |
| 15:04.09 | coppice | [TK]D-Fender: why is HD a waste? |
| 15:04.50 | *** join/#asterisk LuXten (n=pippo@87.19.42.145) |
| 15:05.12 | [TK]D-Fender | coppice: how much real gain will you have for only taking advntage of it within your enterprise? And then adding transcoding load everywhere else? Then factor in the COST. |
| 15:05.14 | ManxPower | SoloFlyer: Asterisk defaults to USA signalling. USA signalling does not do polarity reversal and so by default asterisk won't use it |
| 15:06.13 | coppice | if its just improving calls within your organisation, its a substantial benefit. What does it cost, anyway? |
| 15:06.17 | afrosheen | [TK]D-Fender, I decided the same thing when our CIO got hyped on it |
| 15:06.31 | Uatec | is it possible to be in multiple pickup groups? and so pickup multiple calls? |
| 15:06.35 | aydiosmio | has anyone seen IP desk phones with bluetooth capability yet? |
| 15:06.43 | afrosheen | coppice, it only works between hd-capable phones, when it hits the PSTN that advantage disappears = pointless |
| 15:07.04 | coppice | most VoIP phones can do G.722 |
| 15:07.08 | [TK]D-Fender | coppice: $250 USD for the lowest Polycom phone supporting it, And upwards from there |
| 15:07.13 | *** join/#asterisk CVirus (n=GoD@196.205.193.14) |
| 15:07.29 | CVirus | What's a PSTN pass-through port ? |
| 15:07.39 | [TK]D-Fender | coppice: Most? Snom / GS are marginal with G.711 ! |
| 15:07.53 | ManxPower | CVirus: if power fails it will hardware the PSTN port to the phone port. |
| 15:08.03 | ManxPower | might to the same if the VoIP server is down. Don't know. |
| 15:08.05 | *** join/#asterisk red9012 (n=marc3234@206-248-174-34.dsl.teksavvy.com) |
| 15:08.09 | coppice | what do you mean by marginal? |
| 15:08.18 | [TK]D-Fender | coppice: And the IP 550 is little better than an IP 501 which costs $170. |
| 15:08.55 | CVirus | ManxPower: PSTN Failover == If the VOIP is down, it switches to the PSTN ? |
| 15:09.02 | coppice | listening to people speak through a 3.5kHz bandwidth pipe is like something from the dark ages |
| 15:09.10 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:09.22 | [TK]D-Fender | coppice: Polycom kills everything but Cisco on G.711 audio quality. What would G.722 do for a phone than can't keep up with the basics? |
| 15:09.26 | Uatec | under what situations are you expecting the voip to be down? |
| 15:09.27 | ManxPower | CVirus: You would have to check to be sure. The only verson of PSTN failover that I've seen only kicks in when the power is off. |
| 15:09.45 | CVirus | Automatic PSTN Fallback (Loss of power or IP connectivity) |
| 15:09.46 | Uatec | i'm running our asterisk box on a UPS and the phones are powered by power over ethernet |
| 15:09.49 | [TK]D-Fender | CVirus: Don't use a term like that without mention what DEVICE you are referring to. |
| 15:09.52 | CVirus | that does it I guess |
| 15:09.52 | Uatec | which is also powered on the UPs |
| 15:10.04 | Uatec | even if the power goes down the isdn stays up and the phones stay up |
| 15:10.05 | CVirus | [TK]D-Fender: the SPA-3102 |
| 15:10.12 | red9012 | hi, I have a problem in that when I dialout from asterisk (dial()), it takes a while for my provider to connect the line, thereby, I get no ring signals for a few seconds. How can I solve this situation? |
| 15:11.25 | ManxPower | red9012: what port type? |
| 15:11.25 | red9012 | sip |
| 15:11.25 | *** join/#asterisk Geniack (i=geniack@17.15.185.213.dk-hvi.res.sta.perspektivbredband.net) |
| 15:11.25 | [TK]D-Fender | CVirus: In that case, the PSTN failover typically jsut bridges the FXO & FXS ports on power failure, nothing more. |
| 15:11.25 | Uatec | red9012, do Ringing() then Dial() |
| 15:11.25 | Uatec | i think |
| 15:11.25 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 15:11.26 | ManxPower | Uatec: that is the most useless suggestion I've seen all day. |
| 15:11.26 | CVirus | [TK]D-Fender: will you please elaborate ? |
| 15:11.26 | ManxPower | red9012: Do you see the Dial happen on the CLI right away or is there a delay before seeing the Dial run on the CLI? |
| 15:11.31 | tzafrir_laptop | hi, can anybody here tell me (potentially pm) what Large Israely companies use Asterisk and will admit to that fact? |
| 15:11.35 | *** join/#asterisk dacter (n=dlittrel@207.200.33.213) |
| 15:11.39 | [TK]D-Fender | CVirus: Power goes out. the FXS port on the unit get direct switched to the FXO port and you phone gets PSTN dialtone directly. |
| 15:11.46 | shido6 | what in the... |
| 15:11.56 | tzafrir_laptop | The only one I can think of is BezeqInt |
| 15:12.00 | Uatec | ManxPower? |
| 15:12.00 | CVirus | [TK]D-Fender: it says on loss of IP connectivity too |
| 15:12.10 | [TK]D-Fender | CVirus: Ok, add that too then. |
| 15:12.12 | *** join/#asterisk Dimik_ (n=Dimik_@unaffiliated/dimik) |
| 15:12.15 | CVirus | cool |
| 15:12.18 | CVirus | [TK]D-Fender: Thanks |
| 15:12.22 | [TK]D-Fender | CVirus: New feature AFAIK |
| 15:12.28 | LuXten | I'm pretty new to Asterisk. |
| 15:12.33 | LuXten | I have a CentOS with latest asterisk and zaptel, with a TDM400P with 2 FXS and 2 FXO in it, and a b410p. |
| 15:12.38 | LuXten | If I dial a number through an fxo port, 3-4 seconds later the call connects, but the number dialed never rings. |
| 15:12.43 | LuXten | The call cannects even if I unplug the pstn line from the card. |
| 15:12.47 | LuXten | How could I debug this issue? |
| 15:12.52 | LuXten | The b410p works OK, and I tried to configure asterisk both by hand and with freepbx, but the issue with the fxo channels does not change. |
| 15:12.52 | tzafrir_laptop | LuXten, BRI in Israel? |
| 15:13.12 | tzafrir_laptop | well, we have one, but this was generally for the purpose of developing BRI support... |
| 15:13.15 | LuXten | No, Italy. But my issue is on the PSTN card. |
| 15:13.21 | SoloFlyer | ManxPower: yes but, when polarity reversal is enabledn if asterisk receives a polarity reversal from zaptel then happens to get another one before asterisk creates a seperate thread to handle the zaptel channel it appears that and new polarity reverals that happen are ignored |
| 15:13.28 | *** join/#asterisk MindTheGap (n=iote@c9502ba2.bhz.virtua.com.br) |
| 15:13.41 | [TK]D-Fender | LuXten: it is considered "answered" the moment ist is placed. Analog was not designed for call progress supervision. |
| 15:14.04 | Uatec | is it possible to be in multiple pickup groups? and so pickup multiple calls? |
| 15:14.58 | ManxPower | Uatec: tes |
| 15:14.59 | ManxPower | yes |
| 15:15.40 | LuXten | Ah ok. Now I'll check if it's dialing the number, _but_ I fear it isn't: last time I waited with the call connected for some minutes, and my cellular phone(I was rcalling it) never ring. |
| 15:16.02 | [TK]D-Fender | LuXten: Well, that is something else entirely. |
| 15:16.32 | *** join/#asterisk fetcher (n=jnh@ip-209-172-35-240.static.privatedns.com) |
| 15:16.36 | [TK]D-Fender | LuXten: I would plug an analog phone in parallel with your FXO port and listen in. You should meak VERY sure you have wired up the right port |
| 15:18.37 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 15:18.50 | LuXten | How can I make such a cable? Are there docs? Since it is an fxo port, I think the only thing I can crash is my analog phone, not a great problem. Am I right? |
| 15:20.12 | [TK]D-Fender | LuXten: Not a cable, a 1$ splitter bought at a dollar store... |
| 15:20.35 | Qwell[] | or $12.50 at radio shack |
| 15:20.37 | Qwell[] | ... |
| 15:21.55 | *** join/#asterisk digus (n=digus@206.222.110.30) |
| 15:22.01 | LuXten | Ok, I think I have tons of them.. but I'd like to be confirmed I'm not going to break the card (yes, if I plug the cord in an fxo port). |
| 15:22.24 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 15:22.36 | *** join/#asterisk luke-jr_work (n=luke-jr@adsl-76-194-177-181.dsl.ksc2mo.sbcglobal.net) |
| 15:22.44 | luke-jr_work | Anyone know what's up with TollFreeGateway? |
| 15:23.06 | [TK]D-Fender | LuXten: ... just plug a phone in parallel. I jsut suggested a splitter in case you only had 1 jack for that line. This isn't wizardry or rocket science.... |
| 15:23.49 | Uatec | http://img523.imageshack.us/img523/859/sipbz7.jpg <-- i'm trying to configure my phone (snom 190) to present my Name and it's own number, but i have no idea what's going on |
| 15:23.56 | *** join/#asterisk wunderkin (n=wunderki@ip68-108-204-139.ph.ph.cox.net) |
| 15:24.16 | Uatec | i have to set both username and authentication username to the number... |
| 15:24.19 | Uatec | sorry |
| 15:24.24 | Uatec | to the handsets sip name |
| 15:25.01 | *** join/#asterisk tonycarstens (n=tonyc36@206.135.21.162) |
| 15:25.09 | Uatec | but if i set the Desplay name to anything other than - then i get Not Registered. |
| 15:25.23 | Uatec | and all call names are forbidden |
| 15:26.06 | tonycarstens | i'm having issues with recieving calls, i believe my zapata.conf is not right can anyone help |
| 15:26.41 | Uatec | tonycarstens, http://rafb.net/paste <-- show us your zapata.conf for a start |
| 15:26.46 | tonycarstens | ok |
| 15:28.18 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:28.20 | tonycarstens | http://rafb.net/p/8KUKoV54.html |
| 15:29.39 | *** join/#asterisk orn (n=orn@skrifstofa-8.iphive.is) |
| 15:29.48 | tonycarstens | i have a tdm04b |
| 15:30.10 | orn | Hi. I should be able to see calls on the Incoming trunk (that there are active calls) in the FOP, right? |
| 15:30.38 | aptura | tony what are you trying to do combine both context on one channel? |
| 15:30.59 | *** join/#asterisk [hC] (n=hardcore@65-122-15-162.dia.static.qwest.net) |
| 15:31.07 | LuXten | [TK]D-Fender: here we say "last famous words". I don't find any phone cable splitter.. I'm going to the shop to buy one, then I'll return here. |
| 15:31.25 | *** join/#asterisk thinwires (n=thinwire@cpe-76-50-56-82.buffalo.res.rr.com) |
| 15:31.41 | tonycarstens | aptura: ideally what i want is for line 1 to be used for incoming/outbound calls unless it is in use |
| 15:31.48 | tonycarstens | then it will try 2, then 3, etc. |
| 15:31.51 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-96-8.hvc.res.rr.com) |
| 15:33.15 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:33.29 | thinwires | hey i have a question guys, is there anyone for me to write a dialplan that involves call hunting? |
| 15:33.31 | [TK]D-Fender | tonycarstens: You only show 1 channel being configured. In there you define it once with 1 context, and then OVERRIDE the context and reapply over the same channel |
| 15:34.00 | backblue | hi, anyone using iaxmodem? |
| 15:34.29 | tonycarstens | tk: so i can only use one context per channel |
| 15:34.57 | [TK]D-Fender | tonycarstens: correct. That is the context where INCOMING calls get sent |
| 15:36.35 | tonycarstens | so if i have channel 1 used for INCOMING calls would i be able to use it to make outbound calls? |
| 15:38.22 | [TK]D-Fender | tonycarstens: yes. |
| 15:38.42 | [TK]D-Fender | tonycarstens: you just USE the channel in your dial statement. |
| 15:38.52 | tonycarstens | ahhh |
| 15:38.54 | [TK]D-Fender | tonycarstens: Dial(Zap/1/1234567890) |
| 15:38.59 | tonycarstens | got ya |
| 15:39.03 | orn | where is the setting for timeouts when trying to call an extension? i find it is too low. it is only 4 rings, and then an announcement saying an error has occurred |
| 15:39.18 | [TK]D-Fender | tonycarstens: if its available it will dial out. HOW you get around to calling that line in your dialplan is irrelevent |
| 15:39.29 | tonycarstens | i apoligize, its been a while since i last configured a machine, and i already had limited know-how |
| 15:39.34 | [TK]D-Fender | orn: "show application dial" |
| 15:39.45 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-96-8.hvc.res.rr.com) |
| 15:39.56 | tonycarstens | TK: got it going thanks! |
| 15:40.19 | orn | thanks d-fender :) |
| 15:41.01 | Uatec | <PROTECTED> |
| 15:41.01 | Uatec | ! |
| 15:41.11 | Uatec | Snom put their reset and Reboot buttons so close together |
| 15:41.13 | Uatec | and i pressed reset |
| 15:41.20 | Uatec | now i can't get the damn phone working again |
| 15:41.31 | aydiosmio | way to have fat fingers fatty mcfatfat |
| 15:42.24 | thinwires | lol @ aydiosmio |
| 15:42.38 | [TK]D-Fender | aydiosmio: Wow... you must have rules pre-shool with an iron fist.... |
| 15:42.43 | tzanger | hahaha |
| 15:42.49 | *** join/#asterisk redax (n=redax@mail.caracom.hu) |
| 15:42.51 | redax | hi |
| 15:43.02 | aydiosmio | [TK]D-Fender: as long as you got a laugh out of it |
| 15:43.23 | [TK]D-Fender | aydiosmio: Doesn't quite make the grade, but a "B-" for effort |
| 15:43.40 | Uatec | aydiosmio, grrrr |
| 15:43.47 | aydiosmio | I can't just throw around my A material |
| 15:43.50 | thinwires | not to many people know this but aydiosmio is actually O'Doyle spelled backwards... |
| 15:43.53 | Uatec | lol |
| 15:45.24 | thinwires | so anyways, yeah, having issues with my dial config here... |
| 15:45.44 | thinwires | Is there anyway to add a voicemail box to a call queue? |
| 15:46.01 | LuXten | [TK]D-Fender: I found the splitter. When I plug the line in the FXO channel, I can hear a low dial tone. If I dial, I hear dtmf signals, but the number is'nt called. One strange thing to me is that the last digit is dialed with some delay. Maybe it's normal. Then some seconds after, the dial tone vanishes. I can still dial from asterisk, I hear the dtmf tones more clearly, but nothing happens. |
| 15:47.00 | [TK]D-Fender | LuXten: perhaps you need to increase the gain on your card a bit. |
| 15:47.12 | dacter | how stable is asterisk-1.4-current? any serious problems? |
| 15:47.15 | aptura | check for your ignorepat in extensions LuXten and tell us what it says |
| 15:47.30 | file | dacter: setup a test machine and put it through what YOUR usage is |
| 15:47.33 | aptura | did he state he was dialing 9? |
| 15:47.57 | file | for example, I had 1.4 up for 6 weeks fine and dandy on my personal PBX... but my load and usage could be vastly different then yours and thus my answer invalid |
| 15:48.08 | dacter | sigh... |
| 15:48.22 | dacter | your caution is duly noted. |
| 15:50.28 | [TK]D-Fender | LuXten: and you might want to pastebin the CLI output of your attempt at verbose 10. |
| 15:51.07 | [TK]D-Fender | aptura: that is NOT applicable to ignorepat. |
| 15:51.16 | [TK]D-Fender | aptura: for about 5 different reasons |
| 15:52.11 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 15:52.23 | aptura | TK you are right I did not read it right. |
| 15:52.24 | aptura | :) |
| 15:54.49 | *** part/#asterisk SoloFlyer (n=soloflye@202.novadefence.com.au) |
| 15:56.20 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:56.33 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 15:57.40 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:57.55 | orn | at the risk of being slaughtered i will still ask the question :-) So... I know now that the Dial takes the timeout as a parameter, but where in the asterisk configuration files (I'm guessing this would be somewhere in extensions.conf?) do I configure the value? It seems to be defined in that file as ARG1, but I can't find where it gets ARG1 from |
| 15:58.33 | [TK]D-Fender | orn: It is the 3rd parameter passed to Dial. |
| 15:58.51 | orn | oh ok |
| 15:58.54 | [TK]D-Fender | orn: it isn't "configured". Every single place you dial can be different |
| 15:59.12 | orn | ok, so is it set in the extensions and trunks? |
| 15:59.15 | [TK]D-Fender | orn: and I pointed your right to TFM... so go READ! ;) |
| 15:59.37 | LuXten | [TK]D-Fender: I set rxgain to 8.5 (no idea, found on a doc) and now the dialed phone rings. Unfortunately I hung up the sip phone I'm using to dial, but the called phone continued to ring. |
| 15:59.38 | *** join/#asterisk Paavum (n=Dorphals@200.71.58.39) |
| 15:59.40 | orn | i read the command you passed to me :P i guess i'm just not that intelligent :) |
| 15:59.40 | Paavum | Hello |
| 15:59.47 | [TK]D-Fender | orn: "trucks" is a work that should never be used, and "extensions is in your context is dangerously close.... |
| 15:59.55 | Paavum | How can I know if a couple of extensions are busy or not before dialing? |
| 15:59.58 | [TK]D-Fender | trunks* |
| 15:59.59 | redax | btw the default timeout for Dial() is must be ages :) |
| 16:00.10 | [TK]D-Fender | Paavum: "show application chanisavail" |
| 16:00.23 | [TK]D-Fender | redax: default = NONE. |
| 16:00.46 | Paavum | [TK]D-Fender --> Yes, but when I do chanisavail and I have the phone offhook it tells me its available |
| 16:00.47 | [TK]D-Fender | orn: Everything for this is in extensions.conf |
| 16:00.53 | redax | yeah. I know. that == infine |
| 16:01.01 | [TK]D-Fender | Paavum: then you are clearly not using it right :) |
| 16:01.13 | [TK]D-Fender | Paavum: Pastebin <- |
| 16:01.16 | orn | ok, thanks d-fender |
| 16:01.25 | Paavum | Gimme a sec |
| 16:01.50 | Uatec | Weeeee |
| 16:01.57 | Uatec | got my mobile connected to asterisk |
| 16:02.32 | [TK]D-Fender | Uatec: http://www.albinoblacksheep.com/flash/weeee.php |
| 16:02.34 | Qwell[] | Uatec: chan_cellphone? |
| 16:03.08 | LuXten | [TK]D-Fender: I just found a doc suggesting to use ztmonitor to set gain. But, where should the peak levels arrive? 100%, 50%? |
| 16:04.03 | [TK]D-Fender | LuXten: tweak it by hand a little at a time and see if that helps |
| 16:04.11 | Paavum | http://pastebin.ca/483132 |
| 16:05.29 | Paavum | [TK]D-Fender --> http://pastebin.ca/483132 |
| 16:05.44 | [TK]D-Fender | Paavum: And you're telling me that when all 4 are on calls that it does the No-op? |
| 16:06.16 | Paavum | Nope, 321 is offhook (no call, just off the hook) |
| 16:06.36 | Paavum | and when I do ChanIsAvail I get 321 as an avail chan |
| 16:06.40 | [TK]D-Fender | Paavum: if ANY of them are not on calls then it will report that channel back... |
| 16:06.58 | [TK]D-Fender | Paavum: it finds the first AVAILABLE channel |
| 16:07.07 | [TK]D-Fender | Paavum: O all 4 would have to be on calls. |
| 16:07.17 | Paavum | Yes... but SIP/321 is NOT available |
| 16:07.17 | *** join/#asterisk Batimam (i=Sblerght@200-140-70-249.gnace702.dsl.brasiltelecom.net.br) |
| 16:07.19 | Paavum | is off hook |
| 16:07.27 | Paavum | and it reports it available |
| 16:07.30 | [TK]D-Fender | Paavum: If you want to do something if ANY of them are on a call, then you'll have to test each independantly |
| 16:07.48 | Batimam | someone help me with sm56 (clone)? |
| 16:07.51 | [TK]D-Fender | Paavum: And what do you men "off-hook"? |
| 16:07.58 | [TK]D-Fender | Paavum: off-hook != on a call |
| 16:08.43 | Paavum | Off-hook, like when you just enjoy the dialtone and do not care to interrupt the sound with DTMFs :P |
| 16:08.44 | Batimam | asterisk/zaptel/winmodem sm56 (clone digium) |
| 16:09.08 | [TK]D-Fender | Paavum: Again, that is not ON A CALL. |
| 16:09.43 | Paavum | [TK]D-Fender ... bt it aint available... is it? |
| 16:09.49 | [TK]D-Fender | lunch, back in a few... |
| 16:09.55 | Paavum | oki |
| 16:09.55 | [TK]D-Fender | Paavum: Sure it is. |
| 16:10.18 | Paavum | I dial it and get "the person at extension... is on the phone... Please die" |
| 16:11.03 | Batimam | <PROTECTED> |
| 16:11.22 | ghenry | Are Sangoma cards hard to configure? I've been reading: http://wiki.sangoma.com |
| 16:12.43 | afrosheen | ghenry, of course they're a bitch at first but they have a good wiki that helps alot |
| 16:13.01 | ghenry | yeah, double driver stuff, i.e. wanpipe |
| 16:13.04 | afrosheen | ghenry, and if you get hung up you can always call them, just pray to god you don't get the chinese guy |
| 16:13.05 | ghenry | Hope it works ok |
| 16:13.10 | ghenry | ;-) |
| 16:13.12 | Paavum | But you are right... although ppl may be stupid, they just dont leave the phone off the hook |
| 16:13.13 | Paavum | :P |
| 16:13.31 | afrosheen | Paavum, you've never been to america have you |
| 16:13.40 | orn | d-fender: you seem to be somewhat of an expert; incoming and active calls that come through an incoming trunk, should the be visible somehow in the FOP? That is, should you see some indication on the incoming trunk that there are calls on it? |
| 16:14.07 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:14.16 | Zeeek | indeed |
| 16:14.45 | luke-jr_work | Anyone know what's up with TollFreeGateway? They like to get listed in Enums, but don't seem to ever work |
| 16:17.19 | Zeeek | russellb are you there to hook us up by any chance? |
| 16:18.37 | russellb | i can't talk, but if anyone else here wants to join, i can do it |
| 16:18.48 | Zeeek | one never knows, do one? |
| 16:19.05 | *** join/#asterisk jmacz (n=jmacz@190.24.97.247) |
| 16:19.39 | Zeeek | If no one shows up, I have to talk. That's never good :) |
| 16:19.44 | luke-jr_work | anyone have a * box willing to stress test for 4 hours on 24 lines? :p |
| 16:19.53 | luke-jr_work | calling an 888 # |
| 16:20.26 | russellb | luke-jr_work: i have root on iaxtel :-p |
| 16:20.28 | [TK]D-Fender | Paavum: PASTEBIN all of the related dialplan, and CLI output at verbose 10 including a "show channels" prior to test |
| 16:20.41 | luke-jr_work | russellb: :D |
| 16:20.44 | russellb | but file would hurt me |
| 16:20.45 | Batimam | hey |
| 16:20.48 | luke-jr_work | aww |
| 16:20.55 | file | WHAT |
| 16:21.00 | luke-jr_work | even tho it's toll free? |
| 16:21.00 | file | oh |
| 16:21.06 | Qwell[] | russellb: not for like a week :p |
| 16:21.11 | russellb | heh |
| 16:21.12 | Qwell[] | he'll have forgotten by then |
| 16:21.27 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 16:21.38 | luke-jr_work | file: :) |
| 16:22.35 | Batimam | to make my sm56 work as fxo |
| 16:22.52 | luke-jr_work | file: please? |
| 16:22.58 | Batimam | that should first be installed in my os ? |
| 16:23.24 | Batimam | oder zaptel recognize that without configure debian/etch ? |
| 16:23.24 | Paavum | Ok |
| 16:23.48 | Zeeek | talk about your asterisk-related hopes, dreams and disappointments: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 |
| 16:24.24 | thinwires | hey everyone, I'm trying to get my phone to dial out with Voip Jet using http://pastebin.ca/483176 as my dial plan, any idea's why sometimes the numbers come out with a 011 infront of them? |
| 16:24.58 | Zeeek | 011 is international access? |
| 16:25.09 | Qwell[] | talkshoe.com is such a silly domain |
| 16:25.10 | thinwires | yes, that's the issue, these are local calls |
| 16:25.21 | Qwell[] | an overly restrictive filter would quickly block that... |
| 16:25.31 | luke-jr_work | thinwires: they don't, with that dialplan |
| 16:26.02 | thinwires | ? that's the thing, the number are coming out with an 011 |
| 16:26.18 | thinwires | only about half of the time though, it's confusing as hell... |
| 16:26.20 | luke-jr_work | thinwires: not with that part of your dialplan... |
| 16:26.26 | Zeeek | http://x2z.eu if you prefer |
| 16:26.31 | luke-jr_work | pastebin some logs and more dialplan perhaps |
| 16:27.38 | thinwires | Luke: so your saying that my dial plan is written right and it shouldn't be putting the 011 on the front? |
| 16:28.24 | *** part/#asterisk Batimam (i=Sblerght@200-140-70-249.gnace702.dsl.brasiltelecom.net.br) |
| 16:29.05 | Paavum | [TK]D-Fender --> http://pastebin.ca/483186 |
| 16:29.11 | Zeeek | file what happened? You disappeared |
| 16:29.34 | luke-jr_work | thinwires: that part of it, yes |
| 16:29.50 | [TK]D-Fender | Paavum: FAILURE |
| 16:30.02 | file | I'm fighting with my phone |
| 16:30.09 | Zeeek | o |
| 16:30.13 | luke-jr_work | file: plz plz plz ? :p |
| 16:30.14 | [TK]D-Fender | Paavum: I wanted to see "chos channels" PRIOR to the test and I want to SEE it taken off hook, etc |
| 16:30.17 | file | we have a love hate relationship these days |
| 16:30.32 | Zeeek | here are a few PINs free at the moment: |
| 16:30.36 | Paavum | Its a SIP extension |
| 16:30.50 | Paavum | I cant see when I "off the hook" it |
| 16:31.17 | Zeeek | 2007 2007 22# |
| 16:31.23 | Zeeek | 2007 2007 05# |
| 16:31.31 | Zeeek | 2007 2007 00# |
| 16:32.38 | *** join/#asterisk cayorde (n=flexable@host184-111-dynamic.17-87-r.retail.telecomitalia.it) |
| 16:33.30 | [TK]D-Fender | Paavum: I still want the rest, and it might help if you told us which model you're using |
| 16:34.14 | [TK]D-Fender | Paavum: And I don't see the sample of your trying to dial it an FALIING after like you described either |
| 16:34.25 | Paavum | Oh ok ok |
| 16:34.26 | Paavum | sorry |
| 16:34.32 | Paavum | Will show you that also |
| 16:34.41 | *** join/#asterisk jazzanova (n=boris@S010600146cfc7d5b.vc.shawcable.net) |
| 16:34.42 | jazzanova | hi |
| 16:35.21 | jazzanova | i am looking for a canadian DID provider, with flat mothly fee and multi-channels for outgoing calls. |
| 16:36.09 | thinwires | Luke_jr: http://pastebin.ca/483196 here is the CLI call log... do you think it's because I have the extra 1 on the front of the call? |
| 16:37.56 | [TK]D-Fender | jazzanova: www.unlimitel.com |
| 16:38.34 | ManxPower | jazzanova: and I'm looking for a unicorn |
| 16:38.42 | Paavum | [TK]D-Fender --> http://pastebin.ca/483206 |
| 16:38.43 | ManxPower | there is no such thing as REAL unlimited. |
| 16:38.50 | ManxPower | Honest providers tell you that. |
| 16:39.01 | [TK]D-Fender | ManxPower: he never said unlimited... |
| 16:40.37 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 16:40.38 | [TK]D-Fender | Paavum: Well I guess your phone is just retarded.... |
| 16:40.40 | BSD_Tech | hey |
| 16:40.45 | ManxPower | [TK]D-Fender: silly me |
| 16:40.57 | BSD_Tech | is there any plan to make the asterisk http server secure ? |
| 16:41.08 | BSD_Tech | https and not just http ? |
| 16:41.23 | Qwell[] | BSD_Tech: feel free to backport the changes from trunk |
| 16:41.29 | Paavum | LOL |
| 16:41.37 | [TK]D-Fender | BSD_Tech: Proxy it yourself :) |
| 16:41.38 | xheliox | BSD_Tech: Or use stunnel.. |
| 16:41.38 | BSD_Tech | ok its in the trunk |
| 16:41.46 | Qwell[] | or many other methods |
| 16:41.59 | xheliox | Let's play a game and see how many we can list. |
| 16:42.02 | xheliox | ssh tunneling.. |
| 16:42.04 | BSD_Tech | lol |
| 16:42.13 | Qwell[] | listening on localhost only |
| 16:42.44 | *** join/#asterisk [hC] (n=hardcore@65-122-15-162.dia.static.qwest.net) |
| 16:44.16 | thinwires | ok so is there a way for me to block 011 outbound for a specific provider/trunk? |
| 16:46.17 | BSD_Tech | make it require a passcode |
| 16:46.36 | BSD_Tech | likke 1234| |
| 16:47.15 | thinwires | hm, well I'm not even sure if that's the right way to go about it, I'm not even sure if the problem is on my end, I'm sending the call to them perfectly http://pastebin.ca/483196 ... |
| 16:47.18 | BSD_Tech | or just dont map it on that trunk |
| 16:48.39 | thinwires | it's not, thats the wierd part -- I only have local calls mapped out, but when I call my cell phone with that provider it comes through as 011 instead of the number I give to them |
| 16:49.23 | Dr-Linux | anybody tried web-meetme? |
| 16:49.58 | BSD_Tech | so do they still do the friday phone confrences |
| 16:51.07 | BSD_Tech | the only other thing about the http sevre is fast-cgi for putting ari into the gui |
| 16:51.18 | BSD_Tech | need it for php |
| 16:51.36 | BSD_Tech | dont want to have to add a second http server |
| 16:55.14 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 16:55.15 | coolbeans | Anyone seen this with asterisk realtime? It does it in both 1.2 and 1.4 and google doesn't yeild much so maybe it's something I'm doing: chan_sip.c:16611 reload_config: Section 'did.voip.les.net' lacks type |
| 16:55.21 | orn | [TK]D-Fender: did you see my question regarding trunks earlier? |
| 16:55.22 | blitzrage | you haven't got a 'type' column |
| 16:55.23 | coolbeans | did.voip.les.net is just one of our peers, but there is a type=friend in the db. |
| 16:55.34 | coolbeans | var_name=type, var_val=friend |
| 16:56.00 | blitzrage | for SIP peers, there is a built in dynamic object for those peers |
| 16:56.13 | blitzrage | looks like you're using static realtime |
| 16:56.14 | [TK]D-Fender | orn>d-fender: you seem to be somewhat of an expert; incoming and active calls that come through an incoming trunk, should the be visible somehow in the FOP? That is, should you see some indication on the incoming trunk that there are calls on it? |
| 16:56.19 | coolbeans | blitzrage: Yep. |
| 16:56.21 | blitzrage | also, what version of Asterisk are you using? |
| 16:56.22 | orn | yes, that one |
| 16:56.29 | [TK]D-Fender | orn: depends how you set up FOP |
| 16:56.32 | coolbeans | It does in one both my 1.2 and 1.4 boxes. |
| 16:56.44 | blitzrage | because I found a bug in static realtime, in that if you don't have the table in the order Asterisk expects it, then it won't work |
| 16:56.48 | blitzrage | 1.4.x ? |
| 16:56.55 | coolbeans | 1.4.4 |
| 16:56.57 | blitzrage | ya |
| 16:56.58 | coolbeans | and 1.2.18 |
| 16:56.58 | orn | [TK]D-Fender: I see the outgoing calls, but not the incoming ones. I also can't transfer calls that come in through the trunk to other users, which I think is related |
| 16:57.04 | blitzrage | ya -- the bug was fixed after that |
| 16:57.18 | orn | [TK]D-Fender: That is, I can't transfer them via the FOP |
| 16:57.19 | blitzrage | do you have an 'id' field or someting as the first column? |
| 16:57.22 | coolbeans | So nobody is doing static realtime in < 1.4.4?? |
| 16:57.27 | coolbeans | Yep. |
| 16:57.28 | coolbeans | Ahh! |
| 16:57.31 | coolbeans | I see what you're saying. |
| 16:57.36 | blitzrage | they are -- if you have the DB in the order that Asterisk expects it |
| 16:57.39 | [TK]D-Fender | orn: Sorry, these are FOP questions, not * questions, and I don't use FOP |
| 16:57.42 | coolbeans | drop and re-create my ID field so its at the end. |
| 16:57.45 | blitzrage | now (post 1.4.4), Asterisk will look for the specific columns |
| 16:57.48 | blitzrage | yes |
| 16:58.03 | coolbeans | Great! Will try. Thanks :) |
| 16:58.08 | blitzrage | because in 1.4.4. asterisk does a SELECT * instead of SELECT var_name, var_val .... etc.... |
| 16:58.13 | coolbeans | Ahh... |
| 16:58.49 | orn | [TK]D-Fender: Ok, thanks. Is there any irc channel or some other sort of interactive resource I could use? I have been searching the web for this problem for a long time and I haven't found anything (except a forum that talks about users of HUDlite having the same problem) |
| 16:59.27 | [TK]D-Fender | orn: Forums it is... and if other are having the same problem it could be that there isn't a currently viable solution for it. |
| 17:00.39 | tzanger | blitzrage: there are a few places it does that |
| 17:00.43 | tzanger | bad coders. :-) |
| 17:00.45 | orn | [TK]D-Fender: They were talking about that it had stopped working at some trixbox version |
| 17:01.01 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:01.08 | [TK]D-Fender | orn: Don't even MENTION Trixbox around me.... |
| 17:01.13 | tzanger | [TK]D-Fender: heh |
| 17:01.13 | orn | ok :) |
| 17:01.20 | Qwell[] | [TK]D-Fender: TRIXBOX! |
| 17:01.25 | [TK]D-Fender | orn: And they have nothing to do with each other. |
| 17:01.26 | *** join/#asterisk jav___ (n=jan@riker.artis.uni-oldenburg.de) |
| 17:01.36 | blitzrage | I 3> TRIXBOX! |
| 17:01.39 | [TK]D-Fender | ~trixbox |
| 17:01.52 | jbot | Trixbox is a full linux distro that includes , FreePBX, and other 3rd party add-ons. It is these things on top of which make it seriously painful to support and hence you will find little help here for it. Try asking in #trixbox , or their forums & WIKI at http://www.trixbox.org |
| 17:01.53 | orn | [TK]D-Fender: Just meaning that it had worked at some point... so that the functionality is supposed to be there (at least with HUD) |
| 17:01.53 | blitzrage | errr.... <3 |
| 17:01.53 | blitzrage | lol |
| 17:02.26 | BSD_Tech | Trix box now owned by Fonality |
| 17:02.32 | BSD_Tech | and it has become crap |
| 17:02.37 | Qwell[] | become? ha |
| 17:02.40 | Qwell[] | you must be new here |
| 17:02.49 | Qwell[] | it's *always* been crap :) |
| 17:02.56 | BSD_Tech | no I use to work on the project |
| 17:03.06 | Qwell[] | oh, clearly it became crap when you left then |
| 17:03.09 | Qwell[] | ;) |
| 17:03.37 | BSD_Tech | I quit using the linux junk and went back to bsd |
| 17:03.44 | Nugget | linux is poo. |
| 17:03.46 | BSD_Tech | porting things |
| 17:03.51 | Qwell[] | so is bsd |
| 17:03.52 | Qwell[] | solaris FTW |
| 17:04.00 | Qwell[] | YEAH, I SAID IT |
| 17:04.06 | thinwires | lol, half of my servers are Sun :-p |
| 17:04.07 | Nugget | Linux is the x100p of unixes. :) |
| 17:04.11 | blitzrage | Qwell[]: lol |
| 17:04.13 | Qwell[] | Nugget: haha, nice |
| 17:04.15 | blitzrage | THAT JUST HAPPENED |
| 17:04.17 | BSD_Tech | lol |
| 17:04.20 | Qwell[] | blitzrage: what? |
| 17:05.58 | Qwell[] | hmm |
| 17:05.59 | jav___ | Hi there... I'm trying to compile asterisk using './configure --prefix=$HOME/asterisk' .. after make succesful compiled everything, 'make install' complains about not beeing able to write to '/var/lib/asterisk' .. why is my --prefix not honoured? |
| 17:06.03 | Qwell[] | </Qwell[]> |
| 17:06.34 | Qwell[] | jav___: --sysconfdir and ...one more, I forget |
| 17:06.38 | Qwell[] | ./configure --help |
| 17:07.19 | jav___ | well.. ./configure --help says, that --sysconfdir defaults to PREFIX/etc |
| 17:07.27 | Qwell[] | it does |
| 17:07.31 | Qwell[] | but when you change prefix... |
| 17:08.31 | jav___ | it should change as well, shouldn't it? .. at least that happened in every other autotools software I ever compiled using --prefix |
| 17:08.48 | jav___ | but not with asterisk, you say? I see.. |
| 17:08.59 | Qwell[] | a bunch of autoconfified apps don't do that |
| 17:10.06 | BSD_Tech | man I love having a patio and wireless |
| 17:10.17 | BSD_Tech | I can site outside work and get some sun |
| 17:10.54 | Qwell[] | I mean, if somebody from FSF told us "hey, look, you're dumb - this is wrong", then we'd probably change it, but as far as we understand it, it's really left up to the project |
| 17:11.15 | aptura | BSD I bet :) |
| 17:11.19 | Qwell[] | (autoconf is FSF, right?) |
| 17:11.52 | LuXten | After poking a little with rxgain and txgain, I'm able to dialout via my tdm400p. But it dials only after a few seconds when I plug the pstn line in it. Then, when the dial tone goes away, it never reappears. It seems that the card is not requesting the dial tone correctly, or maybe it does not understand the pstn line tones correctly, since it dials even without the dial tone. What can it be? |
| 17:12.29 | jav___ | I see.. I wasn't aware of the fact, that this is sometimes handled differently... but does that mean, I have to specify every single one of the installation directory options? (because they all default to PREFIX/something) |
| 17:12.46 | [TK]D-Fender | LuXten: Possible set for the incorrect zone.... |
| 17:12.47 | Qwell[] | jav___: as far as I know, it's just --sysconfdir and...sec, lemme look |
| 17:13.04 | Qwell[] | --localstatedir? |
| 17:14.06 | jav___ | that worked! thx for looking it up, Qwell[] |
| 17:17.03 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 17:17.15 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 17:18.28 | DarKnesS_WolF | question .. i want to recorde all the internal calls for SIP users.. so i did a micro for this users.. i want the file format to be callerid-distnation-timestamp |
| 17:18.42 | DarKnesS_WolF | but ${TIMESTAMP} is not working |
| 17:18.44 | DarKnesS_WolF | any idea ? |
| 17:19.44 | *** join/#asterisk glogic (n=rm@ool-4571a1cc.dyn.optonline.net) |
| 17:21.12 | Strom_M | what version of asterisk are you using? |
| 17:21.53 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 17:22.02 | DarKnesS_WolF | 1.4.4 |
| 17:22.15 | glogic | i wish to play tones during a call initiated with Dial at various points during the call i've tried the L() command with dial but this isn't flexible enough, is there a way to possibly play a message to a specific channel using AMI? this would have to work on an alarm |
| 17:22.44 | *** join/#asterisk lokkju_wrk_ (n=lokkju@unaffiliated/lokkju) |
| 17:24.07 | Strom_M | DarKnesS_WolF: read the UPGRADE file; ${TIMESTAMP} has been removed |
| 17:24.14 | Strom_M | it was deprecated as of 1.2 |
| 17:24.26 | DarKnesS_WolF | Strom_M: oh ic :-D |
| 17:24.27 | DarKnesS_WolF | thx ;-) |
| 17:25.08 | DarKnesS_WolF | Strom_M: but from where i can get the new values? |
| 17:25.49 | Qwell[] | in UPGRADE.txt |
| 17:26.12 | DarKnesS_WolF | Qwell[]: same file too :P? thx ;-) |
| 17:28.23 | Strom_M | dare i say it? rtfm :) |
| 17:30.07 | blitzrage | you dare! |
| 17:31.45 | DarKnesS_WolF | Strom_M: i did RTFM the UPGRADE.TXT it it didn't show the new vars. what i have noticed it's going to be replaced with a dialplan functions !?? |
| 17:32.43 | adorah | I need for a project mass paging from multi-users to one point but none of them should be able to hear the other-only the one at the end-point. any suggestions? |
| 17:34.44 | BSD_Tech | never yeard of that |
| 17:34.59 | BSD_Tech | normaly paging is from 1 ext - out to many |
| 17:35.05 | BSD_Tech | your talking revers paging |
| 17:35.11 | BSD_Tech | hmmmm |
| 17:36.04 | BSD_Tech | dont think it can be done |
| 17:36.20 | BSD_Tech | but you might look at queues |
| 17:37.52 | coolbeans | Hey blitzrage, do you know what 'order' the mysql realtime static config table columns need to be? |
| 17:38.57 | blitzrage | sorry, not too sure |
| 17:39.00 | blitzrage | I never figured that out |
| 17:39.32 | blitzrage | Qwell[]: what file would that SQL statement be in for static realtime? |
| 17:39.41 | blitzrage | I could look it up in my current code |
| 17:39.42 | Qwell[] | huh? |
| 17:40.00 | blitzrage | static realtime... where would the SQL statement be for it pulling the database from the DB? |
| 17:40.03 | blitzrage | res_... ? |
| 17:40.16 | Qwell[] | config_odbc? |
| 17:40.22 | Qwell[] | I didn't see the question |
| 17:40.27 | Qwell[] | oh |
| 17:40.31 | Qwell[] | order doesn't matter |
| 17:40.36 | blitzrage | yes it does |
| 17:40.38 | Qwell[] | eh? |
| 17:40.45 | blitzrage | it was a bug I found at it360 |
| 17:40.47 | blitzrage | Kevin fixed it |
| 17:40.53 | blitzrage | because Asterisk does a SELECT * |
| 17:40.55 | *** join/#asterisk Putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 17:41.01 | blitzrage | so if you have an id column or something, it throws it all off |
| 17:41.16 | blitzrage | because asterisk expects things coming in at certain places |
| 17:41.32 | blitzrage | so I was gonna lookup the new SQL statement so coolbeans would know what order Asterisk expected it to be in |
| 17:41.38 | coolbeans | Asterisk is parsing the datasent with the columns ordinal value instead of column name. |
| 17:41.46 | coolbeans | datasent = dataset |
| 17:41.47 | blitzrage | so my question was, "where is static realtime code"? :) |
| 17:42.00 | coolbeans | At least up to 1.4.4 |
| 17:42.06 | Putzz | Im Back please someone help me I can rxfax but txfax doesn not work for some reason. Im creating a .call file for auto dial out it dials soon as other side machine answers it hangs up. here is my .call file: http://pastebin.ca/483315 |
| 17:42.07 | blitzrage | oh, pbx_realtime.c maybe |
| 17:42.16 | blitzrage | hrmmm, nope |
| 17:42.46 | denon | Putzz: im pretty sure rx and txfax doesn't work for a lot of people :) |
| 17:43.06 | Putzz | well im aware of that |
| 17:43.09 | coolbeans | Putzz: You'll have hit and miss success. Use Hylafax and IAXModem if you're bound to do it all with Asterisk. |
| 17:43.10 | Putzz | but I have manage to compile |
| 17:43.14 | Putzz | wich is a huge step |
| 17:43.21 | Putzz | should have use hylafax long time ago |
| 17:43.24 | Putzz | dammit |
| 17:43.37 | denon | just get an as5400 |
| 17:43.54 | coolbeans | The concept behind rxfax/txfax is great, but spandsp doesn't support a lot of the functionality (codecs?) that commercial fax machines use. |
| 17:43.55 | Putzz | does the .call file look right for those who have uses txfax? |
| 17:44.19 | Putzz | it doesnt even try to negotiate |
| 17:44.29 | Putzz | soon as other side answers it goes as complete call |
| 17:44.35 | coolbeans | The spandsp that comes with IAXModem is better and using hylafax fills in the rest. Hylafax is very light and easy on your system. We use it with great success. |
| 17:44.53 | aptura | denon what is the 5400 priced out at |
| 17:44.57 | [TK]D-Fender | Putzz: I'd bet you have to do an ANSWER first |
| 17:44.58 | _mm_ | cool: how many faxes do you process w/ iaxmodem and hyla? |
| 17:44.58 | coolbeans | No we get 95% of our faxes instead of like 80% with just rxfax. |
| 17:45.09 | [TK]D-Fender | Putzz: make an exten that will ansswer the call first |
| 17:45.13 | coolbeans | Hrm... I'd think 50-100 a day |
| 17:45.18 | denon | aptura: check fleabay |
| 17:45.21 | *** join/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
| 17:45.23 | aptura | k |
| 17:45.26 | denon | they're not too bad used |
| 17:45.36 | denon | 5300 or 5400 would work well |
| 17:45.41 | Putzz | [TK]D-Fender: im txfax from asterisk receiving on a normal fax machine |
| 17:45.46 | _mm_ | i'm waiting for my PRIs to come up, but i've got iaxmdoem+asterisk+hylafax all set |
| 17:46.18 | Putzz | how would I answer first then make a .call file. Im confused |
| 17:46.18 | coolbeans | Yep, PRIs will definitely make it easier. |
| 17:46.19 | Putzz | ;-) |
| 17:46.21 | coppice | coolbeans: the spandsp that comes with iaxmodem is the standard one |
| 17:46.31 | _mm_ | hehe, yeah |
| 17:46.36 | aptura | pricy |
| 17:46.43 | Putzz | coppice: I heard u were the man to ask about txfax |
| 17:46.48 | Putzz | hehehe |
| 17:46.51 | coolbeans | coppice: I stand corrected I suppose, I don't claim expertise, just dumping my experiences with hit :) |
| 17:46.52 | blitzrage | coolbeans: SELECT * FROM %s WHERE filename='%s' and commented=0 ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id |
| 17:46.52 | [TK]D-Fender | Putzz: I have just told you what to try. get to it. |
| 17:47.07 | coolbeans | blitzrage: You the man! What file was it in? |
| 17:47.12 | blitzrage | res_config_odbc.c |
| 17:47.19 | coppice | the problem with txfax is something broken in asterisk, so the audio doesn't go out smoothly. |
| 17:47.41 | coppice | for some reason when the audio passes through from iaxmodem it seems to work better |
| 17:48.14 | Putzz | im just going to give up on it and not waste my time and install hylafax and iaxmodem |
| 17:48.33 | [TK]D-Fender | Putzz: I gave you a very quick and direct thing to try.... |
| 17:48.47 | Putzz | Im not sure how fender |
| 17:48.53 | Putzz | ah |
| 17:49.01 | Putzz | context and extension in .call file? |
| 17:49.06 | Putzz | answer |
| 17:49.20 | coolbeans | blitzrage: That shows the sort order, did you by chance see what order the columns need to be in? If not, I can try to dig through it... |
| 17:49.20 | Putzz | then run txfax under program? |
| 17:49.28 | Putzz | is that what u mean? |
| 17:50.18 | [TK]D-Fender | yes |
| 17:50.22 | Putzz | sorry if I seem ignorant I've been searching for a long time |
| 17:50.39 | blitzrage | coolbeans: the sort order would be the order Asterisk expects them to come back in when you do SELECT * |
| 17:50.52 | blitzrage | or, you could just upgrade post-1.4.4 :) |
| 17:51.04 | blitzrage | r62005 seems to work well for me :) |
| 17:51.11 | *** join/#asterisk s3g_fault (i=1000@dump.segv.org) |
| 17:51.24 | jamessan | I'm using * 1.2.16 and it seems that app_voicemail isn't honoring the serveremail option. None of the emails I receive have their From address adjusted according to the option. Is this a known bug? |
| 17:51.32 | coolbeans | Right, the column ordinal values. That particular select statement just shows the column sorting preferences. Or did I miss something? |
| 17:51.55 | coolbeans | blitzrage: lol, it's a high volume production system, I can't risk it ;) |
| 17:51.56 | s3g_fault | potential crash in channel.c:955 |
| 17:52.08 | s3g_fault | vardata is assumed to valid and freed without checking |
| 17:52.27 | coolbeans | The 1.2 system, that is. I tried it on 1.4.4 and thus found the issue which brought me to the channel to ask originally. |
| 17:52.37 | s3g_fault | asterisk 1.2.17 that is |
| 17:53.13 | s3g_fault | anybody? |
| 17:53.23 | mocker | Anyone know any business class SIP trunk providers? |
| 17:53.34 | ManxPower | s3g_fault: I went back to 1.2.15 when I expeirenced crashes in 1.2.17 and 1.2.18 |
| 17:53.36 | coolbeans | moker: Vitelity, Les.Net work well. |
| 17:53.40 | Putzz | [TK]D-Fender: do I put it all in the .call file or should I just dial withing auto dial and make it execute txfax from dialplan? |
| 17:54.08 | mocker | coolbeans: I use Vitelity at home, but I don't know if they're business class.. ;) |
| 17:54.10 | s3g_fault | ManxPower: this exists there too |
| 17:54.17 | mocker | coolbeans: I'll check out Les.Net. |
| 17:54.22 | Dr-Linux | anybody tried web-meetme? |
| 17:54.30 | coolbeans | mocker: What's business class then? |
| 17:54.35 | s3g_fault | there may be something in 1.2.17 that's tickling it, but it's freed without checking |
| 17:54.45 | [TK]D-Fender | Putzz: Call out and dump the call into a dialplan ext once answered. |
| 17:54.59 | s3g_fault | and i've got the stack trace to prove it :( |
| 17:55.04 | Putzz | ok thank you |
| 17:55.26 | Putzz | I will let u know the results |
| 17:55.32 | mocker | coolbeans: Ability to handle high call volume, good/intelligent tech support.. |
| 17:55.41 | coolbeans | mocker: Oh, in that case, none. |
| 17:55.46 | mocker | coolbeans: :( |
| 17:55.47 | coolbeans | mocker: Just get some PRIs. |
| 17:55.54 | mocker | coolbeans: Have PRIs already. :) |
| 17:56.28 | coolbeans | mocker: Vitelity can handle the volume, but their tech support sucks and they don't support re-invites for some reason. Les.Net is great, but geographically too distant for our needs. |
| 18:00.30 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 18:01.51 | Paavum | Hello |
| 18:02.40 | Paavum | I have a GXP2000 |
| 18:02.55 | Paavum | When I call from one phone to the GXP everything works fine |
| 18:03.07 | Paavum | but when I capture the call I only hear one-way audio |
| 18:03.23 | Paavum | and when I do auto answer I get the same behaviour |
| 18:03.27 | Paavum | any ideas? |
| 18:07.21 | *** part/#asterisk s3g_fault (i=1000@dump.segv.org) |
| 18:08.08 | _VoiceMeUp_COM | http://pastebin.ca/483349 |
| 18:08.30 | _VoiceMeUp_COM | can someone see why this number ( modifed ) messes things up |
| 18:08.43 | _VoiceMeUp_COM | its almost alwasy that client that craps it all |
| 18:08.46 | _VoiceMeUp_COM | i think he uses trixbox |
| 18:09.13 | *** join/#asterisk demlak (i=demlak@schwarz-pUnK.de) |
| 18:10.34 | *** part/#asterisk jamessan (n=jamessan@debian/developer/jamessan) |
| 18:11.16 | Paavum | Seems it was a funny codec issue |
| 18:11.34 | _VoiceMeUp_COM | hmmm |
| 18:11.43 | _VoiceMeUp_COM | its from zap/g/1 always |
| 18:11.45 | _VoiceMeUp_COM | cant be codec |
| 18:11.53 | _VoiceMeUp_COM | ah |
| 18:11.54 | _VoiceMeUp_COM | hmm |
| 18:12.16 | _VoiceMeUp_COM | so pribox (ulaw) --> box1 (ulaw) --> clientboz(funkycodec) ? |
| 18:12.44 | _VoiceMeUp_COM | ah you talking about your problem lol |
| 18:13.03 | Paavum | yes :p |
| 18:13.07 | Paavum | sorry what was yours? |
| 18:13.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:13.44 | _VoiceMeUp_COM | http://pastebin.ca/483349 |
| 18:14.41 | Paavum | your trying to call yourself? |
| 18:14.49 | Paavum | I mean what are you trying ta do |
| 18:14.52 | _VoiceMeUp_COM | http://pastebin.ca/483362 |
| 18:14.53 | _VoiceMeUp_COM | hmm |
| 18:14.56 | _VoiceMeUp_COM | check this one |
| 18:14.58 | _VoiceMeUp_COM | that waht it does |
| 18:15.07 | _VoiceMeUp_COM | PRI get s hangup.. but doesnt pass to asterisk |
| 18:15.14 | _VoiceMeUp_COM | so asterisk still ringing |
| 18:15.16 | _VoiceMeUp_COM | is that it ? |
| 18:15.27 | _VoiceMeUp_COM | ill upgrade wanrouter to see |
| 18:19.17 | _VoiceMeUp_COM | so pstn -> asterisk1 --> asterisk2 -> client1 |
| 18:19.49 | _VoiceMeUp_COM | zap is having troubles.. ill upgrade all on that box tonight |
| 18:23.08 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 18:23.37 | DarKnesS_WolF | what function i can get with it the CALLERIDNUM in 1.4 ? |
| 18:23.43 | aptura | what card _Voice |
| 18:24.41 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 18:24.49 | BSD_Tech | he has sangomas |
| 18:25.07 | aptura | interesting. I have always heard thay are generally reliable. |
| 18:25.11 | BSD_Tech | I believe a101 or a102's |
| 18:25.25 | DarKnesS_WolF | got it |
| 18:29.58 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 18:31.28 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 18:35.28 | naitram | anyone know about Originate in the Management Interface |
| 18:35.41 | anonymouz666 | voip-info AMI originate |
| 18:35.43 | anonymouz666 | google for that |
| 18:35.48 | anonymouz666 | it will help you |
| 18:39.15 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:42.12 | coolbeans | Just to settle this once and for all, is there a space in caller ids between the text and <numbers>? i.e., "John Doe"<2225551234> or is is "John Doe" <2225551234> ?? |
| 18:42.12 | naitram | anonymouz666: yeah looked at a lot of that. Trouble is, cant seem to make one client call the other using it. tried: Channel: Sip/MAST, Context: MAST200, Exten: 5000, priority.... But I don't get A call from the device MAST to exten 5000. I get 5000 displayed on my MAST phone and no call to extension 5000. Its like asterisk is telling the MAST device that it has a call from exten 5000 but it... |
| 18:42.13 | naitram | ...doesn't do anything to connect the phones |
| 18:44.03 | Strom_M | coolbeans: callerid="Bell System"<3115552368> |
| 18:46.45 | *** join/#asterisk groogs (n=gregmac@d38-54-164.commercial1.cgocable.net) |
| 18:47.58 | *** join/#asterisk Math` (n=seb@modemcable037.229-56-74.mc.videotron.ca) |
| 18:48.16 | Math` | infinite loops calling chan_local is a pretty bad thing(tm) |
| 18:48.24 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 18:50.35 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:51.29 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 18:56.29 | coolbeans | For logging to mysql, on a high-volume system, would you guys set the my.cnf nice value to a lower priority? It's 0 by default. |
| 18:56.51 | coolbeans | high volume asterisk system, cdr logging. Maybe static config. |
| 18:56.59 | coolbeans | Strom_M: Thanks. |
| 18:57.10 | *** join/#asterisk DrukenHME (n=jdumais@CPE0040f4f9df11-CM0011aea53730.cpe.net.cable.rogers.com) |
| 18:59.00 | *** join/#asterisk uppal (n=uppal@6.157.70.202.dynamic.max.com.pk) |
| 18:59.02 | uppal | pal |
| 18:59.04 | uppal | hi |
| 18:59.29 | [TK]D-Fender | naitram: Pastebin the entire message transmitted, and the CLI output of the attempt, as well your dialplan where applicable |
| 19:00.06 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:00.19 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 19:00.34 | *** join/#asterisk Alan_Hicks (n=alan@208.62.162.112) |
| 19:01.30 | *** join/#asterisk nate12o6 (n=n@216.40.230.5) |
| 19:01.36 | nate12o6 | hey i have a quick voip question i hope someone can answer for me. |
| 19:01.41 | nate12o6 | i see alot of stuff about gsm gateways and such |
| 19:01.48 | nate12o6 | is it possible to use an unlocked cell phone that instead of connecting to cingular servers it will connect to my server and i can control the call? |
| 19:02.17 | Katty | [TK]D-Fender: mew? |
| 19:02.31 | *** join/#asterisk zuez (i=steve@66.103.132.86) |
| 19:02.33 | [TK]D-Fender | Katty: Mew. |
| 19:04.17 | tonycarstens | i'm having problems with the voicemail app |
| 19:04.24 | tonycarstens | it cant read username |
| 19:04.34 | Katty | [TK]D-Fender: so, if i have a working mount command. aka, mount server share to this folder... where do i put that mount command so that when the machine reboots, it will remount that share on startup |
| 19:04.38 | Math` | tonycarstens: check how your dtmf are passed |
| 19:04.48 | tonycarstens | how should they? |
| 19:04.58 | Math` | depends on your settings |
| 19:05.01 | Math` | rfc2833 is a nice way |
| 19:05.15 | pipwerk | Katty: edit /etc/fstab |
| 19:05.25 | Math` | and mark it as auto in that file |
| 19:06.07 | Katty | thanks. |
| 19:06.21 | [TK]D-Fender | Katty: pipwerk 's idea sounds like the proper way. |
| 19:06.39 | [TK]D-Fender | Katty: if it was not service critical I might do it in rc.local or something.... |
| 19:07.03 | Katty | no, it's not service critical. |
| 19:07.09 | Katty | just nice to have if my machine reboots. |
| 19:07.25 | Katty | but, also, let's say this asterisk server reboots |
| 19:07.34 | Katty | and i don't want to do all the modprobe stuff and relaunch asterisk, etc |
| 19:07.40 | Katty | where do i automate that at? |
| 19:07.43 | afrosheen | what pieces need to be in place for a DID to route to an extension |
| 19:07.54 | Mercestes | rc |
| 19:08.20 | Katty | which rc.local? |
| 19:08.31 | Katty | i have...uhh...6 of them. |
| 19:08.34 | tonycarstens | math: thaanks |
| 19:08.36 | Mercestes | I guess. |
| 19:08.45 | Mercestes | I do rc-update add asterisk default myself |
| 19:09.02 | Katty | slow down there panda bear. |
| 19:09.09 | Katty | i've never had one look at rc.local before :P |
| 19:09.22 | Mercestes | I do that on the command line. |
| 19:09.33 | Katty | yes, i'm looking at it now. |
| 19:09.41 | Katty | but, i'm still not sure which one to edit. |
| 19:09.46 | Mercestes | modprobe is under /etc/modules.autoload.d/kernel.2.6 or something like that |
| 19:10.11 | Katty | [TK]D-Fender: do i need to edit the rc.local file that coresponds to my rc level? |
| 19:10.15 | Katty | or the one in /etc/ |
| 19:10.25 | Katty | or the one in /etc/init.d |
| 19:10.30 | [TK]D-Fender | Katty: well You are now asking a few different queswtion. |
| 19:10.39 | Katty | right. |
| 19:10.48 | Katty | for now, i just wanna mount these shares when the machine starts |
| 19:10.51 | [TK]D-Fender | I'm not sure how Debian handles the init process VS RH & Slackware. |
| 19:11.00 | Katty | i have a working mount command. |
| 19:11.07 | [TK]D-Fender | Someone else might be better to advise from here |
| 19:11.24 | Katty | kay |
| 19:11.29 | Math` | is it possible to allow registration from a specific ip only? (since host= is going to decline registration and setting it dynamic allows it from everywhere) |
| 19:12.42 | Alan_Hicks | I am brand-new to the world of VOIP and Asterisk, but I just had a client ask me about it. They have an ancient Merlin phone system that need to replaced within the next year, and they were wondering what Linux and Asterisk can do for them. I'm considering building them an Asterisk server once I get up to snuff on everything, but I saw on Digium's site that they sell an "Asterisk Appliance". Does anyone here have an opinion about the quality of that |
| 19:13.10 | bkruse | ~thebook |
| 19:13.12 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:13.14 | bkruse | ~wiki |
| 19:13.17 | Mercestes | ~docs |
| 19:13.18 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 19:13.18 | bkruse | or is it the wiki... |
| 19:13.26 | bkruse | thanks Mercestes |
| 19:13.29 | Katty | Alan_Hicks: i've always had a /real/ server, never an appliance, so i can't help you out there. |
| 19:13.30 | Mercestes | :) |
| 19:13.40 | Alan_Hicks | Client has approximately 35 extensions and I'm guessing half-a-dozen POTS lines in a hunt group. |
| 19:13.54 | Katty | Alan_Hicks: i can say that a real asterisk server is pretty awesome cause it can excute bash commands too... |
| 19:14.04 | Katty | Alan_Hicks: which paves the way for things like IMs and stuff |
| 19:14.12 | Alan_Hicks | Katty: Thanks. I would rather build a /real/ server myself (if only for the fun of learning Asterisk), but if it would be ready to go out of the box, it might be a better fit for my client. |
| 19:14.14 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 19:14.46 | Katty | Alan_Hicks: that's what we have here, 12 lines and about 30 polycom phones. |
| 19:15.02 | Katty | Alan_Hicks: yeah, i understand that. i had to learn linux and asterisk at the same time. luckily, there's extensive documentation, sample files, and tons of people here that will help. |
| 19:15.12 | Alan_Hicks | I'm not entirely up to speed on the terminology (haven't RTFM'd yet), so I hope I don't embarass myself here too bad. :^) |
| 19:15.25 | Katty | can't do worse than me :P |
| 19:15.31 | Katty | i'm sitting here asking about general linux commands. |
| 19:15.35 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 19:15.55 | Mercestes | Alan_Hicks, there is so such thing as "ready out of the box" then it comes to PBXs |
| 19:15.57 | Alan_Hicks | I'm thinking that I need an FXO (not sure if that's the right thing) card or two so I can use their existing digital phones and not re-pull cable for newer IP phones. |
| 19:16.41 | Mercestes | Alan_Hicks, I doubt that will work. FXO is for POTS not digital PBX turnkey phones |
| 19:16.46 | Katty | Alan_Hicks: voip-supply.com has an array of pci cards for analog lines. ask for a guy named Joe. |
| 19:16.52 | Mercestes | Alan_Hicks, try hooking a buttset to one of those things and see what happens. |
| 19:16.55 | Katty | Alan_Hicks: i can get you his extension, if you want it. he's pretty darn helpful |
| 19:16.58 | Alan_Hicks | Mercestes: Well, there's no such thing as "ready out of the box" anything really, but if it was good stuff, it would be a jumping-off point. |
| 19:17.16 | Mercestes | IT's not a good jumping-off point....unless you meant bridge. |
| 19:17.18 | [TK]D-Fender | Alan_Hicks: I don't believe your handsets are reusable |
| 19:17.19 | Alan_Hicks | Mercestes: Thanks. Like I said, I _really_ don't know the terms right yet, but I will as soon as I start reading the book this weekend. |
| 19:17.48 | Katty | Alan_Hicks: do you know much about linux yet? |
| 19:17.55 | Alan_Hicks | Mercestes: Thanks for the advice. I'll stay away from it then. |
| 19:18.08 | Alan_Hicks | Katty: Sure. http://www.slackbook.org/ ;-) |
| 19:18.14 | [TK]D-Fender | Alan_Hicks: And as for what * can do, you can run it as a full service PBX with call queues, voicemail, interacting with all sort of lines & phones. The choice of which and how is up to your |
| 19:18.22 | Katty | Alan_Hicks: that's half the battle right there :P |
| 19:18.43 | Mercestes | I run my asterisk on a linksys router. |
| 19:19.00 | Katty | Alan_Hicks: it also has a bunch of neat added things people have written. We use an apache/flash thingy here that shows our receptionist what lines are in use, who's on the phone etc (Flash operator panel) |
| 19:19.09 | Alan_Hicks | Really appreciate all your help. I bought the O'Reilly book on * awhile back, but haven't yet had a chance to crack into it. Is this a good book to start with, or do y'all recommend something else? |
| 19:20.31 | Mercestes | It's a great book to start with. |
| 19:20.31 | Mercestes | I would also check the wiki |
| 19:20.31 | Alan_Hicks | Thanks. I'll get right on it this weekend. |
| 19:20.32 | Mercestes | see you Monday. |
| 19:20.32 | Alan_Hicks | hehe |
| 19:20.32 | [TK]D-Fender | Alan_Hicks: And you would be looking at a new cable pull. |
| 19:20.32 | Alan_Hicks | [TK]D-Fender: That's fine. More money for me. ;-) |
| 19:20.32 | Katty | Alan_Hicks: when i started, i really liked the asterisk quick start guides.. |
| 19:20.35 | Mercestes | But, what's wrong with running ethernet over 45 year old cat-3 straight through? It has four wires! that's all Ethernet needs |
| 19:20.35 | Alan_Hicks | Actually, I think their phones were pulled with Cat-5 so it's possible I could just re-terminate. |
| 19:20.56 | Alan_Hicks | Mercestes: Yeah, but it's Cat-3 non-twisted pair. |
| 19:21.08 | Mercestes | Alan_Hicks, So? >.> |
| 19:21.13 | Mercestes | what's a little echo among friends. |
| 19:21.16 | Alan_Hicks | It'll probably run at 10Mb. |
| 19:21.16 | Qwell[] | it'll *work* :P |
| 19:21.21 | Qwell[] | Alan_Hicks: not even |
| 19:21.36 | Alan_Hicks | I've gotten ethernet at 10Mb on cat-3 before. |
| 19:21.47 | Mercestes | not over any real distance |
| 19:21.58 | Katty | Alan_Hicks: http://www.asteriskguru.com/tutorials/ <- i found that place the most helpful of all, second to [TK]D-Fender |
| 19:21.59 | Alan_Hicks | But that was a mistake, and was a short distance in a "quiet" office. |
| 19:22.20 | Alan_Hicks | Katty: TY. |
| 19:22.35 | Mercestes | and me!!!! |
| 19:22.38 | Mercestes | right? |
| 19:22.45 | Alan_Hicks | Right. |
| 19:22.45 | Katty | yes, and Mercestes, of course. |
| 19:22.47 | Katty | and Hmmhesays and file |
| 19:22.49 | Mercestes | yay |
| 19:22.52 | Katty | and half a dozen other awesome people here :P |
| 19:23.12 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 19:23.29 | Katty | i think i might install printers now. |
| 19:23.41 | Katty | tho i can't imagine what i would print on this server |
| 19:24.50 | *** join/#asterisk lunk (n=lunk@cpe-071-068-044-254.carolina.res.rr.com) |
| 19:24.57 | Alan_Hicks | You could store the digital copy of different phone calls, convert to ASCII, print, scan with OCR, gpg encrypt it, and use that for entropy. |
| 19:25.01 | Katty | Alan_Hicks: btw, Joe's ext is 3873. |
| 19:25.06 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 19:25.07 | Alan_Hicks | thanks |
| 19:25.18 | lunk | is there a way to WRITE a running extensions configuration to disk? |
| 19:25.24 | Katty | Alan_Hicks: i think voip supply is the easiest vendor to work with (= |
| 19:26.35 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 19:26.39 | Katty | [TK]D-Fender: if i install sql on this machine, and make asterisk dumps its call log it it so i can query it from another webserver page thingy... |
| 19:26.46 | Katty | [TK]D-Fender: how do i, uhh, backup that database... |
| 19:26.48 | aptura | attacom is good. Fast delivery. |
| 19:26.52 | Katty | [TK]D-Fender: is it a single file, or a folder...or... |
| 19:27.02 | lunk | omg, w00t, save dialplan |
| 19:27.15 | Katty | [TK]D-Fender: is there an automated process of some type for that in case our build gets taken over by enraged aliens. |
| 19:27.33 | Alan_Hicks | Wouldn't that just be a normal sql database? If using MySQL, you could just use mysqldump on that database, right? |
| 19:27.50 | Katty | let's just say i don't know a /thing/ about sql |
| 19:27.56 | Katty | tho i know a billion things about ms access. |
| 19:28.42 | iCEBrkr | mysql_backup.sh |
| 19:28.44 | Alan_Hicks | Well, assuming you're running MySQL, you would just create another database in it with tables setup as needed, and you could back that up with MySQL's integrated tools. |
| 19:28.44 | iCEBrkr | :P |
| 19:29.05 | Katty | i want the backup off the server tho. |
| 19:29.17 | Katty | in case.... it decides to eat itself. |
| 19:29.19 | Katty | and sizzle. |
| 19:29.33 | afrosheen | Katty, and shine |
| 19:29.45 | Katty | yes'r. |
| 19:30.05 | iCEBrkr | Katty: you know anything about bash scripting? |
| 19:30.08 | iCEBrkr | Katty: get that DCC |
| 19:30.17 | Katty | iCEBrkr: no, i think i have everything blocked. |
| 19:30.32 | iCEBrkr | grrr |
| 19:30.57 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/files/mysql_backup.sh |
| 19:31.06 | Katty | oh ah |
| 19:32.42 | iCEBrkr | That should be a good start.. |
| 19:32.43 | iCEBrkr | I think |
| 19:33.32 | Katty | i'm attempting to download it. |
| 19:33.39 | Katty | it seems to not love me tho. |
| 19:34.49 | iCEBrkr | hrrm |
| 19:34.50 | iCEBrkr | hang |
| 19:35.24 | Katty | k |
| 19:35.31 | iCEBrkr | http://www.cyberdyne.org/~icebrkr/files/mysql_backup.sh.gz |
| 19:35.34 | iCEBrkr | but it's still stalled. |
| 19:35.35 | iCEBrkr | hrrm |
| 19:36.08 | Katty | <PROTECTED> |
| 19:36.12 | iCEBrkr | Frick'n nagios |
| 19:36.15 | iCEBrkr | Try again |
| 19:36.17 | iCEBrkr | it's back |
| 19:38.04 | iCEBrkr | work? |
| 19:38.16 | *** join/#asterisk boomerang (n=boomeran@unaffiliated/boomerang) |
| 19:39.39 | hacim | anyone know where I can get good deals on DID origination for brasil? |
| 19:39.50 | hacim | i can't seem to find anything |
| 19:41.44 | Math` | fucking grandstream's new firmware doesnt support md5 challenges |
| 19:43.33 | Katty | iCEBrkr: yes'r, thanks. |
| 19:44.36 | *** join/#asterisk Goodjoke (n=Goodjoke@rrcs-24-97-65-74.nys.biz.rr.com) |
| 19:44.39 | iCEBrkr | cool |
| 19:44.46 | iCEBrkr | Katty: You'll have to edit the script a bit |
| 19:44.51 | iCEBrkr | Katty: user/pass type stuff |
| 19:44.53 | Katty | all good. |
| 19:45.10 | iCEBrkr | Katty: oh and some directory paths, etc. It should be fairly self-explanatory |
| 19:45.11 | iCEBrkr | :P |
| 19:45.33 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 19:46.05 | Katty | mm, documentation |
| 19:46.29 | Blackthorn | Hello... "unable to support trunking for user 'remote' without zaptel timing" is this message to be expected if thers no pri/t1 hooked up yet? |
| 19:46.31 | Katty | i've been documenting this server... i'm up to 19 pages. |
| 19:46.48 | shido6 | blackthorn |
| 19:48.21 | Blackthorn | hi shido |
| 19:48.27 | Blackthorn | i'm back :) |
| 19:48.37 | shido6 | u went to sleep too early |
| 19:48.46 | shido6 | you had knodded off |
| 19:48.50 | *** join/#asterisk hansin321 (n=eric@c-71-196-138-170.hsd1.co.comcast.net) |
| 19:48.51 | shido6 | and signed off |
| 19:49.04 | Blackthorn | yea i was dead at the keyboard last night.. it was after midnight and had to get up at 6 this morning. |
| 19:49.29 | shido6 | i called over there... |
| 19:49.33 | coolbeans | SPOOOON! |
| 19:49.34 | shido6 | left a message somewhere |
| 19:49.36 | shido6 | did you get it? |
| 19:49.39 | Blackthorn | e-mail? |
| 19:49.43 | shido6 | voice |
| 19:49.51 | Blackthorn | nope |
| 19:50.47 | Blackthorn | i called degium and got my t1 card isntalled, problem was that it was a very old card. which i knew it would be since it's my backup card to the one that is currently online. |
| 19:51.17 | Blackthorn | now just need to get that iax working... |
| 19:51.42 | shido6 | call me and I'll walk you through it |
| 19:52.45 | gmfm | anyone have a copy of the Telemarketer Torture sound files for http://www.voip-info.org/wiki/view/Asterisk+Telemarketer+Torture ? I would be willing to host a mirror of it so that the download is available |
| 19:54.23 | *** join/#asterisk `Sean (i=Un1x@CPE000c248d137c-CM00111ae601f8.cpe.net.cable.rogers.com) |
| 19:54.39 | iCEBrkr | gmfm: I thought those files were in the default asterisk-sounds archive? |
| 19:54.44 | iCEBrkr | tt-*.gsm |
| 19:57.02 | gmfm | iCEBrkr: i just have a couple... monkeys and such. The torture script has some lengthy stuff about charities and political parties which i might just recreate with tts |
| 19:59.34 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
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| 20:06.18 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.198.244) |
| 20:06.55 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 20:08.10 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 20:08.17 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-173-168-178.socal.res.rr.com) |
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| 20:11.00 | *** join/#asterisk evilcyrus (n=evilcyru@bas3-hamilton14-1096561391.dsl.bell.ca) |
| 20:11.02 | Katty | anyone framilier with asterisk desktop manaer? is it any good? |
| 20:12.53 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:13.19 | BSD_Tech | ? |
| 20:13.27 | BSD_Tech | asterisk desktop manager |
| 20:13.37 | BSD_Tech | never even new there was one |
| 20:13.41 | BSD_Tech | url |
| 20:13.46 | Katty | http://adm.hamnett.org/ |
| 20:13.59 | Katty | i'm playin with it now |
| 20:14.31 | Mercestes | so I can manage my desktop from my phone?? |
| 20:16.08 | sevard | I think it's the other way around. |
| 20:16.36 | *** part/#asterisk lunk (n=lunk@cpe-071-068-044-254.carolina.res.rr.com) |
| 20:17.01 | thinwires | ok, any idea's as to why my asterisk is telling me all lines are congested? nothing changed it just stopped working |
| 20:17.28 | luke-jr_work | maybe they are :) |
| 20:17.35 | Mercestes | add more lines |
| 20:17.48 | sevard | you have monkeys in your tubes |
| 20:17.49 | evilcyrus | i like the Wi-fi phone thing the best |
| 20:17.53 | thinwires | but they arent, I have two providers and noe one is using the phones at all |
| 20:17.53 | DrukenLPY | use thicker wires... :) |
| 20:17.54 | evilcyrus | thats sweet |
| 20:18.05 | evilcyrus | <---new |
| 20:18.10 | BSD_Tech | wifi phone thing ? |
| 20:18.24 | evilcyrus | i'm watchn the video |
| 20:18.29 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 20:18.41 | BSD_Tech | wifi phone thing ? |
| 20:18.44 | evilcyrus | systm |
| 20:18.47 | BSD_Tech | explain |
| 20:18.51 | BSD_Tech | english |
| 20:19.00 | evilcyrus | like a cell phone |
| 20:19.06 | evilcyrus | but wi-fi |
| 20:19.32 | BSD_Tech | ok well gsm phones and a gsm gatway is the best |
| 20:19.33 | DrukenLPY | i have a wifi phone... usually works good |
| 20:19.37 | thinwires | how do I adjust the verbosity of the CLI? |
| 20:19.39 | DrukenLPY | but my AP is a peice of shit... |
| 20:19.48 | BSD_Tech | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvgr |
| 20:20.26 | evilcyrus | this stuff is cool |
| 20:20.30 | evilcyrus | jsut getting into it |
| 20:21.55 | BSD_Tech | voip has issues |
| 20:22.05 | evilcyrus | so... does landlines |
| 20:22.09 | BSD_Tech | some day they will fix these issues and voip will rule |
| 20:22.17 | evilcyrus | canada Rogers sux |
| 20:22.25 | DrukenLPY | ? |
| 20:22.31 | DrukenLPY | what part of rogers? |
| 20:22.38 | evilcyrus | Roges in canada sucks ....and Bell canada too |
| 20:22.54 | evilcyrus | service sucks and too high of a price |
| 20:23.10 | DrukenLPY | hehe |
| 20:23.24 | evilcyrus | i talked to them and they said Voip is gettin large |
| 20:23.31 | evilcyrus | and there losing alot of customers |
| 20:23.31 | DrukenLPY | well, luckily.. my rogers service in barrie, is better than the rogers service in hamilton :) |
| 20:23.51 | evilcyrus | Rogers Homephone sucks |
| 20:23.58 | evilcyrus | cell phones suck |
| 20:24.05 | evilcyrus | cable is crap. |
| 20:24.16 | DrukenLPY | cable internet is the only service i use |
| 20:24.22 | evilcyrus | me too |
| 20:24.25 | DrukenLPY | i fully agree with the other comments |
| 20:24.27 | evilcyrus | small cable company |
| 20:24.38 | evilcyrus | Mountaincable |
| 20:24.40 | coolbeans | asterisk static realtime rocks. |
| 20:24.44 | evilcyrus | wicked service |
| 20:24.48 | evilcyrus | wicked speeds |
| 20:25.06 | evilcyrus | they offer intronet and phone 80 bucks |
| 20:25.08 | evilcyrus | a month |
| 20:25.15 | DrukenLPY | but limited availability |
| 20:25.34 | evilcyrus | Rogers just fukd up my home phone |
| 20:25.43 | evilcyrus | due to them switchn accounts and stuff |
| 20:25.46 | evilcyrus | 3 days now |
| 20:25.48 | evilcyrus | no phone |
| 20:25.55 | DrukenLPY | :) |
| 20:26.04 | *** join/#asterisk angom (n=angom@red-corp-201.143.54.251.telnor.net) |
| 20:26.05 | evilcyrus | yah i'm pissed and switched |
| 20:26.10 | evilcyrus | had engough |
| 20:26.12 | DrukenLPY | i do my own phone.. so if it breaks, it's usually my own damn fault |
| 20:26.26 | evilcyrus | ahah i'm getting into voip |
| 20:26.31 | coolbeans | Hey Qwell[]: You guys run Linux on the desktop at Digium or Windows? |
| 20:26.31 | evilcyrus | my buddies all have it |
| 20:26.37 | evilcyrus | and work for mountancable |
| 20:26.48 | Qwell[] | coolbeans: why would we run Windows? |
| 20:27.01 | Mercestes | evilcyrus, I think you cr more than anything else. |
| 20:27.14 | evilcyrus | huh |
| 20:27.22 | coolbeans | I meant at the desktop (i.e., accounting dept, etc, etc) |
| 20:27.31 | thinwires | ok, now my phone just started working for no reason.... langangkjnaegjanhk |
| 20:27.36 | evilcyrus | i just want service with a smile |
| 20:27.37 | Qwell[] | coolbeans: I'd say probably more than 80% of the company runs Linux |
| 20:27.47 | glogic | anyone know a way to have multiple prompts play at pre-determined times during a call? the L() command doesn't really do a whole lot that's useful if you want to play multiple prompts and keep the call bridged |
| 20:27.50 | evilcyrus | if i pay 46/m for home phone i want home phone service |
| 20:27.56 | coolbeans | Cool, just curious. |
| 20:28.01 | evilcyrus | and i'm not getting it |
| 20:28.06 | *** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net) |
| 20:30.07 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 20:32.35 | *** join/#asterisk rpetre (n=petre@83.166.220.142) |
| 20:33.29 | glogic | hrrm this doesn't look reall possible |
| 20:34.38 | rpetre | it might be a stupid question, but where could i find some example (and complex) dialplans to learn to organize mine nicely? i know enough to make stuff work, but it always looks uglier than my first firewalls :) |
| 20:36.32 | thinwires | rpetre: what dialplan are you trying to write? your inbound call? |
| 20:37.20 | redax | anybody knows how to configure the S0 bus length with mISDN ? |
| 20:40.09 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:40.10 | rpetre | thinwires: i'm not sure i understand the question. i currently have a working dialplan (incoming and outgoing), but i want to improve stuff in the future some more, and it currently looks like hell, and i pity the guy that will have to maintain it (especially if it's me) |
| 20:40.42 | *** join/#asterisk furibondox (n=linux_us@host-84-223-109-155.cust-adsl.tiscali.it) |
| 20:40.48 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 20:40.50 | rpetre | so i just want to look how more experienced people organize their rules |
| 20:41.55 | thinwires | right, I was asking if your looking for Dial Plan or Calling rules... |
| 20:42.30 | thinwires | http://pastebin.ca/483176 that is a link of my calling rules... but then agai I'm not sure how your setup is... |
| 20:43.16 | Strom_M | catsex |
| 20:43.33 | Strom_M | does asterisk support b-channel transfers on ISDN PRI? |
| 20:44.25 | rpetre | i have 10 numbers from my sip provider and some sip phones (soft and hard). i intend to do fun stuff based on callerid and hours and whatever else i may think of :) |
| 20:44.35 | KuJaX | How can I setup an extension where if someone dials it during the IVR that it will ring into my EXTERNAL CELL PHONE? |
| 20:45.18 | Strom_M | KuJaX: the same way you place outbound calls from internal stations |
| 20:45.29 | thinwires | exten = 1,2,Dial(IAX2/trunk_1/18666119434) for me that dials into my External 1866 number for tech support |
| 20:45.38 | Strom_M | Dial(ZAP/G1/3115552368) or whatever |
| 20:46.18 | KuJaX | i've tried that but it only works part of the time |
| 20:46.24 | KuJaX | sometimes it doesn't ring my cell. (lol) |
| 20:46.58 | afrosheen | maybe your cell provider sucks |
| 20:47.41 | Katty | how do i make asterisk manually parse the manager.conf file? |
| 20:47.58 | thinwires | KuJax: have you monitored the CLI when it fails to dial your cell phone? I'd start there |
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| 20:59.14 | demlak | anyone know a standalone MTA which is very smal? something like sendmail.. for embedded devices... "esmtp" is smal.. but needs a 1mb big lib =( |
| 21:01.40 | galeras | sometimes, moh is stopped on client side many seconds before connection with the agent. Any idea how can i prevent this? (all agents are available via agentlogin) |
| 21:04.39 | *** join/#asterisk Mavvie (n=edwin@ppp39-68.lns3.syd7.internode.on.net) |
| 21:07.08 | *** part/#asterisk boomerang (n=boomeran@unaffiliated/boomerang) |
| 21:07.27 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-065-125.pools.arcor-ip.net) |
| 21:09.29 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:10.09 | galeras | :-$ |
| 21:11.17 | aptura | strange. heard a click in the middle of a call and some line noise was injected into the call and some quiet echo was introduced. |
| 21:14.53 | *** join/#asterisk BSD_Tech (n=bsdtech@adsl-69-230-166-20.dsl.irvnca.pacbell.net) |
| 21:18.58 | galeras | in my case, moh stopped on zap chan is reported by cli |
| 21:18.58 | galeras | and many seconds after, call is linked to the agent |
| 21:21.39 | *** part/#asterisk furibondox (n=linux_us@host-84-223-109-155.cust-adsl.tiscali.it) |
| 21:31.57 | galeras | <PROTECTED> |
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| 21:55.30 | Strom_M | Is there a trick to getting the TC400 working under 1.4 apart from compiling the necessary firmware and kernel driver? |
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| 22:00.39 | *** join/#asterisk Mavvie (n=edwin@ppp7-107.lns4.syd7.internode.on.net) |
| 22:01.10 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:01.42 | *** join/#asterisk zapp-branigan (n=zapp-bra@81.202.214.78.dyn.user.ono.com) |
| 22:03.22 | hansin321 | Can I comment out (#) the modules I don't need in /etc/modprobe.d/zaptel? |
| 22:04.11 | aptura | sure |
| 22:06.40 | hansin321 | aptura: Thanks. I figured so, but just wanted to be sure. |
| 22:10.38 | xpot | can someone help w/ voicemail? |
| 22:11.31 | Strom_M | xpot: just ask you question |
| 22:11.58 | Strom_M | s/you/your/ |
| 22:13.19 | xpot | rgr, I have users set up in SQL and instead of creating a huge voicemail.conf I would like to do the following: ${EXTEN} => ${PASS},${FNAME} ${LNAME},${EMAIL}} --> is this possible? |
| 22:13.48 | Math` | xpot: you can use realtime and put your voicemails in the db too.... |
| 22:14.13 | xpot | Math: is there documetion you can point me to? |
| 22:14.24 | xpot | *documentation |
| 22:14.30 | Math` | sure http://www.google.ca/url?sa=t&ct=res&cd=2&url=http%3A%2F%2Fvoip-info.org%2Fwiki%2Fview%2FAsterisk%2BRealTime%2BVoicemail&ei=tepERpW2D6imgAKO7entDA&usg=AFrqEzeRjE0v2MCiseo-bYOmy7ZeIQUAuA&sig2=1Tjr5-i8fieATw5DMm_jmg |
| 22:14.37 | xpot | thanks |
| 22:15.38 | CVirus | Is the grandstream or the sipura more recommended for production usage ? |
| 22:16.30 | Math` | I'd use sipura |
| 22:16.36 | CVirus | hmm |
| 22:16.42 | [TK]D-Fender | ~phones |
| 22:16.44 | jbot | i heard phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/auggestibility: Polycom (any), SNOM, Aastra 480i, Linksys SPA-9XX, Grandstream, Cisco. |
| 22:16.45 | [TK]D-Fender | ^^^^ |
| 22:18.58 | [TK]D-Fender | ~gs |
| 22:19.02 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 22:19.16 | [TK]D-Fender | ~phones |
| 22:19.19 | jbot | it has been said that phones is http://bani.anime.net/phones/. SIP Hardphones in order of quality/auggestibility: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, (Everything else), and finally ... Grandstream. Type ~gs for more on the latter... |
| 22:19.19 | galeras | grandstream was a disaster for me, we are now using Polycom |
| 22:19.31 | [TK]D-Fender | There we go.... |
| 22:20.07 | Mercestes | I think it should be Aastra then polycom |
| 22:20.24 | Math` | I just had a bug with grandstream's gxp2000 |
| 22:20.40 | Math` | a version of their firmware doesnt work with md5 challenge auth in sip register |
| 22:20.51 | Mercestes | mostly because POlycom customer service is non-existant. If it did exist it would suck |
| 22:20.52 | galeras | forget grandstream |
| 22:21.42 | [TK]D-Fender | Mercestes, Polycom has superior call handling, audio quality, and general feel. Aastra has better soft-keys, attendant options, enhanced presence, and cheaper backlight. |
| 22:21.44 | *** join/#asterisk nhudson (n=nhudson@68.113.120.148) |
| 22:22.04 | galeras | Polycom!!! |
| 22:22.13 | Mercestes | Aastra! |
| 22:22.29 | [TK]D-Fender | Mercestes, I had an IP 600 as my desk phone at work, and chanced an "upgrade" to a Aastra 57i CT. |
| 22:22.29 | galeras | ok, buy 50-50 |
| 22:23.03 | [TK]D-Fender | Mercestes, I wanted the wireless handset to just ring for its own reg and can't stop the base from ringing / hijcking calls. |
| 22:23.24 | [TK]D-Fender | Mercestes, So now MY phone rings every time our warehouse shipping manager gets a call. |
| 22:23.26 | Mercestes | And the polycock is compatible with so many wireless headsets |
| 22:23.30 | [TK]D-Fender | Mercestes, UBER piss-off. |
| 22:23.37 | hansin321 | I am downloading source and compiling. I just downloaded to /usr/local/src and untarred and did a ./configure | make | make install. I had to do all as root. I know usually you should only need to be root for "make install". Is it smarter to do the compiling in my home directory, so I don't need to be root? Any other suggested locations? I'm a little new to the compliling thing. thanks. |
| 22:24.01 | [TK]D-Fender | Mercestes, Also no way to merge independent calls for loca conference, the speakerphone is ... *meh*, and the handset has NO weight |
| 22:24.30 | [TK]D-Fender | hansin321, /usr/src is the usual place |
| 22:24.51 | [TK]D-Fender | Mercestes, the handset cord pulls at it too much |
| 22:25.07 | [TK]D-Fender | Mercestes, I feel rather betrayed by my hopes for it |
| 22:25.45 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 22:26.25 | Mercestes | my one and only point is that Polycom will eventaully get pwned by a company that isn't staffed by a bunch of bitches, that is all. |
| 22:26.58 | Mercestes | of course if my supplier wasn't retarded I wouldn't have to deal with polycom so....I guess someone will pwn them too |
| 22:27.04 | hansin321 | [TK]D-Fender: thanks. looks like I may need to add myself to the src group, and I will have write priveleges to that folder. or something like that, but I will figure it out. |
| 22:29.33 | [TK]D-Fender | Mercestes, I've been in touch with Polycom direct on conference through my vendor once. They know shit about their VoIP line but do their job in bridging me for support (which I only needed once) |
| 22:30.43 | Mercestes | yea, I got a nice voIP brick on my desk that' sbeen here for 3 weeks |
| 22:30.48 | Mercestes | anywauys I'm out. lates |
| 22:40.50 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 22:51.33 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:52.26 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:53.20 | kn0x | okay, I'm trying to build a session border controller here |
| 22:54.37 | kn0x | it'll need to act as a UA server (radius auth.), do LCR, terminate to a SIP trunk based on LCR, and then give radius the accounting info.... all of this is not to challenging in Asterisk... |
| 22:54.54 | kn0x | what I'm curious is if Asterisk could handle, say 700 concurrent calls |
| 22:55.04 | kn0x | no transcoding |
| 22:55.19 | kn0x | just a B2BUA relaying the RTP... |
| 22:56.06 | kn0x | I'm planning on running it on a dual core xeon with 4gb of ram... |
| 22:57.11 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 22:57.28 | mvanbaak | hhmm |
| 22:57.38 | mvanbaak | best to test it with sipp or something |
| 22:58.39 | mvanbaak | if there's no real dialplan involved you can have a look at openser |
| 23:01.52 | *** join/#asterisk ToyMan (n=Stuart@74-32-9-93.dsl1.mdl.ny.frontiernet.net) |
| 23:05.52 | *** join/#asterisk billasterisknewb (n=bill4242@66.60.191.180) |
| 23:06.59 | billasterisknewb | hello.. Got a question just got Asterisk installed in VMWare, I'm trying to use a digium Zaptel TDM800P FXO/FXS card but the installation doesn't seem to detect it.. are there drivers available for a VM installation? |
| 23:08.28 | Qwell[] | billasterisknewb: no... |
| 23:08.40 | Qwell[] | you can't use pci hardware in vmware.. |
| 23:08.47 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 23:08.48 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 23:09.06 | billasterisknewb | oh, well i was not aware of that. bummer.. |
| 23:09.10 | evilcyrus | are linksys Adpters SIP ... to use Old Analog phones |
| 23:09.18 | mvanbaak | Qwell[]: thanks for the chan_skinny commit |
| 23:09.23 | Qwell[] | mvanbaak: mmhmm |
| 23:10.40 | mvanbaak | I really can say now: chan_skinny > chan_sccp |
| 23:10.52 | mvanbaak | feature wise |
| 23:11.16 | Qwell[] | and stability :p |
| 23:11.21 | Qwell[] | chan_sccp is crap in that regard |
| 23:11.22 | mvanbaak | uhhuh |
| 23:11.44 | mvanbaak | the 2 patches that sergio never accepted did make it work a _bit_ better |
| 23:12.01 | Qwell[] | reload, and realtime? :p |
| 23:12.17 | mvanbaak | chan_sccp???? what's that ? |
| 23:12.19 | mvanbaak | ;) |
| 23:12.22 | Qwell[] | I totally used the realtime patch in production, heh |
| 23:12.30 | Qwell[] | for like...months |
| 23:12.41 | Qwell[] | You realize that it's now been over a year since the last release? |
| 23:12.42 | mvanbaak | nice |
| 23:12.55 | mvanbaak | yeah. I tried to contact sergio |
| 23:13.03 | mvanbaak | and sgofferje |
| 23:13.06 | mvanbaak | both seem dead |
| 23:13.27 | Qwell[] | I'm glad a few of you have joined the dark side :p |
| 23:13.44 | mvanbaak | I'm so glad I did |
| 23:14.21 | BSD_tech | the dark side |
| 23:14.23 | mvanbaak | I now know that when asterisk coredumps it's prolly my own code |
| 23:14.27 | BSD_tech | hell its pitch black |
| 23:14.39 | Qwell[] | mvanbaak: heh |
| 23:15.04 | mvanbaak | it's easier to explain here at home as well |
| 23:15.17 | mvanbaak | 'sorry hon, I was trying $random_feature' |
| 23:15.27 | mvanbaak | svn revert channels/chan_skinny.c |
| 23:15.32 | mvanbaak | 'it's working again' |
| 23:15.39 | mvanbaak | *have sex* |
| 23:15.43 | *** join/#asterisk metabox (n=metabox@modemcable192.65-56-74.mc.videotron.ca) |
| 23:16.20 | BSD_tech | so they fixed chan skinny |
| 23:16.29 | BSD_tech | and plugged its holes |
| 23:17.35 | mvanbaak | BSD_tech: actually, it's a great channel driver for great phones |
| 23:17.55 | mvanbaak | too bad we dont get the new video phone from cisco to get that one working |
| 23:17.57 | BSD_tech | wish I could afford 1 to try |
| 23:18.12 | BSD_tech | I want the cisco wifi phone |
| 23:18.12 | Qwell[] | get a 7910 - dirt cheap on ebay sometimes |
| 23:18.17 | Qwell[] | BSD_tech: No you don't :) |
| 23:18.21 | BSD_tech | ok |
| 23:18.58 | BSD_tech | they are that bad |
| 23:18.58 | mvanbaak | the 7920 isn't that ok |
| 23:18.58 | Qwell[] | 7920 is a cool though, but it's not great |
| 23:18.58 | BSD_tech | they 2 |
| 23:18.58 | BSD_tech | I thought |
| 23:18.58 | Qwell[] | it's alright, but not for like $400 :) |
| 23:18.58 | mvanbaak | and it's fucking expensive |
| 23:18.58 | BSD_tech | ok |
| 23:18.58 | Qwell[] | like the 7985 |
| 23:19.03 | Qwell[] | just buy like 40 grandstreams, heh |
| 23:19.09 | mvanbaak | gheh |
| 23:19.09 | BSD_tech | hehehe |
| 23:19.12 | Qwell[] | it'd be about the same price |
| 23:20.48 | *** join/#asterisk metabox (n=metabox@modemcable192.65-56-74.mc.videotron.ca) |
| 23:20.48 | Qwell[] | 7970 though is definitely worth the price |
| 23:20.48 | mvanbaak | get a 7905 or 7910 |
| 23:20.48 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:20.48 | BSD_tech | I need a good wifi phone |
| 23:20.49 | mvanbaak | they are dirt cheap |
| 23:20.49 | Qwell[] | BSD_tech: there aren't really any |
| 23:20.49 | BSD_tech | I need to go shopping |
| 23:20.49 | Qwell[] | 7905...hmm, I need to get one of those |
| 23:20.49 | mvanbaak | Qwell[]: I have one |
| 23:20.49 | metabox | I search the best Hard Sip Phone for receptionist use. |
| 23:20.49 | lesouvage | When I do Set(MINUTE_PRICE=$[ ${MACRO_RESULT} / 10]) and the outcome should be 6,7 the actual result is 6. Is there a way to count with decimals in Asterisks? |
| 23:20.49 | Qwell[] | 12sp, 30vip, 7910, 7960, are all I have |
| 23:20.49 | mvanbaak | actually, I have a 7905 and a 7960 |
| 23:20.50 | Qwell[] | all 4 were given to me by various people, for various reasons :) |
| 23:20.50 | Qwell[] | anyhow, bbl |
| 23:20.53 | mvanbaak | guess I have to contact Florian Overkamp |
| 23:21.02 | mvanbaak | he did some of the initial dev of chan_skinny |
| 23:21.07 | Qwell[] | why? |
| 23:21.16 | mvanbaak | maybe he still has the hardware |
| 23:21.25 | Qwell[] | which hardware? |
| 23:21.26 | mvanbaak | I know him personally |
| 23:21.33 | mvanbaak | the vip and sp hardware |
| 23:21.35 | Qwell[] | oh, I see what you mean |
| 23:21.57 | Qwell[] | the 30vip is decent... very dated, but decent |
| 23:21.58 | mvanbaak | and his company now sells cisco phones |
| 23:22.05 | evilcyrus | can't wait to play |
| 23:22.13 | mvanbaak | maybe I can get some test hardware |
| 23:22.21 | mvanbaak | actually |
| 23:22.48 | mvanbaak | there's a possibility I'm going to work for him |
| 23:22.53 | Qwell[] | neat |
| 23:22.53 | evilcyrus | anyone use the Wifi phones |
| 23:22.59 | evilcyrus | i think thats a sick IDEA |
| 23:22.59 | mvanbaak | yeah |
| 23:23.12 | BSD_tech | http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-53499572480.htm |
| 23:23.19 | BSD_tech | I want to test that one |
| 23:23.24 | lesouvage | mvanbaak: I'm listening |
| 23:23.34 | mvanbaak | speakup (his company) and covide (the company I work for) do a lot together |
| 23:23.50 | lesouvage | Just joking |
| 23:24.03 | mvanbaak | I really like f.overkamp and his company |
| 23:24.09 | *** part/#asterisk billasterisknewb (n=bill4242@66.60.191.180) |
| 23:24.34 | mvanbaak | hhmm |
| 23:24.48 | lesouvage | mvanbaak: have you registered for the 31 may event. |
| 23:24.49 | mvanbaak | assen <-> enschede |
| 23:25.03 | mvanbaak | lesouvage: my wife is there |
| 23:25.13 | mvanbaak | and she needs a tech guy to be there |
| 23:25.19 | mvanbaak | so I think that's a yes |
| 23:25.24 | mvanbaak | but without registration |
| 23:25.41 | lesouvage | mvanbaak: I know, you can tell here tht there are allready 110 registrations. I guess we will have a full house. |
| 23:26.59 | mvanbaak | cool ! |
| 23:27.09 | mvanbaak | greetings from nancy :) |
| 23:27.54 | mvanbaak | btw, you know nancy's last day for covide is 31 may ? |
| 23:28.15 | lesouvage | give her my greetings. Enig idee hoe ik achter de comma kan rekenen met Asterisk. |
| 23:28.56 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:29.28 | lesouvage | mvanbaak: yes, see will start the next day with the team I'm working with for the 31 may event. They are a very nice team but very busy. |
| 23:29.39 | lesouvage | see=she |
| 23:30.11 | mvanbaak | lesouvage: that's why they hired nancy |
| 23:30.12 | mvanbaak | ;) |
| 23:30.24 | lesouvage | good choice. |
| 23:30.29 | mvanbaak | about the comma calculation in * |
| 23:30.39 | mvanbaak | I never do stuff like that in * itself |
| 23:30.48 | mvanbaak | I always use some agi script for that |
| 23:31.13 | mvanbaak | I'm simply more familiar with C and php then with extensions.ael |
| 23:31.37 | mvanbaak | ael2 made asterisk config way more fun for me |
| 23:32.05 | mvanbaak | but some more complicated stuff (like accounting and LCR) is done in C and/or php |
| 23:32.35 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
| 23:32.37 | kink0 | hi |
| 23:33.03 | mvanbaak | lesouvage: btw, the speakup/f.overkamp stuff is not known by my boss |
| 23:33.09 | mvanbaak | please keep it that way |
| 23:33.39 | lesouvage | mvanbaak: OK, you know that the listings of this channel are on the internet? |
| 23:33.51 | mvanbaak | I know |
| 23:34.02 | mvanbaak | my boss doesn't know ;) |
| 23:34.18 | lesouvage | mvanbaak: My lips are sealed. |
| 23:36.40 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:36.55 | *** join/#asterisk wampie (n=wampie@vanbaak.xs4all.nl) |
| 23:36.58 | wampie | hey |
| 23:37.01 | wampie | :) |
| 23:37.04 | mvanbaak | wampie ! |
| 23:37.09 | mvanbaak | lesouvage: meet wampie |
| 23:37.13 | mvanbaak | wampie: meet lesouvage |
| 23:37.16 | wampie | hey lesouvage |
| 23:37.21 | wampie | <- nvbaak |
| 23:37.22 | wampie | lol |
| 23:37.47 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 23:37.49 | mvanbaak | wampie is my wife |
| 23:38.00 | mvanbaak | the one that will find all the bugs in chan_skinny |
| 23:38.01 | mvanbaak | ;) |
| 23:38.27 | wampie | indeed!!! calling with my mom |
| 23:38.30 | wampie | :P |
| 23:38.35 | lesouvage | There was a proposal on the Dutch forum to start a Dutch channel. Maybe that is not such a bad idea. |
| 23:39.12 | lesouvage | wampie: there are allready 110 registrations for 31 may. |
| 23:39.15 | wampie | lesouvage, sorry for the complete lack of activities on that from my site btw |
| 23:39.39 | wampie | lack of time really |
| 23:39.39 | wampie | same with you and answering email ;) |
| 23:39.41 | wampie | j/k |
| 23:39.54 | wampie | wow! |
| 23:40.05 | wampie | lucky for me i'm not organising it yet |
| 23:40.13 | mvanbaak | lesouvage: you want the channel on freenode ? |
| 23:40.20 | wampie | sounds like it needs a second edtition |
| 23:40.28 | mvanbaak | or can it be on some other network ? |
| 23:40.40 | mvanbaak | <--- core member of another network |
| 23:40.41 | mvanbaak | lol |
| 23:40.55 | wampie | 110 |
| 23:40.57 | wampie | phew |
| 23:40.58 | *** join/#asterisk klasstek (n=nunyobiz@c-67-190-165-254.hsd1.co.comcast.net) |
| 23:41.07 | wampie | better come up with a great story then |
| 23:41.27 | wampie | presentation i mean |
| 23:42.10 | lesouvage | mvanbaak: I don't think it is such a good idea. A channel is only usefull when there are always people joining it and it can be more then just a social thing. I don't have time to spend hours a day helping newcomers and I think nobody of the current members has. |
| 23:42.11 | *** join/#asterisk Dimik (n=Dimik@unaffiliated/dimik) |
| 23:43.16 | mvanbaak | lesouvage: you want me to start the channel ? |
| 23:43.28 | mvanbaak | I can maintain a bot there |
| 23:43.38 | mvanbaak | and be there in after-hours |
| 23:43.47 | mvanbaak | during work hours I'm busy right now |
| 23:43.54 | mvanbaak | and my boss wont allow me to be there |
| 23:44.07 | wampie | mvanbaak is an IRC junky |
| 23:44.08 | wampie | lol |
| 23:44.12 | Math` | lol |
| 23:44.13 | mvanbaak | s/my boss/willem/g |
| 23:44.30 | Math` | remembers me when I started coding, we were modding irc servers and everybody has its own server lol |
| 23:44.43 | mvanbaak | Math` ;) |
| 23:44.53 | mvanbaak | Math`: sounds like Ambernet |
| 23:45.20 | mvanbaak | 10 servers, 17 irc operators and avg. of 50 users |
| 23:45.50 | mvanbaak | but we do have 3 stable hubs |
| 23:46.25 | Math` | lol |
| 23:46.41 | Math` | and out of those 50 users... how many are server-generated bots? lol |
| 23:48.14 | Math` | what flavor of ircd are you using |
| 23:48.32 | mvanbaak | ratbox |
| 23:48.58 | Math` | ah efnet's? |
| 23:49.56 | mvanbaak | close |
| 23:50.03 | mvanbaak | efnet uses hybrid |
| 23:50.12 | mvanbaak | but some servers are ratbox |
| 23:50.22 | mvanbaak | hybrid and ratbox are really close |
| 23:50.52 | mvanbaak | take chanfix. it's a hybrid mod by default. but it also runs on ratbox |
| 23:51.07 | mvanbaak | Ambernet indeed runs chanfix |
| 23:51.23 | Math` | yeah we implemented this one for ircu |
| 23:52.07 | mvanbaak | the Ambernet admins use asterisk to have -dev and -maintain conferences |
| 23:52.27 | mvanbaak | we are working on chan_ambernet.so |
| 23:52.36 | Math` | for what lol |
| 23:52.50 | mvanbaak | so you can dial in, give some pincodes, and be able to alter network settings |
| 23:53.02 | Math` | you dont need a channel driver for that |
| 23:53.04 | mvanbaak | stuff like glines, spoofs, jupes etc |
| 23:53.05 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 23:53.12 | mvanbaak | no, we dont need it |
| 23:53.13 | Math` | you just need a good old agi |
| 23:53.21 | mvanbaak | but it's fun to do it as chan_ambernet.so |
| 23:53.30 | mvanbaak | we have it as agi right now |
| 23:53.33 | Math` | I dont see how all the overhead you're getting into is fun |
| 23:53.45 | mvanbaak | but all the admins want to polish their C skillz |
| 23:53.57 | Math` | I se |
| 23:53.59 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-177-38-240.vic.bigpond.net.au) |
| 23:53.59 | Math` | see* |
| 23:54.17 | mvanbaak | all the agi stuff is in bash and perl right now |
| 23:54.32 | Math` | so then make an application.... |
| 23:54.35 | Math` | not a channel driver |
| 23:54.43 | Mavvie | anybody from the Sydney area here? |
| 23:55.01 | Math` | since no calls are going to be terminated to irc... |
| 23:55.05 | mvanbaak | we do: Dial(Ambernet/<servername>) |
| 23:55.09 | wampie | middle of the night here, so..... no |
| 23:55.38 | Math` | mvanbaak: Ambernet(servername) aint enough? |
| 23:55.47 | mvanbaak | no |
| 23:55.49 | mvanbaak | lol |
| 23:55.56 | Math` | if you say so ... :P |
| 23:56.05 | Mavvie | walhala: you might not have noticed that Sydney is about the other side of the world for you. |
| 23:56.22 | mvanbaak | Math`: it's just done this way for one reason: |
| 23:56.31 | mvanbaak | "why?", "because we can" |
| 23:56.41 | Math` | right |
| 23:57.06 | wampie | Mavvie, (assuming you ment me) it's daytime there right? |
| 23:57.14 | Mavvie | wampie: it's 09:56 here |
| 23:57.17 | mvanbaak | Sat May 12 01:57:07 CEST 2007 |
| 23:57.22 | wampie | thanks mvanbaak |
| 23:57.26 | mvanbaak | ;) |
| 23:57.31 | wampie | that's my time too |
| 23:57.33 | mvanbaak | wampie is my wife |
| 23:57.46 | wampie | so we live in the same timezone :P |
| 23:57.48 | mvanbaak | we are like 2 meters away from eachother |
| 23:58.06 | wampie | like i said: same timezone |
| 23:58.11 | mvanbaak | 2 steps to grab the bottle of vino from her |
| 23:58.18 | wampie | hmmzz.. |
| 23:58.23 | mvanbaak | whehehehe |
| 23:58.23 | wampie | 2 steps to many |
| 23:58.28 | mvanbaak | yup |
| 23:58.48 | mvanbaak | enough for today |
| 23:58.59 | Mavvie | wampie: so you can explain to her that when I ask "is there anybody from Sydney here", I won't be interested in the dutch smartasses who think that it's a misspelling of Groenebeek or something. |
| 23:59.13 | mvanbaak | enough of chan_skinny.c dev |
| 23:59.14 | wampie | rofl mav |
| 23:59.48 | Math` | mvanbaak: making an irc client for cisco phones? :) |
| 23:59.52 | *** join/#asterisk DocHolliday (i=RgRabbit@gateway/gpg-tor/key-0x0E4F6D6C) |
| 23:59.54 | wampie | I will explain to her |
| 23:59.59 | mvanbaak | Math`: indeed |
