| 00:01.37 | elixer | JT: so do you think because i have cards from different vendors, that might be complicating things? |
| 00:01.53 | JT | you never know, it could |
| 00:02.11 | JT | also, you load wanpipe before zaptel |
| 00:02.15 | elixer | right |
| 00:02.19 | elixer | already doing that |
| 00:03.16 | elixer | well i've been at this all day, i'll pick back up tomorrow |
| 00:03.22 | elixer | thanks for the assist. |
| 00:03.49 | JT | just don't shut it down? ;) |
| 00:04.24 | elixer | thanks for your time? |
| 00:04.27 | elixer | better? |
| 00:04.34 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.222.111) |
| 00:04.46 | JT | your pbx |
| 00:04.58 | JT | don't shut it down if that's the only thing that errors |
| 00:05.00 | JT | heh |
| 00:05.05 | elixer | its not the only thing that errors |
| 00:05.17 | elixer | so, not really an option |
| 00:05.23 | elixer | i'll get it worked out. |
| 00:05.26 | elixer | later. |
| 00:05.29 | JT | ok |
| 00:05.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:07.07 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 00:07.23 | *** join/#asterisk knarfly (n=knarfly@c-98-203-55-196.hsd1.fl.comcast.net) |
| 00:23.15 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 00:23.36 | MrTelephone | im having dtmf detection issues on a remote asterisk box that is a sip trunk |
| 00:23.41 | MrTelephone | i switched the dtmfmode=info |
| 00:24.06 | MrTelephone | on the sip trunk |
| 00:32.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:33.10 | MrTelephone | i think the secretary there is a friggin idiot |
| 00:33.25 | MrTelephone | everytime i test it there is no problem with the dtmf |
| 00:36.29 | *** join/#asterisk kkn088 (n=kikoun@84.4.74.213) |
| 00:37.43 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 00:37.49 | JT | MrTelephone: dtmf mode info is not recommended |
| 00:38.29 | tzanger | what's the recommended h323 stack for asterisk these days, to the level that anyone would recommend h323 :-) |
| 00:39.09 | JT | cough, yate ;) |
| 00:39.14 | tzanger | heh |
| 00:44.11 | lmadsen | I think just whatever is in -addons |
| 00:44.27 | lmadsen | since ooh323 I don't think is being maintained by the original authors |
| 00:45.09 | coppice | ooh323 always had a rather woolly status |
| 00:45.35 | coppice | it requires expensive tools for outsiders to work on it |
| 00:46.12 | MrTelephone | the stuff is garbage |
| 00:46.36 | *** join/#asterisk hohum_ (n=dcorbe@dhcp64-134-231-191.shs.nyc.wayport.net) |
| 00:47.12 | coppice | in what sense? |
| 00:47.58 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:50.39 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-154-19-170.red.bezeqint.net) |
| 00:57.27 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 00:58.16 | *** join/#asterisk gardo (n=gardo@121.97.144.15) |
| 00:58.30 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 00:59.18 | Olobola | I wonder which class is the best for agi/php. I need to access a SQL Server database from Lumenvox. I know a little PHP already.. what do you think? |
| 01:00.15 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:01.24 | *** join/#asterisk withnall (n=dave@60-242-0-60.static.tpgi.com.au) |
| 01:01.42 | withnall | greetings all. have a question that's been bugging me for awhile now. |
| 01:02.10 | withnall | my asterisk box successfully registers with my sip provider after a sip reload. then exactly 32 seconds later it destroys the registrations |
| 01:02.25 | withnall | which means i no longer have any incoming calls. |
| 01:02.33 | withnall | does anyone have any idea as to what is going on? |
| 01:02.41 | clyrrad | withnall: do you have more than one box registering that DID? |
| 01:02.58 | clyrrad | withnall: or an IP phone or any other device that might be registering that DID? |
| 01:03.20 | withnall | just the one box. registers with 4 dids |
| 01:03.36 | clyrrad | withnall: are you SURE that nothing else is registering this DID? |
| 01:03.51 | withnall | nothing inside my network is. I don't know about outside |
| 01:04.00 | clyrrad | withnall: I had this problem before when I had a Softphone regisering the same DID as my Asterisk server |
| 01:04.22 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:04.52 | asterisknerds | <PROTECTED> |
| 01:05.08 | withnall | i'll see if I can track something down with the provider. thanks for the insight |
| 01:05.17 | clyrrad | no prob |
| 01:05.31 | clyrrad | withnall: BTW - who is the provider? |
| 01:05.45 | withnall | gotalk |
| 01:05.53 | withnall | australian company |
| 01:05.55 | clyrrad | hrm... never used them |
| 01:06.56 | clyrrad | withnall: the fact that your DID looses its registration at that EXACT interval makes me think something else may be regestering it |
| 01:09.25 | withnall | I get an "Auto destorying call" debug message when it happens. that's how I got the timing interval. I don't know why * would destroy the call though. |
| 01:11.51 | *** join/#asterisk Swat2 (n=bler@218-215-201-174.people.net.au) |
| 01:11.56 | Swat2 | Anyone got a Snom320 phone and can explain why when i try to call it, i get all circuits are busy now message? :/ |
| 01:16.19 | *** join/#asterisk rrittenhouse (n=tad@unaffiliated/rrittenhouse) |
| 01:16.21 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 01:16.45 | *** join/#asterisk Olobola (n=chatzill@74.95.13.57) |
| 01:16.48 | andrew` | hi, any suggestions for a good SIP, easy to use (for newbie) client for os x? |
| 01:19.10 | JT | "a good SIP"? |
| 01:19.16 | JT | softphone |
| 01:19.23 | Swat2 | heh |
| 01:20.10 | andrew` | i know ;) |
| 01:20.13 | andrew` | good is relative... |
| 01:22.23 | hrmphh | good sip? |
| 01:22.26 | hrmphh | try guiness |
| 01:23.54 | rrittenhouse | In asterisknow when you connect to the voicemail (from a softphone..) Doesnt it just prompt for a password on the ext your calling from |
| 01:24.16 | rrittenhouse | Or is it supposed to say "Mailbox <pause till you enter mailbox>" then "Password <pause>" |
| 01:33.57 | JT | andrew`: were you asking for a softphone? |
| 01:34.16 | flenders | andrew`: by now you probably already found it, but x-lite has a mac version |
| 01:34.32 | *** join/#asterisk AJaymn (n=Me@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 01:35.24 | *** join/#asterisk simonkern (n=simonker@p54AAAD79.dip0.t-ipconnect.de) |
| 01:35.40 | simonkern | hi |
| 01:35.42 | andrew` | flenders, yep found that one, didn't realize they had a mac version, but thank you |
| 01:35.54 | JT | idefisk supposedly supports SIP now too |
| 01:36.15 | AJaymn | can you get zaptel already compiled? |
| 01:36.25 | JT | no |
| 01:36.37 | JT | well, it'd need to be for a specific kernel version |
| 01:37.07 | flenders | AJaymn: compile it yourself, always the best option |
| 01:37.36 | simonkern | i've seen support for G.722 in the svn trunk. is it possible to use it in 1.4 too? |
| 01:38.10 | JT | simonkern: isn't the G.722 support simply passthrough? |
| 01:39.31 | simonkern | JT: I'm not sure, but it sounds like a real transcoding support "Add codec G.722 support" (quote from svn) |
| 01:39.45 | JT | heh |
| 01:40.02 | JT | i can't believe it's taking so long for G.722 |
| 01:40.11 | simonkern | JT: and there is a g722_decode.c and g722_encode.c |
| 01:40.28 | simonkern | http://svn.digium.com/view/asterisk/trunk/codecs/g722/ |
| 01:41.26 | jcolp | yes, there is a g722 translator in trunk |
| 01:41.42 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-191.shs.nyc.wayport.net) |
| 01:43.15 | simonkern | so, my question is, is it possible to use it with 1.4 |
| 01:43.33 | jcolp | possible? yes, you could backport the codec... it may work without any modifications even |
| 01:43.37 | jcolp | 'tis only code |
| 01:44.31 | AJaymn | flenders: ive tried to compile it on my box but says i dont have the sources for my kernel which is 2.6.20-1.2944.fc6 and if i try to use Yum it wants to update to 2.6.22 |
| 01:45.25 | simonkern | that sounds very interesting. I'm going to start my "test system". Thx for help |
| 01:45.28 | simonkern | bye |
| 01:45.38 | *** part/#asterisk simonkern (n=simonker@p54AAAD79.dip0.t-ipconnect.de) |
| 01:55.25 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 01:55.55 | asterisknerds | <PROTECTED> |
| 01:58.09 | flenders | AJaymn: pretty sure you can get the kernel-headers rpm for that version |
| 01:59.22 | flenders | don't try to yum the kernel-sources or kernel |
| 02:00.09 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:01.52 | AJaymn | flenders this sucks :P You do not appear to have the sources for the 2.6.20-1.2944.fc6 kernel installed. |
| 02:02.07 | AJaymn | thats my error message, but i dont know what rpm i need.. or where to find that ver |
| 02:03.05 | CCFL_Man2 | anyone here install asterisk on linux sparc running on a sun netra? |
| 02:03.25 | JT | no, but let me know how it goes, CCFL_Man2 :P |
| 02:03.29 | JT | why not solaris? |
| 02:03.36 | Sweeper | slowlaris~ |
| 02:03.41 | CCFL_Man2 | lol |
| 02:03.59 | JT | Sweeper: only on X86. |
| 02:04.03 | JT | not on sparc |
| 02:04.22 | flenders | solaris runs really well on sparc |
| 02:04.29 | CCFL_Man2 | i was told the ztdummy driver for the bsds and solaris suck and i can't do MOH or conferencing |
| 02:04.37 | Sweeper | well, I dunno about the new stuff, but on my ss20 cluster, netbsd ran a lot faster |
| 02:04.44 | JT | haha ss20 |
| 02:04.58 | JT | that's some prehistoric stuff now |
| 02:05.07 | Sweeper | .com liquidations, baby |
| 02:05.53 | CCFL_Man2 | because the solaris zaptel drivers i hear are not written well |
| 02:06.33 | CCFL_Man2 | and proper timing can't be obtained |
| 02:07.03 | Sweeper | ebay the netra, get a pc imo |
| 02:07.13 | CCFL_Man2 | Sweeper: why? |
| 02:07.14 | JT | CCFL_Man2: forget about sun hardware for zaptel. |
| 02:07.17 | JT | CCFL_Man2: waste of time |
| 02:07.29 | JT | CCFL_Man2: it's only designed to work and have drivers written for X86 |
| 02:07.33 | snuff-work | mm.. i want to try out one of those T1000/T2000 boxes |
| 02:07.34 | JT | linux specifically |
| 02:08.01 | CCFL_Man2 | so my best bet is linux x86 and not linux sparc? |
| 02:08.19 | JT | right |
| 02:08.37 | CCFL_Man2 | because i was going to install debian sparc on a netra and built the zaptel drivers |
| 02:09.28 | JT | kernel drivers need to be written for each architecture, they are not cross platform |
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| 02:12.01 | CCFL_Man2 | debian has a sparc package for the zaptel drivers though |
| 02:12.16 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-75-80-248-142.dc.res.rr.com) |
| 02:12.22 | JT | sounds like alpha level code |
| 02:12.43 | CCFL_Man2 | zaptel (1:1.2.11.dfsg-1) |
| 02:13.32 | CCFL_Man2 | http://packages.debian.org/stable/comm/zaptel |
| 02:14.58 | CCFL_Man2 | i can use xbox linux though, thats x86, but a 2.4 kernel |
| 02:15.51 | *** join/#asterisk coinoperatedboy1 (n=Charles@203.177.140.188) |
| 02:15.53 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-191.shs.nyc.wayport.net) |
| 02:16.07 | coinoperatedboy1 | hello! |
| 02:17.25 | CCFL_Man2 | i can get a netra for super cheap |
| 02:18.04 | coinoperatedboy1 | i would like to know what pci card to buy if we are planning to make asterisk as an IVR and still interfaced with our existing PABX |
| 02:18.57 | CCFL_Man2 | a T1 card |
| 02:20.44 | coinoperatedboy1 | does the FXO/FXS do any good for our setup? |
| 02:21.34 | coinoperatedboy1 | this is our setup: POTS <-- Asterisk Server --> PABX |
| 02:21.46 | CCFL_Man2 | if thats how you want to interface to your pabx |
| 02:22.22 | coinoperatedboy1 | its a soft migration to Asterisk :-p |
| 02:22.48 | CCFL_Man2 | how do you connect to your pabx? |
| 02:23.33 | CCFL_Man2 | a CAS T1 or something? |
| 02:23.42 | *** join/#asterisk zutoo (n=zutoo@74.94.57.209) |
| 02:25.06 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-141-155-139-103.ny5030.east.verizon.net) |
| 02:25.30 | coinoperatedboy1 | actually i'm not sure about our current PABX... it still serviced by another company... i'll ask if there still is a T1 interface available |
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| 02:33.52 | CCFL_Man2 | i'm not sure how else it would connect |
| 02:34.17 | JT | well |
| 02:34.30 | JT | CCFL_Man2: it could also be via FXO or FXS ports, or PRI, or BRI |
| 02:34.33 | JT | lots of choices |
| 02:34.39 | CCFL_Man2 | oh yeah |
| 02:34.55 | coinoperatedboy1 | :-) |
| 02:35.18 | CCFL_Man2 | forgot about PRI and BRI |
| 02:35.19 | coinoperatedboy1 | thanks... i'll call up the PABX company... we'll boot them out hehehe |
| 02:35.47 | CCFL_Man2 | or just use asterisk as your pabx |
| 02:36.03 | CCFL_Man2 | well, actually, no |
| 02:36.08 | coinoperatedboy1 | our management are japanese... |
| 02:36.32 | CCFL_Man2 | ahh |
| 02:36.33 | coinoperatedboy1 | they are quite paranoid to changes heheh |
| 02:36.35 | coinoperatedboy1 | :-p |
| 02:37.10 | coinoperatedboy1 | we would like to introduce asterisk to the system slowly |
| 02:37.26 | red9012 | I have a whole bunch of extensions (100+) that I like to have available in different contexts. (ie. in sales/support context should allow the caller to dial any extension) |
| 02:37.45 | CCFL_Man2 | JT: so you think the zaptel drivers for linux sparc won't be good enough and cause significant timing issues with ztdummy that'll cause problems with conferencing? |
| 02:38.00 | coinoperatedboy1 | thanks CCFL_Man2 and JT :-) |
| 02:38.03 | CCFL_Man2 | np |
| 02:38.27 | red9012 | right now, i have a context defined for extensions. (all extensions are defined in that context). However, if the caller is in a different context, then the extensions are not available to the caller. How can I fix this? |
| 02:38.54 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 02:39.41 | *** join/#asterisk jp_ast (n=jeep_jer@209-181-70-21.hlrn.qwest.net) |
| 02:39.45 | JT | CCFL_Man2: it might work, you just have to try it |
| 02:39.54 | Nivex | red9012: write a rule in the other context that uses Goto() |
| 02:40.18 | Nivex | red9012: eg: Goto(myextensionscontext,4321) |
| 02:41.11 | CCFL_Man2 | JT: i'll just be using the ztdummy driver, plus the kernel will be 2.6, the package is already built and available, i don't see why it won't work, but who knows |
| 02:41.31 | Nivex | red9012: assuming your extensions all start with 4, you could write: exten => _4XXX,Goto(myextensionscontext,${EXTEN}) |
| 02:42.24 | Nivex | red9012: Your mileage may vary, batteries not included, void where prohibited ;) |
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| 02:55.58 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
| 02:56.19 | Primer | Trying to find examples of dialing a skinny channel in extensions.conf. Anyone have one handy? |
| 02:56.28 | asterisknerds | <PROTECTED> |
| 02:57.20 | Primer | apparently one needs an @ something |
| 02:57.31 | red9012 | nivex-- ok if I do that... should I disable users from having '4' as a valid keypress? |
| 03:01.03 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:01.31 | Nivex | red9012: I just picked '4' as a random example. I don't know what your extensions are. |
| 03:02.06 | red9012 | my extension are actually going from 100 to 999 |
| 03:02.22 | red9012 | therefore, the above example would not apply. |
| 03:02.53 | Nivex | well the Goto is what you need. Munge the pattern as applicible |
| 03:03.59 | red9012 | ok and what if the '4' is already assigned a goto to a different context ( another menu). |
| 03:05.03 | Nivex | This sounds like an IVR. Sounds like you want a "Press 5 to dial by extension" which sends you to a different context that plays a "Enter the extension now" and has a pattern for the 100-999 |
| 03:05.55 | ManxPower | that is how you would do it when you poorly planned your system |
| 03:06.06 | Nivex | ManxPower: I was being polite :) |
| 03:06.22 | ManxPower | I'm never polite. |
| 03:06.45 | Nivex | but yes, this is why all my extensions are 6XXX |
| 03:06.53 | red9012 | it's an ivr. right with dial 'x' to access extension contenxt..... however, I now need to have the extensions available to all menus. |
| 03:07.46 | Nivex | red9012: that can't be done for the reasons ManxPower eluded to. You can't have an extension that starts with the same digit as a menu item |
| 03:07.55 | red9012 | the option of having a goto on a pattern match on a digit would work, however, that digit would have to be disabled. |
| 03:08.07 | Nivex | now if all your extensions were 8100-8199, then you could do it |
| 03:08.37 | red9012 | for example if press 2 would go to techsupport, then a goto on pattern match _2xxx would not work. |
| 03:09.42 | Olobola | has anyone tried running lumenvox through phpagi? |
| 03:09.58 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:11.05 | red9012 | nivex - in your example of 8100-8199 -- what pattern would be used to execute a goto md? |
| 03:11.12 | red9012 | cmd |
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| 03:30.21 | CoffeeKid | I'm having some issues with using AGI within asterisk. I'm trying to simulate a warm transfer (assisted transfer)... But i keep losing channels, anyone around that could help me out? |
| 03:34.54 | *** join/#asterisk Strom_M (n=strom@bas5-montreal28-1177917181.dsl.bell.ca) |
| 03:35.34 | AJaymn | im having issues compiling zaptel on kernel 2.6.22.1-32.fc6 is there somthing different to do? |
| 03:38.46 | Swat2 | in sip.conf, you have to put the allow= commands directly after the disallow=all dont you? |
| 03:39.06 | jql | the 'directly' part seems unlikely |
| 03:39.13 | jql | 'after' is good, though |
| 03:39.29 | JT | yes, after |
| 03:39.31 | Swat2 | yah |
| 03:39.56 | CoffeeKid | Any AGI guys on tonight? |
| 03:39.56 | Swat2 | if you put 2 allow statements above disallow=all, they would be disregarded, correct? |
| 03:40.13 | JT | i can't say for sure without trying, but probably |
| 03:40.44 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 03:40.45 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 03:43.07 | ManxPower | I always do disallow=all and then allow= lines for each of the codecs I want |
| 03:43.22 | JT | allowing all is stupid |
| 03:43.37 | JT | because then you have clients trying to negotiate g.723 and what not |
| 03:45.52 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 03:46.18 | red9012 | the voicemailmain() app uses wav49 for recording voicemail greetings. is there a way to overwrite that? |
| 03:46.45 | Strom_M | red9012: yeah, see voicemail.conf |
| 03:46.50 | ManxPower | red9012: no it does not. It uses whatever you specified in voicemail.conf |
| 03:48.15 | [T]ank | anyone here :-D my friend says its dead. |
| 03:48.25 | *** join/#asterisk qai (n=qai@216-154-21-96.dsl.look.ca) |
| 03:49.00 | red9012 | can I remove some of the options of voicemailmain. for example, no advanced options |
| 03:49.22 | qai | hello |
| 03:50.19 | qai | seems like the word "options" says it all |
| 03:51.44 | red9012 | not when it comes to asterisk. |
| 03:51.54 | ManxPower | try it and see. |
| 03:52.07 | qai | Anyone here have a simple solution to using ztmonitor in a continuous mode. I want to track down callerid problems and need to have ztmonitor running for a day to catch some samples. I only want to create a file and destroy it if after a minute there is no channel activity |
| 03:55.24 | qai | Actually red9012 I think there is.. just trying to stir up the grey matter to recall where I heard it |
| 03:55.36 | *** join/#asterisk asterisknerds (n=asterisk@66.7.122.93) |
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| 03:58.23 | AJaymn | FATAL: Error inserting ztdummy (/lib/modules/2.6.22.1-32.fc6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter |
| 03:58.29 | CoffeeKid | Anyone around with some experience with AGI? |
| 04:00.11 | CoffeeKid | anyone know if there's a way to emulate what an SIP phone is doing? like sending commands directly to the SIP socket? |
| 04:01.37 | [T]ank | CoffeKid: AGI Blows ;-) |
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| 04:02.32 | [T]ank | CoffeeKid: Your just a script kitty :-D |
| 04:04.10 | qai | Sorry red9012. I think what I recalled was that voicemail is a very old app, and one that people have wanted to rewrite for sometime but since it is a very difficult app to write, there has been little progress. The only link I could find is that people are suggesting to go to source to make mod |
| 04:04.11 | JT | if you're going to bag someone out, at least spell "You're" and "kiddie" correctly |
| 04:04.20 | CoffeeKid | [T]ank, hehe, at this point i would agree |
| 04:04.25 | [T]ank | hehe |
| 04:04.32 | [T]ank | we work together :-D |
| 04:04.45 | CoffeeKid | [T]ank, i hate you |
| 04:04.50 | [T]ank | lol |
| 04:05.20 | [hC] | i hate to say it but now that fonality bought them and used pbxtra to relaunch it, trixbox pro is looking really good |
| 04:06.30 | qai | oohh pretty "good" or functional "good" |
| 04:06.58 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.180.56) |
| 04:09.23 | [hC] | im going on screenshots and specs alone at this point, im downloading it now. ill likely get 15 minutes into it and get pissed off again. |
| 04:09.58 | AJaymn | Is there still a problem compiling Zaptel on 2.6.22 kernel? |
| 04:10.21 | CoffeeKid | [T]ank, your bird is stinky. |
| 04:11.18 | CoffeeKid | Ajaymn, shouldn't be, i've done it on Gentoo |
| 04:11.44 | [T]ank | CoffeeKid: your finger's stinky |
| 04:11.59 | CoffeeKid | sometimes |
| 04:14.59 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 04:16.33 | qai | Anyone here have a simple solution to using ztmonitor in a continuous mode. One where it destroys the output file once a minute if the channel wasn't active, and also won't create the file if the channel is active?? |
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| 04:50.40 | AJaymn | im trying to install 1.4 and during the ./configure configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 04:50.40 | AJaymn | what package am i missing? |
| 04:53.12 | ManxPower | gcc-c++ might be the package name |
| 04:53.50 | AJaymn | yum install gcc-c++ NOTHING TO DO |
| 04:53.51 | AJaymn | ?!? |
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| 04:54.56 | *** part/#asterisk lobstr (i=wbram@unaffiliated/lobstr) |
| 04:55.23 | AJaymn | nevermind think it was a typo :P |
| 04:55.25 | AJaymn | Thanks ;) |
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| 04:56.16 | asterisknerds | <PROTECTED> |
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| 05:31.55 | [T]ank | so what am i doing wrong if after i install the zaptel packages for 1.4 and reboot the server, i have to manually modprobe wct4xxp for the card to load. |
| 05:32.00 | [T]ank | i ran make config from the zaptel src. |
| 05:35.13 | CaT[tm] | hobbs: drat. there goes that cunning plan. :/ |
| 05:35.15 | CaT[tm] | oops |
| 05:36.13 | JT | make config doesn't guarantee the module you want will autoload |
| 05:36.38 | [T]ank | other than adding it to rc.local, can i cause it to autoload in one of the configs? |
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| 05:55.55 | asterisknerds | <PROTECTED> |
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| 06:56.17 | asterisknerds | <PROTECTED> |
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| 07:17.34 | FlatFoot | morning all |
| 07:17.44 | antimoof | good afternoon. |
| 07:17.50 | FlatFoot | afternoon to you |
| 07:18.14 | FlatFoot | antimoof: where are you located to be in the afternoon ? |
| 07:20.56 | antimoof | weeaboo central - aka tokyo, japan. |
| 07:21.10 | creadurex | tohiooo |
| 07:21.44 | FlatFoot | antimoof: nice weather ? , tis raining again here in good old england |
| 07:22.49 | antimoof | I wouldn't call it "nice"; it's 32C with 62% humidity. |
| 07:24.04 | antimoof | I think it'd be nicer if it were to rain, heh. |
| 07:24.08 | FlatFoot | daft question , i have a blind friend and need to change ringtone per callerid ( up to 30 diff numbers ) Q is ... |
| 07:24.40 | FlatFoot | is there an equivilant Select Case ( VB ) func i can use in extensions.conf ? |
| 07:25.20 | antimoof | sadly, I'm just coming up to speed on asterisk, myself. :-/ |
| 07:25.26 | FlatFoot | antimoof: that weather sounds uncomfortable |
| 07:25.37 | FlatFoot | aha |
| 07:26.48 | antimoof | high of 34 tomorrow. yuck. |
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| 07:56.03 | asterisknerds | <PROTECTED> |
| 07:58.11 | foo | asterisknerds: ? |
| 08:08.30 | *** join/#asterisk kv0s (n=kv0s@p4FD21FAC.dip0.t-ipconnect.de) |
| 08:08.32 | kv0s | Hi! |
| 08:11.31 | Sweeper | anyone looking for some consulting? :3 |
| 08:13.47 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 08:15.05 | kv0s | It is possible to limit my outgoing trunks to maximum channels? for example i'll limit the usage of zapg1 so at any time one channel is free for incoming calls. |
| 08:18.06 | Sweeper | kv0s: sure. use a system call to an asterisk command to get the status of all channels, and then only dial if there's more than one open |
| 08:18.49 | yonahw-work | Sweeper: out of sheer curiosity, what are the going rates for consulting? |
| 08:20.00 | Sweeper | yakkop: I charge $50 an hour for sans-retainer consulting, unless it's between 2100 and 0900, when the price is $60 |
| 08:20.12 | creadurex | 10$ overtime |
| 08:20.16 | Sweeper | dunno about otherpeople |
| 08:20.18 | creadurex | thats... well |
| 08:20.25 | creadurex | i guess you dont value your spare time :P |
| 08:20.42 | Sweeper | creadurex: atm, I have too much spare time and not enough work :) |
| 08:21.36 | jeremy_g | Sweeper:where are you based at |
| 08:21.46 | Sweeper | central florida |
| 08:22.01 | jeremy_g | ok |
| 08:23.33 | creadurex | hehe |
| 08:23.36 | creadurex | i see Sweeper |
| 08:27.49 | Jabeeds | Hi guys! |
| 08:29.13 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 08:30.04 | Jabeeds | I have a problem with a setup I have. Basically, I have a box on a RealWorld Address, that has a trunk setup to another box on a real world address (second one is not mine) and I can make calls between extensions on my box, and make calls from extensions on my box, but when trying to make calls from the realworld to an ext on my box, i get silence. I believe I have pretty much exhausted all possibilities of NAT and firewall issues |
| 08:31.06 | Jabeeds | P.S. there is a SIP Trunk between the two boxes |
| 08:32.19 | Sweeper | Jabeeds: if you're getting silence, it's still probably NAT |
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| 08:33.05 | Sweeper | does your box have an external ip address? |
| 08:33.10 | Jabeeds | Yes |
| 08:33.14 | Jabeeds | REal-World |
| 08:33.27 | *** join/#asterisk PierreY (n=pierre@86.77.99.125) |
| 08:33.53 | Jabeeds | Clients can be behind NAT or on realworld PPPoE, but the same issue exists |
| 08:34.38 | Jabeeds | So the same issue exists when NAT is completely removed from the equation. |
| 08:34.48 | Sweeper | err |
| 08:35.07 | Sweeper | ok, so what you're saying is, you're getting silence when trying to dial across the trunk? |
| 08:35.17 | Sweeper | I was thinking realworld was some kind of termination provider :P |
| 08:36.47 | Jabeeds | lol |
| 08:36.59 | Jabeeds | The trunk works. |
| 08:37.05 | Jabeeds | Outgoing calls are fine |
| 08:37.22 | Jabeeds | incoming are fine when routed to VoicemailMain() |
| 08:37.30 | Jabeeds | Can hear and speak |
| 08:37.42 | Jabeeds | I believe it is some form of problem with the Bridging |
| 08:38.26 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:38.53 | Jabeeds | Because its only, PSTN->Trunk->Asterisk->Exten That is the problem |
| 08:39.25 | Jabeeds | Inter-Exten calls are fine too. |
| 08:40.52 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 08:42.08 | Jabeeds | Any Ideas? |
| 08:42.12 | Sweeper | so Exten -> Remote ASterisk -> trunk -> Local Asterisk -> Exten works fine. What about PSTN -> Remote -> Exten? |
| 08:42.25 | Jabeeds | Thats right |
| 08:42.50 | Jabeeds | Exten -> Remote ASterisk -> trunk -> Local Asterisk -> Exten works fine. |
| 08:43.09 | Sweeper | what about PSTN -> Remote -> Exten? |
| 08:43.32 | Jabeeds | Wait sorry, thats incorrect... |
| 08:44.11 | Jabeeds | Exten -> LOCAL * -> Trunk -> Remote * -> Ext is fine |
| 08:44.34 | Jabeeds | Exten -> LOCAL * -> Trunk -> Remote * -> PSTN is fine also |
| 08:45.00 | Jabeeds | PSTN -> Remote * -> Trunk -> Local * -> Ext is problematic |
| 08:45.21 | Sweeper | ok. are you getting anything on the console? |
| 08:45.41 | Sweeper | or the call just gets set up without any errors, but you can't hear anything? |
| 08:45.42 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:46.33 | Jabeeds | Call goes through fine with no Errors |
| 08:46.36 | Jabeeds | Just no sound. |
| 08:46.47 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:48.09 | Jabeeds | That's why I thought it would be a bridge problem. |
| 08:48.48 | Sweeper | Ext -> Remote * -> Trunk -> Local * -> Ext -<<does this work? |
| 08:49.07 | Jabeeds | Unsure, as the Remote * is not mine. |
| 08:49.14 | Sweeper | mkay |
| 08:49.23 | Sweeper | if it were a codec problem, it wouldn't work |
| 08:49.43 | Sweeper | and outbound wouldn't work either |
| 08:50.05 | Jabeeds | Occasionally on one extensions setup with X-Lite, I can get 1Way Audio |
| 08:50.41 | Jabeeds | The Exten can hear what comes from the PSTN, but not the other way. |
| 08:50.41 | Sweeper | occasionally? that complicates things :P |
| 08:50.46 | Jabeeds | Yeah |
| 08:51.23 | Jabeeds | Its on a laptop and depends on where is is setup etc. |
| 08:51.46 | Jabeeds | on a lexed network it will exhibit the one way |
| 08:52.15 | Sweeper | oh, so not occasionally |
| 08:52.19 | Jabeeds | no |
| 08:52.27 | Sweeper | jsut depends where you set it up :) |
| 08:52.32 | Jabeeds | correct |
| 08:52.55 | Jabeeds | I meant, sometimes It can work, Sometimes it can't :P |
| 08:52.56 | Sweeper | ok, turn reinvites off on your local extensions, in sip.conf |
| 08:53.02 | Jabeeds | Tried it. |
| 08:55.38 | Jabeeds | Reinvites are turned off for all local extensions and for the trunk. |
| 08:55.40 | phearless | damned asterisk |
| 08:58.00 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
| 09:00.57 | Jabeeds | I am running * version 1.4.9 |
| 09:03.22 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:04.37 | Jabeeds | No other Ideas Sweeper? |
| 09:07.37 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
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| 09:12.05 | SuurMyy | it seems that there are some unsolved lisence issues w/Asterisk ? |
| 09:12.12 | SuurMyy | Anyone care to comment ? |
| 09:12.30 | SuurMyy | <PROTECTED> |
| 09:14.25 | jql | it's theoretically possible that you don't have permission to redistribute asterisk as a whole |
| 09:14.43 | jql | but that doesn't mean you're stopped from downloading or using it in any way |
| 09:14.57 | SuurMyy | we have a commercial use |
| 09:15.17 | SuurMyy | the question being that do we face potential legal trouble if we use asterisk |
| 09:15.19 | jql | commercial use doesn't change anything. it only matters if you're distributing it |
| 09:15.50 | jql | as in, making an astrick appliance and selling it to people |
| 09:15.55 | jql | or something like that |
| 09:16.06 | HarryR | for commercial internal only use - as long as it stays under GPL v2 you can do almost anything you like |
| 09:16.10 | SuurMyy | yes, we'd like to use asterisk as a part of our product |
| 09:16.37 | SuurMyy | as in installing asterisk pbx linux box into our clients' premises |
| 09:16.45 | HarryR | then include a printed version of the GPL license and the source code for everything on CD |
| 09:16.48 | SuurMyy | to support the needs of our other system components |
| 09:17.09 | SuurMyy | we need to give that to the customer ? or need to give it out to the world ? |
| 09:17.32 | HarryR | or just say it uses GPL asterisk, if you want the source code you can get it off the digium site |
| 09:17.49 | HarryR | unless you modify the source yourself, in which case you must make your changes available in source form to your customer |
| 09:17.56 | SuurMyy | say, I install debian and compile asterisk on it and configure it - is it okay to say that you can download debian and asterisk from their sites ? |
| 09:18.04 | HarryR | yah |
| 09:18.24 | SuurMyy | and if we change the source, we need to give it out to a) customer b) world ? |
| 09:18.32 | SuurMyy | or a&&b |
| 09:18.50 | jql | well, you aren't allowed to prevent b. |
| 09:18.57 | HarryR | only to the customer you distribute it to, but they must have the same rights as you (e.g. the changes you make must be released under the GPL) |
| 09:19.05 | jql | so it's effectively a&&b, although you yourself only have to do a |
| 09:19.09 | HarryR | yah |
| 09:19.15 | SuurMyy | fine |
| 09:19.23 | SuurMyy | how about config files ? |
| 09:19.29 | HarryR | their not covered.. |
| 09:19.35 | HarryR | do whatever you want with them |
| 09:19.40 | jql | config files aren't GPLed |
| 09:19.54 | SuurMyy | So we don't need to give out our customers' internal phone book. :) |
| 09:20.00 | HarryR | ofcourse not |
| 09:20.04 | SuurMyy | That wouldda been rather ridiculous :) |
| 09:20.08 | jql | indeed |
| 09:20.48 | SuurMyy | but, if we change sources, they are gpl-tainted, so we have to give em to our customer. And if willing, they can give them out to the world, if they so wish ? |
| 09:20.55 | HarryR | yes |
| 09:21.01 | SuurMyy | okay - |
| 09:21.18 | Sweeper | Jabeeds: sorry, no. it still seems like you should consider it a NAT issue unless you can verify that remote ext to local ext works |
| 09:21.42 | SuurMyy | let's say I write anonther probram that does Z and interfaces to asterisk, so this is a way around the gpl. I just must not include the sources in source files or linking ? |
| 09:22.12 | jql | asterisk has AGI and AMI protocols which you can use to control it |
| 09:22.22 | SuurMyy | Yes. I think we can live w/that. |
| 09:22.24 | jql | any AGI/AMI programs you write belong to you |
| 09:22.33 | Sweeper | as do any modules you might write |
| 09:22.38 | SuurMyy | oh ? |
| 09:22.41 | SuurMyy | in the apps - dir ? |
| 09:22.45 | Sweeper | yep |
| 09:22.48 | SuurMyy | ah. |
| 09:22.55 | SuurMyy | So gpl not an issue at all here. |
| 09:22.57 | HarryR | Sweeper, nah, the modules need to be GPL2 |
| 09:23.04 | SuurMyy | oh ? |
| 09:23.06 | Sweeper | HarryR: surely not |
| 09:23.13 | SuurMyy | <PROTECTED> |
| 09:23.16 | SuurMyy | <PROTECTED> |
| 09:23.18 | SuurMyy | :) |
| 09:23.38 | HarryR | it directly links with asterisk and is entirely dependant on asterisk - so it needs to be GPL |
| 09:23.54 | Sweeper | oh, n/m |
| 09:24.00 | SuurMyy | okay. |
| 09:24.03 | Sweeper | actually, it just needs to be gpl compatible |
| 09:24.09 | SuurMyy | So apps/ <-> >=gpl2 |
| 09:24.10 | HarryR | argh damn irc |
| 09:24.13 | Sweeper | so it can be lgpl or whatever |
| 09:24.20 | HarryR | nope, has to be GPL :\ |
| 09:24.32 | SuurMyy | Well that's okay, I guess. |
| 09:24.55 | SuurMyy | The thing that sucks a bit is that we're a bleeding edge company, so we don't really want to help the competition. |
| 09:25.12 | jql | yeah. you might be able to write a codec or format file that wasn't GPL (with a small GPL wrapper), but apps look like derivitive works |
| 09:25.52 | SuurMyy | But I don't think it's too bad. I know a lot of ppl who are more than capable of writing C that is write-only ;) |
| 09:26.04 | HarryR | or just do stuff through fastagi/manager instead :) |
| 09:26.31 | jql | or write something that has to be glued together with horrific dialplan voodoo, which you don't license |
| 09:26.48 | jql | heh |
| 09:26.50 | SuurMyy | :) |
| 09:26.57 | SuurMyy | that's evil :D |
| 09:26.57 | Sweeper | most stuff you want to do can probably be done via fastagi/manager, honestly |
| 09:27.06 | SuurMyy | yea |
| 09:27.12 | jql | yeah, I run my operation almost entirely via fastagi & manager |
| 09:27.15 | SuurMyy | thank you all very much |
| 09:27.18 | jql | the only source I change is tweaks |
| 09:27.23 | SuurMyy | you were very helpful |
| 09:27.26 | jql | tweak the nat handling, tweak music on hold |
| 09:27.34 | jql | not really writing new queue functions and stuff |
| 09:27.38 | jql | I can do that via agi |
| 09:27.50 | SuurMyy | ...I needed to know this so I can sell the software to our sales-ppl :) |
| 09:29.04 | SuurMyy | ...I dunno what happened, I was supposed to be in R&D to avoid selling anything, but what can you do. |
| 09:31.31 | *** join/#asterisk shinao1 (n=shinao1@41.219.201.187) |
| 09:34.42 | jeremy_g | jql:dialplan vodoo lol |
| 09:39.10 | SuurMyy | I dunno if it means anything to you, but we'll be using Asterisk in a context where it will actually be a part of a system helping the elderly. |
| 09:39.22 | SuurMyy | Just thought it might be nice to know. |
| 09:41.34 | *** join/#asterisk guillote_GNU (n=guillote@host140.200-117-181.telecom.net.ar) |
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| 09:45.10 | HarryR | SuurMyy, as long as it doesn't go tits up when somebodies life depends on it - you're fine |
| 09:45.32 | HarryR | but either way, their elderly so it's not a big deal |
| 09:45.44 | SuurMyy | lol w/you |
| 09:46.09 | SuurMyy | well actually if the system malfunctions, it just means an extra ambulance being sent over, not a big deal. |
| 09:46.25 | SuurMyy | There will always be humans making the actual decisions. |
| 09:46.48 | SuurMyy | So more ppl will of course die because of it, but at least there's always someone to blame %] |
| 09:47.06 | HarryR | what'd be really cool is to have rfid+heartbeat monitoring |
| 09:47.18 | HarryR | that automagically dispatches an ambulance and notifies staff of which room their in |
| 09:47.38 | HarryR | and a cellphone to get geolocation if their outside |
| 09:49.15 | SuurMyy | Well, I could talk more about it, but then I'd have to kill you or pay 500.000 euros to my company. :) |
| 09:50.23 | jeremy_g | yeah kill him |
| 09:50.28 | SuurMyy | lol |
| 09:51.37 | HarryR | lol |
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| 09:54.45 | phearless | hey guys ! |
| 09:54.54 | phearless | anybody know a GUI to redirect calls ? |
| 09:55.04 | phearless | non-flash based if possible |
| 09:55.17 | phearless | I would like to see some source code that redirect calls |
| 09:55.35 | phearless | because i can't add a redirect feature in my interface |
| 09:56.13 | *** join/#asterisk nighty^ (n=nighty@p3132-adsau16honb13-acca.tokyo.ocn.ne.jp) |
| 09:56.52 | phearless | not redirect, but transfer |
| 09:56.53 | phearless | sorry |
| 10:00.41 | *** part/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 10:02.08 | jeremy_g | phearless:we can add that for you |
| 10:02.10 | jeremy_g | :p |
| 10:02.44 | *** join/#asterisk s1gny|wrk (n=s1gny@91.65.152.80) |
| 10:02.50 | *** part/#asterisk s1gny|wrk (n=s1gny@91.65.152.80) |
| 10:03.51 | phearless | did you read my question ? |
| 10:06.50 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:17.21 | *** join/#asterisk dagrim (n=_dagrim_@unaffiliated/dagrim) |
| 10:17.57 | dagrim | Anybody have a recommendation for an affordable prepaid iax provider jus to mess around with *? |
| 10:18.21 | dagrim | that offers toll free dids? |
| 10:26.49 | HarryR | dagrim, voiptalk.org ? |
| 10:38.55 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:42.26 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:53.33 | dagrim | cool thanks |
| 10:54.28 | Aurs | will bindaddr=0.0.0.0 work with virtual ip addresses in 1.4? I've read something saying that asterisk will always use the first ip on one nic if you don't specify ip addr in bindaddr |
| 10:57.35 | dagrim | hmm not quite what i was looking for.. |
| 11:03.24 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 11:07.34 | Sweeper | bahaha |
| 11:07.41 | Sweeper | yay for creepy intercom capabilities |
| 11:08.34 | dagrim | lol |
| 11:11.01 | dagrim | AA is just as dangerous |
| 11:11.01 | dagrim | lol |
| 11:11.07 | HarryR | AA? |
| 11:11.11 | dagrim | autoanswer |
| 11:11.13 | HarryR | ah |
| 11:11.35 | dagrim | lets just say i share an office with 5 people |
| 11:11.35 | dagrim | lol |
| 11:14.29 | phearless | "Redirect" does not work |
| 11:14.34 | phearless | anybody tried to use it ? |
| 11:15.31 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 11:25.31 | *** join/#asterisk roryclw (n=rory@213.165.225.159) |
| 11:26.09 | roryclw | Hi. I'm an asterisk newbie. I'm trying to configure a new core 2 duo machine with a Sangoma card. I'm having trouble compiling wanpipe |
| 11:26.38 | roryclw | I realise this is an asterisk forum, not a wanpipe forum, but help would be much appreciated! |
| 11:27.21 | roryclw | I wonder if the problem might be that I am using the following kernel: 2.6.21-2-amd64 |
| 11:27.30 | roryclw | which is too new for Sangoma's wanpipe... |
| 11:27.41 | HarryR | it's likely :) |
| 11:28.01 | roryclw | HarryR: are you using wanpipe? |
| 11:28.05 | HarryR | uh, and why are you ru nning an amd64 kernel on a core2 duo? |
| 11:28.21 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca) |
| 11:28.25 | HarryR | nope, I cant use sangoma hardware at work |
| 11:28.39 | roryclw | its a generic 64 bit version for core 2 machines. |
| 11:29.00 | roryclw | I believe |
| 11:29.00 | HarryR | core2duo supports EMT64? |
| 11:30.37 | roryclw | HarryR: "If your PC has a 64-bit AMD or Intel processor, you will most likely need the "amd64" images" : http://www.debian.org/CD/faq/#which-cd |
| 11:31.14 | HarryR | and you have more than 4gb of ram? |
| 11:31.24 | roryclw | yes, 6gb of ram |
| 11:31.30 | HarryR | ah :\ |
| 11:32.46 | roryclw | HarryR: that is a problem? |
| 11:33.28 | HarryR | nah was just suprised that core2duo machines supported emt64 |
| 11:34.53 | roryclw | HarryR: yeah! Any other thoughts about the sangoma/kernel level issue? |
| 11:39.08 | jer | HarryR, yes it supports EM64T |
| 11:39.14 | jer | it's not EMT64 |
| 11:39.58 | SuurMyy | is there a nice timeline somewhere about Asterisk, it's versions from like 0.1 to today ? |
| 11:40.02 | CaT[tm] | just call it amd64 - same difference :) |
| 11:40.10 | CaT[tm] | and easier to type |
| 11:40.59 | roryclw | any thoughts about sangoma's wanpipe and a 2.6.21 kernel? |
| 11:43.06 | jer | CaT[tm], but it's not amd64; it changes some things like the segment based memory model, adds new instructions |
| 11:43.14 | jer | that would be like calling ia64 an alpha =] |
| 11:44.38 | roryclw | quite. But Debian's amd64 kernel versions "The port consists of a kernel for all AMD 64bit CPUs with AMD64 extension and all Intel CPUs with EM64T extension, and a common 64bit userspace." |
| 11:46.20 | CaT[tm] | jer: yes, well... I'll just have to kill everyone who knows of the differences. still easier then the intel name :) |
| 11:46.48 | jer | sorry, reading technical documentation is a hobby =] |
| 11:46.56 | *** join/#asterisk MadUni (n=igor@motorola154-31.ip.PeterStar.net) |
| 11:50.27 | MadUni | hi all.. can someone help me to call googleTalk via Asterisk? I've configured everything like it is written here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk , but when I call to someone, I have the problem that is mentioned in this wiki. No audio, phone keeps ringing... but the solution from wiki didn't help me :( |
| 11:54.15 | HarryR | I was having issues with no audio using Asterisk and Jingle too |
| 11:54.49 | HarryR | but not going through googletalk, just a local jabber server |
| 11:56.05 | *** join/#asterisk vgster (n=vgster@91.103.131.90) |
| 11:59.03 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:59.27 | *** join/#asterisk RGi_26 (i=RGi@62.97.247.44) |
| 11:59.57 | RGi_26 | I have a problem calling out with a Cisco 7920 Wireless phone.. using SCCP.... when I call out It calls out.. but when it starts to ring at the other end.. asterisk is ending the call.. |
| 12:02.30 | *** join/#asterisk jmav8 (n=jmav@201.242.58.134) |
| 12:02.40 | jmav8 | Good morning |
| 12:03.10 | MadUni | HarryR, and did you solve it? |
| 12:03.46 | jmav8 | i have a question its anyway posible to limit the time a xten its on a trunk call and hang up? |
| 12:05.42 | *** part/#asterisk roryclw (n=rory@213.165.225.159) |
| 12:06.41 | *** join/#asterisk joeTSUNAMI (n=nnjoeTSU@unaffiliated/joetsunami) |
| 12:06.57 | joeTSUNAMI | good morning! |
| 12:07.44 | joeTSUNAMI | does anyone know how to reject calls without caller id info or marked "private"? |
| 12:10.39 | *** join/#asterisk potsboy (n=chrisg@dsl-146-24-120.telkomadsl.co.za) |
| 12:11.28 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:12.11 | *** part/#asterisk potsboy (n=chrisg@dsl-146-24-120.telkomadsl.co.za) |
| 12:13.54 | *** join/#asterisk ehaupt (n=ehaupt@unaffiliated/ehaupt) |
| 12:15.47 | ehaupt | i changed bindport to 8080 but there are still some other ports asterisk listens on. those are 2000/tcp, 1720/tcp, 2727/udp, 4520/udp, 4569/udp |
| 12:16.15 | ehaupt | can i change those as well? how? |
| 12:16.59 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 12:17.10 | ehaupt | or at least the ones that are relevant for a softphone to connect |
| 12:18.47 | HarryR | does sip prescense & instant messaging work with asterisk? |
| 12:19.18 | HarryR | e.g. if I have a softphone that supports it, at the moment we're using SER which works fine, will asterisk be able to do the same? |
| 12:19.20 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 12:27.38 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:27.38 | *** join/#asterisk menil (n=root@line103-8.adsl.actcom.co.il) |
| 12:27.38 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 12:37.54 | keulin | asterisk is a very good software |
| 12:39.48 | iCEBrkr | LIES |
| 12:42.08 | [TK]D-Fender | HarryR: Presence = yes, IM = no |
| 12:42.15 | HarryR | yah I just figured it out |
| 12:42.36 | [TK]D-Fender | ehaupt: those are MGCP, IAX2, jabber, etc |
| 12:44.36 | ehaupt | [TK]D-Fender: those won't be necessary for xlite, right? |
| 12:44.45 | ehaupt | [TK]D-Fender: i know jabber isn't |
| 12:44.54 | [TK]D-Fender | ehaupt: correct |
| 12:45.14 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 12:46.25 | jeremy_g | keulin:yes it is a very good software. LOL |
| 12:46.37 | *** part/#asterisk PierreY (n=pierre@86.77.99.125) |
| 12:46.51 | jeremy_g | keulin:kindergar..? |
| 12:50.23 | *** join/#asterisk lirakis (n=eric@69.24.142.1) |
| 12:50.30 | lirakis | morning |
| 12:53.46 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 12:55.23 | *** join/#asterisk y7n (n=na@office.intercea.co.uk) |
| 12:58.35 | y7n | If a call reaches TIMEOUT(absolute): exten => s,1,Set(TIMEOUT(absolute)=3600) How would I invoke a system command when this TIMEOUT has been reached? |
| 12:59.36 | y7n | Would exten => s,2,System() be executed, or would this be ignored? |
| 13:00.52 | [TK]D-Fender | y7n: it would be executed NEXT |
| 13:01.14 | [TK]D-Fender | y7n: After about 1/10 of a second it takes the FIRST priority to execute |
| 13:01.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:01.36 | y7n | thanks :) |
| 13:03.05 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 13:07.52 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.36 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:09.36 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:16.55 | *** join/#asterisk Modcuts (n=modcuts@lan.proporta.com) |
| 13:18.59 | *** join/#asterisk Ebola (n=Ebola@host86-139-49-76.range86-139.btcentralplus.com) |
| 13:19.34 | *** join/#asterisk zydrunas (n=Miranda@216.49.228-ip-184.ckt.net) |
| 13:23.31 | *** join/#asterisk lbow (n=lbow@dsl-146-5-187.telkomadsl.co.za) |
| 13:24.42 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:28.09 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:30.42 | *** join/#asterisk guillote_GNU (n=bancaria@host128.201-253-17.telecom.net.ar) |
| 13:30.54 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:31.40 | mosty | when a caller is put on hold, they hear the hold music file from the start of the file. is there a way for asterisk to remember where the music was up to last time they were on hold, and play from that position instead of the start? |
| 13:32.41 | GromiTM | The expression "exten => s,1,NoOp(${ARG3})" give me only "pbx.c: Launching 'GotoIf'" with "app_macro.c: Executed application: NoOp" but not the content of ARG3. |
| 13:33.39 | mosty | GromiTM, what level is verbose and debug set to? |
| 13:33.42 | zeeesh | WARNING[31553]: chan_sip.c:15690 handle_common_options: insecure=very at line 0 is deprecated; use insecure=port,invite instead? |
| 13:33.46 | iCEBrkr | mosty: I thought it only played from the start of the file the first time MoH starts up. |
| 13:33.58 | GromiTM | mosty: set to 128 |
| 13:34.06 | iCEBrkr | mosty: after that, I swear it just continuously "plays" |
| 13:34.31 | mosty | iCEBrkr, that's what i thought also, but a client with an asterisk pbx claims otherwise |
| 13:35.43 | iCEBrkr | I suppose that's the way it used to work.. I really don't know the deal with MoH these days. |
| 13:35.51 | iCEBrkr | hell, I've been pretty much out of the loop anymore. |
| 13:36.06 | iCEBrkr | Life got busy and my Asterisk box just runs. |
| 13:36.30 | mosty | mine too. but my customers seem to have issues from time to time |
| 13:36.34 | JT | mosty: i believe you need to use mpg123 instead of the native player |
| 13:37.14 | lmadsen | zeeesh: message is pretty straight format -- change insecure=very to insecute=port,very in sip.conf |
| 13:37.37 | iCEBrkr | JT: Yeah, that's kinda what I was thinking, I know back in the day with mpg123/321 it just "played" in the background continuously. |
| 13:39.04 | mosty | jt: hmm, i'll try that out when i can test the system |
| 13:40.05 | mosty | is there something i can use to play a ulaw file instead of mpg123/mp3? |
| 13:41.35 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-4c637a3763810fa2) |
| 13:49.33 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 13:49.59 | kombi | and I thought polycoms were less of a pain then ciscos.. |
| 13:50.34 | kombi | polycoms boot from ftp, not tftp <- correct? |
| 13:50.55 | JT | tftp, ftp, http, https, whatever you want |
| 13:51.23 | kombi | really? meaning whatever they find first in dhcpd.conf? |
| 13:52.13 | kombi | tftpd would make more sense since the phone coming out of the box does not know any credentials for like an ftp server.. |
| 13:53.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:54.34 | *** join/#asterisk Strom_M (n=strom@bas5-montreal28-1177917181.dsl.bell.ca) |
| 13:55.22 | zeeesh | <lmadsen>: before changing insecure=very into port ... i was getting atleast error but after changing there is no even error? |
| 13:56.33 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 13:57.22 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-126-140-84.dsl.irvnca.pacbell.net) |
| 14:01.11 | [TK]D-Fender | kombi: They default to FTP, but you set that in the bootrom. And I'd be amazed if your Polycom phone could actually READ dhcpd.conf... how would it do that? Guessing your root PW on SSH? ;) |
| 14:02.03 | Sweeper | well, sip firmware 3.0 contains a quantum processor and an AI |
| 14:02.13 | Sweeper | that's technology for ya! |
| 14:03.03 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 14:03.37 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 14:04.14 | Sweeper | obviously kombi has been hacking into polycom's "secure" ftp! |
| 14:04.19 | kombi | Hi Fender! did I say read? I meant get it from there..(as in next-server) |
| 14:04.29 | *** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:04.29 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 14:05.07 | [TK]D-Fender | kombi: You can specify the address, but the server TYPE in DHCP? Not that I know of... |
| 14:05.16 | kombi | Sweeper: I tamed the bloody 7941, on to the ip430.. |
| 14:05.44 | Sweeper | 430 doesn't need taming, you just need to whisper sweet xml into its ear... |
| 14:05.46 | kombi | Fender: correct |
| 14:06.02 | [TK]D-Fender | GromiTM: Have you considered the possibilty that ARG# is *NULL*? |
| 14:06.03 | kombi | phone whisperer is on duty! |
| 14:06.07 | [TK]D-Fender | ARG3 |
| 14:06.09 | Sweeper | dhcp hands out addresses for tftp, iirc |
| 14:06.31 | [TK]D-Fender | Sweeper: You can do a LOT more with Polycom & DHCP |
| 14:06.44 | [TK]D-Fender | Sweeper: You can set SIP user & PASS and much more. |
| 14:06.57 | Sweeper | [TK]D-Fender: ew |
| 14:08.23 | *** part/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 14:09.03 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:09.20 | [TK]D-Fender | But I don't see a boot server TYPE option. Everything points to TFTP. However Polycoms will use the ADDRESS provided for TFTP server as the ip/host for whatever kind it was TOLD to use. |
| 14:10.28 | kombi | do config files differ for 430 and 501? |
| 14:12.59 | [TK]D-Fender | kombi: Nope. There are some model specific options, but if left to defaults works jsut fine. Make sure to configure your line-key usage & regs properly and the rest is fine |
| 14:15.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:15.46 | *** join/#asterisk prizark_ (i=prizark_@bb121-6-133-90.singnet.com.sg) |
| 14:19.11 | *** join/#asterisk sheppard (n=sheppard@209.89.51.41) |
| 14:23.36 | *** join/#asterisk dioedu (n=dioedu@c915230f.virtua.com.br) |
| 14:24.10 | Strom_M | [TK]D-Fender: so...where in montreal should I go for killer poutine? :) |
| 14:24.38 | [TK]D-Fender | Strom_M: Looking close now :) |
| 14:24.40 | dioedu | hi all, does someone know, in asterisk moh, how * "choose" the first file to play when i have random=no in my configuration ? |
| 14:25.03 | Strom_M | [TK]D-Fender: ? |
| 14:25.10 | [TK]D-Fender | Strom_M: Usual best places : "Lafleurs" or "La Belle Province" |
| 14:25.15 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:25.19 | [TK]D-Fender | Strom_M: You're looking to be here already :) |
| 14:25.19 | dioedu | i have music1.raw, music2.raw ... music9.raw |
| 14:25.27 | Strom_M | [TK]D-Fender: yes, i arrived yesterday |
| 14:25.42 | [TK]D-Fender | Strom_M: Neato. Work or vacation? |
| 14:25.51 | *** join/#asterisk ManxPower (n=manxpowe@032-447-153.area7.spcsdns.net) |
| 14:25.52 | Strom_M | vacation |
| 14:25.57 | Strom_M | going back to vermont tonight |
| 14:26.11 | [TK]D-Fender | Strom_M: I oughtta try that sometime before I burn out here... |
| 14:26.16 | dioedu | in some tests, * got music1.raw, other time, got music9.raw and now get music7.raw |
| 14:26.49 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:27.09 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 14:28.03 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:28.03 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 14:28.31 | *** join/#asterisk Liquinn (n=Dream@unaffiliated/liquinn) |
| 14:28.42 | [TK]D-Fender | Strom_M: What time you heading back? |
| 14:28.52 | *** part/#asterisk Liquinn (n=Dream@unaffiliated/liquinn) |
| 14:29.10 | [TK]D-Fender | Strom_M: You'll want to keep an eye out for rush hour on the bridges. 3-7pm = suck |
| 14:30.03 | *** join/#asterisk davidm870 (n=davidm87@66.150.12.150) |
| 14:30.10 | Strom_M | [TK]D-Fender: my friend and I are discussing departure...we may stay another night |
| 14:32.10 | kombi | Fender: Config file error 0x10020 |
| 14:33.29 | *** join/#asterisk madgyver (n=madgyver@gprs-pool-1-012.eplus-online.de) |
| 14:33.48 | madgyver | hello there |
| 14:34.43 | madgyver | is there a way to use my cellphone with asterisk? |
| 14:35.14 | kombi | you can call.. |
| 14:36.06 | madgyver | the idea is to attatch multiple cellphones to a asterisk-server and use the cheapest provider to create the connection. could I use blootooth for this? |
| 14:36.30 | mvanbaak | madgyver: yup, chan_mobile in asterisk-addons |
| 14:37.06 | madgyver | I'll have a look (is this correct english?) |
| 14:37.18 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:38.38 | [TK]D-Fender | kombi: Look at your app log |
| 14:39.16 | kombi | 0x4020.. tail -f is on |
| 14:39.34 | [TK]D-Fender | kombi: Go double check everything |
| 14:39.53 | [TK]D-Fender | kombi: And start pastebinning if you expect any further help... |
| 14:40.05 | kombi | ok.. |
| 14:40.17 | madgyver | find someone else to doublecheck... doing it yourself won't solve problems ;) |
| 14:40.44 | madgyver | someone in RL! .... |
| 14:40.56 | kombi | I seems I don't have the proper configs to start from, anywhere I can dl a set? |
| 14:41.19 | ManxPower | kombi: you can't. |
| 14:41.36 | kombi | I try the ones on voip-info then.. |
| 14:41.43 | ManxPower | The sample asterisk configs are designed to show every available option. They are not designed to be working config files. |
| 14:42.16 | GromiTM | [TK]D-Fender: I will check it. |
| 14:43.01 | kombi | Manx: I meant the polycom cfg ones, not anything asterisk.. |
| 14:43.04 | madgyver | voip-info is a great ressource. we got asteriskup and running in about 90 minutes (45 minutes hacking SUSE... BAH!) |
| 14:43.20 | GromiTM | [TK]D-Fender: I want to say, that I have to check, wethere the variable is NULL or not. You mean, if I do not get anything with NoOp(), the variable is NULL? |
| 14:43.26 | [TK]D-Fender | kombi: From your reseller |
| 14:43.45 | ManxPower | GromiTM: a varible is null ifit is empty. |
| 14:43.50 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:43.51 | [TK]D-Fender | GromiTM: Correct |
| 14:44.11 | ManxPower | GotoIf($["${MYVAR} = ""]:5?9) |
| 14:44.13 | GromiTM | hmmm ok |
| 14:44.22 | ManxPower | ..er. |
| 14:44.29 | ManxPower | GotoIf($["${MYVAR}" = ""]:5?9) |
| 14:44.43 | ManxPower | that is from memory |
| 14:44.44 | GromiTM | That would answer some questions and gives me a new one.... |
| 14:45.05 | Corydon76-dig | ManxPower: swap the ? and : |
| 14:45.07 | GromiTM | I will check it... *try* |
| 14:45.12 | [TK]D-Fender | mvanbaak: I don't see chan_mobile in my asterisk-addons (1.4.2)..... |
| 14:45.22 | GromiTM | Corydon76-dig: yep |
| 14:45.36 | ManxPower | Corydon76: I would have to look it up |
| 14:46.05 | Corydon76-dig | ManxPower: no, I'm saying the ? needs to be first |
| 14:46.27 | ManxPower | Corydon76-dig: I'm sure you are correct. |
| 14:46.33 | [TK]D-Fender | GotoIf($["${MYVAR}"=""]?5:9) |
| 14:48.04 | ManxPower | [TK]D-Fender: won't that only work with 1.4 (the lack of spaces around the =)? |
| 14:48.26 | [TK]D-Fender | ManxPower: 1.0+ <--- |
| 14:48.52 | [TK]D-Fender | ManxPower: And thats only because I can't remember pre 1.0 personally :) |
| 14:49.16 | mvanbaak | [TK]D-Fender: it is in trunk |
| 14:49.55 | [TK]D-Fender | mvanbaak: Oh you mean the "Install this if you don't value your job" trunk? :) |
| 14:50.25 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 14:50.41 | mvanbaak | gheh, isn't that the case with all asterisk versions ? |
| 14:51.05 | Mercestes | mvanbaak, I dunno, 1.0.9 was pretty nice. |
| 14:51.23 | techman97_andy | mornin' all. I'm having a weird issue with a TE212P / PRI. I have my timeout set in the dialstring to be 300 seconds, but every single time, to the second...my calls hang up on me with a DIALSTATUS=CANCEL after 60 seconds. Any ideas of what I can check? |
| 14:51.47 | techman97_andy | the provider has sworn up and down they don't have any kind of timeout on their PRI. |
| 14:51.51 | *** join/#asterisk masterisk (n=mascool@70.88.122.206) |
| 14:51.54 | *** join/#asterisk LeKstor (n=tdavid@84.5.38.209) |
| 14:52.02 | LeKstor | salut |
| 14:52.09 | Mercestes | techman97_andy, And if you increase your timeout from 300 to 600?? Does it have any effect? |
| 14:52.09 | [TK]D-Fender | techman97_andy: "By the tongues of a thousand serpents this time I do not lie!" |
| 14:52.09 | masterisk | salut |
| 14:52.16 | LeKstor | je sais que y en a 2 3 qui touchent asterisk ici |
| 14:52.17 | ManxPower | techman97_andy: pastebin the CLI output of one of those calls. |
| 14:52.18 | Mercestes | slut |
| 14:52.20 | techman97_andy | rofl |
| 14:52.30 | ManxPower | and I don't want to see and sissyass macros. |
| 14:52.30 | LeKstor | quand on parle du loup ... |
| 14:52.32 | techman97_andy | Mercestes: no effect. 60 seconds, each and every time. |
| 14:52.39 | Mercestes | techman97_andy, and if you remove timeput? |
| 14:52.45 | Mercestes | s/timeput/timeout/ |
| 14:52.45 | masterisk | is astdb still in DB2 format in asterisk 1.4.4 ? |
| 14:52.50 | mvanbaak | Mercestes: uhhuh. we are still running that on some machines |
| 14:53.05 | LeKstor | hmm sorry, wrong language |
| 14:53.07 | techman97_andy | jbot: hmmm, never thought of *removing* the timeout |
| 14:53.08 | LeKstor | hi all |
| 14:53.19 | Mercestes | LekStro: Ah, êtes-vous français ? |
| 14:53.20 | [TK]D-Fender | techman97_andy: Pastebin your dialplan and CLI out put at verbose 10 & PRI debug enabled. |
| 14:53.38 | LeKstor | oui |
| 14:53.42 | Mercestes | LeKstor, Est-toi un garçon ou une fille ? |
| 14:53.48 | ManxPower | [TK]D-Fender: seems like overkill at this point. |
| 14:54.02 | masterisk | is astdb still in DB2 or DB3 format in asterisk 1.4.4 ? |
| 14:54.11 | russellb_ | db1 |
| 14:54.13 | LeKstor | Mercestes : geek |
| 14:54.17 | Mercestes | LeKstor, Quel âge avez-vous ? |
| 14:54.19 | masterisk | oh ok |
| 14:54.22 | masterisk | sucks |
| 14:54.23 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:54.28 | masterisk | for me |
| 14:54.36 | LeKstor | i'm facing the following error " |
| 14:54.38 | LeKstor | " |
| 14:54.43 | Mercestes | :) |
| 14:54.46 | [TK]D-Fender | ManxPower: I like to think of it as "just enough kill". That would hold all the answers, as it always does and people come in here whining and don't even have that handy. How retarded is that? "No I didn't think to LOOK at the problem or have backup ready!" |
| 14:54.56 | LeKstor | "No route to destination" |
| 14:55.08 | [TK]D-Fender | Mercestes: va-t'ens tabarnac :) |
| 14:55.18 | Mercestes | [TK]D-Fender, me and google translate are dangerous together. |
| 14:55.34 | LeKstor | i have only 2 users registreted |
| 14:55.42 | Mercestes | I can offend in over 100 languages. |
| 14:55.43 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 14:55.44 | LeKstor | quite simple configuration for testing |
| 14:55.47 | [TK]D-Fender | Mercestes: You are dangerous all by yourself, Google just allows you to butcher another language in the process :p |
| 14:55.52 | ManxPower | LeKstor: "sip show peers" does the output show the IP addresses of the 2 devices? |
| 14:55.53 | Mercestes | [TK]D-Fender, Yes! |
| 14:55.59 | LeKstor | yes |
| 14:56.20 | LeKstor | io have the two users, with the status "lagged" |
| 14:56.21 | ManxPower | LeKstor: paste the just the actual Dial line in your CLI output. |
| 14:56.31 | [TK]D-Fender | This weekend's plan : Server upgrade to CentOS 5.0 |
| 14:56.37 | ManxPower | LeKstor: lagged means "the connection is too slow, I won't send calls to them) |
| 14:56.45 | [TK]D-Fender | LeKstor: Sounds like a qualify timeout.. not a good sign |
| 14:56.52 | LeKstor | the port seems to be correct, i'm using stun because my 2 clients |
| 14:56.54 | LeKstor | oh |
| 14:56.59 | ManxPower | This weekend's plan: Have fun, get laid. |
| 14:57.01 | LeKstor | great |
| 14:57.08 | Mercestes | This weekend's plan: Rig 60v of DC flowing upward my cable line into the Comcast routers in a terrorist attempt to get back at them for sucking so very hard. |
| 14:57.15 | LeKstor | so with le lagged status, no calls are routed |
| 14:57.18 | LeKstor | that's it ? |
| 14:57.19 | [TK]D-Fender | LeKstor: You likely have "qualify=yes" set for them. Try increasing it (default is 2000ms) to "qualify=3000" and see if that helps |
| 14:57.26 | LeKstor | hmm |
| 14:57.43 | LeKstor | i've put a really short qualify |
| 14:57.51 | Mercestes | LeKstor, don't do that. |
| 14:57.53 | ManxPower | LeKstor: don't do that! |
| 14:57.55 | [TK]D-Fender | LeKstor: It'll "help" if you're jsut marginally over the line. I wouldn't suggest past 5 s. |
| 14:57.56 | Mercestes | :D |
| 14:58.14 | [TK]D-Fender | Go get'em guys! |
| 14:58.39 | ManxPower | the qualify= does NOT set how often to qualify -- it sets the max response time to get a response from the device when sending a SIP OPTIONS (qualify) packet. |
| 14:58.43 | lirakis | is there a application to log agents off? |
| 14:59.03 | ManxPower | lirakis: "show applcations" did not help? |
| 14:59.08 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 14:59.08 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 14:59.18 | *** join/#asterisk saftsack (n=saftsack@217.224.91.120) |
| 14:59.28 | LeKstor | the status is now OK |
| 14:59.31 | LeKstor | looks better |
| 15:00.11 | LeKstor | thanks you so much |
| 15:00.23 | LeKstor | "sip show peers" seems to be my new friend |
| 15:00.36 | LeKstor | the sip part of the call seems okay |
| 15:00.45 | LeKstor | now I need to check the RTP |
| 15:00.47 | lirakis | ManxPower: .. ive looked at the application reference and there.. i dont see any sort of "logoff" or "logout" |
| 15:00.49 | ManxPower | LeKstor: it was always your friend -- you were just not enlightened enough to realise it. |
| 15:00.57 | LeKstor | :) |
| 15:01.05 | [TK]D-Fender | lirakis: login app = logout app. |
| 15:01.14 | LeKstor | i'm quite a beginner in the asterisk field |
| 15:01.42 | LeKstor | and i found nothing on google linking the "no route" to the qualify attribute |
| 15:02.14 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:02.20 | [TK]D-Fender | LeKstor: Please provide full CLI output for your failed call attempt at verbose 10 |
| 15:02.26 | [TK]D-Fender | ~pb |
| 15:02.27 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:02.27 | lirakis | [TK]D-Fender: thats what i thought.. but i log in.. then .. i dial in again.. and do the process.. and it doesnt say "agent logged off" .. it says "agent logged in" again .. :\ |
| 15:02.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 15:02.48 | [TK]D-Fender | lirakis: pastebin how you're doing it. |
| 15:03.14 | lirakis | [TK]D-Fender: will do.. going to try one thing first |
| 15:03.29 | madgyver | thx for the hint with "chan_mobile"! did anyone actually use this? can I forward the audio from the handy to the computer? |
| 15:04.40 | LeKstor | <PROTECTED> |
| 15:04.45 | LeKstor | now i have an RTP pb |
| 15:04.56 | LeKstor | but I have a complicated configuration with NAT |
| 15:05.02 | LeKstor | and I need to do some checks |
| 15:05.08 | *** join/#asterisk mog (i=mog@nat/digium/x-9325ee21b2861377) |
| 15:05.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:05.23 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 15:07.22 | lirakis | Hmm.. soo.. it looks like .. the agent logs in.. but as soon as they hangup.. they get logged out.. |
| 15:07.28 | lirakis | that.. seems wierd |
| 15:08.28 | [TK]D-Fender | lirakis: Don't pass it the exten on the CL, and when it asks you for one, just enter "#" |
| 15:08.55 | putnopvut | lirakis: sorry I'm joining this chat late, but are you using AgentLogin in the dialplan? |
| 15:09.13 | lirakis | [TK]D-Fender: .. i am not passing the exten when i call the app.. but i have been entereing it when asked |
| 15:09.20 | lirakis | im doing a pastebin now |
| 15:09.26 | errr | I am trying to setup a dial plan in our Irving Texas office but I am not from the area and do not know what numbers are local and what is long distance. Is there some web site that would have this kind of information on it? |
| 15:09.49 | [TK]D-Fender | lirakis: then jsut hit # and see |
| 15:09.55 | lirakis | i will |
| 15:10.00 | ManxPower | errr: your best bet is send the call to the telco and let them figure it out./ |
| 15:10.48 | lirakis | [TK]D-Fender: .. when i hit # for agent number.. it just says "login incorrect" immediately |
| 15:11.04 | [TK]D-Fender | lirakis: for EXTENSION, not AGENT <- |
| 15:11.04 | errr | ManxPower: Ok, I guess I will pack me a lunch and prepare for a long day of transfers in hopes of finding the 1 person who knows whats going on over there =) |
| 15:11.14 | lirakis | [TK]D-Fender: .. okay.. it doesnt ask for an extension |
| 15:11.25 | lirakis | putnopvut: yes |
| 15:11.26 | [TK]D-Fender | lirakis: THAT is what you need to change |
| 15:11.46 | lirakis | [TK]D-Fender: .. mm ... okay (lirakis wonders how) |
| 15:12.23 | lirakis | http://pastebin.com/d70b45843 |
| 15:13.02 | masterisk | ok, I'm lost..is there a way to access astdb from PHP ? |
| 15:13.03 | [TK]D-Fender | lirakis: Waitasec... thats the LIVE Agent login where they have to STAY on that call to get calls |
| 15:13.12 | [TK]D-Fender | lirakis: Have you only just begun using queues? |
| 15:13.23 | lirakis | [TK]D-Fender: yes... first time |
| 15:13.24 | lirakis | ha ha |
| 15:13.32 | [TK]D-Fender | masterisk: Yes, there are BDB PHP libraries out there that can do this. |
| 15:13.38 | [TK]D-Fender | lirakis: Wrong tool. |
| 15:13.39 | lirakis | should i be using addqueuemember? |
| 15:13.44 | putnopvut | lirakis: yes. |
| 15:13.47 | lirakis | okay |
| 15:13.58 | [TK]D-Fender | lirakis: AgentCallbacklogin. Get reading... |
| 15:14.13 | lirakis | [TK]D-Fender: i thought they were depreciating that... |
| 15:14.16 | masterisk | [TK]D-Fender: could you name one that works? php has deprecated dbm and ndbm, gdbm gives me a "magic number error" |
| 15:15.12 | [TK]D-Fender | masterisk: I haven't tried this personally yet. |
| 15:15.31 | [TK]D-Fender | masterisk: You could cheat and text-dump the DB for parsing. |
| 15:15.40 | masterisk | [TK]D-Fender: I could :D |
| 15:16.18 | masterisk | but I also need to write stuff back to astdb |
| 15:16.34 | masterisk | anyway, thanks for the tip |
| 15:16.50 | [TK]D-Fender | masterisk: And you can AMI / CLI do that too :) |
| 15:17.06 | FlatFoot | asked a question earlier ... is there a Select Case ( VB ) func in * ? Basically i have a blind friend and i need to change the ringtone depending on the incoming number.# |
| 15:17.29 | [TK]D-Fender | FlatFoot: "show application GotoIf" |
| 15:18.02 | [TK]D-Fender | FlatFoot: There's your condition test. As for changing the ring, that depends on your phone's capabilities. |
| 15:18.23 | [TK]D-Fender | FlatFoot: Then again if its simply CID based you can do that DIRECT entirely on the phone itself on any decent phone. |
| 15:18.32 | FlatFoot | condition test is what i was after THANKS , the snoms handle _ALERT_INFO fine |
| 15:18.38 | FlatFoot | thanks for the info |
| 15:19.35 | [TK]D-Fender | FlatFoot: Ok, then you've got what you need. On a polycom you could just add them to the phone directory and put a ringer of your choice on it and forget the dialplan :) |
| 15:20.24 | madgyver | thank you very much for maintaining and supporting this awsome software! bye |
| 15:22.13 | lirakis | .. soo .. do i need to have an agent logged in .. then add them to a queue with queue add member? ... |
| 15:22.45 | [TK]D-Fender | lirakis: No, adding = logged in |
| 15:22.57 | [TK]D-Fender | lirakis: Completely forget about "agentlogin" |
| 15:23.35 | [TK]D-Fender | lirakis: "show application addqueuemember" |
| 15:25.07 | lirakis | [TK]D-Fender: .. im not using agentlogin any more .. im using AgentCallbackLogin |
| 15:25.31 | [TK]D-Fender | lirakis: ok, then rad up on how to use it. |
| 15:25.33 | [TK]D-Fender | read* |
| 15:25.42 | [TK]D-Fender | lirakis: And set up your agents.conf |
| 15:26.38 | lirakis | [TK]D-Fender: i setup agents.conf... and i did read.. but .. if i have multiple queues .. and i dont want all agents to be a member of all queues.. i am missing how to specify that |
| 15:27.04 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 15:27.15 | [TK]D-Fender | lirakis: you specify MEMEBERS for QUEUES. (queues.conf) |
| 15:27.18 | lirakis | [TK]D-Fender: i can log in.. and log out.. all that works fine |
| 15:27.29 | lirakis | [TK]D-Fender: okay.. so it has to be hard coded? |
| 15:27.35 | [TK]D-Fender | lirakis: "member => Agent/100" , etc |
| 15:27.39 | [TK]D-Fender | lirakis: Yes |
| 15:28.11 | techman97_andy | ok, so I'm back. Looking through my CLI and PRI debug messages, I'm getting back a Hangup Code of 41 (temp failure) from my telco. I'm on the phone with those stinkers now. |
| 15:28.17 | lirakis | [TK]D-Fender: thats where i was missing things.. |
| 15:28.33 | [TK]D-Fender | lirakis: Or you can use the "AddQueueMember" method which has a few different complications, but looks workable. |
| 15:28.44 | lirakis | <PROTECTED> |
| 15:28.48 | techman97_andy | the telco SAYS that error code 41 means that our number is being blocked by the called's person's telco |
| 15:28.49 | techman97_andy | :S |
| 15:29.06 | [TK]D-Fender | lirakis: But you should be fine to hard-code them. You're building your setup 100% yourself so minor maintenance shouldn't be a problem. |
| 15:29.35 | [TK]D-Fender | techman97_andy: You're still just whining about your probelm instead of providing the backup to help us SOLVE IT. |
| 15:29.56 | lirakis | [TK]D-Fender: yeah it will be fine.. i was just confused b/c i thought it was redundant to log in.. and then to be added to the queue |
| 15:30.03 | [TK]D-Fender | techman97_andy: Block AFTER 60s? |
| 15:30.19 | techman97_andy | [TK]D-Fender: I'm not whining - I'm trying to solve it here - I didn't remember to look at the PRI debug, so I'm still trying to work it myself before bringing it here. |
| 15:30.54 | [TK]D-Fender | techman97_andy: No worries, I bit a little hard on that one actually... |
| 15:31.02 | techman97_andy | [TK]D-Fender: NP man...=P |
| 15:31.08 | [TK]D-Fender | techman97_andy: Still go replicate it for us will you... |
| 15:40.10 | lmadsen | FYI: TFoT2 has been sent off to the printers!! |
| 15:40.20 | lmadsen | I'm seeing about getting a PDF released either today or tomorrow |
| 15:43.42 | Mercestes | TFoT2? |
| 15:43.51 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 15:44.35 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 15:45.04 | lmadsen | Asterisk: The Future of Telephony (2nd Edition) |
| 15:45.18 | lmadsen | i.e. "the book" |
| 15:45.30 | Mercestes | Isn't that Teliax's catch phrase? |
| 15:45.42 | lmadsen | maybe they stole it from us :) |
| 15:45.46 | Mercestes | "Thank you for calling Teliax, the future of telephony." |
| 15:45.47 | [TK]D-Fender | lmadsen: So... how long till PDF? :) |
| 15:45.57 | lmadsen | <lmadsen> I'm seeing about getting a PDF released either today or tomorrow |
| 15:46.01 | lmadsen | and I quote... :) |
| 15:46.18 | [TK]D-Fender | lmadsen: Don't mind me.... I'm not all here today... |
| 15:46.25 | Mercestes | s/today// |
| 15:46.25 | lmadsen | [TK]D-Fender: and you are on other days? :) |
| 15:46.36 | lmadsen | and it's proven :) |
| 15:46.37 | Mercestes | s/today/ever/ |
| 15:47.16 | Mercestes | .... |
| 15:47.23 | Mercestes | they should reprogram Jbot so you can edit other people's statements. |
| 15:47.28 | Mercestes | that'd be funny! |
| 15:47.38 | Mercestes | I am an asterisk consultant |
| 15:47.41 | lmadsen | [TK]D-Fender: s/today// |
| 15:47.41 | Mercestes | s/consultant/nubcake/ |
| 15:47.46 | Mercestes | see? |
| 15:47.50 | Mercestes | Fender could pwn me daily. |
| 15:48.01 | FlatFoot | [TK]D-Fender , can ya help ? ( a sad retch of a person ) |
| 15:48.03 | lmadsen | hrmmm... could have sworn you could edit other statements if you prefixed it with the persons name |
| 15:48.18 | *** join/#asterisk saftsack (n=saftsack@217.224.91.120) |
| 15:48.29 | Mercestes | [TK]D-Fender: s/Mercestes/Mom's/ |
| 15:48.33 | Mercestes | nah, doesn't work. |
| 15:48.34 | [TK]D-Fender | lmadsen: Don't give him any ideas |
| 15:49.25 | Mercestes | Would be even better as a stored procedure. |
| 15:49.47 | Mercestes | Then my name could be permanently replaced with "nubcake" |
| 15:50.03 | GromiTM | Hmmm... with http://paste.debian.net/34706 I get alway the playback of invalid. |
| 15:50.18 | GromiTM | May anyone say me or give me a hint why, please? |
| 15:51.07 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:51.35 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 15:53.28 | CCFL_Man2 | anyone run asterisk on linux sparc? |
| 15:54.05 | [TK]D-Fender | GromiTM: Perhaps you should provide some SLI output at verbose 10. |
| 15:54.14 | [TK]D-Fender | CLI* |
| 15:56.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:57.19 | *** join/#asterisk anthm (n=anthm@64.241.37.140) |
| 15:57.19 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:57.19 | CCFL_Man2 | i want to run asterisk on a sun netra, but i want the ztdummy driver to provide proper timing for asterisk |
| 15:57.48 | CCFL_Man2 | so i can host conferencing |
| 15:58.06 | tzafrir | CCFL_Man2, hmm... I know it has been done |
| 15:58.11 | *** join/#asterisk JunK-Y (n=junky@modemcable223.205-56-74.mc.videotron.ca) |
| 15:58.46 | tzafrir | Zaptel should run there. At least the Debian packages compile. No idea about ztdummy, though |
| 15:58.56 | GromiTM | [TK]D-Fender: http://paste.debian.net/34711 |
| 15:59.06 | GromiTM | hmmm I did not get more ... |
| 15:59.13 | JunK-Y | tzafrir: use the force for ztdummy :) |
| 15:59.41 | [TK]D-Fender | GromiTM: And why aren't I seeing ANY dialplan executing at all? |
| 16:00.12 | *** join/#asterisk De_Mon (n=de_mon@fl-71-55-191-178.dhcp.embarqhsd.net) |
| 16:00.21 | [TK]D-Fender | JunK-Y: Strom is in town, might ahve a chance to meet up for a beer later if your interested |
| 16:00.22 | GromiTM | [TK]D-Fender: I hoped you can say me this. Sorry, I totally have no clue. |
| 16:00.42 | JunK-Y | sure, where? |
| 16:00.50 | [TK]D-Fender | GromiTM: Well last I checked you can't do [alpha here] in your exten. |
| 16:00.51 | CCFL_Man2 | tzafrir: yeah, i'm going to use debian, they already have a zaptel drivers package, and i'm going to use the 2.6 kernel, so you think it should work reliably? |
| 16:00.58 | De_Mon | Asterisk(1.4.6) isn't picking up my .call file |
| 16:01.10 | HarryR | De_Mon, where are you putting them? |
| 16:01.11 | [TK]D-Fender | JunK-Y: I'll see if he's sticking around late enough if I see him back here. |
| 16:01.17 | [TK]D-Fender | JunK-Y: PM me your cell # |
| 16:01.22 | De_Mon | The only thing I can think of is that the root filesystem got full where the calls are normally put |
| 16:01.35 | De_Mon | but its got 2 gigs free now, and still isn't picking up the call |
| 16:01.46 | Deeewayne | De_Mon: where are you putting the .call files ? |
| 16:01.47 | *** join/#asterisk onats (n=julian@122.53.139.111) |
| 16:01.52 | De_Mon | /var/spool/asterisk/outgoing/ |
| 16:02.12 | JunK-Y | are you sure you have pbx_spool loaded? |
| 16:02.16 | De_Mon | files are owned by asterisk and all that, it used to work. |
| 16:02.19 | Deeewayne | you are moving them, right? |
| 16:02.27 | De_Mon | yeah |
| 16:02.37 | De_Mon | let me reload pbx_spool and see what happens |
| 16:02.40 | [TK]D-Fender | De_Mon: Are the files still in the spool folder? if so check the DATESTAMP |
| 16:02.45 | Mercestes | [TK]D-Fender, Half of his pastebin is him turning his debug on, lol. |
| 16:02.48 | jcolp | russellb: that's sweet that you didn't think of me as one until after <3 |
| 16:02.52 | jcolp | whoops |
| 16:02.54 | GromiTM | [TK]D-Fender: what excactly did you mean with [alpha here] ? |
| 16:02.57 | De_Mon | unable to unload |
| 16:03.00 | jcolp | silly mouse following |
| 16:03.14 | [TK]D-Fender | GromiTM: You're trying to pattern-match LETTERS. I don't believe that is legal. |
| 16:03.19 | Deeewayne | silly jcolp |
| 16:03.20 | techman97_andy | [TK]D-Fender: OK! I have it figured out, now I need to ping this off the experts. |
| 16:03.24 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:03.26 | De_Mon | cant reload and cant unload pbx_spool.so |
| 16:03.29 | De_Mon | crap |
| 16:03.42 | [TK]D-Fender | De_Mon: "restart now" <- that oughtta do it... |
| 16:03.51 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.169.196) |
| 16:03.54 | De_Mon | yeah.. and drop the calls that are active |
| 16:04.05 | [TK]D-Fender | De_Mon: You and your silly conniptions! |
| 16:04.15 | elixer | [TK]D-Fender: you can pattern match letters |
| 16:04.20 | elixer | at least according to the wiki |
| 16:04.22 | Mercestes | De_Mon: try reload? |
| 16:04.26 | GromiTM | [TK]D-Fender: hmmm ok, I will try it with numbers. |
| 16:04.35 | Mercestes | elixer, only if your phone passes numbers. |
| 16:04.37 | techman97_andy | the deal is....I would call...the call would ring for 60-120 seconds, and then disconnect. My telco was showing a normal clearing code of 16 from the remote side, but they were passing a 41, temp network failure. What we saw was an inband message being sent (this number is disconnected, etc), it would play for that 60-120 seconds, and then issue a normal clearing. Now...the question.... |
| 16:04.40 | Mercestes | s/numbers/letters/ |
| 16:04.49 | techman97_andy | how the heck do I access that inband audio during a call? |
| 16:04.58 | De_Mon | Mercestes pbx_spool.so doesn't support unloading or reloading |
| 16:05.01 | *** join/#asterisk seldon75 (n=chatzill@69.77.161.2) |
| 16:05.07 | elixer | Mercestes: obviously |
| 16:05.38 | elixer | i have this in my dialplan -> exten => _[A-Z][A-Z][A-Z]_XXXX,1,Set(ENTITY_PREFIX=${CUT(EXTEN,_,1)}) |
| 16:06.08 | elixer | and it works fine with our SIP phones |
| 16:06.17 | De_Mon | restarting fixed the spooler problem. but why was it a problem in the first place |
| 16:06.29 | Mercestes | I'm afraid to paste anything about my dialplan. =/ |
| 16:06.34 | Mercestes | Fender and Manx make fun of me everytime I do |
| 16:06.37 | De_Mon | I'm about to get an angry phone call. its okay though I know the call was personal ;) |
| 16:06.53 | Mercestes | De_Mon, they only get angry when it's a personal call. |
| 16:06.58 | JunK-Y | De_Mon: suggestion: use the manager if you have many .call files. |
| 16:07.03 | Deeewayne | angry calls are a good time to restart asterisk |
| 16:07.23 | De_Mon | JunK-Y how do you mean? |
| 16:07.37 | JunK-Y | asterisk manager interface to originate ur call. |
| 16:07.39 | De_Mon | can I call the manager from dialplan and create call files? |
| 16:08.06 | De_Mon | never touched it, will have to see what I can divine about it |
| 16:08.28 | De_Mon | I hate callfiles, so if I can, I will |
| 16:08.37 | *** join/#asterisk Tako-san (n=Tako-san@154.5.212.245) |
| 16:09.22 | *** join/#asterisk glacid (i=unknown@evool.com) |
| 16:09.23 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 16:09.31 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 16:09.56 | glacid | when someone calls Zap/1 with blocked or no callerid, my extensions are getting calls from "Asterisk" |
| 16:10.10 | glacid | did i miss a setting where they can get "Unknown" or "Anonymous"? |
| 16:10.16 | JunK-Y | De_Mon: that would be great if pbx_spool supports reload huh ? :) |
| 16:10.40 | Mercestes | glacid, I think "asterisk" is the default instead of "unknown" |
| 16:11.05 | elixer | De_Mon: i think he was suggesting using the manager to Originate the calls rather than creating .call files |
| 16:11.11 | glacid | Mercestes: ahhh, so i should just put a rule into extensions.conf to switch it? |
| 16:11.25 | glacid | anonymous should still come through as anonymous though? |
| 16:11.34 | De_Mon | can I use the asterisk manager api from the 'asterisk' binary? |
| 16:11.37 | glacid | er.. "blocked" |
| 16:11.42 | Mercestes | glacid, you could do a gotoif($[${CALLERID(name)foo = "foo"]?0:1) to set it to "unknown." |
| 16:12.01 | De_Mon | of the languanges listed on voip-info.org I don't see 'bash' :) |
| 16:12.15 | elixer | De_Mon: no, the manager is used over via a network connections |
| 16:12.23 | elixer | s/connections/connection/ |
| 16:12.29 | Mercestes | De_Mon, lol |
| 16:12.41 | caio1982 | actually you can use bash scripts to access the manager API |
| 16:13.25 | elixer | caio1982: how's that? |
| 16:13.38 | Mercestes | telnet localhost:manager_port |
| 16:13.39 | Nugget | telnet is eeeeeeevil! |
| 16:13.46 | elixer | ugh |
| 16:13.46 | Mercestes | ... |
| 16:13.48 | Mercestes | telnet |
| 16:13.49 | elixer | have fun with that |
| 16:13.55 | mvanbaak | gheh |
| 16:13.59 | s34n | I've read that ztdummy has to be modified for a virtual environment like Xen. Is that true even if it runs on the host? |
| 16:14.11 | mvanbaak | Nugget: we need an ssh interface ;) |
| 16:14.23 | Mercestes | I like to use telnet to telnet into my telnet sites to download telnet clients for use with future telnet telnetting. |
| 16:14.43 | mvanbaak | whehehehehehe |
| 16:14.58 | De_Mon | caio1982 what elixer said |
| 16:15.21 | [TK]D-Fender | Mercestes: Its timed... 1 time / 24 hours |
| 16:15.30 | caio1982 | using telnet means it's possible, the problem of that being ugly, hackish and stupid is a different thing :) |
| 16:15.32 | Mercestes | [TK]D-Fender, killjoy. |
| 16:15.39 | [TK]D-Fender | Mercestes: Just the messenger... |
| 16:15.51 | [TK]D-Fender | s34n: Virtualized timing... LOL |
| 16:17.11 | Mercestes | s34n, as cool as that would be, I would have to suggest against vmc asterisk boxes. I can barely get it to run right on a physical server. |
| 16:18.07 | s34n | Mercestes: if you don't need timing, there's nothing special about a virtual environment for * |
| 16:18.16 | Mercestes | s34n, I heard a rumor that the timing methods will change someday soon so maybe it won't be as much of an issue in the near future. |
| 16:18.22 | Nugget | my head a splode. |
| 16:18.52 | Mercestes | s34n, oh, I agree totally. However, most * installs need a timing source. |
| 16:19.05 | Mercestes | g1powermac, I'm sorry. |
| 16:19.10 | Nugget | can anyone give me a hand debugging an issue with pri callerid name? pri debug shows that I'm getting caller name info, but it's not making it into the variable. |
| 16:19.24 | Mercestes | Nugget, CLI or AGI? |
| 16:19.25 | g1powermac | was trying for the life of me to get it to connect to my wifi router, and then realized I had MAC address filtering setup |
| 16:19.25 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-113-67.lns10.syd6.internode.on.net) |
| 16:19.28 | Nugget | cli |
| 16:19.38 | Mercestes | g1powermac, heh, I do that all the time. :D |
| 16:19.43 | s34n | Mercestes: I have seen a ztdummy modified for xen to not use rtc |
| 16:19.46 | Mercestes | g1powermac, It won't work more than 10 ' away tho. |
| 16:19.47 | g1powermac | and actually I find the reception of the phone not too bad |
| 16:20.17 | [TK]D-Fender | Nugget: I've seen that in cases where a line is forwarded to a PRI on DMS100 Its in a secodary identifier. |
| 16:20.17 | Mercestes | Nugget, I can give a stab at it. |
| 16:20.17 | g1powermac | Mercestes, heh, you haven't seen the wifi antenna I got hooked up to my router :-) |
| 16:20.17 | JunK-Y | s34n: any link? |
| 16:20.17 | *** join/#asterisk engrxyz (n=fgsfgfs@82-34-18-23.cable.ubr03.basi.blueyonder.co.uk) |
| 16:20.28 | Mercestes | g1powermac, you haven't seen my small pile of wifi phones I have stacked in the corner that wouldn't go more than 10'. |
| 16:20.29 | Mercestes | ... |
| 16:20.33 | Mercestes | s/small/large/ |
| 16:20.33 | g1powermac | Mercestes, http://www.fab-corp.com/product.php?productid=3069&cat=250&page=1 |
| 16:20.55 | Mercestes | oh, I had industrial grade antennae as well. |
| 16:21.11 | Mercestes | ~wifi |
| 16:21.12 | jbot | it has been said that wifi is see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 16:21.16 | s34n | JunK-Y: let me find it |
| 16:21.19 | techman97_andy | [TK]D-Fender: It ended up being the remote telco (not my provider) playing an in-band message, issuing a normal call clearing, and then my telco getting confused that we were still in call progress and they issued a normal call clearing. I'm switching my zapata.conf to priindiction=inband from outofband to see if I can grab that inband message. |
| 16:21.20 | g1powermac | Mercestes, well, currently, I can ping the phone and its about 60 ft away from AP |
| 16:21.31 | Mercestes | g1powermac, Ok, don't listen to me. |
| 16:21.47 | g1powermac | its only one bar of reception, but still pingable |
| 16:21.59 | Nugget | it looks like maybe the name is coming in late or something. |
| 16:22.07 | [TK]D-Fender | techman97_andy: That sounds like trouble for normal calls... but run with it... |
| 16:22.23 | Mercestes | he said indiction. |
| 16:22.39 | techman97_andy | [TK]D-Fender: We'll see what happens. Now that I have a steady number to test against, I can run a full battery of tests after hours today. |
| 16:22.40 | Mercestes | Nugget, maybe add a wait or two to allow CID transfer? |
| 16:22.44 | Nugget | http://pastebin.ca/658424 <-- ref line 48 (which is noop "...XO PRI ${CALLERID(all) for...") and then line 80. |
| 16:22.53 | Nugget | worth a shot |
| 16:22.55 | citats | Nugget: assuming its coming over with the right IE, have you tried adding a Wait for a few seconds? |
| 16:23.54 | Mercestes | Hey Nugget, is that your cell # in your pastebin? |
| 16:23.55 | Nugget | shit, that did it. thanks. :) |
| 16:24.02 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:24.33 | Mercestes | >.> |
| 16:24.36 | Mercestes | Your welcome. :D |
| 16:25.14 | Nugget | that's fine, I'm only randomly here. |
| 16:25.32 | Mercestes | lmao |
| 16:25.36 | Mercestes | Nice. |
| 16:28.51 | s34n | JunK-Y: here's a list entry from one of the devs -> http://lists.digium.com/pipermail/asterisk-dev/2007-May/027470.html |
| 16:31.03 | *** join/#asterisk pusanggala (n=a@58.69.243.203) |
| 16:31.34 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 16:32.25 | JunK-Y | thx |
| 16:33.03 | *** join/#asterisk r0b- (i=rob@adsl-76-235-179-178.dsl.klmzmi.sbcglobal.net) |
| 16:33.10 | r0b- | can someone help me |
| 16:33.10 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-189-7-234.lsanca.fios.verizon.net) |
| 16:33.25 | r0b- | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 16:33.26 | r0b- | configure: error: *** termcap support not found |
| 16:33.55 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 16:34.04 | [TK]D-Fender | rob go install libtermcap |
| 16:34.25 | r0b- | this is Fedora |
| 16:34.42 | Nugget | ok, go install libtermcap.rpm |
| 16:34.48 | [TK]D-Fender | lol |
| 16:34.49 | Mercestes | Nugget, lol |
| 16:34.54 | r0b- | lol |
| 16:34.58 | Mercestes | I was wondering where he was going with that. |
| 16:35.05 | Mercestes | r0b-, Or better yet, install gentoo. |
| 16:35.20 | r0b- | but i already got fedora running! |
| 16:35.24 | Mercestes | Congratulations! |
| 16:35.33 | Mercestes | my 7 year old got Fedora running too. |
| 16:35.40 | r0b- | lol |
| 16:35.51 | Mercestes | It installs like windows XP. "next, next...next...default...more next. ok done." |
| 16:36.10 | Mercestes | lunchy time...bbl |
| 16:38.05 | [TK]D-Fender | r0b-: You've been told what to install, get to it |
| 16:38.06 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 16:38.21 | r0b- | i tried |
| 16:38.28 | r0b- | it said it was old :) |
| 16:38.29 | r0b- | :( |
| 16:39.35 | [TK]D-Fender | r0b-: Show us something useful.... |
| 16:39.46 | [TK]D-Fender | r0b-: PASTEBIN is your friend. |
| 16:39.48 | [TK]D-Fender | ~pb |
| 16:39.49 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:40.02 | r0b- | nevermind |
| 16:40.04 | *** part/#asterisk r0b- (i=rob@adsl-76-235-179-178.dsl.klmzmi.sbcglobal.net) |
| 16:43.12 | [TK]D-Fender | another person who was clearly expecting this to be #drphil |
| 16:43.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 16:48.00 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 16:48.11 | seldon75 | what do we thinhk of SIP Trunks? |
| 16:48.34 | Merlin83b | We prefer IAX2 trunks, I believe. |
| 16:48.50 | *** join/#asterisk Steven_elvisda (n=Elvisda@202.47.107.60) |
| 16:49.09 | *** join/#asterisk docelmo (n=vircuser@c-76-99-154-203.hsd1.de.comcast.net) |
| 16:49.21 | *** join/#asterisk docelmo (n=vircuser@c-76-99-154-203.hsd1.de.comcast.net) |
| 16:50.11 | [TK]D-Fender | seldon75: Who's "we" and "trunk" is NOT an appropriate term for a SIP CHANNEL. |
| 16:50.32 | seldon75 | well the reseller is calling it a SIP Trunk |
| 16:50.39 | seldon75 | "we" means everyone |
| 16:51.27 | seldon75 | http://www.bandwidth.com/content/voip?page=voip_sip_trunking_faq |
| 16:52.08 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:52.21 | seldon75 | anyone using / has used this? |
| 16:53.35 | De_Mon | WE haven't agreed on a term |
| 16:53.42 | Steven_elvisda | hello i'm newbie in here |
| 16:53.57 | Steven_elvisda | i want to use my pc with internet call to phone |
| 16:54.31 | seldon75 | Skype-out |
| 16:54.55 | [TK]D-Fender | seldon75: Them specifically? Never heard mention of them. Plenty of other people use SIP/IAX for PSTN access. |
| 16:55.49 | seldon75 | so the term SIP-Trunk is a misnomer? is the term used by other resellers of the same technology |
| 16:56.01 | [TK]D-Fender | seldon75: So skip the terminology. Is it jsut the company you want our opinion on? |
| 16:56.22 | seldon75 | no, the technology - using them with * |
| 16:57.03 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:57.37 | [TK]D-Fender | seldon75: I use SIp to connect through my office, many of my clients have used ITSP's. |
| 16:57.54 | [TK]D-Fender | seldon75: Many would recommend against VoIP for business. |
| 16:58.06 | [TK]D-Fender | seldon75: Just because of the number of points of failure. |
| 16:58.44 | seldon75 | interesting |
| 16:58.47 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 16:58.58 | GromiTM | [TK]D-Fender: Problem solved, the problem was in counting the argumentent. |
| 16:59.13 | GromiTM | [TK]D-Fender: but its running now. |
| 16:59.35 | seldon75 | we have found that our POTS trunks suck. THey have impedance problems that the telco is still trying to fix. Hence the SIP-Trunk idea |
| 16:59.57 | GromiTM | Ok .. work finished, sse you. |
| 17:00.11 | coppice | telcos have a lot of impudence problems |
| 17:00.45 | masterisk | what woukd you guys pick for interfacing with AGI? PERL, PHP, JAVA ? |
| 17:00.52 | *** join/#asterisk Op3r (n=Op3r@125.212.50.210) |
| 17:00.56 | hmmhesays | eh, whatever you want |
| 17:01.20 | coppice | I want SNOBOL |
| 17:01.39 | Op3r | can anyone explain me the 6sec pulse billing? |
| 17:02.23 | coppice | what is there to explain? |
| 17:02.37 | [TK]D-Fender | seldon75: Your call. Keep bandwidth and failover in mind. |
| 17:03.31 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 17:04.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:04.44 | _ViperNetworks | Hello all.... Anyone can help me with Callback DTMF problem? |
| 17:04.54 | Op3r | coppice: whats the difference between getting billed in minutes than in 6 sec pulse |
| 17:06.35 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-232.3g.vodacom.co.za) |
| 17:07.15 | coppice | those pulses do not define the billing step |
| 17:07.22 | seldon75 | So, SIP-Trunks with Asterisk = no (extra) problems |
| 17:07.50 | _ViperNetworks | I have Callback setup and it works fine when tested within US. Works fine meaning the DTMF tones are passed when prompted to enter the phone number. But when I test with some international countries, callback works but DTMF tones are not passed... |
| 17:08.08 | _ViperNetworks | Can anyone suggest where is the problem? |
| 17:08.37 | _ViperNetworks | Under sip.conf I have "dtmfmode = auto". and i am running Asterisk 1.2.18 on CentOS 4.5 |
| 17:09.37 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-189-7-234.lsanca.fios.verizon.net) |
| 17:09.37 | russellb | _ViperNetworks: please don't put an underscore in front of your name so it's at the top of the list . |
| 17:09.43 | dahunter3 | I keep trying to place a call with a softphone and I keep getting Call rejected; 491 request pending. Where should I start looking? |
| 17:10.06 | russellb | i don't care until it's the name of a company |
| 17:10.21 | _ViperNetworks | OK...will change it |
| 17:10.23 | coppice | russellb: he isn't at the top of my list |
| 17:10.24 | dahunter3 | Like would you think from reading that, that I misconfigured asterisk/SIP or that I perhaps have a firewall problem on my home network? |
| 17:10.26 | russellb | thanks! |
| 17:10.34 | russellb | coppice: close to the top then :) |
| 17:11.11 | coppice | russellb: he's near the bottom, near vodak |
| 17:11.18 | russellb | really? |
| 17:11.25 | russellb | smart IRC client, then |
| 17:11.29 | coppice | use a smarter tool :-) |
| 17:11.34 | russellb | heh |
| 17:11.36 | coppice | chatzilla |
| 17:11.54 | [TK]D-Fender | seldon75: Sure, plenty of extra potential problems. Bandwidth, jitter, everything your call goes over as points of failure, etc. |
| 17:11.59 | denon | telnet irc.freenode.net 6667 - then your client is as smart as you are :) |
| 17:12.55 | Nugget | heh |
| 17:13.05 | lmadsen | xchat here |
| 17:13.26 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 17:14.14 | *** join/#asterisk _ViperNet (n=Nitesh@66.184.39.174) |
| 17:14.37 | *** join/#asterisk potsboy (n=chrisg@vc-196-207-32-232.3g.vodacom.co.za) |
| 17:14.46 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 17:16.26 | Nugget | anyone who cared what irc client you were using would have just looked for themselves. |
| 17:17.11 | aut | anyone who cared for your opinion on the matter would have asked themselves :) |
| 17:17.15 | aut | j/k :) |
| 17:17.59 | potsboy | somebody is grumpy with there asterisk system so they take it out on irc .... nice |
| 17:18.16 | masterisk | viper networks .. :) |
| 17:18.16 | Nugget | It's spelled "their" :P |
| 17:18.28 | masterisk | i know those guys |
| 17:21.09 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:25.22 | masterisk | has anyone worked with asterisk-java and php? is asterisk-java really that much slower than phpagi ? |
| 17:25.23 | *** join/#asterisk ViperNetworks (n=Nitesh@66.184.39.174) |
| 17:28.31 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:32.50 | *** join/#asterisk Steven_elvisda (n=Elvisda@202.47.107.60) |
| 17:32.54 | Steven_elvisda | can i use asterisk to make server like net2phone ? |
| 17:33.08 | [TK]D-Fender | Steven_elvisda: Yes |
| 17:33.31 | [TK]D-Fender | masterisk: How many simultaneous calls do you figure on processing in that way? |
| 17:33.59 | Steven_elvisda | pc --> voip server --> phone |
| 17:34.28 | Steven_elvisda | how many stuff that i need to build a voip server? |
| 17:35.25 | potsboy | Steven_elvisda: phone is a broad term |
| 17:35.48 | Steven_elvisda | what you mea potsboy? |
| 17:35.55 | prizark_ | Steven_elvisda, If you have to process lot of SIP messages/sec then a SIP server would be better |
| 17:35.56 | Steven_elvisda | i not yet understand what you mean |
| 17:36.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 17:36.10 | potsboy | what kind of phone analogue/digital/voip |
| 17:36.21 | potsboy | voip server -> phone |
| 17:36.22 | Steven_elvisda | ohhh |
| 17:36.30 | Steven_elvisda | analog and digital potsboy |
| 17:36.43 | prizark_ | can the outgoing caller ID be programmed on an ISDN30 line? |
| 17:36.52 | Steven_elvisda | pc --> voip server(*nix + asterisk) --> phone |
| 17:37.20 | potsboy | prizark_: yes, but is telco dependant |
| 17:37.26 | *** join/#asterisk saftsack (n=saftsack@217.224.91.120) |
| 17:37.40 | potsboy | i.e. not legal our here |
| 17:37.57 | Steven_elvisda | ermm |
| 17:38.09 | prizark_ | postboy, really, I thought all VoIP gateways use that feature |
| 17:38.32 | prizark_ | but not sure whether it can be done over ISDN30 though, now that you have confirmed it |
| 17:38.50 | Steven_elvisda | ermm |
| 17:39.02 | Steven_elvisda | everyone i'm newbie of voip server |
| 17:39.05 | masterisk | [TK]D-Fender: 100 simultaneous calls |
| 17:39.17 | potsboy | cli is part of the protocol.. our telco over rides all changes made to the id though |
| 17:39.19 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 17:39.20 | Steven_elvisda | but my idea is want to do like net2phone server |
| 17:39.23 | [TK]D-Fender | masterisk: Ouch, well then it may be relevent. |
| 17:39.29 | Steven_elvisda | that provide all pc can call to phone |
| 17:39.37 | Steven_elvisda | if i want to do like this |
| 17:39.45 | Steven_elvisda | how should i do first everyone? |
| 17:39.56 | Steven_elvisda | -how +what |
| 17:40.14 | prizark_ | Steven_elvisda: you can setup a asterisk server and then purchase some VoIP termination service |
| 17:40.23 | [TK]D-Fender | Steven_elvisda: Go read THE BOOK. And then the WIKI |
| 17:40.24 | [TK]D-Fender | ~book |
| 17:40.25 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:40.26 | [TK]D-Fender | ~wikis |
| 17:40.27 | jbot | methinks wikis is http://www.voip-info.org |
| 17:40.37 | [TK]D-Fender | Steven_elvisda: And go install * and start playing around with it. |
| 17:40.42 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-189-7-234.lsanca.fios.verizon.net) |
| 17:41.07 | masterisk | [TK]D-Fender so you think java would slow things down compared to php when talking about that many calls ? |
| 17:41.22 | hmmhesays | any scripting language is going to slow things down |
| 17:41.28 | hmmhesays | you have to fire up an interpreter for every call |
| 17:41.36 | hmmhesays | why would you do this? |
| 17:41.42 | carrar | Write it all in assembly language! |
| 17:41.45 | Steven_elvisda | prizark_: can you recomment the cheapest voip termination service one? |
| 17:41.47 | masterisk | hmmhesays: thank you Captain Obvious, I was wondering how much slower asterisk-Java would be compared to phpagi |
| 17:41.51 | potsboy | hmmhesays: deadagi would resolve the speed issue |
| 17:42.40 | prizark_ | Steven_elvisda: that depends on what volume are you expecting and how much are you willing to pay upfront. Also depends on the location you want to access most, like a specific country |
| 17:42.57 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 17:43.02 | hmmhesays | why would deadagi have any impact on speed? |
| 17:43.04 | punkgode | I'm having trouble capturing the result of a get_option command from an AGI, the AGI locks up reading STDIN... a very simple example:http://rafb.net/p/sgk6GK25.html |
| 17:43.16 | punkgode | btw I'm using perl |
| 17:43.18 | [TK]D-Fender | Steven_elvisda: www.teliax.com |
| 17:43.35 | potsboy | hmmhesays: deadagi runs on a remote server.. i have this for db lookup's etc.. with out load on my tdm gateway |
| 17:43.43 | potsboy | err. oops |
| 17:43.46 | potsboy | fastagi |
| 17:43.47 | potsboy | :( |
| 17:44.12 | prizark_ | Steven_elvisda, also do check the latency of the internet route from your server to the termination provider, which can make or break your business |
| 17:45.19 | *** join/#asterisk shinao1 (n=shinao1@41.219.202.82) |
| 17:47.00 | Steven_elvisda | prizark_ how can i check? |
| 17:47.10 | Steven_elvisda | buy my server ip or ... |
| 17:47.49 | shinao1 | hi.. can anyone point me at any guide to integrate an asterisk server with a legacy PBX and phone extensions layout? |
| 17:48.25 | prizark_ | Steven_elvisda: a very basic way of checking latency is by pinging the remote server from your server, this is not exactly accurate but it does the job |
| 17:48.27 | shinao1 | the company im working for doesnt want to dispose of their existing PBX (a Panasonic) and they have about 25 extensions on it already |
| 17:48.40 | potsboy | shinao1: if the link is digital use cpe signalling otherwise its pretty much the same as a telco link |
| 17:50.23 | shinao1 | potsboy: can you point me at any links? im really confused as to how to setup the dialplan and users |
| 17:51.33 | shinao1 | sorry extensions |
| 17:51.41 | potsboy | hmmmm, i dont have on hand.. i remeber finding a nortel one on the new a while back... sorry |
| 17:51.44 | potsboy | *net |
| 17:52.51 | shinao1 | k thanks anyway |
| 17:52.53 | Steven_elvisda | prizark_: have any way beside buy voip terminator service? |
| 17:53.01 | potsboy | google is your friend: www.voip-info.org/wiki/index.php?page=Asterisk-Panasonic |
| 17:53.04 | Steven_elvisda | why voip terminator services can do? |
| 17:53.09 | Steven_elvisda | why we can not as them? |
| 17:53.59 | prizark_ | Steven_elvisda: as postboy suggest do some googling and read all you can on www.voip-info.org please, most of these questions are very well answered over there |
| 17:54.07 | prizark_ | suggested* |
| 17:55.35 | Steven_elvisda | alot of information |
| 17:55.43 | Steven_elvisda | i don where should i read first |
| 17:56.30 | potsboy | Steven_elvisda: have a look at trixbox... perhaps the best place to start.. and has a gui |
| 17:57.12 | prizark_ | Steven_elvisda: AsteriskNow is also a good place to start. Just mull over the information at voip-info.org |
| 18:00.09 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 18:01.14 | [TK]D-Fender | Steven_elvisda: If you looking to learn anything at all you'd be wise to avoid BOTH |
| 18:02.26 | [TK]D-Fender | shinao1: It is generally not cost effective to attempt to reuse legacy equipment. You would probably be best off starting from scratch |
| 18:03.28 | shinao1 | [TK]D-Fender: i have tried to speak to management about it |
| 18:03.47 | shinao1 | but they arent swinging that way so im kinda stuck |
| 18:04.30 | [TK]D-Fender | shinao1: Not yet. Basically it comes down to what you get. It is CHEAPER and more FUNCIONAL to start over. Attempting to retrofit your old digital sets would work like garbage and cost MORE. |
| 18:04.37 | [TK]D-Fender | shinao1: Whats your budget on this? |
| 18:05.06 | prizark_ | Steven_elvisda: try this, http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html , it has an exmaple that will let you configure two users and make calls between them. It is very detailed. |
| 18:05.38 | shinao1 | [TK]D-Fender: about $2K |
| 18:05.54 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:05.58 | [TK]D-Fender | shinao1: For 25 phones, thats jsut not realistic. |
| 18:06.12 | [TK]D-Fender | shinao1: Heck, the phones alonse cost a bit more than that. |
| 18:06.39 | [TK]D-Fender | shinao1: Factor in resale of your old equipment, etc. |
| 18:06.49 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:06.55 | [TK]D-Fender | shinao1: But before we even start there : Why are you looking to switch? |
| 18:07.07 | [Airwolf] | I'm trying to compile asterisk-addons-1.4.2, but it's giving errors that it can't find a mysql directory. And that isn't there. Does anyone have a tip what to do ? |
| 18:07.12 | [T]ank | what would a yellow alarm on zap show status indicate? something provider side or server side? |
| 18:07.33 | potsboy | [T]ank: remote |
| 18:07.38 | [T]ank | thank you |
| 18:07.53 | shinao1 | they need to communicate with a new branch office |
| 18:08.07 | [TK]D-Fender | shinao1: What is the other side using now? |
| 18:08.15 | *** join/#asterisk saftsack (n=saftsack@pD9E05B78.dip.t-dialin.net) |
| 18:08.17 | [TK]D-Fender | shinao1: And how many channels to them do you want? |
| 18:09.00 | shinao1 | the main office has 25 extensions , the branch office has 5 |
| 18:09.23 | shinao1 | maybe 2 channels outside |
| 18:09.54 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 18:09.56 | potsboy | shinao1: think like a suite... put the system in with lone cards and blow them away with the feature list... as suites they are prolly concerned about a *soft* switch |
| 18:10.21 | [TK]D-Fender | shinao1: how many analog line/extension ports do you have available on each system? |
| 18:10.59 | shinao1 | hmmm |
| 18:11.39 | shinao1 | 3 analog lines, 30 exts |
| 18:12.12 | [TK]D-Fender | shinao1: Sorry, let me rephrase, how many FREE ports (and what type), and EACH side? |
| 18:12.20 | punkgode | anyone uses get_option with Perl AGIs ? |
| 18:12.23 | [TK]D-Fender | at* |
| 18:13.19 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 18:14.16 | [Airwolf] | I'm trying to compile asterisk-addons-1.4.2, but it's giving errors that it can't find a mysql directory. And that isn't there. Does anyone have a tip what to do ? |
| 18:14.38 | *** join/#asterisk mog (i=mog@nat/digium/x-40c851a1ea3d8fe9) |
| 18:14.38 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:15.10 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 18:15.37 | [TK]D-Fender | [Airwolf]: Please pastein your attempt + the failure. |
| 18:15.39 | [TK]D-Fender | ~pb |
| 18:15.39 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:16.17 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 18:16.17 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:17.19 | [Airwolf] | [TK]D-Fender: ok on moment |
| 18:17.19 | [Airwolf] | one |
| 18:17.52 | *** join/#asterisk Yarq (n=Nwm@88.250.219.95) |
| 18:17.53 | [Airwolf] | http://pastebin.com/m4712bbe |
| 18:17.57 | Yarq | Hello |
| 18:18.09 | [Airwolf] | I'm just running OpenBSD4.1 generic. |
| 18:18.30 | Yarq | I am using Centos and trying to install freepbx and asterisk but I get a pear error |
| 18:19.14 | [Airwolf] | Yarq: #trixbox |
| 18:19.16 | [Airwolf] | :) |
| 18:20.26 | [TK]D-Fender | [Airwolf]: Do you have/need MySQL? |
| 18:20.43 | [Airwolf] | I have mysql and yes I need realtime support. |
| 18:20.46 | potsboy | [Airwolf]: do you have the dev libs installed... if so check you path... |
| 18:21.05 | [Airwolf] | potsboy: what dev libs ? |
| 18:21.31 | potsboy | mysql development libaries |
| 18:21.31 | *** join/#asterisk ShaunD (i=Shaun@cblmdm72-240-16-15.buckeyecom.net) |
| 18:22.11 | [Airwolf] | pots, hmm ok. Have to check if there is a package for. |
| 18:22.40 | [TK]D-Fender | [Airwolf]: No idea about what to do with OpenBSD. |
| 18:23.35 | ShaunD | What would be a good cheap 4 port fxo interface? |
| 18:23.38 | [Airwolf] | [TK]D-Fender: thank you anyway |
| 18:23.56 | [TK]D-Fender | ShaunD: cheap != good. |
| 18:24.06 | [TK]D-Fender | ShaunD: Good = $$ |
| 18:24.54 | [TK]D-Fender | ShaunD: If you feel like being a bit more of a beta tester you could try Rhino's 4 port HWEC analog FXO card. It IS very inexpensive (relatively) and MIGHT be good. But noone I've ever heard of has used one. |
| 18:24.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:25.53 | ShaunD | I'll take a look |
| 18:25.57 | [TK]D-Fender | ShaunD: http://www.telephonydepot.com/product_p/105-069-005.htm |
| 18:27.07 | [TK]D-Fender | ShaunD: The better choice would be a Sangoma A200d which could cost about $573 |
| 18:27.18 | [TK]D-Fender | ShaunD: http://www.telephonydepot.com/product_p/105-052-a200brme.htm |
| 18:28.00 | ShaunD | I could get a digium card for cheaper than that |
| 18:28.21 | Tako-san | ShaunD: Yes, cheap != good. |
| 18:28.31 | [TK]D-Fender | ShaunD: Yes, without hardware echo cancellation. That automatically disquaifies it as "good" |
| 18:29.08 | [TK]D-Fender | ShaunD: If you want to be a cheapskate and take your chances on serious suffering and long term time lost etc, then by all means. |
| 18:29.27 | *** join/#asterisk Strom_M (n=strom@bas5-montreal28-1177917181.dsl.bell.ca) |
| 18:29.39 | [TK]D-Fender | Strom_M: So, whats the verdict? |
| 18:30.04 | Strom_M | [TK]D-Fender: i stumbled across a poutine place |
| 18:30.10 | Strom_M | and we're going to stay another night |
| 18:30.15 | [TK]D-Fender | Strom_M: Recall the name? |
| 18:30.27 | Strom_M | patata patati or something along those lines |
| 18:30.59 | [TK]D-Fender | Strom_M: Nice & greesy? |
| 18:31.12 | Strom_M | yes |
| 18:31.17 | DarylVOIP | Anyone know of a cheat sheet/command for determining if a callerID is NANP or not? I want to pass callerID if its valid, but replace it with someting valid if its not. |
| 18:31.21 | [TK]D-Fender | Strom_M: McD style fries jsut don't cut it... |
| 18:31.33 | DarylVOIP | I'd like to do it in extensions.conf if at all possible. |
| 18:31.42 | Strom_M | DarylVOIP: NXX-NXX-XXXX is the format for NANP numbers |
| 18:31.55 | Strom_M | do you want to know if it's a valid NPA-NXX combo? |
| 18:32.01 | [TK]D-Fender | DarylVOIP: Or 1-NXX-NXX-XXXX |
| 18:32.07 | DarylVOIP | Right.....but how do I figure that out is an IF statement in extensions.conf is the real question |
| 18:32.09 | ShaunD | I've been using a TDM04B for awhile and I haven't had any problems with it |
| 18:32.19 | DarylVOIP | out IN and IF statement, that is |
| 18:32.37 | Strom_M | how about "figure that out in an if statement" |
| 18:32.44 | Strom_M | third time's the charm |
| 18:33.04 | [TK]D-Fender | DarylVOIP: Place a pattern match into a SEPARATE context for those 2 patterns, then use Chanisavail(Local/${EXTEN}@othercontext) to see if it matches |
| 18:33.10 | DarylVOIP | Yeah...you got it right for me. I'll leave it at that. |
| 18:33.32 | hmmhesays | well I just ordered my hifi ear plugs |
| 18:33.53 | [TK]D-Fender | ShaunD: Then whatever. You asked for good, feel free to run on with your already solid opinion. |
| 18:34.21 | ShaunD | What would you recommend? |
| 18:34.45 | Corydon76-dig | TDM804B |
| 18:34.45 | [TK]D-Fender | ShaunD: I've already told you. |
| 18:34.54 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:34.58 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:35.39 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 18:35.43 | ViperNetworks | Can anyone help me with the Callback DTMF problem...? |
| 18:36.05 | ViperNetworks | I have Callback setup and it works fine when tested within US. Works fine meaning the DTMF tones are passed when prompted to enter the phone number. But when I test with some international countries, callback works but DTMF tones are not passed... |
| 18:36.10 | DarylVOIP | OOooohhh...there's a REGEX function in extensions.conf. That might do it. |
| 18:36.19 | ViperNetworks | Under sip.conf I have "dtmfmode = auto". |
| 18:36.26 | DarylVOIP | Viper - I just went through the same thing. |
| 18:36.32 | DarylVOIP | Some carriers just suck at that. |
| 18:36.41 | ViperNetworks | So is it VoIP Carrier problem? |
| 18:37.14 | DarylVOIP | Could be....I'd also try to set RCF2833 in sip.conf for dtmf |
| 18:37.23 | DarylVOIP | But if that doesn't work, you may be looking at bad routes. |
| 18:37.42 | DarylVOIP | For example - MCI isn't passing any DTMF for me to Oman Mobile and a few other UAE destinations. |
| 18:37.50 | ViperNetworks | OK.. well it didnt work with RCF2833 |
| 18:37.54 | ViperNetworks | so I changed to Auto |
| 18:37.56 | ViperNetworks | and still it didnt work |
| 18:38.04 | DarylVOIP | Broadvox is failing to pass it from 1215 |
| 18:38.41 | DarylVOIP | Looking at my termination provider setups, I actually do have dtmfmode=auto for all of them. |
| 18:38.46 | DarylVOIP | So that should be fine. |
| 18:39.01 | DarylVOIP | And, as I mentioned, it's not an all or nothing. Most of my MCI routes pass DTMF fine. |
| 18:39.30 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 18:39.41 | DarylVOIP | TeleGlobe seems to be passing DTMF from Oman Mobile at the moment, but that could be arbirtaged into oblivion at any point. Gotta love voip termination. |
| 18:40.53 | DarylVOIP | Are you using a SIP DID and entering you number on the initial call? |
| 18:41.09 | DarylVOIP | Or is your inital call never answered and you enter your destination on the callback? |
| 18:43.41 | [TK]D-Fender | Strom been talking with JunK-Y and option is open to meet up for food/beer. You interested? |
| 18:45.11 | ViperNetworks | hmm... |
| 18:45.15 | ViperNetworks | I am using SIP DID |
| 18:45.18 | hmmhesays | food and beer good |
| 18:45.21 | ViperNetworks | Calls ae answered |
| 18:45.42 | ViperNetworks | but when prompted to enter the phone number... thats the time it fails |
| 18:45.50 | ViperNetworks | I get back NULL |
| 18:45.55 | DarylVOIP | And the prompting - that happens when you call the DID? |
| 18:46.28 | ViperNetworks | call to a normal land line phone... |
| 18:46.35 | *** part/#asterisk potsboy (n=chrisg@vc-196-207-32-232.3g.vodacom.co.za) |
| 18:46.59 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:47.07 | DarylVOIP | I'm just trying to figure out which call leg your DTMF issue is at. |
| 18:47.20 | ViperNetworks | Second Leg... |
| 18:47.38 | DarylVOIP | In my callback, you call a SIP DID and it never answers. It gets your ANI and drops the call. Then the * box calls you back, and you enter your destination number. |
| 18:47.50 | DarylVOIP | ...second leg...so yours works the same way as mine? |
| 18:48.29 | ViperNetworks | First the Caller calls a DID number and hangs up... then * will call back and Caller will pick up the call... * will provide a prompt to enter a phone number... but then it fails... |
| 18:49.13 | DarylVOIP | OK...gotchya - just making sure we're talking about the same call leg. In that case, my assumption was correct. I'd be looking at you termination provider(s) for an answer as to why they aren't passing DTMF. |
| 18:49.53 | ViperNetworks | this thing only happens with trying outside USA.... within USA DTMF works with same VOIP provider. |
| 18:49.59 | DarylVOIP | Have you checkd SIP debugs to see what you're negotiating between your * box and the provider? |
| 18:50.10 | DarylVOIP | That doesn't mean much. |
| 18:50.37 | DarylVOIP | Routes are resold over and over from carrier to carrier. You may have 3 different Voip providers and be using the exact same route to a given destination. |
| 18:50.39 | ViperNetworks | hmm let me see wat debug has to say.. |
| 18:50.57 | ViperNetworks | true... |
| 18:51.01 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:51.34 | DarylVOIP | Pastebin the SDP portion - I'll see if I can see anything in it for you. |
| 18:51.36 | ViperNetworks | We have XO for USA... and for international is some other random carrier.. |
| 18:52.01 | ViperNetworks | So I believe its the random carrier.. |
| 18:52.06 | ViperNetworks | ok I will do that |
| 18:52.10 | DarylVOIP | Probably. |
| 18:56.19 | *** join/#asterisk saftsack (n=saftsack@217.224.91.120) |
| 19:00.00 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 19:03.18 | *** join/#asterisk rsilva (n=rsilva@bl4-229-90.dsl.telepac.pt) |
| 19:03.24 | rsilva | hi |
| 19:03.42 | rsilva | I need some help in call transfer in asterisknow is anyone available? |
| 19:04.05 | karleeto | rsilva: whats the problem? |
| 19:04.16 | karleeto | rsilva: you should just directly ask the question |
| 19:04.23 | karleeto | rsilva: then maybe someone will answer ;) |
| 19:05.03 | rsilva | How can I do the call transfer on asterisk now? |
| 19:05.18 | rsilva | I have configured the features.conf, but nothing works. |
| 19:06.12 | karleeto | uhhmm, what do you mean by transfer? |
| 19:06.22 | karleeto | transfering a current call to another extension? |
| 19:06.32 | rsilva | I have the extensions 100,101 and 102 |
| 19:06.56 | karleeto | ok... |
| 19:07.01 | rsilva | I do an call from 101 to 102 and then I need to transfer the call to 102, how can I do that. |
| 19:07.55 | rsilva | simple question, but complication resolution, according to what I saw. |
| 19:07.57 | karleeto | if you called from 101 to 102, then you are already talking to 102, why would you want to transfer a call to them when you are already talking to them? |
| 19:08.15 | rsilva | sorry. |
| 19:08.27 | rsilva | 100 to 101 and then to 102 |
| 19:08.32 | rsilva | sorry. |
| 19:08.51 | *** join/#asterisk iBuMp (n=ibump@cpe-66-68-37-190.austin.res.rr.com) |
| 19:10.27 | karleeto | ok, you press transfer, then 102, then tell the person on 102 you have a transfer for them, they say, OK i'll take the call, then you press transfer again, and they are now connected to 100 and not you anymore |
| 19:11.31 | rsilva | I do not have any "transfer" button, that is the problem. |
| 19:12.20 | [TK]D-Fender | rsilva: You should start by telling us what hardware/software you are using so we don't waste a lot of time on useless advice |
| 19:12.25 | rsilva | I'm using one analog, X-Lite (soft phone) and ovislink S100 sip device. |
| 19:12.37 | rsilva | ok. |
| 19:12.41 | rsilva | sorry |
| 19:13.04 | karleeto | my god.. i have no idea then, i use all real stuff ;) |
| 19:13.41 | rsilva | Yes, I understand. |
| 19:13.47 | rsilva | That is my problem. |
| 19:13.51 | *** join/#asterisk sakic (n=sakic@adsl-227-157-12.clt.bellsouth.net) |
| 19:14.05 | rsilva | The ovislink is the S1000e (sorry) |
| 19:14.31 | rsilva | none of this devices has the transfer button, and the X-Lite does not enable it. |
| 19:14.58 | [TK]D-Fender | rsilva: I would suggest switching from X-Lite to ZOIPER. It has a transfer button. |
| 19:15.07 | [TK]D-Fender | rsilva: Amongst other benifits. |
| 19:15.56 | J4k3 | zoiper = idefisk = wtf is an idefisk? |
| 19:16.05 | J4k3 | sounds like something you'd use to tear up old IDE cables or something |
| 19:16.19 | J4k3 | "that little kid came in and used an idefisk on my pc and now I can't get anything to work!" |
| 19:16.50 | rsilva | But how can I put the other devices to transfer calls among them? |
| 19:21.42 | Nivex | zoiper vs jackeniax ? |
| 19:23.30 | DrAk0 | is it possible deny fordwarding from sip clients? |
| 19:23.33 | rsilva | but the other devices can anyone help me? |
| 19:23.41 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:28.41 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.180.56) |
| 19:28.59 | rsilva | ok, zoiper it worked fine to transfer calls, how can I do that in the sip device and in the analog device, is that possible? |
| 19:30.21 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 19:31.18 | *** join/#asterisk DeepY0X (n=DeepY0X@201.240.79.22) |
| 19:33.00 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 19:34.12 | [TK]D-Fender | rsilva: Go read that device's manual to see how you're supposed to transfer calls with it |
| 19:34.45 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:35.00 | lesouvage | Is anybody of you using the aupix video phone in combination with asterisk? |
| 19:36.31 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:37.54 | [TK]D-Fender | rsilva: Is that the ePhone-1000S? |
| 19:40.17 | *** join/#asterisk guillote_GNU (n=bancaria@host128.201-253-17.telecom.net.ar) |
| 19:41.47 | Aces1Up | i have a agi script that executes on incoming calls, I want to do some testing, how could i set up a softphone so that it initiates an incoming call to by box so that i can use if for testing as i do not have any DID's to use right now? |
| 19:42.06 | lesouvage | I mean the Aupix ap-100 phone |
| 19:42.37 | tzanger | holy shit |
| 19:42.42 | tzanger | I have IRC logs here from 2004 |
| 19:42.49 | tzanger | when lmadsen was in school still |
| 19:43.13 | jcolp | scary |
| 19:43.26 | tzanger | I see he changed his name too |
| 19:43.37 | tzanger | let me guess, he's got some peach fuzz on his face to try and look more mature too |
| 19:44.39 | Aces1Up | or how could i set up my soft phone so that it immediately executes the agi script? |
| 19:44.59 | [TK]D-Fender | Aces1Up: This is serios 101 stuff. Set up your softphone on * and dail an exten for your AGI. end of story. |
| 19:45.39 | lmadsen | tzanger: wow... when I was in school? that's nuts |
| 19:46.12 | [TK]D-Fender | Aces1Up: Just set up an exten and DIAL it. |
| 19:46.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:47.58 | tzanger | lmadsen: yep |
| 19:48.03 | *** part/#asterisk davidm870 (n=davidm87@66.150.12.150) |
| 19:48.39 | tzanger | --- Log opened Thu Dec 09 08:56:10 2004 |
| 19:48.45 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 19:48.46 | tzanger | 09:03 < blitzrage> my irssi client doesn't support extended characters :) |
| 19:48.46 | tzanger | 09:04 * blitzrage goes to school |
| 19:49.33 | tzanger | $ wc -l \#asterisk.log |
| 19:49.33 | tzanger | 3613049 #asterisk.log |
| 19:49.38 | tzanger | that's a LOT of inane banter |
| 19:53.03 | *** join/#asterisk rsilva1 (n=rsilva@bl4-229-90.dsl.telepac.pt) |
| 19:53.10 | rsilva1 | sorry to take so long |
| 19:53.22 | rsilva1 | yes it is the ePhone-S1000e |
| 19:54.38 | lmadsen | tzanger: haha... ya -- it is :) |
| 19:54.53 | tzanger | lmadsen: why the name change? |
| 19:54.57 | tzanger | trying to look more professional? |
| 19:55.48 | lmadsen | I guess... :) |
| 19:57.24 | rsilva1 | How can I transfer call from an analog phone to another phone on asterisk now? |
| 20:00.28 | *** join/#asterisk doughecka (n=doug@unaffiliated/doughecka) |
| 20:00.37 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 20:00.42 | [TK]D-Fender | rsilva Check your dilaplan and make sure you are using the 'tT" options and read up on doing transfers via DTMF |
| 20:01.03 | doughecka | anyone have any experience getting the distingtive ring to work on the grandstream gxp2000? |
| 20:01.19 | doughecka | distinctive* |
| 20:02.09 | Mercestes | ~gs |
| 20:02.10 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:02.13 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 20:02.14 | Mercestes | ~phones |
| 20:02.15 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 20:02.36 | rsilva1 | "tT"? no I did not find anything that explains my problem on the web to do the transfer via DTMF, can you give me some website to check that? |
| 20:02.59 | doughecka | thanks |
| 20:04.32 | [TK]D-Fender | doughecka: You're best served checking the WIKI page on them. I'm pretty sure I saw it listed there. |
| 20:04.51 | *** join/#asterisk Ebola (n=Ebola@host86-139-49-76.range86-139.btcentralplus.com) |
| 20:05.08 | doughecka | yea, I saw it mentioned there |
| 20:05.30 | doughecka | I tried setting the options, but so far nothing can get the dumb things to ring anything other than the default |
| 20:05.32 | [TK]D-Fender | rsilva : "core show application dial" and the WIKI |
| 20:05.48 | [TK]D-Fender | rsilva : They manufacturers page says noting about transfering calls, 3-way, or ANY reasonable functionality. It looks loike the lowest end junker out there. |
| 20:07.23 | fetcher | With Polycom IP 501's, is there a way to have a Transfer'ed call (transferred via the phone) carry through its original Caller-ID to the target phone? (instead of always showing the receptionist's extension, etc)? |
| 20:07.54 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 20:08.22 | [TK]D-Fender | fetcher: Yes, do a blind transfer instead of an attended one |
| 20:09.22 | errr | if I get a complaint from time to time from someone who is calling into our system saying they hear static on the call would that be something where their line calling into us had a problem, or more likely a problem with my pbx system? |
| 20:09.47 | [TK]D-Fender | errr: Could be either |
| 20:10.10 | errr | [TK]D-Fender: every time it happens the call sounds clear on our end (if that matters) |
| 20:10.39 | fetcher | [TK]D-Fender: using the "Blind" soft-key? thanks, hadn't tried that |
| 20:10.56 | [TK]D-Fender | fetcher: yes. |
| 20:10.58 | errr | [TK]D-Fender: is there a way to find out if it was my pbx? |
| 20:11.15 | Mercestes | errr: might want to try a ztmonitor. |
| 20:11.28 | errr | Mercestes: OK, thanks |
| 20:11.42 | [TK]D-Fender | errr: plug an analog phone in parallel. When they complain of static, pick up the parallel phone and hang up on *. Disconnect the analog line from your card. Then ask them again. |
| 20:11.47 | Mercestes | the tx side would stay "high" due to the static you are sending them, i fyou were sending them static. |
| 20:13.22 | errr | we dont have any analog stuff here |
| 20:13.26 | rsilva1 | D-Fender can you tell me what is the command to enable that feature? I have configure the features.conf, please. |
| 20:16.40 | *** join/#asterisk Advant (i=advant@cpe-075-181-103-211.carolina.res.rr.com) |
| 20:17.31 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 20:18.48 | rsilva1 | D-Ferder: the command is this? "console dial 102 T" |
| 20:18.55 | Defraz | When I get an inbound call from my SIP provider I get this in my log 12084039164;phonecontext=+1 |
| 20:19.11 | Defraz | what does it mean and can I strip it off? |
| 20:19.19 | Defraz | the phonecontext=+1 |
| 20:20.08 | De_Mon | Jon Webster" <sip:jonfoo@pbx.elephantoutlook.com>;tag=d7cff87b84a79ee2o0 |
| 20:20.09 | De_Mon | To: "Jon Webster" <sip:jonfoo@pbx.elephantoutlook.com>;tag=as4dc3f3f9 |
| 20:20.12 | De_Mon | gah |
| 20:20.19 | De_Mon | Jon Webster" <sip:jonfoo@pbx.elephantoutlook.com>;tag=d7cff87b84a79ee2o0 |
| 20:20.19 | De_Mon | To: "Jon Webster" <sip:jonfoo@pbx.elephantoutlook.com>;tag=as4dc3f3f9 |
| 20:20.25 | De_Mon | damn you windows!! |
| 20:20.59 | Corydon76-dig | I love elephants. Especially pink ones. |
| 20:21.28 | De_Mon | I changed around the NAT setup for a phone, and now it refuses to REGISTER. I changed the user/password to bogus values and now asterisk says "Device does not match ACL" |
| 20:21.39 | *** join/#asterisk lbow (n=lbow@41-195-77-226.access.uunet.co.za) |
| 20:22.25 | De_Mon | restored phone to factory default and it still says that, I was expecting a "Not Authorized" type of error, not this ACL stuff... |
| 20:23.13 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 20:23.54 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 20:24.17 | De_Mon | When I try to register with a valid user/pass it says "SIP/2.0 401 Unauthorized |
| 20:24.29 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-134-148.buff.east.verizon.net) |
| 20:24.50 | Mercestes | De_Mon, pastebin your sip.conf |
| 20:26.13 | *** join/#asterisk jcolp (n=file@asterisk/developer-and-muffin-lover/file) |
| 20:26.13 | *** join/#asterisk JunK-Y (n=junky@modemcable223.205-56-74.mc.videotron.ca) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk AndyGraybeal (n=andy@casanueva.wifi.frognet.net) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) [NETSPLIT VICTIM] |
| 20:26.13 | *** join/#asterisk znoG (n=gs@130-215-114-200.fibertel.com.ar) [NETSPLIT VICTIM] |
| 20:26.13 | *** mode/#asterisk [+o jcolp] by irc.freenode.net |
| 20:26.24 | Yarq | Hello |
| 20:26.26 | *** part/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:26.29 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 20:26.35 | doughecka | foom |
| 20:26.48 | Mercestes | Holy.... |
| 20:28.59 | De_Mon | Mercestes http://pastebin.ca/658763 |
| 20:36.00 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-191-178.dhcp.embarqhsd.net) |
| 20:36.25 | De_Mon | Mercestes see anything? |
| 20:37.24 | *** join/#asterisk jart (n=user@ool-43509aa5.dyn.optonline.net) |
| 20:37.57 | Mercestes | I would look for random things in your sip.conf which would indicate some level of ACL or origination restriction, like host= or something. |
| 20:38.18 | Mercestes | nothing jumped out at me. Domain= had me scratching my head but ...anyways, that's what I'm looking for, is stuff of that nature. |
| 20:38.19 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 20:38.31 | Mercestes | you can google each sip.conf entry to see if any of that matches what you are looking at. |
| 20:42.16 | De_Mon | the config worked untill I made changes to the router at the phone station |
| 20:42.31 | fetcher | When transferring calls to the parking extension, can Asterisk be made to immediately hang up after announcing the park position? (Giving transferring party their dialtone back?) |
| 20:42.41 | De_Mon | but why theres an ACL error for an invalid auth attempt is just weird |
| 20:44.49 | Yarq | -- Got SIP response 481 "Call Leg/Transaction Does Not Exist" bback from ipaddr. |
| 20:44.55 | Yarq | ? |
| 20:45.02 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:47.41 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:50.37 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 20:51.51 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:54.31 | *** join/#asterisk travesty (n=user@ool-43509aa5.dyn.optonline.net) |
| 20:55.54 | geoff_k | mm citrix aquiring xensource |
| 20:56.59 | *** join/#asterisk shinao1 (n=shinao1@41.219.201.1) |
| 20:57.27 | *** join/#asterisk AJaymn (i=AJaymn@71-82-218-158.dhcp.mdsn.wi.charter.com) |
| 20:57.50 | De_Mon | I hate NAT |
| 20:57.58 | De_Mon | or maybe its just sip phones... |
| 20:58.06 | AJaymn | with the new Zaptel 1.4 and kernel 2.6 do you still need to have USB for timing? |
| 20:58.57 | Mercestes | I never realized USB was required for timing. =/ |
| 20:59.17 | De_Mon | it was a long time ago, but I don't think its required for 1.2... |
| 20:59.18 | elixer | AJaymn: no |
| 20:59.35 | travesty | are there any example configurations for two * machines with e1 cards to talk to each other? |
| 20:59.43 | travesty | i can't find anything on voip-info |
| 21:00.03 | travesty | because i'm confused by pritimer, pri_cpe/pri_net, etc |
| 21:00.06 | AJaymn | elixer: do you know where the howto is to install it? |
| 21:01.01 | Mercestes | Uh, voip-info.org maybe? |
| 21:01.04 | elixer | AJaymn: install USB? the zaptel configure/make will pick up the fact that you have 2.6 and use its internal timing (if i remember correctly) |
| 21:01.58 | De_Mon | AAAAAAAAHHHHHHHHHHHHH |
| 21:02.18 | De_Mon | I just did another factory reset and re-entered the exact same settings I have been for the last 3 hours. and IT WORKS |
| 21:02.47 | *** join/#asterisk saftsack (n=saftsack@pD9E05B78.dip.t-dialin.net) |
| 21:02.50 | *** part/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 21:04.00 | Deeewayne | travesty: try configuring for CAS if PRI is too confusing |
| 21:04.46 | travesty | must have pri :( |
| 21:04.54 | wunderkin | travesty, one should be net, the other cpe |
| 21:04.59 | travesty | one really weird thing that has been happening |
| 21:05.41 | [TK]D-Fender | travesty, Why on earth would you use E1 cards to connect 2 *'s? Collosal waste of money |
| 21:05.59 | Mercestes | [TK]D-Fender, to run it into a fax server with a T1 faxboard? |
| 21:06.00 | shinao1 | hi guys.. i was here earlier.. asking for help with integration Asterisk and a legacy PBX..Panasonic..not sure of the brand |
| 21:06.09 | travesty | i can't get e1 to work so i'm trying to setup t1. on a box with a digium card i changed the configuration, ztcfg -vvvv shows a correct t1 configuration, but when i load up asterisk it tells me: Unable to open channel 16: Device or resource busy |
| 21:06.09 | travesty | here = 0, tmp->channel = 16, channel = 16 |
| 21:06.09 | *** join/#asterisk MdeP (n=mdep@56-92-22-190.adsl.tie.cl) |
| 21:06.12 | shinao1 | [TK]D-Fender: hi |
| 21:06.18 | lirakis | rarg! .. polycoms can be so evil! |
| 21:06.24 | travesty | because voip is illegal in india :) |
| 21:06.30 | [TK]D-Fender | shinao1, So how many ports on each side, and of what type? |
| 21:06.38 | Mercestes | travesty, D-Chan is 24 on a T1, not 16 like on a E1. |
| 21:06.49 | Mercestes | travesty, and use encryption to hide yoru VoIP traffic from those Indian commies. |
| 21:06.53 | kombi | where do you set the web-interface credentials in polycom phones? |
| 21:06.59 | lirakis | ive tried to factory default this dang 301 several times and it has some remnants of the previous config.. enough for it to d/l from the boot server |
| 21:07.06 | travesty | but here's the weird thing, it gives me that error when it reads "channel => 1-23" from zapata.conf |
| 21:07.06 | lirakis | .. maybe i have to disable dhcp |
| 21:07.35 | [TK]D-Fender | kombi, You'll never use it period if you know whats good for you... |
| 21:07.36 | Mercestes | travesty, ok, then your circuit is likely an E1 |
| 21:07.47 | travesty | it's not plugged in to anything |
| 21:07.56 | travesty | i'm just connecting two boxes i setup |
| 21:07.59 | pigpen | [TK]D-Fender, hey, remember the "bell" solution I was talking about, where a gsm file would play into a Meetme generated by the page app? |
| 21:08.01 | travesty | to test this machine i'm building |
| 21:08.09 | pigpen | couldn't I use the AGI_BACKGROUND? |
| 21:08.13 | [TK]D-Fender | pigpen, Yup. What'd you do for it? |
| 21:08.24 | travesty | let me pastebin this |
| 21:08.39 | Mercestes | Yea, I was about to suggest that actually |
| 21:08.40 | shinao1 | sorry [TK]D-Fender im not sure what you mean, but, the head office site has 2 PSTN phone numbers coming out of the PBX, and there are supposedly 25 extensions.. the branch office either has nothing installed, or they have a smaller PBX and 3 extensions.. is that helpful? |
| 21:08.40 | pigpen | Well, I have been so dam busy, I am just not getting to implement it. |
| 21:08.42 | kombi | Fender: ok, then how do I stop the bloody time from flashing on display.. |
| 21:08.45 | Mercestes | Usually Fender beats me to it. |
| 21:08.57 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 21:08.59 | pigpen | but wondered what you thought of possibly using the agi_background. |
| 21:09.15 | pigpen | I know you mentioned one way, but I was unsure of the process. |
| 21:09.19 | [TK]D-Fender | shinao1, No, I need to know what kind of jacks are AVAILABLE (not used!) and of what kind on EACH SIDE. |
| 21:09.39 | [TK]D-Fender | shinao1, What does it matter if you have 100 line but no ports FREE? |
| 21:09.47 | shinao1 | im sorry but the most experience i had with asterisk is setting up SIP for the office, now im elected telephony officer! :( |
| 21:09.50 | shinao1 | oh RJ11 Ports |
| 21:10.10 | Mercestes | shinao1, Want my resume? |
| 21:10.15 | shinao1 | shoot sorry :) |
| 21:10.17 | shinao1 | <PROTECTED> |
| 21:10.23 | g1powermac | shinao1, heh, I got that just because I run everything else here at our business :-) |
| 21:10.28 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 21:10.33 | g1powermac | shinao1, never touched asterisk in my life |
| 21:10.36 | Mercestes | however many you want to put plugs in, usually. |
| 21:10.43 | shinao1 | i have yet to see the actual equipment, im from another sub-company |
| 21:11.00 | shinao1 | ports free on the PBX, i get it |
| 21:11.19 | Mercestes | Why are you trying to integrate * and panasonic? |
| 21:11.25 | shinao1 | :) cut it out guys |
| 21:11.47 | Mercestes | no, really, I'm being serious |
| 21:11.53 | shinao1 | because management wants to save on getting astribanks , which i suggested Mercestes |
| 21:12.16 | Mercestes | ~cheap |
| 21:12.16 | jbot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 21:12.16 | travesty | here isa good description of the pri problem i'm having: http://pastebin.com/m184c855c |
| 21:12.28 | shinao1 | very cheap i agree |
| 21:12.34 | Mercestes | shinao1, here is your main problem. Imagine with me, for a moment.... |
| 21:12.42 | Mercestes | your phones go offline.... |
| 21:12.55 | Mercestes | no inbound..no outbound..management is *screaming* at you...your gonig to be fired in the next 30 seconds. |
| 21:13.02 | Mercestes | Asterisk, as far as you can tell, is *fine* |
| 21:13.21 | Mercestes | Panasonic vendor shows up....takes one look at all the crap you have wired to his perfect, precious system, making it look like a telecom borg.... |
| 21:13.21 | shinao1 | ok so how do i get out of a bad sitch? im already ostracized by being the only Linux user in the nearest 5 miles |
| 21:13.21 | shinao1 | hmm |
| 21:13.23 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 21:13.24 | shinao1 | i see Mercestes |
| 21:13.28 | Mercestes | and he points at it, screams "DEVIL!" and blames you entirely. |
| 21:13.55 | shinao1 | yeah, because i use Lye-nux, apparently |
| 21:14.05 | kombi | How do you soft-reboot a polycom? |
| 21:14.11 | Mercestes | well, no, just because you have something wired to his system. |
| 21:14.19 | Mercestes | kombi: links |
| 21:14.30 | Mercestes | kombi, sip notify polycom-chk-config peer |
| 21:14.35 | shinao1 | so what do you advise, Mercestes |
| 21:14.46 | Mercestes | kombi, menu, 3, 2, 456, enter, 3, 2, yes. |
| 21:14.51 | Deeewayne | travesty: is chan_zap already loaded ? |
| 21:14.52 | kombi | Mercedes: wicked! |
| 21:14.54 | Mercestes | shinao1, what end result do you want? |
| 21:15.16 | travesty | Deeewayne: the same thing happens if i unload and load it again |
| 21:15.29 | travesty | maybe i should try installing 1.2 |
| 21:15.46 | [TK]D-Fender | shinao1, You are running yourself around in circles. |
| 21:16.06 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 21:16.14 | [TK]D-Fender | shinao1, Just figure out what you HAVE (used & free) and come back with that |
| 21:16.22 | Deeewayne | what does zap show channels display ? |
| 21:16.52 | travesty | "no such command 'zap show'" because chan_zap.so halts the load process at the ERROR |
| 21:17.01 | [TK]D-Fender | kombi, You stop the clock from flashing when you give it a SNTP time server to synch with. |
| 21:17.34 | Yourname` | Hello, I installed Asterisk and all of its libpri.zaptel, and all that stuff into /usr/src/asterisk. Now, the AMD tutorial says rm -rf asterisk http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD -> if I do it, it'll delete all the contents of /usr/src/asterisk that happens to have all the other asterisk related stuff.. what do I do? |
| 21:18.14 | Mercestes | Yourname`, well, first.....you don't install into usr/src/asterisk |
| 21:18.24 | Mercestes | Yourname`, you install *from* /usr/src/asterisk |
| 21:18.50 | Yourname` | Mercestes: Ah.. so even if I remove all of that /usr/src/asterisk.. it won't effect the installED applications? |
| 21:18.51 | Mercestes | Yourname`, second.... I tend to suggest running anything involving rm and both -r and -f. |
| 21:19.18 | Mercestes | Yourname`, if you did a successful make && make install from /usr/src/asterisk you can safely delete the src files but......why would you want to? |
| 21:19.36 | Yourname` | Mercestes: Like I said, that tutorial wants to. |
| 21:19.39 | Mercestes | Yourname`, what are you going to do if you ever have to recompile? Or suffer a power loss? bad HDD? Data corruption? |
| 21:20.05 | Mercestes | Yourname`, It's a wiki page, which means it could have been written by a retarded chimp. We don't know. |
| 21:20.41 | Mercestes | Yourname`, Hell, *I* could have editted it just now to tell you to do something inappropriate with dd, /dev/urandom, and your /dev/hda |
| 21:21.05 | Yourname` | Mercestes: Can I copy/paste a couple lines here? |
| 21:21.08 | Deeewayne | travesty: the first message from your pastebin was that ZapSendKeypadFacility was already loaded |
| 21:21.13 | Mercestes | no |
| 21:21.19 | Yourname` | Mercestes: Well, 4 lines. In your msg then? |
| 21:21.21 | Mercestes | you can pastebin them. |
| 21:21.25 | Mercestes | or msg me. |
| 21:21.37 | travesty | Deeewayne: that's just because i didn't run unload chan_zap before it |
| 21:21.40 | Mercestes | People pasting in here obscures my cynicism and abuse and we can't have that. |
| 21:21.52 | Yourname` | lol |
| 21:22.02 | Yourname` | Mercestes: Those are the contents of /usr/src/asterisk |
| 21:22.10 | travesty | Deeewayne: if i unload chan_zap entirely, and load it again, the same thing happens except that first message |
| 21:22.18 | Mercestes | Ahh |
| 21:22.27 | Mercestes | The asterisk subdir was kind of unnecessary but I see what you did. |
| 21:22.38 | Mercestes | no reason to rm -rf asterisk unless your critical on diskspace. |
| 21:22.47 | Yourname` | Yup, and now this tutorial asks me to rm -rf .. |
| 21:22.57 | Yourname` | Which I have no problem doing. It's a 250gig HDD. |
| 21:23.06 | Yourname` | So, nothing should be affected, right? |
| 21:24.26 | Deeewayne | well your pastebin shows you trying to load a module that is already loaded. It's probably not good to consciously do the wrong thing when you are trying to debug something |
| 21:24.58 | shinao1 | Mercestes: end result? I want to be able to achieve IVR, co |
| 21:25.12 | shinao1 | sorry inter office calling on the cheap |
| 21:25.37 | shinao1 | thanks [TK]D-Fender i do think i need more research on this |
| 21:26.03 | Mercestes | shinao1, plugging it into your panasonic won't give you cheap interoffice calling. |
| 21:26.24 | [TK]D-Fender | shinao1, there is very little to research. What kind of ports available on box1? Same for box 2. Whats so complicated about this? |
| 21:26.26 | travesty | Deeewayne: you're missing the real problem, the issue i'm having is that it's crashing when it tries to load channel 16, when everything is perfectly configured to be a T1 |
| 21:26.39 | Yourname` | Mercestes: So if I do rm -rf to that asterisk directory, nothing should be affected, right? |
| 21:26.45 | shinao1 | [TK]D-Fender: i havent physically seen the boxes myself |
| 21:26.47 | Mercestes | Yourname`, why risk it? |
| 21:26.59 | [TK]D-Fender | shinao1, Place a phone call. This isn't rocket science. |
| 21:27.02 | Yourname` | lol ok |
| 21:27.06 | Yourname` | Thanks. |
| 21:27.07 | shinao1 | all i have is info from a coworker whos mandated to head this project.. |
| 21:27.09 | Mercestes | np |
| 21:27.10 | shinao1 | lol |
| 21:27.13 | shinao1 | i hear you |
| 21:27.39 | shinao1 | but i WAS asking, how many ports do i need to make things work? |
| 21:27.39 | travesty | i'm going to try installing asterisk 1.2 |
| 21:28.00 | AJaymn | xlixer: do you know where i can find a howto for installing Zaptel 1.4 on kernel 2.6? |
| 21:28.08 | Mercestes | [TK]D-Fender, That must be a new feature in Asterisk. randomly plug it into any other phone solution, and it instantly provides you with cheaper rates regardless of implementation. |
| 21:28.26 | Mercestes | AJaymn, try the wiki |
| 21:28.55 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 21:30.11 | travesty | thank you for your help Deeewayne |
| 21:30.19 | Deeewayne | did something work ? |
| 21:30.39 | travesty | not yet, installing 1.2, i'll let you know if the issue is solved |
| 21:30.47 | travesty | it's really weird |
| 21:30.58 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 21:31.00 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 21:31.03 | Deeewayne | do you have a rxgain setting in /etc/asterisk/zapata.conf ? |
| 21:31.20 | travesty | yes, it's set to the default of 0.0 |
| 21:31.26 | travesty | that was another kind of weird thing... |
| 21:34.03 | Yourname` | Mercestes: If I don't follow that one stupid rm -rf step, I'm unable to do : /usr/src/asterisk/asterisk-1.2.23/contrib/scripts/astxs -install -autoload app_amd.c |
| 21:34.19 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 21:37.31 | travesty | ah found it |
| 21:38.01 | travesty | i set b8zs signalling in zaptel.conf by mistake, which is for e1 lines, so asterisk freaked out |
| 21:38.32 | travesty | oh wait no.... |
| 21:38.35 | pigpen | b8zs is not only for e1's |
| 21:38.45 | kombi | Fender: Doesn't actually make more sense to use overrides over copies of sip.cfg and phone1.cfg? That way configs persist even an upgrade.. |
| 21:38.52 | kombi | ..it.. |
| 21:39.01 | travesty | n/m b8zs is for T1s, back to the drawing borad |
| 21:39.51 | mog | travesty: most t1 lines are esf,b8zs |
| 21:40.02 | mog | some are d4, ami but its rare |
| 21:40.08 | travesty | mog: that's how i have mine setup, this is really starting to look like a bug |
| 21:40.29 | mog | whats the probe bob. |
| 21:40.32 | travesty | maybe my hardware got cozy being an e1 card, and is resisting me telling it to use t1 signalling now |
| 21:40.38 | *** part/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 21:40.55 | mog | do a cat /proc/zaptel/1 |
| 21:41.02 | mog | tell me how many channels it says it has |
| 21:41.12 | Mercestes | Yourname`, if you *do* do that stupid rm -rf step....you won't have a /usr/src/asterisk/asterisk-1.2.23/contrib/scripts/astxs -install -autoload app_amd.c |
| 21:41.31 | Yourname` | Mercestes: I don't know why, but rm -rf asterisk made it work. |
| 21:41.41 | Yourname` | Anyway, I guess I'll just follow the tutorial. |
| 21:41.50 | Mercestes | That would probably be best.. |
| 21:41.51 | travesty | mog: http://pastebin.com/m7b54e68d |
| 21:43.01 | travesty | which is strange because on the other box it shows the same config only it doesn't bother showing channels after 24 |
| 21:43.13 | mog | and you want i tto be in t1 mode? |
| 21:43.17 | travesty | yes i do |
| 21:43.36 | travesty | i even tried rebooting |
| 21:43.46 | mog | rmmod wcte11xp ; modprobe wcte11xp t1e1overridemode=0 ; ztcfg -vvv |
| 21:43.50 | mog | is what you need |
| 21:44.29 | mog | and then you can just put t1e1override=0 in your modprobe.conf |
| 21:44.53 | mog | no mode |
| 21:45.00 | mog | just t1e1override=0 |
| 21:45.03 | mog | sorry about that |
| 21:45.16 | travesty | that worked |
| 21:45.18 | mog | woot |
| 21:45.25 | mog | and it should be happy now |
| 21:45.32 | mog | card is probably jumpered to e1 mode |
| 21:45.33 | travesty | and asterisk /is/ happy |
| 21:45.42 | mog | you can fix it manually, but doing it in software is just as good |
| 21:46.00 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:46.06 | mog | be sure to add it to modprobe.conf for next reboot |
| 21:46.13 | travesty | it is! i've never seen those jumpers |
| 21:46.42 | mog | its at the top and back of card |
| 21:46.43 | travesty | thank you |
| 21:46.49 | mog | its just one jumper on the single span card |
| 21:46.58 | travesty | <--- used to sangoma cards |
| 21:47.26 | mog | as they keep food on my table |
| 21:47.31 | mog | and work well |
| 21:47.38 | travesty | you work for digium? |
| 21:47.42 | Mercestes | and don't segfault your kernel. |
| 21:47.48 | mog | yes |
| 21:47.51 | mog | and yes |
| 21:48.04 | [hC] | sangoma cards have panic'ed my kernel in the past, but havent recently. |
| 21:48.21 | travesty | digium cards have panicked my kernel in the past ;) |
| 21:48.27 | mog | eep |
| 21:48.32 | [hC] | oh wait, mog is present. s/sangoma/the s word which we will not say/ |
| 21:48.33 | [hC] | :) |
| 21:48.34 | Mercestes | [hC], yea, that's kinda how it goes. If you do it *right* it works fine, but if you screw up in any way (or do something even slightly unexpected) it eats yoru kernel. |
| 21:48.40 | travesty | some spinlock thing back in 2005 |
| 21:48.42 | mog | heh |
| 21:48.58 | mog | i can stand those who must not be name |
| 21:49.00 | mog | ^_^ |
| 21:49.07 | [hC] | mog, im going to scout out the nearest jack in the box from our hotel again this year, just incase. |
| 21:49.09 | travesty | i remember when you guys came out with the 24 port card, we were so happy |
| 21:49.10 | [hC] | :) |
| 21:49.20 | mog | nice |
| 21:49.23 | mog | im hoping to go |
| 21:49.27 | mog | but dont know for sure |
| 21:49.32 | mog | glad you like it |
| 21:49.38 | travesty | we actually managed to get either 5 or 6 4-ports in the same system all on their own interrupts before that |
| 21:49.47 | mog | heh |
| 21:49.49 | [hC] | you might not go this year?? boo hiss! |
| 21:49.51 | mog | that must have been fun |
| 21:50.09 | travesty | go where? astricon? |
| 21:50.15 | [hC] | yuh |
| 21:50.16 | mog | not my decisions send your complaints to bigboys@digium.com |
| 21:50.23 | mog | yes |
| 21:50.30 | mog | i have gotten to all the others |
| 21:50.34 | travesty | i'm supposed to speak there but i think i'm going to cancel |
| 21:50.34 | mog | they are a lot of fun |
| 21:50.35 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 21:50.42 | mog | WHAT?!?!! |
| 21:50.47 | [hC] | travesty: dude.. |
| 21:50.50 | Yarq | how do you enable direct sip connection to asterisk (when you enter the ip the ivr responds) |
| 21:50.55 | travesty | what? |
| 21:50.59 | [hC] | travesty: if you're going to cancel, you better do it like.. right now.. that will throw everything off! |
| 21:51.11 | travesty | ok i will send out an email tonight |
| 21:51.18 | travesty | so i don't cause too much of a problem |
| 21:51.41 | [hC] | yeah man, alot of planning goes into astricon, im su re the less wrenches they're thrown they'd appreciate. |
| 21:51.54 | mog | yup yup |
| 21:51.55 | travesty | i'm unemployed right now so money is tighter |
| 21:52.05 | mog | jobs@digium.com |
| 21:52.15 | [hC] | what were you going to speak about? |
| 21:52.23 | travesty | stuff, you know |
| 21:52.29 | Yarq | do you have to have a soundcard on the server in order to get asterisk work :S |
| 21:52.31 | travesty | i don't know if i can work in IT again for a while |
| 21:52.53 | Mercestes | s/wrenches/wenches |
| 21:53.01 | travesty | my life kind of fell apart with this last job due to burn out |
| 21:53.01 | Mercestes | There should be wenches at astricon |
| 21:53.10 | Mercestes | Hey, can I send my resume there? >.> |
| 21:53.20 | mog | http://www.digium.com/en/company/careers/ |
| 21:54.31 | Mercestes | hey, I fit atleast one of those. |
| 21:54.48 | Mercestes | I'll posty on over. |
| 21:54.49 | mog | nice |
| 21:55.19 | Sweeper | mmm, likewise |
| 21:55.28 | travesty | mog: thanks again for your help |
| 21:55.30 | Sweeper | mog: is the support tech job remote work? |
| 21:55.46 | [hC] | I cant apply. :( |
| 21:55.46 | mog | not sure Sweeper |
| 21:55.52 | mog | no problem travesty |
| 21:55.55 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 21:56.00 | [hC] | I poured over the page looking for lackey, but it just wasnt there. |
| 21:56.03 | flujan | hi guys, little doubt... :) |
| 21:56.08 | mog | tech support is done on site, but not sure if thats same job your looking at |
| 21:56.16 | mog | heh hc |
| 21:56.22 | [hC] | i cant abandon my career dreams!! |
| 21:56.23 | [hC] | :P |
| 21:56.30 | flujan | I i issue a monitor using the manager interface, the channel being monitored will have all incoming and outgoing calls recorded? |
| 21:56.33 | Sweeper | this one just talks about phone and email |
| 21:56.33 | mog | i could use a lackey hc |
| 21:56.37 | mog | how much you looking for |
| 21:56.43 | [hC] | hmm |
| 21:56.45 | mog | thats done on site currently |
| 21:57.01 | flujan | and if this guy is a member of a queue, it will record all calls that the channel answered coming from the queue? |
| 21:57.09 | [hC] | good question mog... i suppose a steady stream of good food and rum and coke will suit me just fine. oh, and some place to sleep. |
| 21:57.11 | [hC] | :) |
| 21:57.20 | [TK]D-Fender | flujan, channel = call, NOT device |
| 21:57.29 | mog | i can meet those requirements |
| 21:57.34 | [hC] | woot! im in! |
| 21:57.34 | mog | in exchange for your freedom ^_^ |
| 21:57.43 | mog | cant have my lackeys goofing off |
| 21:57.51 | flujan | [TK]D-Fender: understood... Thanks for the tip [TK]D-Fender :) |
| 21:57.51 | [hC] | ok cool, just come to astricon and ill sign the papers. :P |
| 21:58.24 | mog | its a deal |
| 21:58.30 | [TK]D-Fender | flujan, What you need is all dialplan |
| 21:59.04 | flujan | I am currently running asterisk on a 2x Dual Core AMD Opteron(tm) Processor 265. running 90 channnels with a te412p card... |
| 21:59.11 | flujan | sometimes the load of the machine hits 4 |
| 21:59.22 | flujan | I need to low this load |
| 21:59.29 | Sweeper | flujan: transcoding? |
| 21:59.35 | mog | you havent lived till you have seen a system hit 137 |
| 21:59.39 | J4k3 | a load average of 4 on a quad processor machine doesn't sound terribly unacceptable |
| 21:59.47 | mog | not my asterisk machine but it was a lot of fun |
| 22:00.00 | flujan | just using sip softphones and E1 links. |
| 22:00.08 | flujan | I am using ulaw both ends... |
| 22:00.18 | Sweeper | hmm |
| 22:00.24 | mog | heh mine was caused by benchmarking my jabber server |
| 22:00.31 | mog | got 10,000 messages through a second |
| 22:00.33 | [hC] | sounds like you just need to reroute the encryptions |
| 22:00.42 | flujan | mog gor sure... |
| 22:00.53 | mog | ? |
| 22:00.56 | J4k3 | jabtastic! |
| 22:00.57 | flujan | do you guys have ideas why this heavy load on the machine? |
| 22:01.24 | flujan | [TK]D-Fender: I am already doing it using the dialplan, but not for the queue members... |
| 22:02.28 | flujan | [TK]D-Fender: I have a queue with 40 members... I just want to monitor 10 of them... I dunno how to monitor this specific channels, so I set monitor-format=wav on the queues.conf and all queue calls are recorded... :( |
| 22:03.17 | flujan | J4k3: but sometimes asterisk delays to deliver the calls... stop recording the queue_log file and so on... |
| 22:03.39 | kombi | anyone know a good xml editor for command line? |
| 22:03.42 | flujan | J4k3: I am having a lot of problem due to this load... when the machine have a low usage, i doesn`t get into this problems... |
| 22:03.53 | J4k3 | ick, is the network otherwise healthy (is it having hell keeping the calls happy?) |
| 22:04.13 | flujan | kombi: ruby :) |
| 22:04.16 | Yourname` | Does anyone have any exprience setting up TeleVantage 9133i SIP phone with Asterisk? (It asks for TFTP and stuff like that!) |
| 22:04.20 | Yourname` | ~televantage |
| 22:04.36 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 22:05.52 | elixer | do you NEED to patch/recompile Zaptel to install a Sangoma card? |
| 22:05.54 | flujan | J4k3: yeap, I think the network is ok. Sometime a call just He__o (i dunno how to say this in english.) I don hear the l's |
| 22:06.05 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 22:06.12 | elixer | its an A400 card which we have 3 FXS modules in |
| 22:06.35 | elixer | reading the patch files they have it doesn't seem like its adding anything other than some echo cancellation stuff, but that is hardware specific |
| 22:06.39 | flujan | how many calls this machine I have would be able to handle? |
| 22:08.23 | [TK]D-Fender | flujan, how are queu memebers being called? |
| 22:08.48 | flujan | [TK]D-Fender: realtime from a queue_members table. |
| 22:09.18 | [TK]D-Fender | flujan, Thats just a DATA source. What kind of CHANNEL? |
| 22:10.35 | Yourname` | So, no one even heard of making a TeleVantage phone work wth Asterisk? |
| 22:10.43 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 22:11.08 | flujan | [TK]D-Fender: :) SIP/40000 up to SIP/40100 |
| 22:13.48 | *** join/#asterisk kkn088 (n=kikoun@84.4.151.213) |
| 22:13.49 | [T]ank | i am trying to make a phone so it cannot take more calls if it is already on one. Here is how I have tried... however i expected the group count to be one... not 0. can anyone see a mistake in how I am doing this? http://pastebin.ca/658886 |
| 22:14.45 | [TK]D-Fender | flujan, You add SIP devices to your queue directly? |
| 22:16.04 | [T]ank | since the group is being set, shouldn't the group count be 1? |
| 22:17.13 | flujan | [TK]D-Fender: Yes, i control the login use a web app. When a user log on the web i add his device to the queue_members table and he starts to receive calls. |
| 22:19.08 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 22:19.38 | aut | does the Polycom 330 have a re4ally good speakerphone? |
| 22:19.43 | flujan | [TK]D-Fender: is there a problem in this aproach? |
| 22:20.57 | [TK]D-Fender | aut, yup |
| 22:21.12 | aut | [TK]D-Fender, any idea what the HD Voice stuff is? |
| 22:21.21 | aut | sounds like a marketing term :) |
| 22:21.23 | [TK]D-Fender | aut, 501+ might be better given the increased size/weight |
| 22:21.42 | Yourname` | Hi, if I see someone trying to register as "Registration from 'Test1 <sip:Station%2026@11.11.11.11>' failed for '12.12.12.12' - ACL error (permit/deny) -> what do I have to add in sip.conf ? username=Station 2062 ? |
| 22:21.56 | [TK]D-Fender | aut, HD = G.722 = only between phones, and not full * support yet. HD = who seriously gives a crap, its a bloody PHONE! |
| 22:22.07 | travesty | mog: how do you feel about the digium clone cards? do they license from you guys or something? or is that top secret.... |
| 22:22.23 | russellb | you can transcode G.722 with codec_g722 in trunk :) |
| 22:22.39 | mog | travesty: clone cards are not licensed |
| 22:22.48 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:23.26 | Mercestes | travesty, everytime you buy a clone card, God kills a kitten |
| 22:23.34 | mog | heh something like that |
| 22:23.36 | russellb | nice |
| 22:23.58 | *** join/#asterisk nirz (n=nirz@79.179.88.51) |
| 22:23.58 | travesty | i never have :) but i have four digium clones i got for free *dances* |
| 22:24.03 | aut | [TK]D-Fender, with 320s, can i use G.729 all the way? from phone to phone and through most SIP providers (teliax, vitelity, etc)? |
| 22:24.17 | [TK]D-Fender | aut, yup |
| 22:24.30 | aut | [TK]D-Fender, even allowing ulaw for my ciscos, it doesnt work when i go from ulaw on the phones to g.729 at teliax |
| 22:24.44 | aut | the second cisco phone always hears messed up audio |
| 22:24.52 | aut | when i do a conf call |
| 22:25.21 | aut | im willing to change phones if it fixes the problem.. |
| 22:25.24 | flujan | [TK]D-Fender: is this aproach I use to add members to my queues bad? |
| 22:28.29 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:32.03 | [TK]D-Fender | flujan, Have them go through Chan_local and you can add your recording there |
| 22:34.13 | JT | [TK]D-Fender: surely G.722 is more than between phones if supported by the SIP server. |
| 22:35.13 | [TK]D-Fender | JT : * can't transcode to/from it. |
| 22:35.22 | [TK]D-Fender | JT : so.. umm.. FAILURE <- |
| 22:35.27 | *** join/#asterisk blinky42 (n=me@c-71-230-47-244.hsd1.pa.comcast.net) |
| 22:35.29 | JT | it can in trunk, also asterisk is not the only thing out there |
| 22:35.42 | [TK]D-Fender | JT : ... What channel are you in? ;) |
| 22:36.25 | JT | anyway, trunk can do it |
| 22:38.17 | [TK]D-Fender | JT : Which means we'll never see it in 1.4 standard FTP release, and we'll be waiting for another year or so for 1.6 (guesstimate) |
| 22:38.34 | JT | that's probably true |
| 22:38.50 | kombi | how do you make a contact list available for all polycom phones connected? |
| 22:40.10 | [TK]D-Fender | kombi, Generally you don't. Each phone should ahve control over its own list. |
| 22:40.43 | [TK]D-Fender | kombi, You can manually change the permissions on theprovisioning server (which you seemed to be avoiding) and lock it in that way |
| 22:40.53 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:41.15 | Nivex | mindCrime!! |
| 22:42.33 | kombi | Fender: got provisioning happening, one could of course symlink all directories. I just wonder how to do a company-wide list as well as an individual one |
| 22:42.51 | mindCrime | Nivex!!! |
| 22:43.17 | mindCrime | what's up? |
| 22:43.50 | Nivex | sitting at the alt LUG meeting :) |
| 22:43.56 | kombi | even got a hang of the "my config file overrides the factory one" concept, havn't conquered timezone and locale yet..;) |
| 22:44.13 | mindCrime | Nivex: cool |
| 22:44.19 | [TK]D-Fender | kombi, No such thing. They only support a single list. Whats the point of adding internal entries in there for you? |
| 22:44.43 | kombi | my future employer..;) |
| 22:45.03 | kombi | anyways, symlinks seem to be the measure of choice then |
| 22:45.04 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 22:46.30 | *** join/#asterisk adeel (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net) |
| 22:46.56 | [TK]D-Fender | kombi, except for presence the concept of a "corporate list" on a Polycom phone is largely counter-productive, and locking them out from changing the phone worse |
| 22:48.33 | adeel | i'm using asterisk 1.4.8 and am noticing a distorted ringing tone when placing outgoing calls...what could be the cause of this? latency and jitter i don't think are the problem because once the call is cleared, there is no more distortion |
| 22:50.14 | lirakis | extensions can be alpha characters right? |
| 22:50.31 | lirakis | .. like exten => abcd,1,Answer() |
| 22:53.30 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:55.49 | *** join/#asterisk rvhi (n=chatzill@66.135.230.96) |
| 22:56.15 | rvhi | hi, can I do "sip reload" through asterisk manager? |
| 22:56.35 | kombi | Fender: would have a hint on the timezone? I set it in dhcp and in phone1.cfg (the file derived from it), set override dhcp back and forth but nothing does it.. |
| 22:57.26 | [TK]D-Fender | kombi, show me what you did and I'll show you what you did wrong |
| 22:57.53 | kombi | coming up.. |
| 22:58.21 | outtolunc | action: command |
| 22:58.21 | outtolunc | command: sip reload |
| 23:00.48 | kombi | Fender: http://pastebin.ca/658930 |
| 23:01.24 | [TK]D-Fender | kombi, tcpIpApp.sntp.gmtOffset="1" <---- lol |
| 23:01.41 | kombi | that was quick.. 3600 better? |
| 23:01.58 | [TK]D-Fender | kombi, todays catch-phrase is "Unit Of Measure" ;) |
| 23:02.16 | kombi | please, not miliseconds..;) |
| 23:02.34 | [TK]D-Fender | kombi, No, only seconds :) |
| 23:03.41 | kombi | ..rebooting now |
| 23:06.18 | kombi | WORKING! What a difference a difference makes.. would you care to look at my language settings as well? <-> http://pastebin.ca/658934 |
| 23:07.29 | [TK]D-Fender | kombi, I highly recommend leaving that part alone from stock install. |
| 23:07.57 | kombi | how else would one change locales? |
| 23:08.13 | kombi | (putting it back to where it was) |
| 23:08.55 | [TK]D-Fender | kombi, let them set it on the phone itself |
| 23:09.11 | [TK]D-Fender | kombi, quick & easy |
| 23:09.23 | kombi | point taken. Tried that though and it refuses to change it.. |
| 23:09.25 | [TK]D-Fender | kombi, some things really are better off left alone |
| 23:10.01 | [TK]D-Fender | kombi, Maybe you locked down the provisioning folder too much and it didn't want to write back the local config. |
| 23:10.16 | [TK]D-Fender | kombi, Also, do you have the language folder set up right? |
| 23:11.28 | kombi | lang folder is there with the configs and everything is even world writable for time being. I read somewhere that one needs to copy the appropriate lang.xml to base dir? |
| 23:12.50 | *** join/#asterisk jeremi_g_laptop (n=tze@c213-100-17-48.swipnet.se) |
| 23:13.13 | jeremi_g_laptop | amigoes |
| 23:13.31 | jeremi_g_laptop | i have not come here to listen to the silence |
| 23:13.32 | kombi | Fender, you were right! |
| 23:13.37 | kombi | language changed.. |
| 23:13.47 | jeremi_g_laptop | good move kombi, talk talk... |
| 23:14.09 | jeremi_g_laptop | thou hath come to silence thee all |
| 23:14.26 | jeremi_g_laptop | okay |
| 23:15.41 | [TK]D-Fender | kombi, you're welcome |
| 23:16.10 | kombi | sleeping time in my locale, cheers Fender, over and out |
| 23:16.35 | jeremi_g_laptop | Hi [TK]D-Fender :) |
| 23:16.50 | jeremi_g_laptop | I watched that 20min documentary |
| 23:17.10 | jeremi_g_laptop | checked out to see if they teach that in stockholm |
| 23:17.14 | [TK]D-Fender | jeremi_g_laptop, Cool stuff huh? |
| 23:17.29 | [TK]D-Fender | jeremi_g_laptop, Indeed you have a MAJOR dojo there IIRC |
| 23:17.33 | jeremi_g_laptop | wud have been even more cooler had it involved some katas and stuff |
| 23:17.56 | [TK]D-Fender | jeremi_g_laptop, What are you talking about? Thats ALL kata... |
| 23:18.13 | jeremi_g_laptop | true, but i mean physical, non weapon |
| 23:18.31 | jeremi_g_laptop | amazing reflexes they got uhh |
| 23:18.52 | [TK]D-Fender | jeremi_g_laptop, ah, well there is a jui-jutsu component, but I have VERY little footage of that. |
| 23:19.01 | jeremi_g_laptop | yup i saw that |
| 23:19.08 | jeremi_g_laptop | the jui-jitsu component |
| 23:19.11 | jeremi_g_laptop | :) |
| 23:19.13 | jeremi_g_laptop | i love it |
| 23:19.45 | jeremi_g_laptop | i ll check out the dojo here may be tomorrow |
| 23:20.01 | jeremi_g_laptop | is it possible that i could compile asterisk on one linux station and do a make install after chrooting into another distro e.g. knoppix |
| 23:21.43 | *** join/#asterisk boch (n=fran@190.48.240.92) |
| 23:21.58 | boch | hi all, |
| 23:22.20 | boch | do you know if calls originated from manager exec the 'h' ext ? |
| 23:23.16 | [TK]D-Fender | boch, a channel is a channel is a channel. |
| 23:23.46 | jeremi_g_laptop | hehe |
| 23:24.30 | boch | so when any channel hangs it runs h, right |
| 23:24.48 | *** part/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 23:28.18 | [TK]D-Fender | boch, if your dialplan is set up right, yes |
| 23:28.33 | boch | thanks |
| 23:29.45 | jeremi_g_laptop | cute |
| 23:30.07 | jeremi_g_laptop | boch: or else exten=_.,1,Hangup |
| 23:31.02 | jeremi_g_laptop | place it properly |
| 23:31.20 | jeremi_g_laptop | i really need some sleep now |
| 23:31.25 | *** join/#asterisk fribuntu (n=fritz@xdsl-87-78-224-148.netcologne.de) |
| 23:31.33 | fribuntu | Hi all |
| 23:31.38 | jeremi_g_laptop | Hi fribuntu |
| 23:32.31 | jeremi_g_laptop | say it fribuntu while i am alive |
| 23:32.41 | boch | can i originate a call from the manager using a sip account with user/pass auth ? |
| 23:32.48 | fribuntu | Can someone tell me which config files are essential for a poor european with ISDN BRI and an AVM card and SIP? |
| 23:32.52 | *** join/#asterisk anthm (n=anthm@mb60736d0.tmodns.net) |
| 23:32.52 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:33.24 | *** join/#asterisk perf3kt (n=perf3kt@ppp-69-210-3-243.dsl.ipltin.ameritech.net) |
| 23:33.25 | fribuntu | jeremi_g_laptop: I am glad you are alive and awake :) |
| 23:33.27 | jeremi_g_laptop | boch:why not |
| 23:33.38 | jeremi_g_laptop | whats stopping you from originating a call |
| 23:34.16 | boch | jeremi_g_laptop, i mean without the acct user & pass defined in the sip.conf |
| 23:34.34 | fribuntu | I have 57 config files of which very few make sense to me. |
| 23:35.12 | fribuntu | boch: I think you will have trouble registering a SIP client without user/pw, unless you allow guest accounts, no? |
| 23:36.10 | boch | something like SIP/user:pass@host/ext ? |
| 23:36.46 | jeremi_g_laptop | boch:you gotta create an outgoing peer in sip.conf to call outwards at least |
| 23:37.14 | jeremi_g_laptop | ok i am dead |
| 23:37.21 | jeremi_g_laptop | bye bye |
| 23:37.54 | boch | no way without using sip.conf? |
| 23:38.15 | fribuntu | boch: I am not sure I understand what you are trying to do |
| 23:38.40 | fribuntu | boch: and I only have asterisk installed (and not yet up and running) since 10 min ago :) |
| 23:39.33 | boch | i want all my providers only in a db, no sip.conf |
| 23:39.41 | fribuntu | boch: But generally, to reach someone on their SIP you do not need a password. Just user@registrar will do. You only need your own password to register to asterisk or some other SIP gateway. |
| 23:39.55 | boch | to format the dial string with an agi from the db |
| 23:41.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 23:41.27 | boch | i want to auth on every call, instead of being registered for X secs trough sip.conf |
| 23:41.55 | mvanbaak | SIP/user:pass@host/ext |
| 23:41.59 | mvanbaak | that not working ? |
| 23:47.04 | *** join/#asterisk rhiliam (n=user@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 23:48.07 | rhiliam | Is there a way to increase the volume of incoming calls on a zap channel? I can barely hear them, but hey can here me fine. I have played with rxgain, to no avail. |
| 23:48.16 | luisjose | is it possible to deny call redirection? |
| 23:49.20 | [TK]D-Fender | rhiliam, what are you using? |
| 23:49.41 | rhiliam | * 1.2.23 and TDM card with 2 FXO and 2 FXS |
| 23:50.08 | [TK]D-Fender | boch, jsut set up your peer and don't do a register. |
| 23:50.44 | [TK]D-Fender | rhiliam, rxgain really is it. In order to take effect you have to restart * or reload chan_zap.so |
| 23:51.32 | rhiliam | I have reload chan_zap.so, but it makes no difference. I have add rxgain as high as 15, with no difference. Is rxgain channel/zap specific? |
| 23:52.08 | [TK]D-Fender | rhiliam, yes. pastebin your zapata.conf |
| 23:52.10 | Mercestes | <PROTECTED> |
| 23:52.27 | luisjose | yah |
| 23:52.34 | luisjose | i don't want the pones to do that |
| 23:52.36 | Mercestes | so, yes. |
| 23:52.59 | rhiliam | http://pastebin.ca/658966 |
| 23:53.03 | luisjose | Mercestes, how? |
| 23:53.35 | Mercestes | luisjose, which side of my car is my gas tank on? |
| 23:53.59 | luisjose | Mercestes, i don't have access to the phones |
| 23:54.07 | Mercestes | then your screwed |
| 23:55.23 | rhiliam | the main incoming line is zap/1 and the main phone is zap/3 |
| 23:56.25 | *** join/#asterisk Yourname` (n=chatzill@76-10-128-178.dsl.teksavvy.com) |
| 23:56.27 | Yourname` | ~sipnat |
| 23:56.28 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:57.58 | luisjose | though that |
| 23:58.01 | luisjose | but still |
| 23:58.23 | Mercestes | still what? |
| 23:58.35 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 23:58.40 | luisjose | sux :P |
| 23:59.28 | [TK]D-Fender | rhiliam, try to put the gain on ONLY your FXO channels |