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| 00:06.07 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 00:15.10 | raden_work | jeremy_g, and whats this suppose to accomplish ? |
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| 00:16.31 | drudge` | i think he wanted an excuse to say molest |
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| 00:21.55 | jeremy_g | raden_work:quality check |
| 00:22.21 | jeremy_g | distorted audio due to bad network,etc. would not get converted into text |
| 00:22.46 | jeremy_g | known text |
| 00:22.55 | raden_work | LOL |
| 00:22.55 | jeremy_g | i must sleep |
| 00:23.03 | raden_work | there are so many flaws in your theory |
| 00:23.10 | jeremy_g | raden_work::) really |
| 00:23.30 | raden_work | introducing a lot of variables |
| 00:23.36 | raden_work | find a more practical approach |
| 00:23.44 | raden_work | like maybe a network analyzer |
| 00:23.52 | jeremy_g | raden_work:howz that, please prove it - you sound like a proponent of MOS algorithms |
| 00:23.54 | raden_work | or make a program to simulate a SIP stream |
| 00:24.07 | jeremy_g | :D |
| 00:24.36 | jeremy_g | raden_work:where will the quality analyzer fail |
| 00:24.47 | jeremy_g | its me at the both ends |
| 00:25.00 | raden_work | ok for one you are relying on that fact that the text to speech will work 100% of the time on a clear audio stream doubtful |
| 00:25.32 | raden_work | speech to text 100% of the time |
| 00:25.38 | jeremy_g | i play the audio and i receive it and if the network has screwed it for some reason, wav to text will fail to produce the known text |
| 00:25.57 | raden_work | maybe the program just failed ? |
| 00:26.07 | raden_work | maybe the wav file did not play clearly |
| 00:26.22 | raden_work | impractical |
| 00:26.30 | jeremy_g | :D |
| 00:26.54 | jeremy_g | raden you are loosing my respect due to your light weight arguments |
| 00:27.03 | jeremy_g | i can train the damn sphinx engine to my wav file |
| 00:27.22 | *** join/#asterisk s14ck (~s14ck@190.142.77.144) |
| 00:27.44 | raden_work | then go figure it out |
| 00:27.50 | jeremy_g | and i can see at the moment from the previously recorded wav files that they are same in terms of listening experience as the one played back at the source |
| 00:28.32 | jeremy_g | even the file size is the same |
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| 00:28.57 | raden_work | jeremy_g, I dont see what your trying to accomplish |
| 00:29.07 | raden_work | great i can go text to speech to speech to text |
| 00:29.07 | jeremy_g | even thats a good parameter, if the file size changes, something must have happened and i set a tolerance level on file size |
| 00:29.43 | jeremy_g | raden_work:affect of voip network on the sound quality |
| 00:29.46 | jeremy_g | effect |
| 00:30.23 | raden_work | I can tell you the effects by watching the jitter and latency between two points |
| 00:30.34 | jeremy_g | under different conditions e.g. bandwidth,etc. |
| 00:30.42 | *** join/#asterisk BeeBuu (~chatzilla@58.252.73.164) |
| 00:31.03 | jeremy_g | i measure that already |
| 00:31.06 | raden_work | good luck with your project |
| 00:31.14 | BeeBuu | my E1 Card still does not detect busy.......anyone help please? |
| 00:32.03 | jeremy_g | BeeBuu:i would have i am tooo sleepy now |
| 00:32.24 | jeremy_g | bye mates |
| 00:32.31 | BeeBuu | jeremy_g: oh...no |
| 00:32.36 | BeeBuu | help me please~~~ |
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| 00:48.27 | BeeBuu | oh...no one can help me? |
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| 01:40.50 | pabelanger | BeeBuu: what protocol? |
| 01:41.00 | BeeBuu | ISDN |
| 01:41.38 | pabelanger | BeeBuu: Then enable ISDN debugs and look at the IE. Should be easy enough to figure out why |
| 01:41.54 | BeeBuu | At CLI? |
| 01:43.07 | pabelanger | or setup logger.conf to send it to a file |
| 01:45.17 | BeeBuu | ok,let me try |
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| 01:47.35 | pabelanger | BeeBuu: BTW: how is it not detecting busy? And inbound or outbound call? |
| 01:47.36 | BeeBuu | pabelanger: core set isdn debug? |
| 01:47.43 | BeeBuu | outboud |
| 01:47.51 | pabelanger | with the dial command? |
| 01:47.54 | BeeBuu | yes |
| 01:48.17 | pabelanger | err... DIALSTATUS variable? |
| 01:48.46 | pabelanger | BeeBuu: pri set debug |
| 01:48.49 | pabelanger | if I remember |
| 01:48.58 | BeeBuu | when i dial(dahid/g11/xxxx,,Ttr),it even go next goto(${DIALSTATUS}) |
| 01:49.22 | pabelanger | ~collectdebug |
| 01:49.22 | infobot | methinks collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 01:49.36 | pabelanger | BeeBuu: ^^ show us a complete call |
| 01:49.39 | pabelanger | pb the results |
| 01:49.55 | BeeBuu | i'm trying |
| 01:57.44 | BeeBuu | the fact is ,when i dial out ,callee hangup,but the caller( sip ) still ringing.... |
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| 02:03.38 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 02:05.24 | *** join/#asterisk jaybinks (~jaybinks@bh2s1.overthewire.com.au) |
| 02:05.42 | jaybinks | hey... i have cisco SPA504g handsets ... how can I configure them to let me use # in the number ??? |
| 02:05.58 | jaybinks | ( These stupid things use # as a dial terminator ... ) |
| 02:10.01 | drmessano | jaybinks, are you sure about that? |
| 02:10.20 | drmessano | # usually forces the number to be dialed when there isn't an acceptable pattern match |
| 02:10.50 | drmessano | Such as when using a dialplan of xxxxxxxxx, and you dial 123.. the phone awaits the remaining digits before passing |
| 02:11.33 | BeeBuu | ~pastbin |
| 02:12.22 | jaybinks | hmm ok.. ill take a look |
| 02:12.30 | jaybinks | and your right ... it is default dialplan ... |
| 02:12.31 | thehar | sounds like you need to edit the digit map |
| 02:12.48 | jaybinks | thing is I need to implement # in some feature codes |
| 02:12.56 | jaybinks | 99#12345789# |
| 02:13.01 | jaybinks | to turn on call forwarding etc... |
| 02:13.12 | jaybinks | and that just sends 99 to the phone :( |
| 02:13.22 | jaybinks | ill see what I Can get, by playing with the dialplan stuff |
| 02:16.44 | jaybinks | hmmm even if I have # specifically in the dialplan... it dosnt do as it should |
| 02:17.04 | jaybinks | read somewhere that # ALWAYS sends the call |
| 02:20.54 | BeeBuu | pabelanger: http://pastebin.com/MJ5tgrUh ,please help |
| 02:23.29 | pabelanger | BeeBuu: pri set debug on, make call again |
| 02:25.20 | ChannelZ | OT: Should a CPU fan be blowing into the heat sink or out from it? |
| 02:25.36 | pabelanger | away |
| 02:26.01 | pabelanger | to pull heat from the CPU |
| 02:26.06 | ChannelZ | that's what I thought, to be drawing cool air in through the heat sink |
| 02:26.11 | BeeBuu | pabelanger: it;s on |
| 02:26.18 | ChannelZ | just making sure my brain is working |
| 02:26.24 | pabelanger | BeeBuu: pb new call |
| 02:29.06 | BeeBuu | oh,no,No such command?! |
| 02:29.52 | BeeBuu | what's wrong? |
| 02:46.01 | BeeBuu | what's i missed install? |
| 02:53.20 | BeeBuu | pabelanger: http://pastebin.com/fg6We2je i pasted all i can. please help |
| 02:54.24 | pabelanger | BeeBuu: Need to see the complete call, not just the last 50 lines |
| 02:59.14 | BeeBuu | how can i set more in logger.conf? |
| 03:00.01 | BeeBuu | debug=debug? |
| 03:00.18 | BeeBuu | debug=>debug? |
| 03:06.28 | *** join/#asterisk bougyman (bougyman@pdpc/supporter/gold/bougyman) |
| 03:12.16 | drmessano | It isn't a matter of being more verbose, you need to paste more of the CLI |
| 03:15.10 | BeeBuu | pabellanger: http://pastebin.com/iLX6wGdw is it ok? |
| 03:15.51 | drmessano | [22:17] <jaybinks> read somewhere that # ALWAYS sends the call <-- Yes, # always sends the call... but you should never have to FORCE the DIAL.. PROPER DIALPLAN and the phone will execute the call properly |
| 03:23.01 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 03:25.43 | pabelanger | BeeBuu: According to your take, the number is not busy. In fact you can see Asterisk Ringing. Now, it does indicate inband call progress, so you could record the line and try to hear what is going on. |
| 03:26.57 | BeeBuu | what's your mean? |
| 03:27.16 | BeeBuu | mixmonitor without b? |
| 03:27.18 | pabelanger | Also, PB you DADHI .conf |
| 03:27.45 | BeeBuu | ok,wait,please |
| 03:28.34 | pabelanger | BeeBuu: Is priindication=inband set in chan_dahdi.conf? |
| 03:28.42 | pabelanger | if not set it, restart asterisk then retry your call |
| 03:28.59 | BeeBuu | in dahdi-channels.conf? |
| 03:29.48 | pabelanger | yes |
| 03:29.51 | pabelanger | no |
| 03:29.55 | pabelanger | chan_dandi.conf |
| 03:30.06 | BeeBuu | ok,let me try |
| 03:34.40 | BeeBuu | inbound or outband? |
| 03:35.48 | pabelanger | Do what you have been doing |
| 03:35.51 | pabelanger | outbound |
| 03:36.27 | BeeBuu | asterisk -rx "core restart now" ? |
| 03:37.49 | BeeBuu | i hangup. but it still show ringing~~~~ |
| 03:39.11 | BeeBuu | i will crazy on it~~~~ |
| 03:40.35 | pabelanger | stop and start asterisk to be safe, otherwise time to record the line and see what is happening. |
| 03:42.41 | BeeBuu | callee is a cell phone,when it hangup,it playback a voice message :the phone is busy.... |
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| 03:45.55 | *** part/#asterisk jaybinks (~jaybinks@bh2s1.overthewire.com.au) |
| 03:46.51 | BeeBuu | is it cause the problem? |
| 03:47.00 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 03:52.07 | titter | Odd question, but in the past I would use dadhi/g1/1234567890 in my queues.conf file to dial to outside numbers. I now have SIP trunking setup, and I am using SIP/pstn/1234567890 in queues.conf. In this example, what outbound dialing rules would queues.conf use? I don't see anything in the CLI that shows what context it used to dial out from. |
| 03:52.58 | titter | Is it just using the Dial() cmd? |
| 03:54.37 | [TK]D-Fender | titter: there is no such thing as context.The queue dials the device you list. The end |
| 03:56.11 | titter | [TK]D-Fender: Thanks, wasn't thinking clearly, I understand now. |
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| 04:41.10 | Justman | hi guys |
| 04:41.23 | Justman | anybody use announceoverride in the app_queue ? |
| 04:42.11 | Justman | I trying to use Queue(queue_name,t,,/var/spool/..../filename); but nothing changes, thanks for any help |
| 04:45.47 | *** join/#asterisk ranjan (~ranjan@122.172.123.124) |
| 04:46.28 | ranjan | Hi all |
| 04:46.50 | ranjan | can i used the zaptel to make my V92 modem work with asterisk? |
| 04:48.38 | [TK]D-Fender | ~savemoney |
| 04:48.39 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 04:48.45 | [TK]D-Fender | ranjan: NO. |
| 04:49.21 | carrar | hahah |
| 04:49.27 | carrar | almost missed that |
| 04:49.41 | carrar | How about a hays smart modem 300! |
| 04:49.53 | carrar | turn that sucker into a ATA |
| 04:51.34 | ranjan | [TK]D-Fender: i have read that an intel V92 modem can be used. |
| 04:51.58 | ranjan | [TK]D-Fender: and they are saying that its just a X100P clone. |
| 04:52.15 | carrar | should just work then |
| 04:52.39 | ranjan | carrar: [TK]D-Fender and my modem is Conexant Systems, Inc. SoftV92 SpeakerPhone SoftRing Modem with SmartSP (rev 01) |
| 04:52.40 | carrar | It's that easy |
| 04:52.55 | carrar | is it working? |
| 04:53.34 | ranjan | carrar: no i want to test it. For that can you give me some links on how to configure zaptel? |
| 04:53.58 | carrar | You not able to reach google? |
| 04:54.05 | ranjan | carrar: :) |
| 04:54.07 | carrar | look for X100P configs |
| 04:54.13 | ranjan | carrar: ok :) |
| 04:54.20 | ranjan | carrar: let me see |
| 04:54.40 | carrar | http://tinyurl.com/2wgukv6 |
| 04:55.04 | ranjan | carrar: wow what was that !!! |
| 04:55.23 | carrar | THE POWER OF THE INTERNETS |
| 04:55.35 | ranjan | carrar: how did you do that :) please let me know :) |
| 04:55.58 | carrar | You will need to travel to Japan and study the ways of Ninjahood |
| 04:56.12 | ranjan | carrar: :P |
| 04:58.27 | ranjan | carrar: thank you man :) i got it ;) this is awsome :) |
| 04:58.39 | ranjan | carrar: now let me try with my modem :) |
| 04:58.57 | drmessano | That Conexant isn't gonna work |
| 04:59.06 | carrar | shhh |
| 04:59.11 | ranjan | drmessano: oh :( have you tried that? |
| 04:59.13 | *** join/#asterisk cjk (~ldidelot@vodsl-11065.vo.lu) |
| 04:59.37 | drmessano | Wait, you mean I have a box of Conexant Asterisk FXO cards and didnt even know it?? |
| 04:59.43 | drmessano | O.o |
| 04:59.54 | carrar | determination is the key to insanity |
| 04:59.58 | drmessano | ZOMG |
| 05:00.19 | carrar | ZOMG YES!!!!!!!!! |
| 05:00.32 | carrar | You are sitting on a GOLD MINE! |
| 05:01.53 | drmessano | ranjan, complete waste of time.. Get a real card. There is a modem chipset, an Agere, I believe, that works in Asterisk that is the basis for the "Clone X100P" card you see millions of on ebay.. you likely would never find one in a junkbox you bought as a modem years ago and now "ZOMG I CAN HAS AKERISK" |
| 05:02.30 | drmessano | It's certainly not going to be that Conexant |
| 05:02.36 | ranjan | drmessano: ok thank you for your infromation man :) |
| 05:03.02 | carrar | he has more information |
| 05:03.10 | ranjan | drmessano: by the way does an agere chipset based modem work? |
| 05:05.07 | drmessano | ranjan, no.. it's not that easy.. it's one very specific chipset, and again, you are VERY LIKELY to NEVER EVER find one out there as a 99 cent junkbox modem |
| 05:05.10 | drmessano | Forget it |
| 05:05.13 | drmessano | Buy a real card |
| 05:05.44 | ranjan | drmessano: i,will take your advice instead of wasting my time :) |
| 05:09.24 | carrar | it's not complete waste of time |
| 05:09.36 | carrar | You will learn configs |
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| 05:16.47 | shamelessn00b | lol'd |
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| 06:25.58 | SeTTleR | hi |
| 06:31.13 | ChannelZ | ahoy |
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| 07:08.47 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 07:08.54 | Chodorenko | Hi All |
| 07:09.34 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:09.37 | schmidts | good morning |
| 07:15.10 | Chodorenko | i try use srtp on asterisk , read /doc/tex/secure-calls.tex, create sip.conf which encryption=yes in peer configuration, create dialplan which exten => 123,n,Set(CHANNEL(secure_bridge_signaling)=1) , configure two gxp-2000 for srtp call . if i call from one to two phone i have error -- Executing [8888@default:2] Set("SIP/test2-00000024", "CHANNEL(secure_signaling)=1") in new stack |
| 07:15.11 | Chodorenko | [Oct 14 10:14:27] WARNING[18811]: func_channel.c:491 func_channel_write_real: Unknown or unavailable item requested: 'secure_signaling' |
| 07:15.11 | Chodorenko | <PROTECTED> |
| 07:15.56 | Chodorenko | or [Oct 14 10:14:27] WARNING[18811]: func_channel.c:491 func_channel_write_real: Unknown or unavailable item requested: 'secure_brige_signaling' if use bidge variables |
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| 07:23.07 | ChannelZ | And you really are using 1.8? |
| 07:24.27 | Chodorenko | Yes |
| 07:25.25 | Chodorenko | call on 600 is encrypted good |
| 07:27.54 | *** join/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 07:28.00 | schmidts | you miss a "d" on bridge ;) |
| 07:28.02 | ChannelZ | well check for typeos then |
| 07:28.22 | ChannelZ | because what you showed didn't really line up (your dialplan example was extension 123 but your failure was 8888, etc) |
| 07:28.31 | schmidts | and there is no item called secure_signaling only secure_bridge_signaling or secure_bridge_media |
| 07:28.50 | hurdman | hi, what soft do you use for voice recognition with asteris with french from france language ? |
| 07:29.12 | ChannelZ | illegal aliens |
| 07:29.23 | schmidts | chinese coworkers? |
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| 07:43.03 | DND | guys wht's the command to show loaded modules? |
| 07:43.24 | schmidts | module show |
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| 08:56.19 | mahiti-irc | hi |
| 08:57.00 | mahiti-irc | i have a Digium TE122P PRI card, Centos 5.3, asterisk 1.4.26, and latest dahdi all setup |
| 08:57.09 | mahiti-irc | but i have no idea |
| 08:57.22 | mahiti-irc | how to configure the echo cancellers? |
| 08:57.43 | mahiti-irc | pci:0000:03:02.0 wcte12xp+ d161:8001 Wildcard TE122 |
| 08:57.53 | mahiti-irc | is the result of dahdi_hardware |
| 08:58.06 | mahiti-irc | can anyone help me on this? |
| 08:59.19 | mahiti-irc | no one? |
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| 09:01.19 | mahiti-irc | how to configure the echo cancellers in dahdi? |
| 09:01.19 | mahiti-irc | can anyone help me on this? |
| 09:01.47 | SeTTleR | hi schmidts, i saw your post on the issue at https://issues.asterisk.org/view.php?id=18134 |
| 09:01.59 | SeTTleR | i think, your comment is correct. thats exactly the line. one must only check ERRNO and print a correct error message |
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| 09:02.48 | mahiti-irc | guys, anyone knows how to configure the echo cancellers in dahdi? |
| 09:03.28 | SeTTleR | in 1.6.13 a system() call was used, where the comment from davidw applies, but this was obviously changed |
| 09:05.47 | SeTTleR | s/1.6.13/1.6.2.13/ |
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| 09:07.36 | schmidts | settler hello ;) |
| 09:08.13 | schmidts | the only problem i see a execl only returns 0 on success or -1 on failure but ERRNO should normaly exists |
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| 09:22.17 | schmidts | settler that would be a little tricky ;) with execl its used to call a /bin/sh start with the parameter to start whatever you want |
| 09:23.27 | SeTTleR | yes i see that. maybe a stat of /bin/sh before calling execl can solve the problem |
| 09:27.14 | mahiti-irc | guys, anyone knows how to configure the echo cancellers in dahdi?? |
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| 09:33.27 | schmidts | we can use a fopen to /bin/sh but i dont know if this would work |
| 09:35.51 | WIMPy | I'm not exactely convinced if its Asterisks task to check for basic system functionalety. |
| 09:36.52 | WIMPy | But if you want to check for a working shell, I'd do it upon registration of app_system, not every time it's used. |
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| 09:37.37 | exoca | hi |
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| 09:38.13 | b_d | hi all |
| 09:38.23 | exoca | i've got a problem witch is common but i don't find the goods words to discribe it |
| 09:38.36 | exoca | i use asterisk 1.6.x |
| 09:38.43 | exoca | when i play a sound |
| 09:38.48 | exoca | before each sound |
| 09:38.55 | b_d | does anyone have any experience with the audiocodes mp-118 fxs? |
| 09:38.58 | exoca | i've a "scritch" |
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| 09:39.21 | exoca | if someone can help me, thanks |
| 09:39.51 | SeTTleR | hi WIMPy, maybe this is a better way. yes, i know this is a minor thing and maybe not asterisks thing to do, but i've wasted a lot of time to figure this out (yes, i know STUPID :D) and i want to prevent other people from having the same problem, when this could be checked by asterisk.. |
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| 09:41.17 | Chodorenko | Developers Give Normal Manual for setup SRTP on asterisk 1.8 Thanks |
| 09:46.51 | schmidts | settler the question is how does this happen, cause i think /bin/sh should be checked with configure |
| 09:47.33 | SeTTleR | huh? i started asterisk in a chroot environment and forgot to put the sh binary in that env |
| 09:48.34 | SeTTleR | i don't know, it's really a corner case. i just wanted to prevent other users from making the same mistake :-) |
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| 09:52.09 | schmidts | i know but you didnt have compiled it with chroot right? |
| 09:54.13 | SeTTleR | mmh i don't know, let me check back. never heard about that. can you give me a link? |
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| 09:55.17 | DND | guys anyone tried free fax for asterisk? |
| 09:55.22 | DND | im having a problem receiving fax |
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| 10:04.26 | mahiti-irc | thanks for you help guys |
| 10:07.45 | exoca | nobody has this problem before? playing a sound "beep" and listing a little sound "scritch" before "beep"? |
| 10:08.08 | exoca | same thing with others sound |
| 10:08.26 | exoca | giving a bad render ? |
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| 10:39.39 | Jasnejac | exoca: I have the same problem but only with phones attached to "slow" connections and apparently only with old Cisco 7960 phones |
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| 10:45.17 | elliot98 | takes his place |
| 10:45.28 | elliot98 | how does SRTP play with FAX T.38? |
| 10:45.41 | elliot98 | is it possible to encrypt fax? |
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| 10:51.12 | schmidts | settler i will have a patch for you soon |
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| 10:53.52 | schmidts | or not, hmm do you think it would be a good idea to stop asterisk if /bin/sh could not be found? |
| 11:00.46 | Tim_Toady | it would break windows combatibility :P |
| 11:05.44 | schmidts | possible ;) |
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| 11:14.58 | schmidts | settler ok forget what i have said, its not that easy at all |
| 11:15.11 | wdoekes2 | schmidts: you could warn upon loading of app_system |
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| 11:15.40 | schmidts | yes but then the module will be showed as it was loaded and you cant unload or reload it again |
| 11:16.00 | schmidts | if i return -1 to the module loader asterisk will stop |
| 11:16.11 | wdoekes2 | yes.. you should warn only |
| 11:16.15 | schmidts | not crashing just an exit condition from the main |
| 11:16.15 | wdoekes2 | not abort |
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| 11:18.54 | wdoekes2 | does execl(2) return if argv[0] is not found? if so, you could _exit(63) or some other unlikely number and check for that from the main process, and then do a check to see whether /bin/sh is executable.. but it feels like a hack too large for the feature it provides |
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| 11:22.53 | schmidts | there is something in the errno variable but i am with wimpy that it should be checked on startup if /bin/sh is found or not |
| 11:23.33 | SeTTleR | schmidts, sry, was at lunch. The behaviour as it is now is totally sufficient. i only like to have an error message in the log that says "/bin/sh not found" instead of only SYSTEMSTATUS being APPERROR |
| 11:25.18 | SeTTleR | i don't think, that's that so serious that you have to stop asterisk or something. it just took me a while to find out that /bin/sh was missing when using the System() app and i want to prevent other users from wasting as much time as i did :-) |
| 11:27.10 | schmidts | was on the wrong path for this ;) |
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| 11:30.35 | schmidts | patch is in the issue if you could reproduce this it would be nice if you can try this |
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| 11:32.06 | SeTTleR | yes, i took a look at that and that should totally fix the issue, thx :-) |
| 11:33.05 | SeTTleR | testing is a little bit complicated here. i will see, if i can do some tests |
| 11:33.39 | schmidts | np |
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| 12:14.42 | E-bola | Hello |
| 12:16.50 | b_d | anyone here using audiocodes fxs atas? |
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| 12:44.57 | bip | hello i have this problem here: http://pastebin.com/zExs9DBj |
| 12:45.47 | bip | dmesg shows this card here: http://pastebin.com/ZHtxq2k4 |
| 12:45.59 | bip | any hint will be appreciated |
| 12:46.39 | bip | i m tryn to figure if i can use asterisk with the cards we already have or not ...need to know before selecting cards for purchase |
| 12:47.27 | [TK]D-Fender | bipFunny you don't consider it important to tell us precisely WHICH card you have since you're wondering if it should work. |
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| 12:48.17 | bip | well [TK]D-Fender i m not telling because i m not sure what card it is exactly ... |
| 12:48.30 | bip | i might go and have a look at it |
| 12:48.32 | WIMPy | bip: You can't use I$L with Asterisk. But you can use misdn. |
| 12:48.51 | WIMPy | I4L |
| 12:49.08 | bip | is misdn a standard kernel module ? |
| 12:49.22 | bip | or do I need to install something ? |
| 12:49.25 | WIMPy | Yes |
| 12:49.29 | schmidts | bip: what does your lspci says |
| 12:49.37 | bip | i have a plain debian lenny box btw |
| 12:49.48 | bip | one second schmidts thank you :) |
| 12:49.53 | WIMPy | misdn2 has been in the standard kernel for some time. |
| 12:50.17 | bip | sorry I m totally ignorant as far as telephony goes |
| 12:50.31 | bip | i m a jack all tarde sysadmin |
| 12:50.48 | bip | I know a little about a lot ;-) |
| 12:50.55 | bip | s/tarde/trade |
| 12:51.58 | schmidts | is this not the job description for sysadmins? |
| 12:52.12 | *** join/#asterisk exoca (~exoca@87.240.77.170) |
| 12:53.42 | bip | this is my lspci output schmidts http://pastebin.com/KTa1j1Yn |
| 12:54.07 | bip | depends schmidts my romomate thinks he is a security expert ;-) |
| 12:54.24 | bip | s/romomate/roommate |
| 12:54.39 | schmidts | bip lol :D |
| 12:55.13 | bip | nothing worth laughing here schmidts ;-) |
| 12:55.24 | schmidts | i cant see a card which i would call a isdn interface card, maybe anyone else can identify one ;) |
| 12:55.56 | bip | yeah i ca tell ya ... |
| 12:55.59 | bip | one sec |
| 12:55.59 | WIMPy | bip: From a picture I found, it looks like it's a bog standard HFC-S card. So it will work with misdn2, misdn or patched dahdi. |
| 12:56.31 | bip | so i just modprobe one of those and zaptel once more ? |
| 12:56.53 | bip | Digi International Datafire Micro V (Europe) (rev 02) |
| 12:57.19 | bip | fenrir:/etc# modprobe misdn |
| 12:57.19 | bip | FATAL: Module misdn not found. |
| 12:57.20 | bip | fenrir:/etc# |
| 12:57.22 | WIMPy | You could also use a patched zaptel. But zaptel has been replaced by dahdi some time ago. |
| 12:57.31 | bip | lemme apt-file search that ... |
| 12:57.41 | bip | i m a old timer ... |
| 12:57.52 | bip | and i m following a zaptel based howto now ... |
| 12:57.55 | WIMPy | The debian should fit :-) |
| 12:58.03 | bip | lemme give ya the link |
| 12:58.04 | WIMPy | Then.. |
| 12:58.22 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 12:58.24 | bip | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-4-SECT-5.html |
| 12:58.32 | WIMPy | But it will most probably give you some extra headache. |
| 12:58.43 | elliot98 | how does SRTP work with faxing, esp. T.38? |
| 12:58.47 | bip | headache is my second name WIMPy ;-) |
| 12:59.09 | WIMPy | bip: That is for an FXO card. |
| 12:59.19 | tzafrir | bip: don't follow an old HOWTO... |
| 12:59.38 | WIMPy | He's on debian :-) |
| 13:01.20 | bip | is a fxo card a card that plugs into a telco line ? |
| 13:01.30 | bip | sorry if I sound dumb ... |
| 13:01.39 | bip | I might actually be dumb ;-) |
| 13:02.11 | WIMPy | FXo/FXS are POTS, i.e. analogue. You want ISDN, i.e. digital. |
| 13:02.28 | WIMPy | Or you actually have... |
| 13:02.30 | bip | ah ok |
| 13:02.44 | bip | no no that was plugged into isdn |
| 13:02.59 | bip | thanks from steering me in a different direction WIMPy |
| 13:03.33 | bip | what we have here is sip phones plugged into lan dialing trough isdn |
| 13:03.55 | tzafrir | bip: there are backports of dahdi (dahdi-tools, dahdi-source, etc.) |
| 13:04.17 | bip | ok tzafrir i will check backports repo ... |
| 13:04.20 | *** join/#asterisk af_ (~getsmart@78.134.21.149) |
| 13:04.20 | tzafrir | But I'm a bit confused. Which card is the HFC card? |
| 13:04.39 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 13:04.45 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 13:04.51 | WIMPy | That Micro V thing. |
| 13:05.06 | bip | this one tzafrir : 00:0a.0 Network controller: Digi International Datafire Micro V (Europe) (rev 02) |
| 13:05.27 | bip | i will google that and linux driver and see if google is helpful |
| 13:05.45 | bip | expect me back here often today ... |
| 13:05.48 | nicola_pav | hello. I wrote a script that scans the active IP addresses in my network. AS a result I will have a txt file including ip addresses and MAC addresses |
| 13:05.59 | WIMPy | Don't bother to search for the card itself. Just look out for HFC-S. |
| 13:06.06 | nicola_pav | I want also to have the vendor of the IP phones connected |
| 13:06.10 | nicola_pav | is that possible? |
| 13:06.19 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:06.38 | nicola_pav | can vendors be differentiated against their mac addresses? |
| 13:07.09 | SeTTleR | nicola_pav, yes, they can.. every vendor has its own prefix of mac addresses |
| 13:07.16 | bip | HFC-S AND "linux driver" WIMPy ? |
| 13:07.27 | nicola_pav | is it published anywhere? |
| 13:07.31 | SeTTleR | there are lists available in the internet. just search in google |
| 13:07.35 | nicola_pav | or do i have to figure it on my own? |
| 13:07.40 | nicola_pav | ok, thanks |
| 13:07.50 | WIMPy | nicola_pav: There are lists out threr, but they won't neccessarily give you the detail you might hope for, |
| 13:08.40 | WIMPy | bip: I already gave you the options: misdn2, misdn1 or patched dahdi (or zaptel). I4L, what you have working now won't do with Asterisk. |
| 13:08.41 | [TK]D-Fender | HFC-S .... High-Fructose Corn Syrup. Doesn't sound healthy..... |
| 13:09.29 | bip | ok i will see if those are avilabel with backports WIMPy , apt-file serach knows nothing about em |
| 13:09.40 | nicola_pav | WIMPy: all i need is the prefix of mac address |
| 13:10.42 | SeTTleR | WIMPy, why not use chan_capi? as far as i've seen, the card supports capi |
| 13:10.57 | WIMPy | bip: misdn1 is seperate, misdn2 is part of the standard kernel. |
| 13:11.12 | SeTTleR | nicola_pav, yes, go for google, you will find lists of mac prefixes pointing you to the vendor |
| 13:11.14 | WIMPy | Right. CAPI would also be an option. |
| 13:11.18 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:11.35 | bip | well if i do modprobe misdn2 or misdn a get a module not found output |
| 13:12.04 | bip | modprobe misdn2 |
| 13:12.04 | bip | FATAL: Module misdn2 not found. |
| 13:12.05 | bip | fenrir:/etc# |
| 13:12.19 | WIMPy | bip: How many years is your kernel old? |
| 13:12.33 | WIMPy | They're both called misdn, unfortunaletely. |
| 13:12.38 | bip | it s brand new |
| 13:12.42 | bip | one sec |
| 13:12.59 | bip | fenrir:/etc# uname -a |
| 13:12.59 | bip | Linux fenrir 2.6.26-2-686 #1 SMP Thu Sep 16 19:35:51 UTC 2010 i686 GNU/Linux |
| 13:12.59 | bip | fenrir:/etc# |
| 13:13.28 | bip | modprobe misdn |
| 13:13.40 | WIMPy | I call that quite old, but I think that already should have misdn2. Maybe it's just not enabled. |
| 13:13.43 | bip | FATAL: Module misdn not found. |
| 13:13.59 | bip | ok ... |
| 13:18.28 | bip | anybody run asterisk with debian here ? |
| 13:18.44 | bip | there are some debian specific isssues I think ;-) |
| 13:19.38 | WIMPy | Maybe there's a repossitory somewhere with less outdated stuff? :-) |
| 13:19.42 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 13:20.17 | bip | yest the backports I m looking there while whining in #debian-it WIMPy ;-) |
| 13:23.26 | *** join/#asterisk ee99ee (~ee99ee@187.114.203.125) |
| 13:24.02 | ee99ee | Hi, I have a SIP phone configured here with an extension. This phone is on a poor connection and is randomly disconnecting and causing hangups... I think it is becasue it's higher latency than our other extensions (~200ms) |
| 13:24.11 | ee99ee | I just get random hangups mid-conversation on the phone |
| 13:24.20 | ee99ee | Is there a way to adjust timeout settings or something for a particular extension? |
| 13:25.49 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:27.08 | bip | maybe i should remove zaptel and use dahdi |
| 13:27.50 | SeTTleR | ee99ee, what phone? grandstream? :-) |
| 13:28.00 | leifmadsen | ee99ee: qualify=no |
| 13:28.09 | ee99ee | SeTTleR: Polycom IP430 |
| 13:28.24 | WIMPy | bip: Search for zaphfc then or for bristuff. |
| 13:28.25 | ee99ee | leifmadsen: thanks, will try that now |
| 13:28.42 | leifmadsen | ee99ee: also set the re-registration to a lower value, but either set qualify=no or set it to a number higher than 2000 |
| 13:28.50 | bip | ok lemmme have a look WIMPy |
| 13:29.08 | SeTTleR | aha, i don't know that. i only know of such problems with grandstream phones... |
| 13:29.31 | WIMPy | A failed qualify doesn't terminate an active connection. |
| 13:29.37 | Katty | morning |
| 13:30.07 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 13:30.25 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-171-237.dsl.stlsmo.sbcglobal.net) |
| 13:31.06 | ee99ee | WIMPy: then what does? |
| 13:31.10 | *** join/#asterisk deonv (~adium@41.219.94.148) |
| 13:31.24 | ee99ee | I adjusted qualify=no, but it still disconnects... I didn't re-register the extension after the chnage, so I'm rebooting the phone now |
| 13:31.55 | WIMPy | ee99ee: Possibly rtptimeout, but that's rather high by default. So maybe it's a network problem. |
| 13:32.40 | ee99ee | WIMPy: do you know what the trptimeout is? I'm at our brazil office, so the latency back to our Astrisk server in Atlanta is about 200-230ms |
| 13:32.44 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:33.14 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 13:33.34 | WIMPy | ee99ee: Can't remember the default, but it must be at least 20s, as I wanted to decrease it. |
| 13:33.45 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 13:33.53 | ee99ee | WIMPy: according to voip-info.org, it's 0 (no timeout) by default (http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf) |
| 13:34.13 | *** join/#asterisk vader- (vader@c-71-225-201-226.hsd1.nj.comcast.net) |
| 13:34.14 | WIMPy | That's quite long then :-) |
| 13:34.22 | ee99ee | hehe |
| 13:35.26 | ee99ee | still disconnects with qualify=no :-/ |
| 13:36.18 | ee99ee | ah ha: chan_sip.c: Maximum retries exceeded on transmission |
| 13:38.00 | ee99ee | is there a way to just drop the SIP packet if retransmit fails? |
| 13:39.34 | *** join/#asterisk Woody4286 (~Woody2143@machine76.Level3.com) |
| 13:40.05 | leifmadsen | Katty: yo! |
| 13:40.17 | WIMPy | ee99ee: i think that actually really meand a SIP packet, not an RTP packet, but I'm sure someone else will know better than me. |
| 13:41.11 | ee99ee | hmm |
| 13:41.47 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:41.47 | kaldemar | ee99ee: sip is the signaling of the call, so that would be bad and a no. |
| 13:42.11 | ee99ee | there is some slight packet loss here today -- about 4% -- perhaps that is related |
| 13:42.24 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 13:42.30 | *** join/#asterisk firstpoint (~firstpoin@19-163.5-85.cust.bluewin.ch) |
| 13:42.35 | firstpoint | Hi everybody |
| 13:42.43 | firstpoint | Hope you're all doing well |
| 13:43.08 | firstpoint | I've got a huge problem with installing asterisk-addons |
| 13:43.10 | firstpoint | Anyone available ? |
| 13:43.19 | FreezeS | hi |
| 13:43.23 | WIMPy | ~ask |
| 13:43.23 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:43.23 | FreezeS | what exactly is the problem ? |
| 13:43.55 | firstpoint | I'm currently running Asterisk 1.4.1 |
| 13:43.57 | firstpoint | on Debian |
| 13:44.12 | firstpoint | I did an svn checkout of asterisk-addons 1.4 |
| 13:44.18 | firstpoint | then ./configure |
| 13:44.42 | firstpoint | and when I run make |
| 13:44.57 | firstpoint | menuselect/menuselect --check-deps menuselect.makeopts |
| 13:44.57 | firstpoint | Generating embedded module rules ... |
| 13:45.17 | firstpoint | -/bin/grep: /usr/include/asterisk/buildopts.h: No such file or directory |
| 13:45.33 | firstpoint | about 10 times the same error |
| 13:45.43 | firstpoint | can't find anything very helpful on google or on asterisk forums |
| 13:45.53 | firstpoint | i also have, after: chan_ooh323.h:19:22: error: asterisk.h: No such file or directory |
| 13:46.09 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 13:46.13 | firstpoint | many ".h" errors, and also "ooh323cDriver.h:35: warning: âstruct ast_codec_prefâ declared inside parameter list" |
| 13:46.35 | FreezeS | what debian version are you running ? |
| 13:46.40 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 13:46.57 | [TK]D-Fender | firstpoint: 1.4.1? Pardon? ANCIENT |
| 13:47.53 | firstpoint | Debian 5.0.6 |
| 13:48.34 | bip | http://pastebin.com/1dAwVEeR :) |
| 13:48.37 | firstpoint | yes 1.4.1, that's actually the version installed by apt-get... I've been running fine until now so I don't really need to upgrade |
| 13:50.01 | firstpoint | I'm just willing to install mySQL support for RealTime. |
| 13:50.26 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-yimpuvpdfwfmcuhm) |
| 13:51.01 | kaldemar | firstpoint: that's strange, debian lenny has a version of 1.4.21.2 in its repositories. |
| 13:52.26 | kaldemar | firstpoint: there is the asterisk-mysql package too if that's what you're after. |
| 13:54.19 | WIMPy | bip: What did you use? What does dmesg say? |
| 13:55.01 | bip | one sec lemme see |
| 13:55.09 | bip | i did modprobe zaptel |
| 13:55.15 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:55.56 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 13:55.58 | bip | this is dmesg output about zaptel WIMPy : |
| 13:55.59 | anonymouz666 | coppice: how hard is to detect inband DTMF on compressed codecs? Like GSM |
| 13:56.06 | bip | [14082.651751] Zapata Telephony Interface Registered on major 196 |
| 13:56.06 | bip | [14082.651751] Zaptel Version: 1.4.11 |
| 13:56.07 | bip | [14082.651751] Zaptel Echo Canceller: MG2 |
| 13:56.27 | bip | i thjink I m moving foward but ... |
| 13:56.39 | bip | http://pastebin.com/S0EVaP7w |
| 13:58.00 | bip | why do I get this when i run "/sbin/ztcfg -vv " : ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 13:58.03 | bip | ?? |
| 13:58.20 | coppice | anonymouz666: it sort of works with GSM and G.729 when the DTMF is clean, but it can be rather intolerant of poor quality DTMF from the source |
| 13:59.53 | firstpoint | kaldemar, gonna tryin installing asterisk-mysql |
| 13:59.57 | firstpoint | keep you updated thanks |
| 14:00.19 | anonymouz666 | coppice: it works but not in a robust way with the current Asterisk code |
| 14:00.26 | anonymouz666 | :/ |
| 14:01.21 | firstpoint | kerframil -> did apt-get install mysql seems better now !! |
| 14:01.27 | firstpoint | [Oct 14 16:00:59] ERROR[29241]: res_config_mysql.c:629 mysql_reconnect: MySQL RealTime: Failed to connect database server on (err 2002). Check debug for more info. |
| 14:01.36 | firstpoint | before it said it couldn't find the engine |
| 14:01.36 | anonymouz666 | I am about to move from chan_dahdi to another channel driver to make this work better |
| 14:03.08 | coppice | well, the DTMF detector in * was a bit broken the last time I tested it. with a good DTMF detector, and DTMF tones bang on spec it seems to work reliably through G.729 and GSM 06.10, but in the real world a lot of phones produce poor DTMF tones, and the reduced margins caused by GSM and G.729 makes detection rather hit and miss |
| 14:03.29 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-217-046.pools.arcor-ip.net) |
| 14:04.41 | anonymouz666 | when you tested? |
| 14:04.58 | leifmadsen | coppice: how do you test that? There is a DTMF test in the testsuite now, but it might be worth it to review and verify the test is done reliably. |
| 14:05.15 | leifmadsen | s/what you tested?/what did you test that last?/ |
| 14:05.25 | leifmadsen | gah, copy fail |
| 14:05.36 | leifmadsen | s/what/when/ |
| 14:05.39 | leifmadsen | heh |
| 14:05.52 | coppice | digium were given a proper test suite with the original DTMF decoder, and threw it away - same with most test code they have been given |
| 14:06.13 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 14:06.23 | leifmadsen | sounds rather dramatic |
| 14:06.54 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 14:07.34 | kaldemar | bip: in addition to zaptel, you need to load a module that supports your card. |
| 14:07.39 | *** join/#asterisk SuPrSluG (~SuPrSluG@host-64-179-106-158.ind.choiceone.net) |
| 14:08.28 | ee99ee | I think the problem is that port 5060 is being forwarded to another host by the NAT here... is there a way to have Asterisk listn on two ports (5060 and 5061, for example)? |
| 14:09.03 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 14:09.42 | bip | kaldemare now i have tried a differnt approach i did : "genzaptelconf -s -d -c it |
| 14:10.16 | bip | kaldemare now i have tried a differnt approach i did : "genzaptelconf -s -d -c it" and now "/sbin/ztcfg -vv " gives no error |
| 14:10.30 | leifmadsen | coppice: how many years ago was the test suite provided? Is there a place to review that code? Just because it may not have been used back then doesn't mean it can't be reviewed and reimplemented now |
| 14:11.11 | bip | http://pastebin.com/tfR5raqN |
| 14:11.21 | coppice | if anyone actually cared, your question wouldn't even arise |
| 14:11.22 | bip | will this work ? |
| 14:11.39 | WIMPy | bip: Standard zaptel won't work. You need either need a bristuffed version or the zaphfc module. |
| 14:11.40 | leifmadsen | coppice: I care and I knew nothing about that test suite for DTMF detection |
| 14:11.59 | leifmadsen | coppice: that's the second time in two days I've seen you mention it, and I'd like to see if there is something that could be done |
| 14:12.04 | bip | ok WIMPy i will keep reading docs |
| 14:12.25 | coppice | its in spandsp. along with a DTMF decoder that works properly |
| 14:12.33 | bip | i have seen some bristuff reference in the docs i have read so far ... |
| 14:12.44 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:12.53 | bip | zaptel-source was telling something about it ... |
| 14:13.59 | coppice | the talk off test audio is copyright, so its not provided with spandsp. perhaps digium still have that |
| 14:14.18 | tzafrir | bip: sorry for jumping in now: what do you have installed right now? |
| 14:14.31 | tzafrir | Do yourself a favour: don't install bristuff |
| 14:14.50 | bip | well the README:debian says: |
| 14:14.52 | bip | This version has the the bristuff patch and kernel modules. The bristuffed |
| 14:14.52 | bip | modules require a bristuffed Asterisk to work. For the older asterisk 1.2 |
| 14:14.53 | bip | this is the package asterisk-bristuff . Current asterisk 1.4 packages |
| 14:14.54 | bip | include support for bristuff. |
| 14:15.09 | bip | so i guess I already have bristuff tzafrir |
| 14:15.15 | tzafrir | bip: what card do you have? |
| 14:15.21 | *** join/#asterisk garymc (~chatzilla@host81-139-127-32.in-addr.btopenworld.com) |
| 14:15.26 | bip | i have ... one sec |
| 14:15.58 | bip | 00:0a.0 Network controller: Digi International Datafire Micro V (Europe) (rev 02) |
| 14:16.09 | WIMPy | never had success with bristuff, either. |
| 14:16.15 | bip | hcf-s I have been told |
| 14:16.27 | bip | if does not work we will get new cards |
| 14:16.35 | WIMPy | But then none of the options are a real success, IMHO. |
| 14:16.39 | bip | but we have got 2 of those lying around |
| 14:16.50 | bip | so i m supposed to give em a try |
| 14:17.12 | bip | of course if something surely working and cheap exist please lemme know |
| 14:17.17 | WIMPy | bip: That's the most common hardware, you can get. |
| 14:17.40 | bip | yeah our windows based pbx use em |
| 14:17.48 | bip | but we must replace it |
| 14:17.59 | bip | for lots of reasons |
| 14:18.05 | bip | obsolte os |
| 14:18.10 | bip | obsolete os |
| 14:18.10 | WIMPy | At the moment, I'd recommend using misdn2/lcr/chan_lcr. But that won't happen with debian packages. |
| 14:18.14 | bip | windows sucks |
| 14:18.21 | bip | and so on ;-) |
| 14:18.39 | bip | u mean i shoul use centos on that machine ? |
| 14:18.47 | bip | we are a debian shop ... |
| 14:18.52 | coppice | friends don't ask friends to use misdn2 |
| 14:19.01 | bip | we have some rhel based appliances |
| 14:19.12 | WIMPy | You can use debian, if you update it manually. |
| 14:19.22 | bip | but me and and my cell mate prfer debian ;-) |
| 14:19.36 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 14:19.36 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:19.39 | bip | we do everything by hand here ... |
| 14:19.55 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 14:20.03 | [TK]D-Fender | Behold the contradiction.... |
| 14:20.07 | bip | fabolous handmade italian appiances ;-) |
| 14:20.17 | WIMPy | Get Kernel, Asterisk and LCR from source and you should get it going. |
| 14:20.40 | bip | recompile the whole bunch ? |
| 14:20.58 | WIMPy | yes |
| 14:20.59 | bip | and changin card wont be quicker cleaner ? |
| 14:21.01 | tzafrir | bip: I don't recall this supported with dahdi. What's the output from lspci -nn (only the line for that device) |
| 14:21.12 | bip | one sec tzafrir |
| 14:21.14 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 14:21.25 | tzafrir | bristuff: the Debian packages or installing from the bristuff distro? |
| 14:21.36 | WIMPy | bip: You could try CAPI. But it's been ages since I tried that way. |
| 14:21.43 | bip | 00:0a.0 Network controller [0280]: Digi International Datafire Micro V (Europe) [114f:0071] (rev 02) |
| 14:22.12 | WIMPy | bip: It's basically the same for most single port cards out there. |
| 14:22.17 | tzafrir | I don't think this has DAHDI drivers. |
| 14:22.18 | bip | well i stick to packages if I can tzafrir or backports |
| 14:22.51 | bip | so same sweat blood and tears which ever card i use WIMPy ? |
| 14:23.08 | tzafrir | DAHDI is in backports . But as I mentioned, that PCI ID is not something I recall from DAHDI |
| 14:23.33 | firstpoint | So I installed asterisk-mysql |
| 14:23.44 | WIMPy | CAPIs are diffrent from verndor to vendor. Most cards are HFC-S, though. At least all the cheap ones. But they're prefectely ok. |
| 14:23.50 | firstpoint | I got some errors concerning crd_mysql.conf, so I added the basic configuration in the file |
| 14:23.56 | tzafrir | If it's an HFC card, it should be trivial to patch the driver to accept it |
| 14:23.59 | bip | so lemme think about my options: |
| 14:24.07 | firstpoint | However it now says that cdr table cannot be found |
| 14:24.14 | bip | stick to debian but compile all the stuff I need |
| 14:24.19 | tzafrir | (or simpler: 'bind' in sysfs, without any patching) |
| 14:24.36 | firstpoint | I don't want to install mysql cdr with asterisk, but I just want to activate realtime on asterisk, with a2billing database... am I on the right way ? |
| 14:24.43 | bip | option 2 change distro centos and yum install all I need ? |
| 14:24.56 | bip | any other option ? |
| 14:24.57 | tzafrir | bip: try capi |
| 14:25.06 | bip | like 3 4 and 5 ? |
| 14:25.14 | bip | ok otion 3 try capi |
| 14:25.32 | bip | any other suggestion ? |
| 14:26.13 | bip | ado you know any good capi tutorial ? |
| 14:26.15 | WIMPy | ^^ Get Kernel, Asterisk and LCR from source and you should get it going. |
| 14:26.23 | bip | yeah WIMPy |
| 14:26.35 | bip | but i d rather use binaries if i can |
| 14:26.42 | firstpoint | any idea anyone ? |
| 14:26.53 | bip | i need very very serious reasons to work from sources |
| 14:26.56 | WIMPy | Old misdn1 is more straightforward, but doesn't have the best reputation. |
| 14:27.05 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 14:27.20 | kaldemar | firstpoint: you can disable the module. add noload => cdr_addon_mysql.so in your /etc/asterisk/modules.conf. |
| 14:27.22 | WIMPy | bip: Like to get it going? :-) |
| 14:27.27 | bip | i do compile only when non binary is available |
| 14:27.38 | bip | well ... |
| 14:27.45 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 14:27.55 | bip | i d rather change distro the compile just to get it going |
| 14:28.14 | bip | imho it s easier to mantain when u use prepackaged binaries |
| 14:28.30 | firstpoint | kaldemar -> will try that thanks |
| 14:28.50 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 14:29.01 | bip | sso option inpriority order are: |
| 14:29.07 | bip | 1 try CAPI |
| 14:29.31 | bip | 2 switch to centos and see if the works |
| 14:29.40 | bip | 3 compile everything |
| 14:29.47 | bip | am I correct ? |
| 14:29.57 | WIMPy | Your choice :-) |
| 14:30.00 | bip | or there is something I m missing ? |
| 14:30.17 | WIMPy | is away for an hour or two now. |
| 14:30.20 | bip | yeah i know it s my choice |
| 14:30.27 | firstpoint | Connected to mya2billing@127.0.0.1, port 3306 with username a2billinguser for 3 seconds. |
| 14:30.29 | firstpoint | thanks ! |
| 14:34.14 | *** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr) |
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| 15:05.54 | E-bola | http://www.listware.net/201009/asterisk-users/40259-asterisk-users-doing-dnsmgrlookup.html |
| 15:06.03 | E-bola | ^^ do anybody have the answer for him? I have the same issue |
| 15:06.36 | E-bola | I dont understand why asterisk needs to constantly do dns lookups when no new hostnames are being accessed |
| 15:08.29 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 15:11.15 | Qwell | bip: your nick is confusing my proxy :p |
| 15:11.23 | Qwell | (really, it's confusing ME, heh) |
| 15:11.37 | bip | why ? |
| 15:12.14 | Qwell | because when I do proxy stuff, it responds as -bip |
| 15:12.19 | Qwell | <-bip> /BIP (RELOAD|LIST|JUMP|BLRESET|HELP) |
| 15:12.34 | Qwell | carry on. just a random comment. |
| 15:15.54 | vinhdizzo | Hi everyone. I've never used Asterisk and my knowledge about telephony is quite limited. I was wondering whether Asterisk could be used with Google Voice (phone number) and Gizmo5 (VoIP provider) to get an auto attendant when someone dials my GV #. Would the call forwarding from Asterisk count as minutes from Gizmo5? Or does the minute counting stop after the call is forwarded? Thanks. |
| 15:16.43 | Qwell | vinhdizzo: how does gizmo come into that? |
| 15:16.46 | Letoric | Qwell: Is there any issue that you are aware of, with installing festival for linux on a aystem with asterisk present? |
| 15:16.47 | Qwell | just use GV |
| 15:16.58 | Qwell | Letoric: why would there be a problem? O.o |
| 15:17.12 | Letoric | Qwell: I was hoping there wouldn't be, but wanted to double check ;) |
| 15:17.32 | vinhdizzo | Qwell: GV would have to be forwarded somewhere, so Gizmo5's SIP? |
| 15:17.41 | Qwell | why would it have to be forwarded somewhere? |
| 15:17.59 | vinhdizzo | OK, so maybe I'm not understanding Asterisk then |
| 15:18.16 | *** join/#asterisk fofware (~Fabian@host199.190-31-51.telecom.net.ar) |
| 15:18.31 | vinhdizzo | Qwell: Was following this http://www.pbxer.com/build-your-pbx-step-5-add-a-voip-provider/ and the last step says to add a VOIP provider, so i figured one was needed |
| 15:18.48 | vinhdizzo | So GV can be connected to Asterisk directly? |
| 15:19.03 | Qwell | vinhdizzo: do yourself a favor and blackhole that domain, and anything controlled by him :) |
| 15:19.21 | Qwell | actually, wait what? |
| 15:19.31 | Qwell | I believe he ripped that content off. |
| 15:21.01 | vinhdizzo | I see. Any recommendation on a website then? I guess I'm still confused on what Asterisk is exactly. My impression now is it's a "phone program" that a phone number can connect to, either through VoIP or a landline (via some hardware) |
| 15:22.40 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 15:23.34 | [TK]D-Fender | vinhdizzo: * is a PBX and telephony toolkit |
| 15:23.44 | [TK]D-Fender | vinhdizzo: You can do whatever you want with it |
| 15:24.26 | [TK]D-Fender | vinhdizzo: Some people make PBX's with it. I use mine as a jukebox and coffee timer |
| 15:25.10 | vinhdizzo | [TK]D-Fender: thanks for your response. How does it connect to the phone system worldwide? Post me a link if you don't have time to explain |
| 15:25.38 | beek | ~book |
| 15:25.38 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:25.53 | beek | vinhdizzo: ^^^^^^^^^^^^^^^^^^ |
| 15:26.02 | vinhdizzo | thanks |
| 15:26.02 | *** join/#asterisk ANurmi (~Adam@63.230.70.254) |
| 15:27.22 | [TK]D-Fender | vinhdizzo: GV provides service, and there are regular ITSPs. You can also by hardware interfaces to use real lines (BRI, PRI, T1, analog, etc) |
| 15:27.25 | [TK]D-Fender | ~itsp |
| 15:27.25 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 15:28.48 | *** join/#asterisk Marquel (~Marquel@84.16.240.149) |
| 15:28.49 | vinhdizzo | [TK]D-Fender: Thanks. I'll read more into the documentation. |
| 15:29.32 | [TK]D-Fender | vinhdizzo: What do yuo want to accomplish? |
| 15:29.43 | vinhdizzo | So if a call is forwarded from a Call Attendant, would minutes continue to count from the VoIP after the call is forwarded? Or are the calls connected? |
| 15:30.30 | vinhdizzo | [TK]D-Fender: I just want to set up such that my GV number goes to an auto attendant. From there, have options to listen to a few pre-recorded messages, or select numbers to be forwarded to |
| 15:30.40 | [TK]D-Fender | vinhdizzo: There is no "forward". * sits in the middle of calls. |
| 15:30.53 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:30.53 | *** join/#asterisk jamko (~chatzilla@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 15:31.04 | Marquel | morning, little problem with includes. i have three includes in my context, two of them in the form 'include => other|08:00-19:30:mon-fri|*|*' - dialplan show does not use these includes. what is wrong with that? |
| 15:31.10 | vinhdizzo | [TK]D-Fender: Tha |
| 15:31.17 | vinhdizzo | Thanks. thats what i wanted to know |
| 15:31.38 | vinhdizzo | So * sits in the middle of the call. That means I *should* get an unlimited VoIP plan |
| 15:34.20 | [TK]D-Fender | vinhdizzo: that depends on your actual needs |
| 15:34.47 | [TK]D-Fender | vinhdizzo: most people don't actually use enough "per-minute" value to warrant use of an "unlimited" type of service |
| 15:34.57 | [TK]D-Fender | vinhdizzo: You would ahve to examine your needs very closely |
| 15:34.59 | vinhdizzo | [TK]D-Fender: yea. I guess I don't expect THAT many calls. |
| 15:35.23 | vinhdizzo | I'll look into Ooma |
| 15:37.17 | vinhdizzo | One more question. Can Asterisk make automated telephone calls? |
| 15:38.08 | [TK]D-Fender | vinhdizzo: Yes. You can trigger it to call out |
| 15:38.24 | [TK]D-Fender | vinhdizzo: And process that call the same ways you would a call that came in. |
| 15:39.32 | vinhdizzo | Actually one more. Appears you guys have explored all the options with GV. I know this is a long shot, but can we make free dial out calls from GV using Asterisk like in gmail? I doubt just thought I'd ask |
| 15:40.27 | [TK]D-Fender | vinhdizzo: there (were) means of doing this. I don't know if those are closed off yet. Get Google-ing |
| 15:41.10 | vinhdizzo | [TK]D-Fender: Once everything is set up, how easy would it be to select an mp3 and list of phone numbers (txt?) each week for call automation? |
| 15:41.56 | [TK]D-Fender | vinhdizzo: there is no such thing as a "list" you will have to trigger the calls with a script of your own making |
| 15:42.06 | *** part/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 15:42.08 | [TK]D-Fender | vinhdizzo: there is no "queueing" followup-etc. |
| 15:42.29 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-123-250.lsanca.fios.verizon.net) |
| 15:42.40 | [TK]D-Fender | [11:41]<[TK]D-Fender>vinhdizzo: there is no such thing as a "list" you will have to trigger the calls with a script of your own making |
| 15:42.40 | vinhdizzo | accidentally clsoed window |
| 15:42.41 | [TK]D-Fender | [11:42]<[TK]D-Fender>vinhdizzo: there is no "queueing" followup-etc. |
| 15:43.09 | vinhdizzo | thanks [TK]D-Fender |
| 15:44.02 | vinhdizzo | Can you give me a link to an example hardware I would need if I wanted to use my landline with Asterisk? |
| 15:44.31 | [TK]D-Fender | vinhdizzo: that clearly depends on WHAT hardware you'd use |
| 15:44.49 | vinhdizzo | Have to take mom to the dr's now. I'll read up and ask more questions later. Thanks [TK]D-Fender |
| 15:44.50 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 15:47.20 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:47.23 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 15:48.39 | timeshell_atwork | Greetings. Anyone know why I'd get a "Call/Transaction Does Not Exist (481)" error after making a call using a SIP extension connected by TCP? |
| 15:50.02 | timeshell_atwork | Version 1.6.0.28 |
| 15:50.59 | *** join/#asterisk trelane` (~trelane@funtoo/staff/trelane) |
| 15:55.39 | timeshell_atwork | Found out why. Pedantic doesn't work properly with TCP SIP connections. |
| 15:55.46 | timeshell_atwork | At least not with 1.6.0.28 |
| 15:56.52 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:02.16 | leifmadsen | o.O |
| 16:03.32 | Marquel | nobody? |
| 16:07.00 | WIMPy | fuck |
| 16:07.09 | WIMPy | I hate PC sports. |
| 16:07.11 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:07.25 | WIMPy | Just carried a 22" CRT to the 2nd floor. |
| 16:07.35 | WIMPy | 76lbs :-( |
| 16:07.41 | timeshell_atwork | leifmadsen Was that directed at me? |
| 16:08.04 | leifmadsen | timeshell_atwork: ya :) |
| 16:08.12 | timeshell_atwork | Well, is it fixed yet? |
| 16:08.12 | WIMPy | bip: What's going on? No news? :-) |
| 16:08.14 | timeshell_atwork | :D |
| 16:08.15 | leifmadsen | WIMPy: ya those things are heavy |
| 16:08.20 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:08.24 | leifmadsen | timeshell_atwork: no, because you're using 1.6.0.28 :) |
| 16:08.26 | timeshell_atwork | I'm trying out 1.6.2.14-rc1 right now |
| 16:08.29 | timeshell_atwork | I'll let you know |
| 16:08.46 | leifmadsen | I'm not really all that interested :) file an issue you think there's a bug |
| 16:08.47 | timeshell_atwork | I'm using 1.6.0.28 because I KNOW it WORKs with asterisk-gui |
| 16:08.59 | timeshell_atwork | Of course I will. |
| 16:09.04 | timeshell_atwork | Just don't reject it on me. |
| 16:09.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 16:09.06 | timeshell_atwork | :D |
| 16:10.14 | leifmadsen | timeshell_atwork: I promise nothing :) |
| 16:10.35 | leifmadsen | timeshell_atwork: as long as you don't file it against 1.6.0 or 1.6.1, then I will triage as appropriate :) |
| 16:10.41 | *** join/#asterisk erinspice (~erin@207.98.195.107) |
| 16:10.41 | drmessano | All bug reports should be filed against 1.12.x. That is all. |
| 16:10.48 | *** part/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 16:11.12 | leifmadsen | heh, that is 2 major versions away! |
| 16:12.31 | drmessano | Yes, well, not to step on your toes, but I am already working on the book based on 1.12.x |
| 16:13.50 | leifmadsen | drmessano: awesome! it will probably take you 2-3 years anyways :) |
| 16:15.01 | drmessano | Yes, at least.. it's hard documenting app_facebook and app_twitter when they've not even been conceived yet. |
| 16:17.02 | *** join/#asterisk asw (~kostas@109.178.255.180) |
| 16:17.59 | asw | hello,is there any available .deb for asterisk?thanks |
| 16:18.06 | seanbright | an app that integrates with facebook and twitter!? thats revolutionary |
| 16:18.28 | coppice | its called firefox |
| 16:20.27 | Marquel | nobody with knowledge about timed includes? |
| 16:21.42 | Letoric | Marquel - you might look at 'GoToIfTime' |
| 16:23.53 | Marquel | Letoric: so no timed includes in asterisk 1.6 anymore? |
| 16:24.20 | bip | i m lost WIMPy googling n trying |
| 16:24.27 | bip | capi |
| 16:24.29 | bip | misdn |
| 16:24.43 | bip | i know somwhere there is a solution |
| 16:24.53 | bip | but I have not yet found it |
| 16:25.05 | WIMPy | Many solutions :-) |
| 16:25.11 | bip | yeah |
| 16:25.17 | WIMPy | But no perfect one. |
| 16:25.18 | bip | compile mISDN |
| 16:25.35 | bip | replace zaptel with dadhi maybe |
| 16:25.38 | WIMPy | The old one? |
| 16:25.39 | Letoric | Marquel: they are still there, according to the documentation I see, I'm just not familiar with them personally |
| 16:25.44 | bip | figure how capi works |
| 16:26.10 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:26.21 | bip | well i have not found any info about mISDN 2 for debian |
| 16:26.39 | bip | just a couple of howtos about compilin 1.9 ... |
| 16:26.54 | Marquel | Letoric: okay, so i guess i'm doing it wrong... :( |
| 16:27.05 | WIMPy | It's probably too new. Only 2 or 3 years. |
| 16:27.06 | bip | the i have installed a asterisk-chan-capi debian Ãpackage |
| 16:27.19 | bip | but cannot figure how it works |
| 16:27.20 | *** join/#asterisk Jareeta (~ahailes@WHITE-METEO.MIT.EDU) |
| 16:27.20 | Letoric | Marquel: According to what I see on voip-info.org, it's pretty easy to do - just make sure to use the comma as the delimiter, | isn't correct with 1.6 |
| 16:27.42 | bip | then i have found a lenny dadhi backport thta might work |
| 16:27.43 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 16:27.47 | WIMPy | You also need the capi driver for the card |
| 16:28.04 | bip | but today the debian backports repos seem to be offline |
| 16:28.16 | WIMPy | Even with dahdi you still need extra support for zaphfc. |
| 16:28.22 | bip | how do I find the capi driver for the card WIMPy ? |
| 16:28.38 | bip | dadhi isnt installing right now |
| 16:29.01 | joshaidan | Hey, quick question, what is Asterisk trying to do when it outputs "chan_sip.c: Oooh, format changed to 4" in the logs? |
| 16:29.01 | bip | i have found 3 packages in debian backports: |
| 16:29.05 | WIMPy | Either it's in the kernel or on the vendors CD or website. |
| 16:29.45 | bip | dahdi-linux dahdi-source dahdi |
| 16:30.10 | bip | vendor might be out of business as far as i know |
| 16:30.35 | WIMPy | Not in the kernel. |
| 16:30.56 | bip | we just have those 2 cards theat were shipped with this legacy windows pbx |
| 16:31.20 | bip | now i have been asked to replace it with asterisk so one card was pulled out |
| 16:32.08 | bip | how i figure if there is the right capi driver in the kernel WIMPy ? |
| 16:32.44 | WIMPy | The Kernel only has CAPI for AVM and Eicon cards. |
| 16:33.45 | bip | so i might need to change card eventually ? |
| 16:34.51 | WIMPy | To what? |
| 16:35.14 | bip | to one that produce less headache |
| 16:35.24 | bip | if that card exist ;-) |
| 16:35.53 | bip | hello do you have a cheap non headache producing isdn card ? |
| 16:35.55 | bip | ;-) |
| 16:35.59 | WIMPy | Multi-port cards are supported natively by dahdi. |
| 16:36.17 | bip | mmm |
| 16:36.26 | bip | one question ... |
| 16:36.33 | WIMPy | It's just that Asterisk and ISDN never liked each other much. |
| 16:36.36 | Qwell | bip: BRI? |
| 16:36.40 | bip | yep |
| 16:36.47 | bip | isdn is what we have got here |
| 16:36.56 | Qwell | Digium BRI410p |
| 16:37.00 | WIMPy | But those el cheapo cards are actually fine if you know how to do it. |
| 16:37.05 | bip | and in europe i guess that mean bri |
| 16:37.11 | Qwell | or...whatever the model # is |
| 16:37.16 | bip | how much Qwell ? |
| 16:37.29 | WIMPy | B410p |
| 16:37.30 | Qwell | oh, B410P |
| 16:37.30 | Qwell | http://www.digium.com/en/products/digital/ |
| 16:37.35 | bip | i work for some cheap bastards ;-) |
| 16:37.51 | Qwell | or an HA8/HB8 |
| 16:38.20 | WIMPy | What's that? |
| 16:38.26 | Marquel | Letoric: okay.... * does not include it at all... :/ |
| 16:38.29 | Qwell | http://www.digium.com/en/products/hybrid/h8.php |
| 16:38.36 | bip | will this one work: |
| 16:38.40 | bip | http://www.voipango.com/en/ISDN-Interfaces/Digium/TDM410P-Bundles/Digium-TDM402BF-2x-FXO.html |
| 16:38.45 | Qwell | no |
| 16:38.47 | Qwell | that's analog |
| 16:38.50 | bip | it sorta fits in our budget |
| 16:38.54 | Letoric | Marquel: pastebin the appropriate areas of the dialplan, and the output from the console during a call |
| 16:38.54 | bip | ah ok |
| 16:39.23 | Letoric | Marquel: once you do that, somebody can provide more assistance (I'll try, but with the right info, you'll get a lot more involvement in here) |
| 16:39.39 | WIMPy | Ah, an answet to the sangoma combothingies, I guess. |
| 16:40.10 | bip | way too expensive |
| 16:40.20 | bip | anything worth 25% of that ? |
| 16:40.26 | [TK]D-Fender | Qwell: Congrats on you guys bridging that functionality gap your lineup had |
| 16:40.33 | bip | http://www.voipango.com/en/ISDN-Interfaces/Digium/BRI-PRI-E1-T1-J1/Digium-TE122PF-T1/E1-PRI-PCI.html |
| 16:40.39 | Qwell | bip: also no |
| 16:40.51 | bip | that cost 400 euros |
| 16:40.53 | Qwell | Please look at the model numbers I said. Don't just pick random stuff. |
| 16:41.11 | bip | B410P ? |
| 16:41.21 | Qwell | or H8 |
| 16:41.33 | *** join/#asterisk cutlass (~cutlass@c-67-165-132-67.hsd1.il.comcast.net) |
| 16:41.35 | WIMPy | There are tons of 4 or 8 port BRi cards outside. But they are all the same technology, you already have. |
| 16:41.50 | bip | quad bri is too much |
| 16:41.57 | bip | we only have 2 lines |
| 16:42.18 | WIMPy | Then you already have the optimum. |
| 16:42.31 | WIMPy | Hmm. Beronet did 2 port, IIRC. |
| 16:42.50 | bip | its sangoma cheaper ? |
| 16:42.59 | cutlass | I have an outbound proxy configured and I'm trying to send SIP OPTIONS to the endpoints (qualify=yes)....I'm finding that the OPTIONS messages are bypassing the proxy and going directly to the endpoints...does anyone know how to get the OPTIONS to route through the proxy also? Is this expected behavior? |
| 16:43.12 | bip | yes we do but we need to have em working with asterisk |
| 16:43.34 | WIMPy | CCD (the maker if the HFC chips) more or less has a monopoly on BRI stuff now. |
| 16:44.18 | WIMPy | Take a recent kernel, LCR and a recent stable Asterisk. |
| 16:44.28 | Marquel | Letoric: http://paste.pocoo.org/show/275478/ - extensions.conf (relevant part) |
| 16:44.53 | coppice | bri is a dying market. there probably isn't enough volume left for the other makers, like winbond, to continue |
| 16:45.29 | bip | is this recent enuff WIMPy : |
| 16:45.33 | bip | fenrir:/usr/share/doc/asterisk-chan-capi# asterisk -V |
| 16:45.33 | bip | Asterisk 1.4.21.2~dfsg-3+lenny1 |
| 16:45.51 | bip | i might have a more recent one in backports |
| 16:45.53 | Qwell | only 15 releases behind. |
| 16:45.55 | WIMPy | Don't think there's anything dying yet. Actially with the migration to voip I think sales are higher than ever before. |
| 16:46.29 | bip | 15 releases doesn t matter if it works Qwell |
| 16:46.30 | coppice | huh? bri cards mostly sold for data use. nobody does data by bri any more |
| 16:46.33 | Marquel | Letoric: using that configuration leads to * using nighttime context all the time. specifying (mutually exclusive) times for nighttime context produces http://paste.pocoo.org/show/275479/ |
| 16:46.34 | WIMPy | No. Thats nothing near recent. |
| 16:46.42 | bip | ok |
| 16:46.56 | bip | lemme see if backports is better in your opinion |
| 16:47.15 | WIMPy | They are still used for data, but they weren't used much for voice which happens now. |
| 16:47.21 | bip | asterisk (1:1.6.2.9-1~bpo50+3) |
| 16:47.27 | Letoric | Marquel: you are not specifying time for the nighttime, so yes, it will always be included |
| 16:47.30 | bip | is that any better WIMPy ? |
| 16:47.40 | WIMPy | Fax servers are probably the same as ever. |
| 16:47.49 | WIMPy | Yes, that's like it. |
| 16:47.55 | bip | i can get that if backports gets fixed |
| 16:47.55 | Qwell | bip: Just the security vulnerability. But who cares about those, right? |
| 16:48.02 | Marquel | Letoric: yes, but according to docs, worktime should be tried first. and it isn't tried at all. |
| 16:48.03 | coppice | how many people still use BRI for data? |
| 16:48.29 | Marquel | Letoric: and specifying times for nighttime renders any incoming call not-working. |
| 16:48.30 | bip | we should we are a security consultancy firm Qwell ;-) |
| 16:48.35 | WIMPy | POS terminals. |
| 16:48.47 | Qwell | bip: well, you clearly don't, if you're 15 releases behind. |
| 16:49.21 | bip | well i care about getting thing done sometimes |
| 16:49.22 | Letoric | I don't see where you placed a call in that console output |
| 16:49.28 | bip | receving calls |
| 16:49.32 | bip | making calls |
| 16:49.48 | bip | after I get that i start thinking about the erst |
| 16:49.50 | bip | oops |
| 16:49.51 | WIMPy | bip: Why didn't you start with a non-historic distro two hours ago, tehn? |
| 16:49.54 | bip | after I get that i start thinking about the rest |
| 16:50.05 | Marquel | Letoric: according to docs "dialplan show" is equivalent to an incoming call. |
| 16:50.18 | bip | well i use debian and usually it does whhat i need to do ... |
| 16:50.32 | coppice | last time I communicated with someone at cologne it sounded like they were close to abandoning BRI |
| 16:50.33 | bip | but if we want to do a distro war, why not ;-) |
| 16:50.40 | Letoric | Marquel: Set verbosity to 3, call the system, watch the system process through the call |
| 16:50.59 | Marquel | Letoric: did so. and it does not try worktime at all. |
| 16:51.03 | bip | let s say i use the backported asterisk |
| 16:51.16 | bip | shall i get a backported kernel as well ? |
| 16:51.26 | bip | what else ? |
| 16:51.31 | Letoric | ok, well do dialplan show isdn-in |
| 16:51.36 | WIMPy | coppice: There's no doubt, the phone system is being shut down, but as I said: ATM there is a high demand for interfacing the old and the new. |
| 16:52.09 | Letoric | don't forget dialplan reload if you are editing the file as we're going through this fun |
| 16:52.10 | Marquel | Letoric: it lists all includes. |
| 16:52.21 | Marquel | of course not. |
| 16:52.26 | WIMPy | And it still makes sense to use ISDN phones locally, as IP phones are pretty expensive to run. |
| 16:52.44 | Qwell | expensiwhatnow? |
| 16:53.11 | bip | well cost is varriable depend for the local market |
| 16:53.18 | WIMPy | How many words is that? |
| 16:53.21 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:53.31 | bip | we do not all live in northern europe or usa ... |
| 16:54.42 | anonymouz666 | I live in Hong Kong coppice is my neighbour |
| 16:54.44 | anonymouz666 | rs |
| 16:54.45 | WIMPy | It's not the purchase price. I said the running costs. They draw much more power and switches draw a lot of power as well. That's somehow the exact opposite of green IT. |
| 16:54.45 | anonymouz666 | heh |
| 16:54.59 | anonymouz666 | j/k |
| 16:55.27 | cutlass | I have an outbound proxy configured and I'm trying to send SIP OPTIONS to the endpoints (qualify=yes)....I'm finding that the OPTIONS messages are bypassing the proxy and going directly to the endpoints...does anyone know how to get the OPTIONS to route through the proxy also? Is this expected behavior? |
| 16:55.39 | WIMPy | And the comparison between a classical PBX and a PC is even worse. |
| 16:55.51 | coppice | I wonder how many ISDN phones were ever made. the number must be really small. |
| 16:55.58 | bip | we already have apc based pbx |
| 16:55.59 | Marquel | Letoric: okay, both worktime and nighttime includes are listed in dialplan show, just worktime is never executed. |
| 16:56.18 | bip | we just want to get rid of windows and use a linux based pbx |
| 16:56.37 | WIMPy | I'd guess that tehere a more ISDN phones than VOIP phones so far. |
| 16:56.40 | bip | we have file server mail server and web server running debian |
| 16:56.58 | bip | we already have sip phones WIMPy |
| 16:57.08 | bip | only the outbound line is isdn |
| 16:57.17 | *** join/#asterisk joako_ (~joako@opensuse/member/joak0) |
| 16:57.18 | Letoric | Marquel: Unfortunately, I just don't enough to help you through this. The best thing I can offer, is how I handle time testing in my dialplan. Maybe it will help you.. |
| 16:57.20 | Letoric | http://paste.pocoo.org/show/275485/ |
| 16:57.25 | bip | and it will eventually be phased out |
| 16:57.58 | WIMPy | bip: Yes, that's what ppl are usually building these days. And hence the need for ISDN cards. |
| 16:58.29 | coppice | WIMPy: Not even close. BRI phones only sold in a few markets, and only in small numbers in those few markets. Most of the world's major offices are now full of IP phones. |
| 16:58.34 | bip | we have some old usb based phones that are dying |
| 16:58.51 | bip | now we got a cheap offer on some cisco sip phones |
| 16:59.08 | WIMPy | But there have been many millions of ISDN phones. |
| 16:59.31 | bip | here all phones are sip phones |
| 16:59.44 | coppice | WIMPy: where? |
| 16:59.50 | WIMPy | It was hard enough to get a quote on ISDN chips if you wanted to buy only 100000. The price lists started at 1000000. |
| 16:59.58 | bip | and since most server I manage run debian I d rather have debian in my asterisk server as well |
| 17:00.32 | bip | but yes i might use the asterisk lemmy-backports version |
| 17:00.55 | bip | what is LCR WIMPy ? |
| 17:01.20 | bip | do i need that and a more recent kernel as well ? |
| 17:01.22 | WIMPy | http://linux-call-router.de/ |
| 17:01.26 | drmessano | They named a Debian version after Lemmy from Motorhead? |
| 17:01.28 | drmessano | How awesome |
| 17:01.31 | bip | is writing down shopping list |
| 17:01.36 | Qwell | drmessano: Toy Story, newb |
| 17:01.47 | WIMPy | Also check http://misdn.org/ |
| 17:01.52 | bip | ok lcr = linux call router ? |
| 17:01.57 | Qwell | I wonder when they'll release Buzz |
| 17:02.01 | WIMPy | Yes |
| 17:02.05 | drmessano | Qwell, wouldn't that be LENNY, NEWB? |
| 17:02.59 | Qwell | drmessano: no |
| 17:03.24 | Marquel | Letoric: i'd say yes, but my contexts are a little more complex and i don't intend to throw GoTo* in the dialplan at it.... |
| 17:03.51 | bip | well i had already found that page WIMPy but every doc about debian is about compiling misdn 1.x |
| 17:04.15 | *** join/#asterisk oryxtec (~test@119.152.76.97) |
| 17:04.32 | WIMPy | Sure. As misdn2 is in the standard kernel, there's nothing special about that any more. |
| 17:04.45 | *** join/#asterisk BesticlesWork (~larry@209-58-227-178.static-ip.telepacific.net) |
| 17:04.54 | WIMPy | You just need the new misdnutils as well. |
| 17:05.26 | BesticlesWork | is there a place where i can view the changes from 1.6.2.9 to 1.6.2.13. I'm having issues with AMI. |
| 17:05.43 | bip | well all misdn stuff is considered eperimental in debian ... |
| 17:05.59 | bip | experimental ... |
| 17:06.13 | bip | and not advised for production |
| 17:06.23 | bip | ok lemme rephrase |
| 17:06.31 | WIMPy | misdn1 or misdn2? |
| 17:06.40 | bip | what distro do you run asterisk on WIMPy ? |
| 17:07.02 | bip | it s not specified WIMPy but i might check |
| 17:07.10 | WIMPy | misdn1 never went beyond experimental. That's why it was replaced by misdn2 which then went into Linux. |
| 17:07.20 | WIMPy | 's using Slackware. |
| 17:07.40 | anonymouz666 | Slackware still exist? |
| 17:07.43 | anonymouz666 | that's a mistake. |
| 17:07.47 | anonymouz666 | hehe |
| 17:07.51 | WIMPy | But anything less rock stable than Debian should do. |
| 17:08.05 | bip | well our chief engineer loves slackware |
| 17:08.08 | WIMPy | Works very well for me. |
| 17:08.19 | bip | but he is too busy to do this project I m afraid |
| 17:08.23 | bip | ;-) |
| 17:08.29 | *** part/#asterisk espiceland (~erin@nat/digium/x-zfoczztsbusudihf) |
| 17:09.06 | bip | anything less rock stable than Debian should do == ubuntu ? |
| 17:09.25 | bip | i don t love it, but i can adapt if i need to ;-) |
| 17:09.29 | WIMPy | Anything that comes with a kernel less than two years old. |
| 17:09.43 | WIMPy | The rest is configure&&make&&make install. |
| 17:09.44 | [TK]D-Fender | Nothing so stable as a dead-stop... |
| 17:09.57 | bip | well i can install a more recnte backported kernel in debian WIMPy |
| 17:10.09 | bip | goes checking |
| 17:10.22 | WIMPy | At least tehy don't document bugs instead of fixing them like Cisco :-) |
| 17:11.33 | WIMPy | But the fix needs to be documented for at least two years, which isn't much better in practice. |
| 17:11.48 | bip | Ãwill that work WIMPy : linux-image-2.6-686 (2.6.32+27~bpo50+1) |
| 17:12.21 | WIMPy | Should be ok. |
| 17:12.35 | bip | ok thanx everybody |
| 17:12.48 | bip | goes running home |
| 17:12.51 | bip | :) |
| 17:13.22 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 17:13.23 | cusco | hi |
| 17:13.30 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:14.23 | drmessano | Ok now, lets stop mocking Debians ultra long release cycle |
| 17:14.30 | drmessano | No, carry on |
| 17:15.03 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 17:15.16 | bip | well u love it if u do basic stuff with a single hand ;-) |
| 17:15.44 | cusco | its ready when its ready |
| 17:15.44 | bip | might not work for 2 handed cutting edge ;-) |
| 17:15.45 | cusco | :-) |
| 17:15.49 | *** join/#asterisk drudge` (drudge@unaffiliated/drudge/x-837452) |
| 17:16.11 | drmessano | Do they support CD-ROM's yet? |
| 17:16.11 | bip | beside we wear capes ... |
| 17:16.18 | drmessano | These floppies are getting heavy |
| 17:16.26 | bip | so it s not worth discussing with us ;-) |
| 17:16.49 | *** join/#asterisk thansen (~thansen@12.8.216.84) |
| 17:16.57 | cusco | can sendFax() app return receipts ?? |
| 17:17.13 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 17:17.32 | *** join/#asterisk Michael_On_Jira (~chatzilla@199.84.137.3) |
| 17:17.59 | nny | looking into an interesting issue with one extension using a Polycom 320. I have swapped the phone out with other phones but the problem returns. have a user who is experiencing lockup/reboot on a second call. Going to investigate firmware updates and diff the config. any experience others may have had appreciated... |
| 17:22.15 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 17:35.06 | nny | so I found this line <OVERRIDES reg.1.ringType="16"> in the MAC-phone.cfg. No other phones use ring type 16. Could this be the issue? |
| 17:35.22 | nny | other phones have this override line, but not with that ring tone... |
| 17:35.42 | nny | sorry <OVERRIDES reg.1.ringType="16"/> |
| 17:41.18 | *** join/#asterisk ukine_work (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 17:41.43 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 17:45.27 | coppice | cusco: what does returning a receipt mean to you? |
| 18:04.01 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 18:07.40 | *** join/#asterisk Ouchi (~kvirc@200.150.66.162) |
| 18:16.16 | *** join/#asterisk slidesinger (~slidesing@c-174-57-6-126.hsd1.nj.comcast.net) |
| 18:16.42 | Katty | WARNING I HAVE CUPCAKES |
| 18:17.34 | *** join/#asterisk jolly23 (~haryv@154.5.144.198) |
| 18:17.41 | [TK]D-Fender | Katty: This is a Muffin-Only zone. BE WARNED! |
| 18:18.22 | coppice | english or american? |
| 18:18.35 | [TK]D-Fender | Stud :p |
| 18:18.36 | Katty | english muffins are good for breakfast |
| 18:18.45 | jolly23 | Is there a known list of linksys routers that will pass rtp and sip? Since we changed ISP's telus router will not pass the configure ports to my server and wonder if its not on a list of known routers that will work. |
| 18:18.52 | Katty | i'll stud your muffin in a minute. |
| 18:19.23 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 18:19.31 | thehar | WOOF |
| 18:20.29 | Jareeta | Mrow? |
| 18:20.37 | leifmadsen | jolly23: any of them? it doesn't require anything special |
| 18:21.02 | [TK]D-Fender | jolly23: jolly23 Did you set up the forwarding or not? |
| 18:21.14 | jolly23 | yes I did |
| 18:21.45 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 18:22.03 | jolly23 | BTW, I am also upgrading 1.4 to 1.6 since the DNS bug was still in 1.4 the unability of 1.4 to see the dns ISP killed my server. |
| 18:22.32 | jolly23 | Not sure why. Everything is configured and can ping internet hostname in the asterisk server. |
| 18:22.55 | anonymouz666 | DNS bug? |
| 18:23.13 | anonymouz666 | what DNS bug? |
| 18:23.19 | jolly23 | yes, 1.4 is it loose internet connectivity will bring down all the phones. Its a old one. |
| 18:23.45 | anonymouz666 | very old |
| 18:23.48 | jolly23 | local lan phones will go down :) |
| 18:24.24 | anonymouz666 | what changes in 1.6 regarding to this issue ? do we have now async DNS support? |
| 18:24.35 | anonymouz666 | and not lock the main thread |
| 18:25.05 | jolly23 | btw, will asterisk 1.6 work with dahdi-linux-2.1.0.4 |
| 18:26.40 | *** join/#asterisk `ph8 (ph8@unaffiliated/ph8) |
| 18:27.21 | jolly23 | Ill just remove it |
| 18:29.35 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:33.54 | jolly23 | getting yum mirror time outs errors on this yum download at http://www.asterisk.org/downloads/yum |
| 18:34.12 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-3.ngn.onecommunications.net) |
| 18:34.20 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 18:35.53 | fullstop | I am replacing our existing (ancient) voicemail / directory system with app_voicemail and app_directory. Our current directory allows you to enter the first 4 letters of the last name, and figures out who it might be. When you select a user, it will say "For future reference, #NAME# is extension XXXX." |
| 18:36.07 | fullstop | Is there any way to do the "For future reference" part with app_directory? |
| 18:37.18 | [TK]D-Fender | fullstop: it dumps you back into the dialplan at the target exten.. so say it in there before continuing to app_voicemail |
| 18:37.45 | fullstop | Thanks |
| 18:37.52 | fullstop | For some reason, I didn't consider that. |
| 18:42.58 | *** join/#asterisk oDesk (~f@94.99.102.139) |
| 18:46.54 | fullstop | [TK]D-Fender: is there any way to query the greet.g729 / greet.ulaw file for a particular voicemail user / context? |
| 18:47.50 | fullstop | app_directory.c gets it with: res = do_directory(chan, cfg, ucfg, args.vmcontext, args.dialcontext, res, digit, &flags, opts); |
| 18:47.58 | fullstop | but I do not have such luxuries from my dialplan. |
| 18:48.54 | Letoric | fullstop: You could just specifically play it, using variables to assist in the path to the file |
| 18:49.30 | fullstop | Stupid question, but does Playback allow me to pass an absolute path? |
| 18:49.31 | [TK]D-Fender | fullstop: Its a recording like any other. |
| 18:50.09 | fullstop | I was more interested in a way to put together a path from a voicemail context... but let me see what I can find. |
| 18:50.12 | [TK]D-Fender | fullstop: And yes, path is relative to the4 sounds folder unless prefixed with a "/" for a standard absolute path |
| 18:50.23 | fullstop | okay, so I want the / |
| 18:50.26 | fullstop | will do |
| 18:50.27 | fullstop | thanks |
| 18:50.39 | [TK]D-Fender | fullstop: /var/spool/voicemail/default/${exten}/greet |
| 18:50.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 18:53.47 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 18:56.04 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 18:57.24 | vinhdizzo | [TK]D-Fender: I searched for "asterisk google voice" and am running into PBX in a flash and Incredible PBX. Any warning regarding these? |
| 18:57.37 | Kobaz | anyone know where to get updated audiocodes firmware |
| 18:57.49 | Qwell | vinhdizzo: They are wrong. |
| 18:58.00 | Qwell | remember what I said about blackholing? that is where that applies. |
| 18:58.05 | vinhdizzo | Qwell: so stay away? |
| 18:58.06 | vinhdizzo | ok |
| 18:58.14 | jolly23 | Its my router. Lame isp likes to hug control over it so now need to edit it. |
| 18:58.47 | vinhdizzo | Qwell: and can you elaborate on "they are wrong"? |
| 18:58.53 | jolly23 | PBX in a flash is a take off from the days of asterisk at home. I use AAH long ago and it worked fine. |
| 18:58.59 | Qwell | he doesn't know what he's talking about, with any subject. |
| 18:59.35 | Qwell | vinhdizzo: part of this will probably be quite useful. http://www.davidvossel.com/?p=28 |
| 18:59.39 | Qwell | The_Boy_Wonder: ^ :D |
| 18:59.50 | The_Boy_Wonder | oh gosh |
| 19:00.02 | The_Boy_Wonder | :p |
| 19:00.02 | Qwell | also I just saw the statusmessage comment. |
| 19:00.05 | Qwell | that's awesome |
| 19:00.34 | [TK]D-Fender | ? |
| 19:10.59 | oDesk | i'm having problem running script using the dialplan |
| 19:11.01 | *** join/#asterisk aquarat (~cowalski@mail.aquarat.za.net) |
| 19:11.12 | MACscr | anyone recommend a low profile fxo card? Preferably a cost effective one. |
| 19:11.24 | oDesk | the scripts runs and execute when starting "asterisk -cvvvddd" |
| 19:12.10 | oDesk | but if started the normal way asterisk -f -U xxx -G xxxx |
| 19:13.25 | oDesk | MACscr: i'm using x100p and it's perfect for family use only problem is the CallerID |
| 19:14.18 | thehar | mahna mahna! |
| 19:14.36 | MACscr | oDesk: this is for a small office, but its really only going to be used if the sip trunks were out. Though i would still prefer it to be pretty decent |
| 19:15.04 | oDesk | MACscr: i've got it from here http://www.x100p.com/, don't expect the customer service to be responsive as they never replied to my emails but they sent my items very quickly |
| 19:15.39 | oDesk | MACscr: i've got the 2nd edition one |
| 19:16.01 | oDesk | MACscr: sorry special edition http://www.x100p.com/products/FXO.php |
| 19:17.09 | oDesk | MACscr: i'm doing near 4000+ minutes on it per month |
| 19:17.25 | MACscr | but no caller id huh? |
| 19:18.11 | *** join/#asterisk darkskiez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 19:18.30 | oDesk | MACscr: actually for my country yes.. but for your country you can find it works, even though you can buy CallerID converter from DTMF to FSK or whatever |
| 19:18.41 | MACscr | im in the US |
| 19:19.19 | oDesk | MACscr: other device to advice you which is really perfect the Grandstream HandyTone 503 and you can get it for 54 USD from telephonydepot |
| 19:19.43 | oDesk | MACscr: it has 1 FXO and 1 FXS and you can trunk it to your asterisk directly |
| 19:19.50 | [TK]D-Fender | MACscr: Linksys SPA-3102 |
| 19:19.55 | [TK]D-Fender | ~gs |
| 19:19.55 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:19.56 | [TK]D-Fender | ~grandstream |
| 19:19.57 | infobot | grandstream is probably the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 19:20.26 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 19:20.54 | MACscr | what sucks is that I had planned on spending $100 and getting the TDM410P, but because of the size, its not going to work. I have an inspiron 530S that im going to throw it into. |
| 19:21.03 | *** join/#asterisk iulhk (~adil@115.186.42.84) |
| 19:21.24 | timeshell_atwork | leifmadsen Ping FYI Pedantic does seem to be fixed somewhere between 1.6.0.128 and 1.6.2.14-rc1 |
| 19:21.45 | leifmadsen | timeshell_atwork: rock on |
| 19:21.53 | timeshell_atwork | Heh |
| 19:21.55 | timeshell_atwork | yah!! |
| 19:22.00 | timeshell_atwork | rocks on |
| 19:22.25 | timeshell_atwork | Now I have to do some extensive 1.6.2 testing in the gui. |
| 19:22.41 | [TK]D-Fender | 1.6.0.128 ? zOMG.... I should keep up more.... |
| 19:23.02 | MACscr | oDesk: man, those X100P seemed very well priced. $30? |
| 19:23.18 | [TK]D-Fender | ~x100 |
| 19:23.22 | [TK]D-Fender | ~x100p |
| 19:23.22 | infobot | x100p is probably an obsolete card. You don't want to bother trying to make it (or any of the "digium compatible" clones) work. Get a TDM01B, and you will save your sanity, your hair, and countless other things. |
| 19:24.14 | MACscr | well crap. Those are like 10 x the cost and arent low profile |
| 19:24.30 | Qwell | but they actually work. |
| 19:24.52 | MACscr | they wont work for me because of the size |
| 19:25.07 | MACscr | my budget is $100 for this card |
| 19:25.44 | [TK]D-Fender | [15:19]<[TK]D-Fender>MACscr: Linksys SPA-3102 |
| 19:26.00 | MACscr | yeah, i might have to consider that, but i was hoping for a pci card |
| 19:26.06 | oDesk | MACscr: it wooorks perfectly, i'm passing 4000+ minutes per month on it very good voice quality |
| 19:26.25 | [TK]D-Fender | MACscr: Sangoma A200 |
| 19:29.51 | oDesk | MACscr: the driver is wcfxo and it's known by DAHDI driver easily , if you don't care much about the caller id then you'll not have any problem, if this is your problem you could have Tone Converter from DTMF to FSk or FSk to DTMF easily |
| 19:32.25 | MACscr | [TK]D-Fender: thanks for the suggestion, a little bit closer on the price at around $160, but still a bit higher. oDesk, did you create your asterisk system from scratch or did you use a "package" like elastix or piaf? |
| 19:42.50 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 19:43.26 | zoid_ | Hi, I just bought an openvox A400 card, and trying to set up with dahdi_genconf, but I get a blank conffile, lspci sees the card |
| 19:43.30 | zoid_ | why can this be? |
| 19:45.59 | oDesk | MACscr: i've used trixbox, piaf and elastix, all works fine even dahdi updates all going smoothly |
| 19:46.35 | oDesk | MACscr: now i'm residing with Elastix, only for the backup/restore option they have which is perfect tool |
| 19:47.48 | *** join/#asterisk Jamuel (~jamuel@209.116.43.2) |
| 19:48.18 | MACscr | oDesk: thanks. Just ordered one. At $39 after shipping, its worth at least a try |
| 19:48.27 | Jamuel | Hi. Does anyone know if its possible to prevent multiple concurrent logins to an app_voicemail mailbox? |
| 19:48.40 | Jamuel | from within app_voicemail? |
| 19:49.28 | *** join/#asterisk dmast (~chatzilla@exchange.newpointe.org) |
| 19:50.13 | dmast | Is this the right place for asking 1.8 RC related questions? |
| 19:50.47 | leifmadsen | dmast: yep |
| 19:50.48 | Qwell | dmast: sure |
| 19:50.58 | oDesk | MACscr: great, you'll love it, actually i've TDM400 at work for heavy calls, but i can't differ the call quality from x100p the quality is the same, good luck |
| 19:51.40 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.73) |
| 19:52.09 | dmast | I've been experimenting with the unsolicited_mailbox option, and I'm not quite clear on how to properly use it when I've got more than one SIP user looking at that peer for MWI notifications. |
| 19:53.05 | vinhdizzo | Qwell: I compiled and installed asterisk per that davidvossel.com website you shared. How do I know it installed successfully? I dont see an "asterisk" command |
| 19:53.32 | Qwell | umm |
| 19:53.52 | Qwell | I do believe you're in over your head. |
| 19:53.57 | Qwell | ~book |
| 19:53.58 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:54.18 | dmast | For example, I've got an Exchange 2010 server set up as a peer, with "unsolicited_mailbox=1500" in the context. I've got two SIP users set up with "mailbox=1500@SIP_Remote" in their contexts. Yet both users receive MWI if only one user actually has a new voicemail message. |
| 19:54.27 | *** join/#asterisk George555 (~George555@188.24.123.199) |
| 19:54.32 | George555 | hello |
| 19:54.47 | vinhdizzo | ahh it's in /usr/sbin |
| 19:54.49 | George555 | I have a problem with a variable in asterisk |
| 19:54.54 | *** join/#asterisk breardo (~no@234-200-29-134.hcc.mnscu.edu) |
| 19:55.32 | George555 | i'm trying to use: |
| 19:55.34 | George555 | exten=_X! ,1,Monitor(wav,${CALLERID(name)}%d%t,m) |
| 19:55.47 | George555 | but the time and the won't appear on the filename |
| 19:55.49 | George555 | onli the name |
| 19:55.54 | George555 | only |
| 19:56.12 | George555 | :( |
| 19:57.10 | George555 | can someone please help me |
| 19:57.41 | zoid_ | nevermind, I got it |
| 19:59.09 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 19:59.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:59.43 | George555 | i tried with timestamp but still didn't worked |
| 20:01.57 | George555 | how can I gget the time and date variable in there / |
| 20:01.59 | George555 | ? |
| 20:03.41 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 20:05.56 | *** join/#asterisk ramyavenkat (~quassel@cpe-76-179-115-51.maine.res.rr.com) |
| 20:07.59 | *** join/#asterisk jolly23 (~haryv@154.5.144.198) |
| 20:11.52 | vinhdizzo | Qwell: you the owner of that davidvossel site? |
| 20:13.29 | nny | anyone ever seen a polycom crap the bed (hard reboot on a call or second call) if the OVERRIDE in phone.cfg is pointed at a possibly broken ringtone? |
| 20:14.10 | nny | have a client with one extension (tested multiple phones) that was doing just that, diffed the configs. Only difference outside of auth was the OVERRIDE line pointing at ring "16" |
| 20:14.13 | jolly23 | has the dialplan logic changed from 1.4 to 1.6? |
| 20:15.27 | [TK]D-Fender | jolly23: UPGRADE.TXT |
| 20:16.02 | jolly23 | k |
| 20:19.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:24.00 | *** join/#asterisk ramyavenkat (~quassel@cpe-76-179-115-51.maine.res.rr.com) |
| 20:24.21 | ramyavenkat | hi |
| 20:26.24 | ramyavenkat | I would like to know if asterisk can be installed on spa2102... instead of a separate host.. ? |
| 20:26.42 | [TK]D-Fender | ramyavenkat: No. |
| 20:26.59 | [TK]D-Fender | ramyavenkat: That device can't run a full OS and *. It is an ATA. |
| 20:27.18 | ramyavenkat | Can you let me know why? |
| 20:27.38 | [TK]D-Fender | ramyavenkat: Why can't a coffee-pot run *? Because it isn't a full computer. |
| 20:27.42 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 20:27.53 | ramyavenkat | As per google.. it runs linux.. |
| 20:28.12 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 20:28.58 | [TK]D-Fender | ramyavenkat: Where do you see that you can DO anything with that fact? |
| 20:29.12 | [TK]D-Fender | ramyavenkat: Where is the aceess to it? What about space to install anything? |
| 20:29.42 | [TK]D-Fender | ramyavenkat: Just because I give the keys to a car doesn't mean an elephant can fit in it |
| 20:30.09 | ramyavenkat | <[TK]D-Fender: Ok.. |
| 20:31.24 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 20:32.01 | kerx | hi all, i'm trying to hook up my computer's modem into a card so that i can do dialup through a sip provider. Any suggestions on what hardware I need and type of configuration? I'm kinda in the blue about it |
| 20:32.01 | ramyavenkat | <[TK]D-Fender I am not aware of the specs of the device.. |
| 20:32.42 | [TK]D-Fender | kerx: IP over Voice over IP? |
| 20:32.53 | [TK]D-Fender | kerx: RETARDED, and I know of nothing that will survive this |
| 20:32.55 | kerx | lol, i guess you can look @ it that way |
| 20:33.01 | kerx | well, think of this |
| 20:33.11 | [TK]D-Fender | kerI have... and it's dumb :p |
| 20:33.13 | kerx | is it possible to hook up a Fax machine into a card and do fax? |
| 20:33.26 | [TK]D-Fender | kerx: If' you're happy at 9600 baud |
| 20:33.36 | kerx | well, i'm doing this for a specific reason |
| 20:33.44 | kerx | obviously i'm not doing it for bandwidth :) |
| 20:33.55 | [TK]D-Fender | And lucky considering fax requires T.38 ... and there is no T.38 for "data" |
| 20:34.01 | MACscr | i need to search for sip providers that allow cid "spoofing" as i need to be able to forward calls to my cell phone and use the original callers info. What keywords should i be searching for? |
| 20:34.04 | kerx | well, i'm not doing fax |
| 20:34.17 | kerx | i'm basically trying to dial out using a modem |
| 20:34.23 | [TK]D-Fender | kerYou are missing the poitn. The only reason faxing over IP works is BECAUSE of it. |
| 20:34.23 | kerx | do i need a standard ATA card? |
| 20:34.25 | MACscr | im assuming there is a different term than spoofing that is used |
| 20:34.34 | [TK]D-Fender | kerx: And there is nothing to help "data" |
| 20:35.05 | [TK]D-Fender | And... it's checkout time, BBIAB |
| 20:35.10 | kerx | l8r |
| 20:35.13 | carrar | MACscr, you might look at toll avoidance :) |
| 20:35.49 | MACscr | that doesnt seem right |
| 20:36.49 | carrar | it's also called diversion |
| 20:37.03 | carrar | there's a header for that |
| 20:38.50 | *** join/#asterisk MiserySoft (~elende@92.41.120.176.sub.mbb.three.co.uk) |
| 20:42.59 | *** part/#asterisk MiserySoft (~elende@92.41.120.176.sub.mbb.three.co.uk) |
| 20:47.00 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 20:52.22 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:53.01 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 21:06.30 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 21:06.39 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:07.55 | titter | What would cause this many packets to be dropped during a call? Latency to the user seems normal (65ms) from Atlanta to Michigan. I could hear her normally, she could not hear me. http://pastebin.com/JdWYgnBD |
| 21:08.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:11.48 | The_Boy_Wonder | vinhdizzo: did you post a comment on my blog?! |
| 21:14.43 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:15.53 | *** join/#asterisk csnook (~chris@va-76-1-132-194.dhcp.embarqhsd.net) |
| 21:16.44 | Jamuel | from within app_voicemail is it possible to prevent multiple concurrent logins to the same mailbox? |
| 21:16.44 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |
| 21:20.55 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 21:23.23 | *** join/#asterisk silvestre_id (~silvestre@200-204-158-49.dsl.telesp.net.br) |
| 21:28.06 | *** join/#asterisk hfb (~hfb@pool-98-119-37-144.lsanca.dsl-w.verizon.net) |
| 21:35.30 | *** join/#asterisk fleixius (~fleixius@unaffiliated/fleixius) |
| 21:35.48 | fleixius | How do I check if my dahdi interface is receiving a call or not. Asterisk doesnt seem to handle the inbound call |
| 21:38.21 | tzafrir_laptop | fleixius, is it analog? isdn? r2? ss7? |
| 21:42.05 | fleixius | tzafrir_laptop: this is analog |
| 21:48.06 | fleixius | tzafrir_laptop: Any ideas? |
| 21:48.46 | tzafrir_laptop | for starters, core set verbose 3 |
| 21:48.55 | tzafrir_laptop | Do you see a channel created? |
| 21:52.48 | *** join/#asterisk mechbangirc (~mechbangi@mbl-82-58-63.dsl.net.pk) |
| 21:53.43 | mechbangirc | hi can anyone give me callcentric [voip provider] working configuration [peer] and register? |
| 21:54.10 | mechbangirc | i am behind nat and dont have static ip |
| 21:56.07 | *** join/#asterisk Marco2010_DE (~Marco@91-66-235-179-dynip.superkabel.de) |
| 22:02.06 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:02.11 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:04.15 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:14.12 | Letoric | Quick question, so I can screw with my boss real fast ;P |
| 22:14.27 | Letoric | how can I set the caller id from the console, so that when I originate a call from the console, it shows what I want |
| 22:14.28 | Letoric | ? |
| 22:14.53 | Letoric | going to make his cell phone call his desk phone |
| 22:18.44 | vinhdizzo | does anyone know whether it's possible to set up fax using asterisk with google voice or gizmo5? |
| 22:19.49 | Marco2010_DE | hey guys |
| 22:20.15 | Marco2010_DE | anyone runnig with an sangoma a500 bri here? |
| 22:20.20 | Marco2010_DE | running |
| 22:21.59 | *** join/#asterisk [cannibalera] (~cannibale@201-35-215-225.fnsce703.dsl.brasiltelecom.net.br) |
| 22:30.24 | *** join/#asterisk csnook (~chris@216.30.179.82) |
| 22:57.51 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 23:11.48 | *** join/#asterisk jsgoecke (~Adium@c-98-207-5-162.hsd1.ca.comcast.net) |
| 23:17.51 | *** join/#asterisk csnook (~chris@138.210.3.1) |
| 23:28.48 | *** join/#asterisk felipe_ (~felipe@my.nada.kth.se) |
| 23:32.42 | *** join/#asterisk [cannibalera] (~cannibale@201-35-215-225.fnsce703.dsl.brasiltelecom.net.br) |
| 23:54.22 | *** join/#asterisk oryxtec (~test@119.152.76.97) |
| 23:54.48 | oryxtec | guys this is 1st time im using asterisk 1.6 |
| 23:54.57 | oryxtec | but it seems to be asterisk 1.6 is not loading up agi script |