| 00:02.16 | e-Regg4e | Math` do you have any experience with linksys pap2? |
| 00:07.28 | Math` | ya |
| 00:07.38 | e-Regg4e | i have a basic question |
| 00:07.47 | aptura | voipjet does not return emails |
| 00:07.57 | e-Regg4e | aptura: after 5 days maybe |
| 00:08.02 | aptura | for me never |
| 00:08.02 | aptura | :) |
| 00:08.13 | e-Regg4e | Math`: i have configured the pap2 |
| 00:08.17 | e-Regg4e | but i want to do this: |
| 00:08.30 | e-Regg4e | i want to connect both lines to an analog pbx |
| 00:08.49 | e-Regg4e | so, if i dial the extension in the analog pbx |
| 00:08.55 | e-Regg4e | of the pap2 lines |
| 00:09.10 | e-Regg4e | i want to get a dial tone and i want to make the pap2 able to dial a voip extension |
| 00:09.12 | e-Regg4e | is that possible? |
| 00:09.20 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 00:10.21 | Math` | sure... |
| 00:10.41 | Math` | oh |
| 00:10.47 | Math` | wait no you cant |
| 00:10.58 | Math` | you need a device with an FXO port |
| 00:11.04 | e-Regg4e | damn! |
| 00:11.13 | Math` | the SPA-3000 has 1 FXO and 1 FXS |
| 00:11.16 | e-Regg4e | :'( |
| 00:11.22 | Math` | or else your PAP2 wont like the voltage :) |
| 00:11.51 | e-Regg4e | damn it :( |
| 00:13.54 | RypPn | afternoon all, I've successfully completed my testing using a x100p-clone and would like to move to production. can someone recommend a replacement with better audio quality for one fxo? |
| 00:14.22 | *** join/#asterisk tinrsh (n=claudiu@81.181.94.112) |
| 00:14.27 | tinrsh | 'nite |
| 00:15.21 | tinrsh | hi there, is there any way to change the permissions that asterisk uses when creating voicemail files or monitor files ? |
| 00:16.28 | tinrsh | anybody awake ? |
| 00:16.50 | angryuser | http://bugs.digium.com/view.php?id=9400#bugnotes have anyone has problem like this? (misdn b10P port go down) |
| 00:20.44 | *** join/#asterisk plasmid (n=noway@c-68-46-97-136.hsd1.pa.comcast.net) |
| 00:23.19 | JunK-Y | tinrsh: which permissions would u like to set? |
| 00:24.07 | tinrsh | those that asterisk uses when recording to voicemail ( the audio files ) |
| 00:24.24 | *** join/#asterisk _mm_ (n=mmclain@75.80.238.180) |
| 00:24.28 | tinrsh | or the monitor files |
| 00:24.35 | JunK-Y | i mean, which permissions, not which files. |
| 00:24.56 | tinrsh | I sorry, I don't understand the question |
| 00:24.58 | Hymie | tinrsh: you can chmod g+s the directories, and then files created in those directories will inherit the group id of the parent directory |
| 00:25.38 | Hymie | tinrsh: this is a long standing issue, I'm surprised there isn't an option in voicemail.conf yet |
| 00:26.23 | tinrsh | Hymie: yes, this is an option, but I would've preffered a conf file option |
| 00:26.25 | tinrsh | :( |
| 00:28.38 | *** join/#asterisk mvand (n=mvand@CPE-65-28-181-127.neb.res.rr.com) |
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| 00:30.55 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 00:31.29 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:32.19 | mvand | Is anyone using chan_mgcp with version 1.4.4? |
| 00:37.24 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:37.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:45.20 | *** join/#asterisk CVirus (n=GoD@196.205.192.216) |
| 00:46.34 | CVirus | I'm compiling zaptel ... In the zconfig.h .. what shall I set #define DEFAULT_TONE_ZONE to ? I live in Egypt |
| 00:47.55 | *** join/#asterisk bkw_ (i=brian@adsl-70-143-39-207.dsl.tul2ok.sbcglobal.net) |
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| 00:50.03 | RageMax | is there a hardened low-latency linux distro out there being developed specifically for asterisk? |
| 00:50.03 | carrar | zaptel/zonedata.c |
| 00:50.52 | carrar | probably 19? Israel? |
| 00:51.05 | CVirus | I don't think so |
| 00:51.15 | carrar | well look in there |
| 00:51.18 | carrar | pick one |
| 00:51.52 | carrar | http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf |
| 00:52.27 | carrar | Egypt is in that doc |
| 00:52.30 | carrar | you can just create one |
| 00:52.50 | carrar | or find one thats like it |
| 00:54.33 | Maxxed | hey any of you guys have any experince working with an answering service? |
| 00:54.34 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 00:55.06 | CVirus | carrar: Thanks |
| 00:55.34 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 00:55.34 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:59.00 | mvand | Is anyone using chan_mgcp with asterisk 1.4.4? I'm having a problem with an ATA that works with version 1.2 |
| 01:00.20 | *** join/#asterisk Caplain (n=shayne@adsl-75-45-231-77.dsl.sfldmi.sbcglobal.net) |
| 01:00.24 | CVirus | Do I have to recompile asterisk and libpri after recompiling zaptel ? |
| 01:00.36 | carrar | can't hurt |
| 01:00.36 | Caplain | where do i get a zaptel? |
| 01:01.17 | carrar | You mean a Zap type interface card? |
| 01:01.48 | Caplain | something that will let me use my land line as a channel |
| 01:01.51 | carrar | http://www.digium.com/en/products/ |
| 01:02.04 | carrar | http://www.digium.com/en/products/hardware/analogcards.php |
| 01:02.13 | carrar | TDM400P |
| 01:02.22 | carrar | with 1 FXO card should work |
| 01:03.26 | Caplain | <PROTECTED> |
| 01:03.33 | carrar | no |
| 01:03.38 | carrar | just a your telephone line |
| 01:03.48 | carrar | FXS modules are for the phone to plug into |
| 01:04.00 | carrar | FXO modules are for the telephone line to plug into from your provider |
| 01:04.36 | carrar | You would need 1 of each if you want to use a analog phone and analog phone service |
| 01:04.58 | carrar | using 2 out of the 4 slots on the TDM400P |
| 01:05.14 | Caplain | those are expensive |
| 01:05.28 | carrar | and they work great |
| 01:05.37 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 01:07.00 | mvand | Caplain: If you're just looking for something cheap to use for exploring asterisk, you might want to try an X100P |
| 01:07.13 | carrar | that will replace a FXO |
| 01:07.17 | aptura | carrar a short few years ago PBX were considerably more pricy. |
| 01:07.21 | carrar | but not for a FXS port |
| 01:07.34 | Caplain | okay |
| 01:07.40 | [TK]D-Fender | Caplain, Linksys SPA-3102 is a decent choice for a single line & phone, considerably cheaper |
| 01:08.43 | [TK]D-Fender | +/- 75$ USD |
| 01:09.19 | mvand | Is the SPA-3102 and FXO and an FXS? Like the -3000 was? |
| 01:10.10 | [TK]D-Fender | yes |
| 01:10.35 | [TK]D-Fender | mvand, Now includes a strong CPU, and can be a router as well. |
| 01:10.40 | Caplain | im ordering the Linksys SPA-3102 |
| 01:10.53 | Caplain | wtf everything is like a cheap commecrial |
| 01:11.14 | mvand | How is the PSTN echo on the 3102? I had problems with my -3000 (Mostly due to my (former) PSTN provider) |
| 01:11.16 | [TK]D-Fender | strong-ER |
| 01:11.56 | [TK]D-Fender | mvand, I'd guess a little better, but its hit or miss quite likely. Have you checked out the voxilla forums for guides on tuning it? |
| 01:13.00 | *** join/#asterisk bbryant (n=Brett@user-24-214-124-177.knology.net) |
| 01:13.46 | mvand | Voxilla is how I finally got acceptable results. I was able to tune my X100 a little better, but was never able to get CallerID to work. I even tuned the rxgain to 2 decimal places by calling the ,illiwatt line |
| 01:15.11 | mvand | analog phones connected directly to pstn got callerID 80-90% of the time, X100 got it 20% of the time, SPA3K got it 25% of the time |
| 01:16.02 | mvand | I ended up firing windstream (formerly alltel) and porting my 15 year old phone number to vitelity: all digital |
| 01:16.12 | killfill_ | hey guys.. asterisk is somehow adding a '0' before in all zap outgoing calls. i see in the logs i.e. Called g1/4201494 but its really calling 0420.... |
| 01:16.26 | killfill_ | where could this setup be?.. cannot find how to make it call normally.. |
| 01:17.02 | carrar | mvand, I had to buy a X100P from Digiam years ago to get callerID to work on a modem card |
| 01:17.34 | explidous | killfill_ sounds like a dialplan error... |
| 01:17.34 | carrar | Haven't used it since |
| 01:17.43 | carrar | PRI's are so much nicer |
| 01:18.07 | killfill_ | hm.. |
| 01:18.20 | explidous | carrar: Yes and some civilised countries even have ISDN in the Home ;-) |
| 01:18.29 | carrar | heh |
| 01:19.01 | Corydon76-home | ISDN to the home? Barbarians! Invade them! |
| 01:19.55 | explidous | Didn't we do that yet? |
| 01:20.41 | killfill_ | explidous: but when it says "Called xxxxx" the number is correct.. :S |
| 01:21.23 | explidous | sorry have to go dinner calling |
| 01:21.28 | *** join/#asterisk karlhaines (n=karl@unaffiliated/karlhaines) |
| 01:21.49 | explidous | killfill_ you might want to post your dialplan for people to look at.... |
| 01:22.12 | explidous | passwords removed ofcourse! |
| 01:22.18 | killfill_ | hehe |
| 01:23.49 | *** join/#asterisk Fieldy (i=60h7MHiR@gentoo/contributor/Fieldy) |
| 01:23.53 | mvand | carrar: I had my X100 working great in my previous residence, but had echo problems when i moved into my current home 3 years ago. I managed to tune it to an acceptable level, but when I added callerID 4 months ago, was never able to get it to work *reliably* on any device. The SPA-3K gave me similar quality to the X100 |
| 01:26.00 | mvand | The thing that's cool about the SPA3K (and I assume the 3102) is that you can address the FXO and FXS independantly, and if asterisk goes down, the phones dial out directly through the PSTN |
| 01:28.58 | *** join/#asterisk anthm][ (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:29.46 | killfill_ | explidous: this is my dialplan: http://www.sofsis.cl/extensions.conf |
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| 01:31.12 | killfill_ | any tips are welcome.. :P |
| 01:32.25 | killfill_ | somehow its prepending a 0.. al numbers get called with a 0 before... :S |
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| 01:38.09 | tinrsh | bye all |
| 01:38.11 | tinrsh | quit |
| 01:38.31 | Corydon76-home | killfill_: set pridialplan=unknown in zapata.conf and restart |
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| 01:40.31 | RypPn | mvand: interesting comment on the x100p, I'd been led to believe it was only useful for testing, and I should be looking for something better for production. |
| 01:43.10 | killfill_ | Corydon76-home: greate! thanks! |
| 01:45.10 | killfill_ | dont know what it means.. but worked.. :P |
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| 01:45.31 | mvand | RypPn: I understand that you are really risking reliability when using more than one X100 per box. If you're only using one, they seem to be quite reliable |
| 01:46.29 | mvand | (How many *production* PBXen need only one FXO?) |
| 01:46.29 | RypPn | mvand: I had more of a fight getting mine correctly configured for the UK, but I'm still not overly convinced by its audio quality |
| 01:47.14 | mvand | Isn't the UK impedance different than the US? |
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| 01:49.32 | RypPn | yeah, I changed it to CTR21 |
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| 01:52.49 | mvand | Where do you do that? I was only able to find rxgain and txgain settings |
| 01:53.07 | RypPn | I fed opermode into modprobe.conf |
| 01:53.11 | RypPn | u want a paste? |
| 01:54.07 | mvand | No, thanks. I got rid of the PSTN line last month. |
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| 01:57.09 | Feroxis | If I set a variable with Set() in an extension, will it still be available in h extension, or are they cleared before entering the h extension? |
| 02:00.00 | *** join/#asterisk noneo (n=ankamins@82-33-8-125.cable.ubr14.newt.blueyonder.co.uk) |
| 02:00.22 | mvand | Is anyone using MGCP with asterisk 1.4.4? I'm having a problem migrating from 1.2.15 |
| 02:01.19 | RypPn | mvand: I'm setting up a test at home, but would like to put it in the office where they are using 4 isdn channels, my thoughts are why do they need isdn at the cost? But analog doesnt sound an option from my testing so far |
| 02:03.23 | mvand | How many desk sets? |
| 02:03.44 | RypPn | mvand: (Just answering the production comment) |
| 02:03.51 | RypPn | umm, 7 or 8 |
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| 02:33.10 | Koshatul | I've been searcing google and the mailing list for a fix for this problem, but I keep getting either posts with "go post this in dev/users" (depending on where it was posted) |
| 02:33.29 | Koshatul | When I receive an inocming call on the Zaptel lines on my TDM400P with two FXS and two FXO |
| 02:33.32 | Koshatul | I get this |
| 02:33.34 | Koshatul | [May 20 12:37:43] WARNING[18486]: chan_zap.c:6879 ss_thread: CallerID returned with error on channel 'Zap/4-1' |
| 02:33.47 | Koshatul | and the line basically "hangs up" exiting non-zero |
| 02:33.54 | Koshatul | has anyone had this before ? |
| 02:34.19 | Koshatul | (i'm in australia, i've tried starting with a fresh zapata.conf, ztcfg -vv shows no errors, i've powered down and up a few times. |
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| 02:37.35 | fastfeet | Could someone point me in the proper direction to learn how to have Asterisk Display CID information when I receive a call from the PSTN line through my Linksys SPA-3102. |
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| 02:56.08 | rkeels | I should untar mpd into /lib/bin on ipod right? |
| 02:56.22 | rkeels | oops sorry |
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| 03:23.38 | killfill_ | how do i tell "if 1<${EXTEN}<60" ? ... |
| 03:28.19 | bkw_ | what? |
| 03:28.27 | bkw_ | can you clarify that? |
| 03:31.05 | killfill_ | i wish to check if the EXTEN number is bewteen 1 and 60, becouse if so, i wish to dial SIP/${EXTEN} |
| 03:31.24 | killfill_ | its for incomming zap calls |
| 03:33.15 | bkw_ | 01 tur 60 or 1 thru 60? |
| 03:33.50 | killfill_ | 01 thru 60 really.. |
| 03:33.51 | bkw_ | best way is to do _Z,1, for 1 thru 9.. then _[1-6]X for 10 thru 60 |
| 03:33.54 | bkw_ | ok |
| 03:33.59 | bkw_ | then _XX, |
| 03:35.13 | killfill_ | ah |
| 03:36.09 | killfill_ | bkw_: is there a way to know if a user uses SIP or IAX phones?.. i got muxed up users.. wish to redirect this calls to their sip/iax phones.. |
| 03:37.00 | bkw_ | go read the docs .. not 100% sure |
| 03:37.31 | killfill_ | any docs recomendation?.. |
| 03:37.32 | killfill_ | :) |
| 03:37.43 | bkw_ | what version? |
| 03:37.45 | bkw_ | 1.4 or 1.2? |
| 03:37.47 | killfill_ | 1.4 |
| 03:37.58 | bkw_ | don't they have chan_user? |
| 03:37.58 | mvand | what difference if they're calling in with sip or iax |
| 03:37.59 | killfill_ | i think im beginning to unserstand thing.. |
| 03:38.50 | killfill_ | mvand: oh no.. i mean.. i wish to redirect ZAP calls to their extensions. but dont know if i shall call them by SIP or IAX... :) |
| 03:40.35 | bkw_ | use SIP or IAX |
| 03:40.35 | bkw_ | dont' mix |
| 03:40.35 | killfill_ | already have mixes up.. SIP hw phones.. and IAX "remote" notebooks.. (out the nat) |
| 03:40.35 | mvand | dial(Local/901) |
| 03:40.36 | killfill_ | err local?.. |
| 03:40.36 | bkw_ | stay away from Local |
| 03:40.36 | killfill_ | local is like "sip or iax"?.. |
| 03:40.43 | bkw_ | local is evil |
| 03:41.11 | coppice | local isn't evil. people who use local are evil |
| 03:41.18 | bkw_ | haha |
| 03:41.45 | mvand | exten => 901,1,Dial(SIP/thatguy) |
| 03:42.16 | mvand | exten => 902,1,Dial(IAX/theotherguy) |
| 03:42.24 | killfill_ | wow.. i think its working.. :) |
| 03:42.35 | killfill_ | im just calling Local.. :) |
| 03:43.01 | killfill_ | oh, well right.. i could call IAX with priority 1, and then SIP.. so if iax doesnt work, try sip.. :P |
| 03:43.05 | mvand | exten => s,1,Dial(Local/901&Local/902) |
| 03:43.15 | bkw_ | killfill_, why on earth are you mixing IAX and SIP |
| 03:43.22 | bkw_ | users can use one or the other |
| 03:43.27 | bkw_ | you're making this harder on yourself |
| 03:43.36 | killfill_ | heh |
| 03:44.03 | killfill_ | well, some users use IAX becouse nat probls with sip, and some has hw phones that are sip..(i.e. secretaries) |
| 03:44.36 | mvand | I have to use both mgcp and sip because my DG-104s won't speak SIP, and my GXP-2000 won't speak mgcp. |
| 03:44.37 | bkw_ | um if they have nat problems then you have done something WRONG |
| 03:45.17 | bkw_ | nat is not a sip problem.. its an RTP problem brought on by dumb ass sip devices that don't do things the right way.. mixed with asterisk that does the NAT thing half assed |
| 03:45.19 | coppice | I understand the GXP-2000's point of view :-) |
| 03:45.24 | killfill_ | oh yah.. well, the net guys decided they dont want to route SIP inside.. |
| 03:45.43 | bkw_ | killfill_, then he should be fired |
| 03:46.08 | coppice | MGCP is a product of the clueless |
| 03:46.13 | killfill_ | thats true. and i think he has no much time left.. :P |
| 03:46.47 | bkw_ | coppice, I can agree with you on that |
| 03:46.47 | coppice | actually, most VoIP protocols are, but MGCP takes this to the extreme |
| 03:46.47 | *** join/#asterisk neuralwind (n=root@201.222.91.81) |
| 03:46.53 | bkw_ | see IAX suffers the same problem behind NAT that all voip protocols suffer from |
| 03:47.02 | coppice | The people who wrote the MGCP speak clearly had no clue where implementation complexity would lie |
| 03:47.09 | mvand | Yeah, Steve I know. But it's a really inexpensive way to get 4 cordless phones attached |
| 03:47.27 | neuralwind | how do u run asterisk? i wanna try my x100p card |
| 03:48.04 | killfill_ | ok.. i have another qeustion.. i have i.e. exten = _123 and exten = _456 .. both of thouse math the different numbers (123, and 456).. i wish to execut on both of them Set(Language()=es).. how could i do this? |
| 03:48.12 | killfill_ | math/match... |
| 03:48.20 | bkw_ | killfill_, well you do it like so |
| 03:48.28 | killfill_ | i mean.. withouth repeating it twice.. |
| 03:48.29 | coppice | mvand: why is it cheap? and MGCP box is only different from a SIP box due to the firmware loaded into it. they are *always* the same hardware |
| 03:48.48 | mvand | eBay! |
| 03:48.52 | bkw_ | _XXX,1,Set(Language()=es) then 123,2,Something and 456,2,Somethingelse |
| 03:49.03 | killfill_ | aah |
| 03:49.21 | coppice | mvand: and SIP firmware can't be downloaded from somewhere? |
| 03:49.23 | bkw_ | _XXX will match both... then the priority of two on each will pickup the rest |
| 03:49.35 | killfill_ | got it. bkw_ thank. |
| 03:49.36 | bkw_ | also don't prefix it with _ |
| 03:49.49 | bkw_ | unless you're doing matching with X,Z or N in the extension |
| 03:49.57 | killfill_ | oh right.. they are just numbers.. yup |
| 03:50.50 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 03:51.17 | kiwoneka | good eveing to all |
| 03:51.25 | killfill_ | greate. |
| 03:52.14 | kiwoneka | i need some quick help with my dial plan |
| 03:52.41 | kiwoneka | i am trying to cal dubai and i just cant get my internation to work prperly |
| 03:52.44 | kiwoneka | help |
| 03:52.59 | *** join/#asterisk bmg505 (n=leon@196.209.176.79) |
| 03:54.25 | mvand | I've not been able to find *any* firmware for a Clarent CPG 201 |
| 03:56.03 | coppice | yeah, the VoIP world is pretty sick. boxes you can't buy unless you are one of the chosen few. boxes riddled with bugs, where the only way to get updates is by being a service provider. There should be a special place in hell for this scum |
| 03:56.14 | kiwoneka | this is what i am trying to dial 0097150XXXXXXX |
| 03:56.43 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:57.51 | mvand | IIRC, the DG-104 has only mgcp and h323 firmware, with SIP additionally being available on the DG-102 |
| 03:58.55 | coppice | mvand: this will almost certainly be something that is only true in the minds of the marketing department |
| 03:59.26 | mvand | If I can get the Clarent box to work with my new asterisk 1.4.4 install (it works with 1.2.15), I will try upgrading my DG-102 to SIP |
| 04:01.04 | mvand | Oh, I agree with you. I have a two port version of the box: the DG-102S. If I recall correctly *that* box has available firmware for each of three protocols: sip, mgcp(yuck), and h323(YUCK) |
| 04:02.02 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 04:02.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:02.25 | coppice | SIP is at least as nasty as H.323, though MGCP is the clear champ for nastiness |
| 04:02.36 | fall0ut | MGCP isn't nasy |
| 04:02.41 | fall0ut | asterisk's MGCP is just nasty |
| 04:03.15 | coppice | MGCP is the most brain dead of all VoIP protocols |
| 04:03.27 | fall0ut | yes |
| 04:03.36 | fall0ut | moves all important functions into the GC or CA |
| 04:03.38 | fall0ut | whatever you wanna call it |
| 04:03.48 | fall0ut | but, that is good |
| 04:04.05 | coppice | it doesn't. that's why it is brain dead. maximum complexity. minimum reward |
| 04:04.25 | fall0ut | minimum reward? |
| 04:04.46 | fall0ut | far less interop shit to deal with |
| 04:04.59 | fall0ut | CA can determine functionality easier |
| 04:05.03 | fall0ut | so you're left with less interop |
| 04:05.12 | coppice | it was intended to permit simpler gateways than other protocols. it actually requires the same complexity of gateway, and really messy interactions |
| 04:05.38 | fall0ut | deployments with NCS/MGCP are a lot easier than SIP |
| 04:05.51 | mvand | And my Clarent ATA is even worse than the D-Link. It won't work with 1.4.4, but it did with 1.2.15. |
| 04:06.01 | fall0ut | although |
| 04:06.07 | fall0ut | I will admit roaming subscribers are better off with SIP |
| 04:06.08 | fall0ut | than MGCP |
| 04:06.13 | coppice | that is mostly because there are far fewer implementations of MGCP, so less variability |
| 04:06.30 | fall0ut | NCS/PacketCable1.0 is teh wins |
| 04:06.40 | fall0ut | newer packetcable specs are teh wins, too |
| 04:06.44 | fall0ut | but they are moving into IMS model |
| 04:07.16 | fall0ut | MGCP business line functionality is pretty badass actually |
| 04:07.24 | fall0ut | especially when comparing proper implementation to a SIP phone |
| 04:07.36 | fall0ut | even SIP-B and newer SIP implementations |
| 04:37.10 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 04:37.10 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.4.4 (April 27, 2007) Asterisk 1.2.18 (April 24, 2007), Zaptel 1.2.17.1, 1.4.2.1 (April 25, 2007) -=- Other fun channels: #asterisk-gui, #asterisknow, #asterisk-commits -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 04:43.43 | arafath | hello Corydon |
| 04:44.49 | arafath | can any one help me to configure 4E1 with one signal channel+libss7 |
| 04:44.56 | *** join/#asterisk Mercestes (n=Merceste@cpe-68-203-137-159.houston.res.rr.com) |
| 04:45.08 | Mercestes | If I wanted to do escalationin a queue with "ringall" how would I do it? |
| 04:45.40 | Mercestes | Say, I have 5 users, 2 on 0, 2 on 1, and 1 on 2. How do I ring 1,2 first, then 1,2,3,4 and finally, 1,2,3,4,5 using priorities in a queue with ringall? |
| 04:45.48 | explidous | arafath: I mmight... whats the problem? |
| 04:45.53 | Mercestes | no, the phones won't necessariliy be "busy" they could be unmanned or simply ignoring their phone. |
| 04:47.20 | explidous | oops sorry, didnt see the ss7... Where do you get ss7 from? |
| 04:47.30 | arafath | pls see the link http://pastebin.ca/497999 |
| 04:47.46 | arafath | from telco |
| 04:48.52 | arafath | they gave me 4 E1 and signal type is ss7, but they gave only one signal channel |
| 04:48.54 | explidous | why do they give you ss7 not ISDN/PRI???? |
| 04:49.13 | arafath | they have only R2 and SS7 |
| 04:49.23 | explidous | that works fine with PRI but I am not sure it works that well with ss7.... |
| 04:49.28 | explidous | what is the card? |
| 04:49.30 | arafath | and i m using Digium TE407 |
| 04:49.50 | coppice | arafath: SS7 normally works with a common signalling channel, although a redundant pair is preferred |
| 04:50.22 | arafath | at the given link i added my zaptel.conf and zapata.conf file |
| 04:51.03 | arafath | http://pastebin.ca/497999 |
| 04:51.15 | explidous | yes, but ss7 mis not the most used protocol for asterisk users ;-) |
| 04:51.40 | arafath | yap i know |
| 04:52.51 | arafath | the problem is that my 1st E1 works fine, but from 2nd to 4th E1 i only get ring but no voice from both side |
| 04:52.57 | coppice | you can always use R2 |
| 04:53.17 | arafath | but digium dont support R2 |
| 04:53.36 | coppice | most * users in south ameria use R2 |
| 04:54.25 | explidous | or use Sangoms driver and Cards... |
| 04:54.44 | coppice | what difference does using sangoma make? |
| 04:55.14 | neuralwind | i have x100p , already installed asterisk and zaptel .I started asterisk, how do i place a call now? |
| 04:55.32 | explidous | you can use theri ss7 lib |
| 04:55.32 | arafath | no diiference |
| 04:56.03 | coppice | sangoma does not have an SS7 lib. they use ss7box |
| 04:56.12 | arafath | yes i m using ss7lib |
| 04:57.42 | explidous | silly question why did you give each trunk its own group? |
| 04:59.32 | arafath | i configurd 1st E1 1-31 port and 2nd as 33-63 and so on |
| 04:59.54 | explidous | yes, but why did you use differnt groups for them? |
| 05:00.05 | coppice | you want them to be one group, don't you? |
| 05:01.03 | explidous | I never tried common signalling over differnt groups, not sure if that works... |
| 05:01.14 | Mercestes | nobody likes my question. :( |
| 05:01.15 | neuralwind | how do u place a call in asterisk? |
| 05:01.31 | Mercestes | neuralwind, with a phone. |
| 05:01.31 | explidous | That and your zone beeing US are the only things that jiump at me.. |
| 05:01.37 | Mercestes | neuralwind, or with a dial command |
| 05:01.53 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 05:02.00 | explidous | neuralwind: DIAL |
| 05:02.31 | neuralwind | im win CLI |
| 05:02.34 | neuralwind | with |
| 05:02.38 | neuralwind | theres no dial command |
| 05:02.48 | neuralwind | u mean i have to assing a dial plan first |
| 05:02.50 | explidous | http://www.voip-info.org/wiki-Asterisk+cmd+Dial |
| 05:03.23 | arafath | for routing my call |
| 05:03.25 | explidous | neuralwind, from the console? |
| 05:04.36 | explidous | arafath, did you try putting them in one group, leaving out the span directives as well see if you can use all four that way... just to confirm that that is not the problem... |
| 05:05.03 | arafath | yes i tried that |
| 05:05.18 | neuralwind | explidous yes from asterisk CLI i just want make a local phone call |
| 05:05.22 | explidous | arafath, and what does it do... |
| 05:05.35 | neuralwind | i dont see where u dial the number |
| 05:06.04 | explidous | neuralwind, so you have OSS or ALSA loaded and configured? |
| 05:06.06 | arafath | 1st E1 work fine but when i try to call using 2nd E1 i can recive only ring but no voice |
| 05:06.31 | neuralwind | explidous i dont know |
| 05:06.55 | explidous | neuralwind, how long are you using Astrerisk? |
| 05:07.09 | neuralwind | i never used |
| 05:07.14 | neuralwind | im trying to start now |
| 05:07.21 | neuralwind | and test it with my x100p |
| 05:07.34 | explidous | OK, you might want to do some reading first... |
| 05:08.18 | neuralwind | i think i have to edit some conf files first right |
| 05:08.50 | explidous | http://safari.oreilly.com/0596009623 its free for online reading and download |
| 05:09.06 | neuralwind | thanks |
| 05:09.09 | explidous | neuralwind, you have to edit A LOT |
| 05:09.33 | explidous | Or you can use something like Trixbox |
| 05:10.04 | neuralwind | i dont want to use all pbx features i just want to place a local call and see the call duration |
| 05:10.22 | explidous | A new version of the book is about to be released as well... |
| 05:10.53 | explidous | You have to do some configuring to make it do anything! |
| 05:11.15 | neuralwind | do u think it will be better to install asterisk gui? |
| 05:11.22 | explidous | Out of the box it is less versatile than a stone |
| 05:11.56 | explidous | asteriskNOW gives you a better start and even that requires some knowledge to operate... |
| 05:12.36 | neuralwind | couldnt install asterisknow i had too many problems |
| 05:12.53 | neuralwind | in any case in that book u pass me is everything i need? |
| 05:12.53 | explidous | neuralwind, what problems? |
| 05:13.24 | arafath | st E1 work fine but when i try to call using 2nd E1 i can recive only ring but no voice |
| 05:13.25 | arafath | neuralwind explidous i dont know |
| 05:13.33 | arafath | sorry |
| 05:14.49 | explidous | arafath, what is your telco, what country are you in? |
| 05:15.44 | *** join/#asterisk n3glv (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 05:15.45 | arafath | my telco using huwaie switch |
| 05:16.16 | n3glv | quick diagnose question |
| 05:16.20 | n3glv | <-- SIP read from 66.225.245.186:5060: |
| 05:16.20 | n3glv | INVITE sip:s@67.41.154.213 SIP/2.0 |
| 05:16.20 | n3glv | Via: SIP/2.0/UDP 66.225.245.186:5060;branch=z9hG4bK4470f638;rport |
| 05:16.27 | n3glv | is this broken at itsp or pbx? |
| 05:18.30 | explidous | I think you have to create a trunkgoup like for NFAS basically with the same settings as for PRI NFAS |
| 05:18.51 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 05:19.03 | explidous | n3glv, sorry not you ;-) |
| 05:19.26 | n3glv | np |
| 05:19.28 | n3glv | didn't look like it was to me.. ;-) |
| 05:19.33 | raphxl | quit |
| 05:19.34 | n3glv | I think we found it |
| 05:19.56 | n3glv | the provider req the /<did> on end of reg, he had 10 digits and they send 1+ |
| 05:20.02 | n3glv | (11 digits) |
| 05:20.02 | coppice | arafath: I don't know how mature that SS7 code is. if you get serious problems, most people in south america are happily using * with R2. |
| 05:20.12 | *** part/#asterisk n3glv (n=k3pc@c-71-60-125-243.hsd1.pa.comcast.net) |
| 05:21.37 | explidous | arafath, as I said as well this is not a heavy traveled path ;-) |
| 05:22.14 | arafath | thanx explidous |
| 05:23.14 | explidous | arafath, i am just reading thru chan_zap, you might try that as well |
| 05:23.30 | explidous | it looks like the NFAS is supposed to be the same |
| 05:24.07 | arafath | any link? |
| 05:25.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 05:25.50 | explidous | try /usr/src/asterisk ;-) |
| 05:27.41 | explidous | in my case /usr/src/asterisk-1.2.18/channels/chan_zap.c |
| 05:35.13 | arafath | what soft phone i can use for linux? |
| 05:36.17 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:40.15 | explidous | there is quite a selection... i like x-lite or twinkle |
| 05:41.05 | explidous | idefisk for IAX... but there are tons more... |
| 05:43.52 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 05:45.41 | arafath | thanx |
| 05:57.41 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:13.58 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 06:16.29 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 06:20.44 | *** join/#asterisk rue_mohr (n=rue@h24-207-90-24.cst.dccnet.com) |
| 06:29.17 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 06:35.59 | rue_mohr | ok help |
| 06:35.59 | rue_mohr | dpo stands for? |
| 06:37.52 | rue_mohr | oh cool |
| 06:37.57 | rue_mohr | http://www.electrodata.com/Acronym.htm |
| 06:38.13 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 06:41.48 | *** join/#asterisk J4k3 (n=jsuter@openwrt.us) |
| 06:42.12 | coppice | of course, they do make up half those acronyms on the spot :-) |
| 06:46.01 | rue_mohr | 48% of all statistics are too |
| 06:46.13 | rue_mohr | ok, I think I know the problem |
| 06:46.43 | rue_mohr | I can set the "function" of any the the set channels on the channelbank to any of.. |
| 06:46.57 | rue_mohr | ls_em, ls, gs, gs_em, plar, plar_fxo, gs_dnis, ls_dnis |
| 06:47.02 | rue_mohr | its an analog phone |
| 06:47.08 | rue_mohr | so iirc I want ls |
| 06:47.14 | rue_mohr | ? sound right? |
| 06:48.11 | rue_mohr | I currently have it set to LGS_DPO, and you cant hang up |
| 06:49.06 | coppice | ground start would not be a good thing for a plain old phone |
| 06:49.11 | rue_mohr | LGS = loop start subsciber = analog phone |
| 06:49.32 | rue_mohr | hmm |
| 06:49.32 | coppice | surely the G is ground |
| 06:49.41 | rue_mohr | yea, gs ground start |
| 06:49.45 | rue_mohr | ls = loop start |
| 06:49.58 | rue_mohr | ls_em = ????? |
| 06:50.09 | rue_mohr | EMElement Manager |
| 06:50.15 | rue_mohr | whatever that is |
| 06:50.23 | coppice | <PROTECTED> |
| 06:50.34 | rue_mohr | *blink* |
| 06:50.47 | coppice | E&M, not S&M :-\ |
| 06:50.48 | rue_mohr | which is... (I feel I should know this) |
| 06:51.13 | coppice | E&M is ye olde ancient signalling scheme |
| 06:51.14 | explidous | ear and mouth |
| 06:51.36 | coppice | it is often called such, but that's not really correct |
| 06:52.11 | explidous | none the less call that quite frequently... |
| 06:52.22 | rue_mohr | ;First 4 channels are the FXO modular card |
| 06:52.22 | rue_mohr | signalling = fxs_ls |
| 06:52.22 | rue_mohr | channel=>1-6 |
| 06:52.23 | rue_mohr | ok |
| 06:52.48 | rue_mohr | so I suppose I set the cahnnelbank to LGS_LS |
| 06:55.41 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 06:55.43 | rue_mohr | interesting |
| 06:55.45 | rue_mohr | <PROTECTED> |
| 06:55.45 | rue_mohr | <PROTECTED> |
| 06:55.45 | bluelinq | hello |
| 06:55.53 | rue_mohr | the status is different |
| 06:55.58 | bluelinq | anyone with 7960 sccp phones? |
| 06:56.06 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 06:59.29 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 07:00.16 | bluelinq | nobody with a cisco phone? |
| 07:03.43 | rue_mohr | not I |
| 07:03.49 | rue_mohr | I can only afford analog ones |
| 07:03.56 | bluelinq | :-0 |
| 07:08.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 07:10.26 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:21.26 | rue_mohr | now what the diff between ls and ls_em |
| 07:21.44 | *** join/#asterisk saftsack (n=oliver@p54a7d5b8.dip.t-dialin.net) |
| 07:22.00 | coppice | _em |
| 07:26.00 | rue_mohr | anyone have a channelbank of a T1 card they can take a look at something for me with? |
| 07:28.15 | rue_mohr | I'm sure if I have debug dialed up I should see digits pressed on a picked up phone |
| 07:33.40 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:42.22 | *** join/#asterisk saftsack (n=oliver@p54a7e346.dip.t-dialin.net) |
| 07:42.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:49.28 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 07:51.36 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 07:54.14 | *** join/#asterisk saftsack (n=oliver@p54A7D254.dip.t-dialin.net) |
| 07:59.13 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 08:12.49 | *** join/#asterisk saftsack (n=oliver@p54A7F48F.dip.t-dialin.net) |
| 08:15.55 | rue_mohr | ok, I have a handset to my ear that asterisk is giving mehte demo on |
| 08:16.06 | rue_mohr | zap show channel 1 |
| 08:16.11 | rue_mohr | Hookstate (FXS only): Onhook |
| 08:16.14 | rue_mohr | erm |
| 08:16.29 | rue_mohr | why is it talking to me if it thinks the phone is on the hook? |
| 08:35.04 | *** join/#asterisk LakeSolon (n=blake@64-83-205-22.dhcp.stcd.mn.charter.com) |
| 08:36.05 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 08:39.25 | *** join/#asterisk john-eman0n (n=eman0n@cpe-76-181-165-26.columbus.res.rr.com) |
| 08:55.01 | yonahw-work | i am using realtime with mysql driver and the information for dbuser and dbpass in res_mysql.conf is not what is being passed to mysql |
| 08:55.05 | yonahw-work | any ideas why? |
| 08:59.45 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:01.33 | yonahw-work | anyone awake? |
| 09:10.20 | *** join/#asterisk _omer (n=_omer@lhr-mp-dig-p11-74.brain.net.pk) |
| 09:10.20 | _omer | hello |
| 09:10.20 | yonahw-work | hi |
| 09:10.22 | _omer | does asterisk support SIP/SIMPLE for instant messaging ? |
| 09:10.36 | yonahw-work | sorry don't know |
| 09:10.50 | _omer | np :) |
| 09:11.14 | yonahw-work | do you have any experience with mysql realtime? |
| 09:11.29 | _omer | yes |
| 09:11.45 | _omer | u mean realtime asterisk with mysql ? |
| 09:11.56 | yonahw-work | yes, with the mysql driver |
| 09:12.10 | _omer | yes. |
| 09:12.32 | yonahw-work | it seems that res_config_mysql.c is not using my res_mysql.conf file to connect |
| 09:12.52 | yonahw-work | is the res_mysql.conf supposed to reside in /etc/asterisk? |
| 09:14.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:15.03 | _omer | you just need to config couple of files... |
| 09:15.04 | _omer | like |
| 09:16.34 | yonahw-work | when i look at the debug it is not connecting to mysql however the info it is using is not what is in res_mysql.conf |
| 09:17.57 | _omer | have you install asterisk-addons?> |
| 09:18.03 | yonahw-work | yes |
| 09:18.07 | _omer | and get res_mysql.conf |
| 09:18.14 | yonahw-work | got it |
| 09:18.35 | _omer | it shud be in /etc/asterisk |
| 09:18.40 | yonahw-work | it is |
| 09:18.56 | _omer | edit it to suit our config.. |
| 09:19.08 | _omer | dbhost, dbname, dbpass |
| 09:19.14 | _omer | your config* |
| 09:19.18 | yonahw-work | i did that but it seems that is not the info being used |
| 09:20.06 | yonahw-work | the connection fails and when i look in the debug it seems that the dbuser, dbpass and socket are all wrong |
| 09:21.19 | _omer | have you change extconfig.conf ? |
| 09:21.29 | yonahw-work | yes |
| 09:22.02 | yonahw-work | the connection to the database is failing so nothing else will work |
| 09:22.38 | _omer | have you allowed your IP Address in mysql ? |
| 09:22.47 | yonahw-work | its on the local machine |
| 09:22.59 | yonahw-work | asterisk can connect for the cdr |
| 09:23.24 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 09:24.02 | _omer | ok |
| 09:24.09 | _omer | what about the tables for realtime ? |
| 09:24.14 | _omer | created? |
| 09:24.50 | _omer | extconfig.conf should have lines something like |
| 09:24.52 | _omer | extensions => mysql,asterisk,extensions_table |
| 09:24.52 | _omer | sipusers => mysql,asterisk,sip_buddies |
| 09:24.52 | _omer | sippeers => mysql,asterisk,sip_buddies |
| 09:25.17 | *** join/#asterisk ploieel (n=ploieel@Fb2a2.f.ppp-pool.de) |
| 09:25.25 | yonahw-work | i got all that done but i dont see how any of that would matter if asterisk is failing to connect to mysql |
| 09:26.38 | yonahw-work | i get "MySQL RealTime: Cannot Connect (2002): Can't connect to local MySQL server through socket '/tmp/mysql.sock" and that isnt even the right socket path |
| 09:27.26 | yonahw-work | i have /var/lib/mysql/mysql.sock in res_mysql.conf |
| 09:28.00 | _omer | and asterisk is able to insert cdr in the same mysql server? |
| 09:28.20 | yonahw-work | yes same database different table |
| 09:31.53 | *** join/#asterisk coppice (n=chatzill@96.199.17.210.dyn.pacific.net.hk) |
| 09:33.07 | *** join/#asterisk newbie1 (n=Administ@202.53.167.31) |
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| 10:02.09 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
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| 10:10.35 | mvanbaak | mornin all |
| 10:11.11 | cnbrk | hi |
| 10:11.24 | mvanbaak | skinny doesn't support regcontext ? |
| 10:17.34 | _omer | does asterisk support SIP/SIMPLE for instant messaging ? |
| 10:25.05 | robin_sz | so, I have an * box ... |
| 10:25.05 | robin_sz | apart from the CLI is there some way to see what calls etc are in progress? |
| 10:25.06 | robin_sz | I looked at one thing once, umm, flash operators panel, but its wasnt really suitable |
| 10:31.35 | *** part/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
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| 10:58.18 | robin_sz | so, is there some way to reset the SIP 'inuse' counters apart from rebooting * ? |
| 10:58.29 | robin_sz | it seems abit crap to have to reboot it all the time |
| 11:05.51 | yonahw-work | can anybody help me with a asterisk real time problem. res_mysql.conf is apparently not being read properly since the debug file shows wrong settings being used |
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| 11:30.08 | omri | is it possible to compile a channel module without the asterisk source? I'm using debian's asterisk package and I hate to replace it... |
| 11:31.52 | pipwerk | no, but you could install the debian asterisk package source as well ;-) |
| 11:33.10 | omri | err I didn't have a deb-src line in my sources.list; that's why apt-get source asterisk didn't work :P |
| 11:33.21 | omri | I was thinking there's no source package for asterisk :P |
| 11:43.35 | *** join/#asterisk oej (n=olle@65-182-39-213.cre.bil.biltmorecommunications.net) |
| 11:45.09 | mvanbaak | 12:11 < mvanbaak> skinny doesn't support regcontext ? |
| 11:45.12 | mvanbaak | now it does :) |
| 11:48.37 | *** join/#asterisk saftsack (n=oliver@p54a7d237.dip.t-dialin.net) |
| 11:50.11 | robin_sz | so, is there some way to reset the SIP 'inuse' counters apart from rebooting * ? |
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| 13:32.31 | philippel | hey all - any thoughts on what would make an em_w T1 setup work fine for inbound, and call outbound but yet the outbound calls are never 'answered' (so dial hangs up after the ring time option expires)? |
| 13:36.12 | *** join/#asterisk Swat2 (n=bler@218-215-192-135.people.net.au) |
| 13:37.04 | Swat2 | how would one set inbound sip and zaptel calls CallerID Name to the CallerID Number (My Telco doesnt do this). |
| 13:38.11 | mvanbaak | exten => incoming,n,Set(CALLERID(name)=${CALLERID(num)}) |
| 13:39.28 | Swat2 | in extensions.conf, just under [from-zaptel] ? |
| 13:40.00 | mvanbaak | if incoming calls go there, yeah |
| 13:40.14 | mvanbaak | put it as a step in the normal handling of incoming calls |
| 13:40.27 | mvanbaak | replace the 'incoming' with the exten you are using |
| 13:41.11 | Swat2 | i want to do it globally for any zap or incomming sip calls |
| 13:49.11 | Swat2 | mmm, nope still not working |
| 13:49.20 | Swat2 | look at it harder tomorrow i spose |
| 13:49.26 | Swat2 | sleep time |
| 13:49.28 | Swat2 | cheers |
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| 13:54.05 | robin_sz | so, is there some way to reset the SIP 'inuse' counters apart from rebooting * ? |
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| 14:22.24 | Greek-Boy | does anyone know of carrier-grade IAX2 capable devices? |
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| 14:35.37 | hmepas | hi. I have such scheme: [HW PHONES]<private net 172.*.*.*>[172.*.*.1<- Asterisk ->Real IP]<internet>[Real IP<-Nat Server->Private IP]<another private net>[SOFT PHONES]. |
| 14:35.43 | hmepas | All phones could call each other. But I could hear voice only when calling from |
| 14:35.43 | hmepas | soft to soft phone or only when call from HW to HW phones. For soft phones I am |
| 14:35.43 | hmepas | using nat=yes in * settings. STUN not using. Any suggestion how to force work HW |
| 14:35.43 | hmepas | <PROTECTED> |
| 14:36.05 | hmepas | help me, safe maillist from another stupid throll =) |
| 14:38.56 | robin_sz | hmepas, you need to forward the rtp ports through your NAT router |
| 14:39.36 | robin_sz | the NAT is passing the SIP control channel, but not the RTP ports, thats all |
| 14:40.12 | robin_sz | easiest answer is to use IAX2 based softphones, as it all goes down the same channel |
| 14:46.44 | *** join/#asterisk Dovid (n=Dovid@bzq-88-155-99-120.red.bezeqint.net) |
| 14:47.02 | Dovid | Ping Tzafrir |
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| 14:51.00 | *** join/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com) |
| 14:51.08 | psi0n | hey all |
| 14:51.40 | psi0n | i have a small question, im guessing quite easy actually.. |
| 14:51.48 | hmepas | robin_sz: ty, could you tell me the name of softphone based on IAX2? Linux version prefered. |
| 14:52.34 | hmepas | robin_sz: and without rtp ports forwarding how both my softphones work each other? It's bcos them both in same network? |
| 14:53.04 | hmepas | robin_sz: bah, you mean all this 16384-32767 to forward? |
| 14:53.26 | Dovid | psi0n: whats the question ? |
| 14:53.51 | psi0n | i need the line "exten => s,n,SIPDtmfMode(inband)" to come after "exten => s,1,Set(FROM_DID=s)" under [ext-did] in extensions_additional.conf, but by using [ext-did-custom] in extensions_custom.conf |
| 14:54.07 | psi0n | i've tried the following: |
| 14:54.19 | psi0n | [ext-did-custom] |
| 14:54.20 | psi0n | exten => s,2,SIPDtmfMode(inband) |
| 14:54.20 | psi0n | ; end of [ext-did-custom] |
| 14:54.27 | psi0n | didnt work |
| 14:54.43 | Dovid | paste ur configs on p |
| 14:54.44 | Dovid | ~pb |
| 14:54.46 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 14:55.02 | psi0n | hmm ok, brb |
| 14:55.19 | robin_sz | hmepas, yes all those ports to forward for SIP |
| 14:55.34 | robin_sz | yes, they work on the same netwrok because the rtp does not go through asterisk |
| 14:56.08 | Dovid | psi0n: I have to run. robin_sz is real good at helping others ;) |
| 14:56.10 | Dovid | :P |
| 14:56.13 | robin_sz | I dont know of any working linux softphoens based on iax2 |
| 14:56.47 | hmepas | robin_sz: what's about windows iax2 based soft phones? |
| 14:57.02 | robin_sz | hmepas, firefly (closed source) |
| 14:57.38 | hmepas | it's need port forwarding too? (sorry i know it's stupid question) |
| 14:57.43 | robin_sz | http://iaxclient.sourceforge.net/iaxcomm/ |
| 14:57.46 | coppice | iaxcomm runs on windows and linux, and supports IAX2 |
| 14:57.56 | robin_sz | IAX2 does not need port forwarding |
| 14:58.39 | robin_sz | coppice, I never managed to get the audio working on linux, but yes, in theory, it shold work |
| 14:59.20 | robin_sz | so, is there some way to reset the SIP 'inuse' counters apart from rebooting * ? |
| 14:59.27 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 14:59.35 | coppice | I use it sometimes. it has its limitations, but I never had an audio problem |
| 15:02.48 | psi0n | ok im back |
| 15:03.15 | psi0n | http://pastebin.ca/498563 |
| 15:03.51 | psi0n | robin_sz: perhaps you could take a look at it if you don't mind? |
| 15:04.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-24-162-48-94.houston.res.rr.com) |
| 15:06.36 | robin_sz | using AMP? |
| 15:07.23 | psi0n | i am using AMP yes (freePBX), but making the changes in question directly in the conf files |
| 15:07.39 | robin_sz | better go and ask on #freepbx |
| 15:08.07 | psi0n | oh.. i thought this would be more of an asterisk question |
| 15:08.12 | coppice | i'm using AMP in conjunction with VOLT on this machine |
| 15:08.32 | robin_sz | I know nothing of AMP and its weird configs that it writes |
| 15:09.03 | psi0n | ok, np. i'll check there.. |
| 15:09.03 | robin_sz | what you cold do is ... |
| 15:09.12 | psi0n | ? |
| 15:09.25 | robin_sz | edit the conf the way you want and then not use AMP again, so it wont overwrite it? |
| 15:09.27 | mvanbaak | remove freepbx |
| 15:09.28 | mvanbaak | ;) |
| 15:09.31 | psi0n | lol |
| 15:10.53 | psi0n | robin_sz: yea, that is an option, or at least insert the line every time i make a change thru AMP, but i'd rather try doing it the proper way if possible |
| 15:10.54 | *** join/#asterisk ta^3 (n=tacvbo@189.136.34.99) |
| 15:11.21 | robin_sz | errr, editing it by hand and not using amp IS the proper way of doingit |
| 15:11.28 | psi0n | hehe |
| 15:11.30 | robin_sz | its basically an AMP problem |
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| 15:59.10 | jangell | I'm using Asterisk and have a T1 with E&M Wink. Asterisk is not recognizing that calls are being answered on outbound calls and hangs up after the 300 second timeout. Any idea? |
| 16:02.52 | ManxPower | jangell: Chances are the telco's wink settings are different from Asterisk's default ones. |
| 16:03.03 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-90-37.w86-206.abo.wanadoo.fr) |
| 16:03.54 | ManxPower | jangell: Try putting in: |
| 16:03.55 | ManxPower | wink=270 |
| 16:03.56 | ManxPower | rxwink=270 |
| 16:04.06 | ManxPower | before the channel lines in /etc/asterisk/zapata.conf |
| 16:04.25 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:04.33 | Strom_M | if you wink at the telco too much, the telco might think you're asking it out on a date |
| 16:07.48 | ixx | how can you make asterisk ignore pressing '#' when calling into some remote automated phone system that is requiring you to press '#' |
| 16:08.05 | ixx | ... local outbound asterisk ignoring it ... |
| 16:08.56 | Strom_M | ixx: turn off the crap in features.conf |
| 16:09.10 | Strom_M | ad then don't pass "t" or "T" as an argument to "Dial" |
| 16:09.45 | Strom_M | s/ad/and/ |
| 16:12.24 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 16:13.06 | kombi | my very first hardware ip phone, do I need dhcp for asterisk to find it? |
| 16:13.49 | ixx | <PROTECTED> |
| 16:14.22 | ixx | I would like to have T sometimes |
| 16:14.28 | ixx | but for now I will leave it off |
| 16:14.44 | Strom_M | ixx: there are better ways of doing transfers |
| 16:14.50 | ixx | yeh? |
| 16:14.53 | Strom_M | hookflashes, the TRANSFER button, etc |
| 16:15.24 | Qwell | dead hookflashes? |
| 16:15.32 | Strom_M | hahahhahahahhahaahhaha |
| 16:15.33 | ixx | ah... i have done the hookflash... have not thought about that for a while |
| 16:15.34 | Strom_M | <3 Qwell |
| 16:16.28 | kombi | will a cisco 7941 work right away or does it need to be tweaked first? |
| 16:17.03 | mvanbaak | kombi: running SIP or Skinny ? |
| 16:19.39 | jangell | ManxPower, I figured it out... adding callprogress=yes worked.....I thought callprogress was only for analog lines? |
| 16:20.11 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-64.plaicmtc01.tx.dh.suddenlink.net) |
| 16:20.36 | ManxPower | jangell: callprogress will make your calls disconnect randomly. You have to fix the basic problem. Call progress only MASKS the real problem |
| 16:20.54 | jangell | ManxPower, ok. i'll try those.. |
| 16:21.11 | ManxPower | jangell: callprogress is for an zap channel type that does not have answer and disconnect supervision |
| 16:22.35 | hmepas | robin_sz: thanks man, you my hero. DNAT didn't helped but IAX2 worked great, now i could fully test my phones. Thanks a lot. |
| 16:22.47 | jangell | ManxPower, 270 doesnt work |
| 16:23.15 | robin_sz | hmepas, no problem, that will be 15 euro, standard fee. |
| 16:23.28 | robin_sz | so, is there some way to reset the SIP 'inuse' counters apart from rebooting * ? |
| 16:25.13 | *** join/#asterisk Mad||Cow (n=madcow@74.94.5.97) |
| 16:25.16 | Greek-Boy | anyone know what kind of voicemail servers the big mobile telcos use? |
| 16:25.40 | Strom_M | Greek-Boy: usually big expensive telco-grade voicemail stuff |
| 16:26.20 | Greek-Boy | yeah so where can I find that kinda stuff? |
| 16:26.31 | Strom_M | why? |
| 16:27.25 | Greek-Boy | coz I'm compiling a biz plan for a wimax mobile operator opportunity |
| 16:27.37 | rue_mohr | I'm having a signaling problem with my channelbank and would like to know more about how they work, any references anyone can suggest? |
| 16:27.48 | Mad||Cow | Anyone ever have any issues with POTS lines being very quite? I have my RXgain up all the way to compensate, but it sounds horrible. Wondering if I have something miss configured that is causing this. |
| 16:28.09 | Strom_M | rue_mohr: what kind of signaling problem? |
| 16:28.17 | Strom_M | Mad||Cow: don't you mean "quiet"? |
| 16:28.40 | Mad||Cow | strom_m: yeah... sorry ;-) |
| 16:28.45 | rue_mohr | Strom_M, mainly that the best I'v been able to do, is to have asterisk think the phone is off the hook when its on, and on the hook when its off |
| 16:29.04 | Strom_M | Mad||Cow: you should never have to set rxgain higher than about 5 or 6 |
| 16:29.09 | rue_mohr | I been scouring google for d channel protocol info and didn't really find squat |
| 16:29.22 | Strom_M | rue_mohr: uh, channel banks != ISDN |
| 16:29.31 | Mad||Cow | Strom_M: I have mine set at 8.... and I still have to turn my headsetups up all the way |
| 16:29.38 | ManxPower | rue_mohr: In the USA the D-Channel uses the Q.931 protocol. |
| 16:29.44 | rue_mohr | mainstreet 3624 |
| 16:29.55 | ManxPower | EuroISDN might use a varient of that,I would think |
| 16:30.20 | rue_mohr | my buddy who gave it to me pioneered the asterisk work so far on it, but didn't finish |
| 16:30.50 | Strom_M | rue_mohr: how many analog ports does it have? |
| 16:30.53 | rue_mohr | Strom_M, sorry if i'm mixed up, I'm new to this :) |
| 16:31.01 | ManxPower | rue_mohr: And no, channel banks use channelized T-1 (CAS, in the USA) and not ISTN |
| 16:31.05 | rue_mohr | I have 6 FXS and 2 FXO |
| 16:31.22 | ManxPower | rue_mohr: what signalling is the channel bank set for? |
| 16:31.35 | rue_mohr | loop start |
| 16:31.53 | Strom_M | what kind of framing and line coding is it supposed to be using? |
| 16:32.04 | rue_mohr | bz... |
| 16:32.16 | ManxPower | rue_mohr: then you need to set Asterisk for loopstart, It would be fxo kewlstart signalling for the FXS ports and fxs kewlstart for the fxo ports. |
| 16:32.31 | Strom_M | ManxPower: are you high or something? |
| 16:32.31 | ManxPower | remember fxo ports use fxs signalling and fxs ports use fxo signalling |
| 16:32.37 | Strom_M | kewlstart != loopstart |
| 16:32.38 | rue_mohr | yup |
| 16:32.44 | jangell | ManxPower, If I call up XO and ask them how many ms for the wink are they gonna have any clue what I'm talking about? |
| 16:32.45 | ManxPower | Strom_M: just waking up. |
| 16:32.45 | rue_mohr | well |
| 16:32.52 | kombi | mvanbaak: sorry, the phone rang, skinny up to now |
| 16:33.21 | ManxPower | Strom_M: no, but most loopstart devices provide far end disconnect supervision and so they would be kewlstart |
| 16:33.21 | rue_mohr | signalling = fxs_ls |
| 16:33.21 | rue_mohr | channel=>1-6 |
| 16:33.21 | mvanbaak | kombi: my 7960 and 7905 work fine with skinny |
| 16:33.27 | mvanbaak | but you need asterisk-svn for it to work good |
| 16:33.31 | mvanbaak | I think |
| 16:33.37 | mvanbaak | dont know the skinny state in 1.4 |
| 16:33.38 | ManxPower | rue_mohr: ports 1-6 are FXO ports? |
| 16:33.56 | mvanbaak | in 1.2 you can use skinny too, but you will need chan_sccp for that |
| 16:33.58 | rue_mohr | no |
| 16:34.03 | mvanbaak | and chan_sccp is not very good |
| 16:34.05 | rue_mohr | wait a sec |
| 16:34.16 | rue_mohr | there are analog phones on ports 1-6 |
| 16:34.25 | rue_mohr | you saying I have the setting backwards? |
| 16:34.31 | ManxPower | rue_then the signalling must be fxo_ls |
| 16:34.32 | kombi | mvanbaak: hmm, what version might the debian package be.. |
| 16:34.42 | kombi | just checking.. |
| 16:34.45 | rue_mohr | that might explain the backwards operation |
| 16:34.47 | ManxPower | (11:32:00) ManxPower: remember fxo ports use fxs signalling and fxs ports use fxo signalling |
| 16:34.48 | mvanbaak | debian packages are 1.2 |
| 16:34.53 | Mad||Cow | If I have a phone line plugged into my pbx (that I am using to place and receive calls), am I using fxs or fxo signaling? |
| 16:35.02 | kombi | mvanbaak: sigh.. |
| 16:35.03 | ManxPower | (11:32:00) ManxPower: remember fxo ports use fxs signalling and fxs ports use fxo signalling |
| 16:35.07 | rue_mohr | ok, I'm go change that up and try it out, thanks |
| 16:35.16 | ManxPower | am I on channel wide /ignore? |
| 16:35.20 | kombi | mvanbaak: I should compile my own then.. |
| 16:35.34 | Strom_M | welcome to #asterisk, land of the balls and home of the cocks |
| 16:35.42 | mvanbaak | yeah. or put the SIP image on the phone |
| 16:35.49 | ManxPower | Strom_M: sounds like torture to me |
| 16:36.10 | rue_mohr | ManxPower, sorry, I did hear you, it just mixed me up is all |
| 16:36.24 | rue_mohr | "a man hears what he wants to" |
| 16:36.30 | ManxPower | rue_mohr: I was referring to Mad||Cow |
| 16:37.14 | kombi | mvanbaak: I wonder what might be more hassle, compiling asterisk or flashing the cisco phones with sip |
| 16:37.38 | ManxPower | kombi: flashing the phone is much more work |
| 16:37.45 | Mad||Cow | ManxPower: sorry... I'm kind of new to this... I just dont understand the difference between fxo and fxs. |
| 16:38.09 | kombi | ManxPower: I am just reading it.. tftp and all.. |
| 16:38.11 | rue_mohr | here is a question I'd like to throw out there if anyone knows but isn't critical |
| 16:38.15 | ManxPower | ~fxofxs |
| 16:38.17 | jbot | extra, extra, read all about it, fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 16:39.00 | kombi | ManxPower, mvanbaak: so, apt-get remove --purge asterisk and start over..;) |
| 16:39.03 | rue_mohr | here is a question I'd like to throw out there if anyone knows, the 'function of a card can be set to ls_em, ls, gs, gs_em, plar, plar_fxo, gs_dnis, ls_dnis, or dpo |
| 16:39.12 | Greek-Boy | is there a way to turn off the "comedian mail" intro when going to mailbox? |
| 16:39.12 | rue_mohr | whats with plar and dpo? |
| 16:39.22 | Strom_M | private line automatic ringdown |
| 16:39.41 | rue_mohr | pickit up and it makes the call automatically? |
| 16:39.43 | ManxPower | rue_mohr: PLAR == Batphone |
| 16:39.43 | Strom_M | Greek-Boy: yes, specify the mailbox number |
| 16:40.00 | Greek-Boy | ok |
| 16:40.16 | robin_sz | so, whenever I do a blind transfer, the transferrign phone is left with an in-use figure, even though it no longer has any calls |
| 16:40.17 | Mad||Cow | ManxPower: hrm. So if I have a bunch of PSTN lines plugged into my phone system, in the zapta.conf defining them as fxsks=1 would be incorrect? |
| 16:40.38 | robin_sz | result, I do a txfr, and then I can;t rx anymore calls |
| 16:40.40 | ManxPower | Mad||Cow: that would be correct, but it only specifies port 1 |
| 16:40.46 | robin_sz | sip show inuse confirms this |
| 16:40.53 | Mad||Cow | ManxPower: understood |
| 16:41.00 | ManxPower | robin_sz: using calllimit ? |
| 16:41.04 | Strom_M | Mad||Cow: ManxPower> (11:32:00) ManxPower: remember fxo ports use fxs signalling and fxs ports use fxo signalling |
| 16:41.06 | robin_sz | err? |
| 16:41.10 | robin_sz | am I? |
| 16:41.13 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 16:41.28 | robin_sz | I have a call limit of 1, which is correct |
| 16:41.28 | ManxPower | robin_sz: The answer would be yes or no |
| 16:41.42 | Strom_M | the answer is balls |
| 16:41.45 | ManxPower | robin_sz: call limit interactly badly with just about everything in asterisk |
| 16:41.53 | robin_sz | riiiight ... |
| 16:42.00 | robin_sz | so ... what IS th answer then? |
| 16:42.14 | ManxPower | transfers don't decriment the call limit, vfor example. |
| 16:42.22 | ManxPower | I'm never ever needed to use call limit |
| 16:42.24 | robin_sz | this I have noted |
| 16:42.36 | robin_sz | OK, so ... how do I not use call limit? |
| 16:42.58 | ManxPower | robin_sz: There are a zillion ways to not use call limit. |
| 16:43.38 | kombi | mvanbaak: just while you're still around, do you just plug the 7941 in and off it goes? |
| 16:43.42 | ManxPower | We turn off call waiting on our phones and have each line register as a different SIP ID. We do the call apprearance aka line hunting using Asterisk and DIALSTATUS or HANGUPCAUSE |
| 16:43.45 | rue_mohr | May 20 03:07:42 ERROR[2664] chan_zap.c: Signalling requested on channel 1 is FXO Loopstart but line is in FXS Loopstart signalling |
| 16:43.47 | rue_mohr | :/ |
| 16:43.47 | robin_sz | ManxPower, well, my desired behaviour is that if I am on the phone, I dont want it trying to ring, it shoudl do "the person you are calling is not available" |
| 16:44.01 | rue_mohr | so it was right |
| 16:44.08 | Strom_M | rue_mohr: NO |
| 16:44.09 | ManxPower | robin_sz: turn off call waiting then |
| 16:44.19 | rue_mohr | so somethign lse is wrong? |
| 16:44.32 | Strom_M | rue_mohr: you have to make the zaptel,conf and zapata.conf files agree |
| 16:44.32 | ManxPower | rue_mohr: you have to change it in /etc/zaptel.conf too and rerun ztcfg -vvv |
| 16:44.35 | robin_sz | so, no call limit, no call waiitng, right? |
| 16:44.41 | ManxPower | robin_sz: correct |
| 16:44.46 | robin_sz | 'k |
| 16:44.56 | rue_mohr | I changed zapata.conf |
| 16:44.58 | ManxPower | robin_sz: and make sure your phone registeres each line as a different SIP ID. |
| 16:45.06 | robin_sz | que? |
| 16:45.14 | ManxPower | rue_mohr: there are TWO files. /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 16:45.23 | ManxPower | robin_sz: how many lines does your phone have? |
| 16:45.45 | rue_mohr | 1 right now |
| 16:45.51 | robin_sz | 1 I think |
| 16:45.59 | Strom_M | today must be Super Stupid Sunday in #asterisk |
| 16:46.02 | rue_mohr | got a second on its way in a few months |
| 16:46.09 | robin_sz | well, it has 9 I think, but I use 1 |
| 16:46.14 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/daviey) |
| 16:46.19 | rue_mohr | hah |
| 16:46.21 | rue_mohr | sorry |
| 16:46.36 | Daviey | Hi, is zaptel-dummy still required for meet-me? |
| 16:46.48 | ManxPower | robin_sz: many phones will register all lines as ONE SIP account. If that is the case, for most phones the call will just roll over to the next available line. |
| 16:47.00 | Strom_M | Daviey: yes, unless you have a zaptel card |
| 16:47.02 | Daviey | And that still requires zaptel kernel patch? |
| 16:47.11 | ManxPower | if the phone registers each line as a DIFFERENT SIP account then you do not have the problem |
| 16:47.15 | Strom_M | kernel patch? |
| 16:47.19 | Strom_M | wtf? |
| 16:47.32 | robin_sz | ManxPower, ahh, when I didnt have call_limit, I did notice that line 2 rang when another call was incoming |
| 16:47.35 | Daviey | hmm - i thought i had to patch the kernel last time with the zaptel driver |
| 16:47.40 | ManxPower | Daviey: The linux kernel has not had to be patched for Zaptel in many many years. |
| 16:47.41 | Strom_M | it's a module you load in. it's not a patch |
| 16:47.58 | Strom_M | jesus, it's Super Stupid Sunday in #asterisk - Party like it's 2002 |
| 16:48.00 | ManxPower | robin_sz: That would be expected since you are not setting up a seperate SIP account for each line. |
| 16:48.11 | Daviey | thank you... |
| 16:48.13 | robin_sz | right |
| 16:48.42 | rue_mohr | heh, I need 5 more analog phones for testing :) |
| 16:48.55 | rue_mohr | BIIIG money there ;) |
| 16:48.58 | ManxPower | robin_sz: you have VERY little control over how the call appears on the phone unless you have each line on the phone register as a different SIP user to Asterisk. |
| 16:49.16 | robin_sz | uhh |
| 16:49.19 | robin_sz | how horrid |
| 16:49.21 | ManxPower | We just use the MAC address of the device as the sip.conf userid and add a -a -b -c etc to indicate the line appearance |
| 16:49.41 | robin_sz | hmmm |
| 16:49.41 | rue_mohr | OOooo |
| 16:49.48 | rue_mohr | says its on the hook, AND IT IS! |
| 16:49.57 | Strom_M | instant cock |
| 16:49.58 | rue_mohr | sweet! |
| 16:49.59 | Strom_M | just add water |
| 16:50.05 | rue_mohr | I pick it up and it talks to me |
| 16:50.05 | robin_sz | ManxPower, that seems to make snese |
| 16:50.18 | rue_mohr | sweet, it knows when I hang up! |
| 16:50.32 | rue_mohr | my guru versaw that one |
| 16:50.34 | ManxPower | robin_sz: It is critically important that you never thing and extension = device |
| 16:50.40 | ManxPower | thing == think |
| 16:50.40 | rue_mohr | I wonder how he missed that |
| 16:50.57 | ManxPower | rue_mohr: I've been using Asterisk for at least 4 years |
| 16:51.04 | rue_mohr | I must speak with the great kb1_kanobe about this |
| 16:51.11 | *** join/#asterisk danw_home (n=dan@94cms.gotadsl.co.uk) |
| 16:51.13 | rue_mohr | yea... |
| 16:51.24 | rue_mohr | he thought the state machine needed to be moded |
| 16:51.31 | rue_mohr | :/ |
| 16:51.37 | rue_mohr | this is sweet |
| 16:51.47 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 16:51.56 | Strom_M | rue_mohr: obviously he's an idiot |
| 16:52.01 | DrukenLPY | afternoon peoples |
| 16:52.02 | ManxPower | programmers always think the fix requires new code. |
| 16:52.08 | robin_sz | ManxPower, while you are on ;) ... I subscribed several blf buttons to other peers, they do seem to light up when the extension is ringing, and if the phone answers an incoming call |
| 16:52.19 | robin_sz | but they dont show if the user makes an outgoing call |
| 16:52.23 | rue_mohr | I promise all my following questions will be related to extensions programming and up :) |
| 16:52.24 | DrukenLPY | anyone here ever used a wrt54g as a server? |
| 16:52.34 | ManxPower | Strom_M: kb1_kenobie wrote one of the Zaptel ECs and also did the tellabs wiki page |
| 16:52.53 | Strom_M | ManxPower: /me puts foot firmly in mouth |
| 16:52.55 | mvanbaak | kombi: sorry, vim took all my attention |
| 16:53.13 | mvanbaak | kombi: I had to setup a tftp server and put some xml configs there to make the 79XX working |
| 16:53.42 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 16:53.44 | kombi | vi rules.., thanks! |
| 16:53.50 | ManxPower | Strom_M: on the other hand I sent him an e-mail titles "Help me kb1_kenobie, you're my only hope" and he never responded to it. |
| 16:54.02 | kombi | mvanbaak: with skinny that is? |
| 16:54.04 | rue_mohr | AHA! and the channelbank stopped flashing 4!!!! |
| 16:54.08 | mvanbaak | kombi: yeah |
| 16:54.12 | ManxPower | Maybe I should have signed it "Princess Leia" |
| 16:54.15 | rue_mohr | its flashing 0 now! I like that! |
| 16:54.18 | kombi | thanks! |
| 16:54.21 | danw_home | I'm sure this is user error, forgive me if it is documented somewhere but I've been trawling for a while with no luck: I'm using Waitexten to wait for a short while for digits from an inbound call on a SIP channel. I can see all four digits arriving in INFO messages from my gateway but for some reason asterisk only registers the first digit ... any ideas why ? |
| 16:54.51 | Strom_M | danw_home: do you have a suitable pattern match in that context? |
| 16:54.53 | ManxPower | danw_home: is asterisk set up for INFO? |
| 16:54.54 | danw_home | the console message is __ast_pbx_run: Invalid extension '1', but no rule 'i' in context 'conference' |
| 16:55.05 | danw_home | where I've actually sent "1234" |
| 16:55.13 | *** join/#asterisk coil (i=scott@24.96.135.212) |
| 16:55.17 | danw_home | yeah, I've set "relax" in the sip.conf |
| 16:55.24 | Strom_M | danw_home: theres' nothing to match "1234" in the context |
| 16:55.25 | Greek-Boy | can asterisk act as an sms centre? |
| 16:55.27 | ManxPower | danw_home: all that does is ause problem |
| 16:55.32 | danw_home | yes, I've tried _XXXX and an explicit 1234 |
| 16:55.45 | ManxPower | dtmfmode=INFO |
| 16:55.53 | ManxPower | relaxdtmf=no |
| 16:55.54 | danw_home | lemme try that |
| 16:56.09 | Mad||Cow | ManxPower: In /etc/zaptel.cof I have fxoks=1; In zapata.conf I have signalling=fxo_ks for channel => 1 however, when I try and start asterisk now, I get chan_zap.c: Unable to register channel '1'. I changed everything over to fxo as you suggested. |
| 16:56.22 | danw_home | funny thing is, if I type "2345" it says Invalid extension '2' ... |
| 16:56.35 | danw_home | so it's obviously getting at least the first digit. |
| 16:56.55 | ManxPower | danw_home: the issue could be any number of problems |
| 16:57.01 | Strom_M | danw_home: pastebin the relevant section of extensions.conf |
| 16:58.00 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 16:58.08 | robin_sz | where can i get asterisk sounds in english? |
| 16:58.22 | *** join/#asterisk pariah (n=j0sh@unaffiliated/pariah) |
| 16:58.25 | rue_mohr | sounds? |
| 16:58.27 | Strom_M | asterisk.org |
| 16:58.37 | Strom_M | or in the freaking tarball |
| 16:58.45 | coil | how do i dial a phone number?? |
| 16:59.12 | danw_home | ManxPower your suggestion fixed it, thanks. |
| 16:59.19 | Strom_M | coil: you use chan_imacompletetard.so |
| 16:59.23 | robin_sz | Strom_M, there is an english version in the tarball? |
| 16:59.28 | coil | how do i use that!?!?! |
| 16:59.31 | danw_home | thanks for your help |
| 16:59.48 | Strom_M | robin_sz: yes |
| 16:59.57 | robin_sz | coo. |
| 17:00.09 | rue_mohr | hey erm, anyone have a simple little dial plan I can leech? |
| 17:00.22 | robin_sz | because this american woman is driving me dippy |
| 17:00.25 | ManxPower | Anyone is welcome to send money via paypal to eric@fnords.org |
| 17:00.40 | Fieldy | spammer? |
| 17:00.46 | Strom_M | robin_sz: technically she's canadian |
| 17:01.01 | robin_sz | thats part of america |
| 17:01.03 | rue_mohr | ManxPower, is that a hint? |
| 17:01.03 | ManxPower | rue_mohr: there really is no such thing as a simple dialplan |
| 17:01.08 | rue_mohr | heh |
| 17:01.18 | Strom_M | robin_sz: you bloody brits and your hash key gibberish |
| 17:01.27 | ManxPower | rue_mohr: It is a hint to anyone I have helped. |
| 17:01.37 | Strom_M | marmite and chutney and quid |
| 17:01.47 | Fieldy | ah. well toss that in what you said :) otherwise some may think it's spam |
| 17:01.50 | robin_sz | quid? |
| 17:01.50 | rue_mohr | ManxPower, are you the digium man? |
| 17:01.57 | robin_sz | oh, the POUND |
| 17:02.05 | robin_sz | or 50c as you say in the USA |
| 17:02.08 | ManxPower | rue_mohr: I do not work for Digium in any way. |
| 17:02.20 | rue_mohr | oh |
| 17:02.28 | ManxPower | Since I don't work for Digium, I can be mean to people that deserve it. |
| 17:02.29 | rue_mohr | just a regular guru eh |
| 17:02.59 | robin_sz | Strom_M, ok, are there english sounds OTHER than alison in the tarball? |
| 17:03.09 | rue_mohr | oh man I really need to think about this |
| 17:03.21 | Strom_M | robin_sz: so when you say "English" you really mean "British English" |
| 17:03.24 | ManxPower | robin_sz: look on the Wiki or the maling list archives |
| 17:03.28 | Fieldy | surely you can look yourself? |
| 17:03.28 | rue_mohr | lets see, the house has 11 rooms, and there is one outside shop |
| 17:03.46 | rue_mohr | so I cant see needing more than 99 extensions |
| 17:03.52 | robin_sz | Strom_M, yes, I guess I do. |
| 17:03.53 | *** join/#asterisk rudholm (i=rudholmm@nat/yahoo/x-5d189cdf88ec7474) |
| 17:03.54 | rue_mohr | so I could use two digits |
| 17:04.03 | Strom_M | +1 ambiguity points |
| 17:04.03 | Strom_M | +1 geocentrism |
| 17:04.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 17:04.23 | Strom_M | rudholm! |
| 17:04.31 | rudholm | Strom_M! |
| 17:04.35 | Strom_M | what are you doing at work? |
| 17:04.50 | rudholm | who says I'm at work? |
| 17:04.56 | Qwell | freenoed |
| 17:05.02 | Qwell | freenode too |
| 17:05.09 | Strom_M | freenoid |
| 17:05.17 | rudholm | ah, no, I just run irssi on my work box |
| 17:05.41 | rudholm | inside 'screen' |
| 17:06.02 | Qwell | robin_sz: feel free to hire somebody to record some, and contribute them |
| 17:06.02 | Strom_M | rudholm: so how about I stop procrastinating and bring all those interface cards over to that area code / rate center boundary of yours? |
| 17:06.19 | rudholm | british sounds? I think I have a recording of the sound of the inside of a pub around here somewhere if you want... |
| 17:06.22 | robin_sz | surely I cant be the only brit using asterisk? |
| 17:06.35 | *** join/#asterisk lee_is_me (n=chatzill@12-201-102-196.client.mchsi.com) |
| 17:06.38 | rue_mohr | ok, how about this |
| 17:06.55 | rue_mohr | I'll make my goal to be able to dial one phone from another |
| 17:06.58 | ManxPower | robin_sz: a 15 second google search turned up many references to birish sounds |
| 17:07.02 | Fieldy | run a google search maybe you can find what you are looking for. else, hey, the sound that come with it are fine and well done ,use em. failing that you could always make your own. |
| 17:07.07 | ManxPower | Results 1 - 10 of about 810,000 English pages for asterisk british sounds. (0.18 seconds)Â |
| 17:07.09 | Fieldy | ManxPower beat me to it doh |
| 17:07.14 | ManxPower | Now get off your lazy ass and look for yourself. |
| 17:07.19 | rudholm | Strom_M: yeah, that sounds good. I don't have any plans for today that I can recall. I need to wake up and stuff first, though. |
| 17:07.29 | Strom_M | rudholm: me too |
| 17:07.32 | Strom_M | im still in bed |
| 17:07.38 | rudholm | ah |
| 17:07.59 | Strom_M | do you want to do the breakfast thing? i havent eaten yet |
| 17:08.50 | rudholm | My landscaper is supposed to show up to talk to me, so I should probably stay here. |
| 17:09.01 | Strom_M | ok |
| 17:09.19 | rudholm | gonna go see Fred |
| 17:09.19 | rudholm | ? |
| 17:09.35 | rue_mohr | is the default demo progrmming 100% in extensions.conf or is alot of it somewhere else, I cant find where it picks up |
| 17:09.42 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:09.52 | Strom_M | either fred, Casa Rudholm, or Cerca de Casa Rudholm |
| 17:10.04 | rue_mohr | aha |
| 17:10.19 | rudholm | heh |
| 17:10.40 | Mad||Cow | ManxPower: If I put fxoks in my /et/zaptel.conf and do a ztcfg -vv it complains about an invlaid argument and says something about "Did you forget that FXS interfaces are configured with FXO signalling". Are you positive that the line running from my phone company should be defined as fxoks in zaptel.conf? |
| 17:10.43 | rue_mohr | heh, I can use hte at&t demo for something phone related |
| 17:11.43 | ManxPower | Mad||Cow: Read the jbot message CAREFULLY |
| 17:11.46 | ManxPower | ~fxofxs |
| 17:11.47 | jbot | methinks fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 17:12.09 | ManxPower | FXO ports use FXS signalling. FXS ports use FXO signalling |
| 17:12.23 | Strom_M | http://www.jerkcity.com/jerkcity2786.html |
| 17:12.54 | *** join/#asterisk BSD_tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 17:14.23 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 17:14.25 | Mad||Cow | ManxPower: So then you are correct. I should deinfe the ports as fxo in /etc/zaptel.conf as the input to them has voltage and dial tone |
| 17:14.38 | ManxPower | *sigh* |
| 17:14.51 | lee_is_me | lol |
| 17:14.56 | ManxPower | Mad||Cow: There is a test: What kind of ports do you plug a phone line into? |
| 17:14.59 | Mad||Cow | ManxPower: sorry ;-) |
| 17:15.15 | Mad||Cow | ManxPower: FXO |
| 17:15.35 | ManxPower | Mad||Cow: Correct! What signalling do FXO ports use? |
| 17:15.44 | Mad||Cow | ManxPower: FXS |
| 17:15.50 | ManxPower | Mad||Cow: correct! |
| 17:16.04 | ManxPower | now you know how to set up /etc/zaptel.conf and /etc/asterisk/zapata.conf |
| 17:16.51 | Mad||Cow | ManxPower: so then zaptel.conf should define my channels as FXO right? |
| 17:17.08 | rue_mohr | Mad||Cow, channelbank? |
| 17:17.19 | ManxPower | Mad||Cow: no. |
| 17:17.32 | ManxPower | in zaptel.conf you specify the SIGNALLING not the PORT TYPE. |
| 17:17.41 | ManxPower | and what kind of signalling do FXO ports use? |
| 17:17.52 | Mad||Cow | ManxPower: FXS gotcha |
| 17:18.02 | Strom_M | FXcocks |
| 17:18.04 | Mad||Cow | ManxPower: then where do i define the port type? |
| 17:18.12 | ManxPower | Mad||Cow: nowhere. |
| 17:18.30 | ManxPower | you can't change the port type, so there is no point in setting it in a config file. |
| 17:19.18 | *** part/#asterisk ManxPower (n=manxpowe@181.sub-75-201-83.myvzw.com) |
| 17:19.21 | Mad||Cow | ManxPower: makes since... just a bit confusing |
| 17:19.31 | Strom_M | makes since what |
| 17:19.35 | Strom_M | or did you mean "makes sense" |
| 17:19.50 | robin_sz | OK, I found british sounds in a-law |
| 17:19.51 | Mad||Cow | Strom_M: #2 |
| 17:19.56 | robin_sz | G711a |
| 17:20.19 | robin_sz | but what about my custom announcements I recorderd as gsm? |
| 17:20.24 | rudholm | robin_sz: yeah, that'd make sense, wouldn't it? |
| 17:20.32 | Fieldy | just use then, you'll be fine |
| 17:20.40 | Strom_M | you should have recorded them as wav or alaw for higher quality ;) |
| 17:20.41 | Fieldy | them |
| 17:20.46 | robin_sz | right |
| 17:21.30 | robin_sz | so I just made sounds_american and sounds_british directories, |
| 17:22.49 | robin_sz | ummm ... but I cant find where /usr/share/asterisk/sounds/ is set in the * configs |
| 17:23.03 | Strom_M | should be /var/lib/asterisk/sounds |
| 17:23.15 | Strom_M | look in /etc/asterisk/asterisk.conf |
| 17:23.15 | robin_sz | wheely? |
| 17:23.38 | robin_sz | nope |
| 17:23.43 | robin_sz | ntothing to do with sounds in there |
| 17:24.54 | pipwerk | astdatadir? |
| 17:26.00 | robin_sz | dont have that as a setting |
| 17:26.11 | robin_sz | astvarlibdir |
| 17:26.12 | robin_sz | ? |
| 17:27.36 | *** join/#asterisk arafath (n=root@203.88.71.234) |
| 17:29.11 | rue_mohr | does everything usually start with [default]? |
| 17:29.35 | rue_mohr | I could use like a 3 line extensions.conf so I know whats fluff and whats not |
| 17:30.18 | robin_sz | bah poxy partially complete soundsets |
| 17:31.02 | robin_sz | "British English Female basic VM system intended to replace the default Asterisk installation" |
| 17:31.06 | *** join/#asterisk |Tiger| (n=Tiger@213.201.58.8) |
| 17:31.10 | robin_sz | no mailbox full .... |
| 17:32.12 | rue_mohr | wonder what a dialplan would look like that barked at you and hung up. |
| 17:32.25 | rue_mohr | "woof" 'click' |
| 17:33.09 | |Tiger| | is there any web-based system for adding user for asterisk? |
| 17:34.37 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 17:36.55 | *** join/#asterisk robin_z (n=robin@rapid2.gotadsl.co.uk) |
| 17:37.03 | robin_z | so, to empty out a load of unused voicemailboxes? |
| 17:37.08 | robin_z | just delete the /var/spool/asterisk/voicemail/default/foo ? |
| 17:37.11 | robin_z | or is there data with it? |
| 17:38.59 | *** join/#asterisk killown (n=killown@unaffiliated/killown) |
| 17:41.39 | [TK]D-Fender | robin_z, Yes, you can somply trash the whole folder |
| 17:41.59 | robin_z | ta |
| 17:42.02 | [TK]D-Fender | robin_z, if the box is called upon again, the whoe structure will be recreated |
| 17:42.12 | robin_z | nice |
| 17:42.16 | robin_z | thanks |
| 17:42.17 | [TK]D-Fender | robin_z, without anything previously recorded of course |
| 17:42.25 | robin_z | but of course |
| 17:43.17 | [TK]D-Fender | rue_mohr, [defaul] only exists in your head and is so generically named as to be stupid. WTH does [default] imply? Is that the kind of security model you want to start with? |
| 17:43.38 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 17:44.07 | [TK]D-Fender | no sample file should ever use [default] as a context.... sets a bad precedent and people begin thinking that the name itself has some magical built-in meaning. |
| 17:45.00 | robin_z | [TK]D-Fender, people probably confuse it with the [general] section in soem files |
| 17:45.10 | robin_z | quite different |
| 17:45.17 | [TK]D-Fender | robin_z, That too. |
| 17:50.33 | robin_z | hmmm |
| 17:50.50 | robin_z | this * doesnt seem to be looking for g711a files for sounds ... |
| 17:50.57 | robin_z | they exist, it ignores them |
| 17:51.04 | [TK]D-Fender | robin_z, pastebin it... |
| 17:51.19 | robin_z | pastebin what? |
| 17:51.59 | [TK]D-Fender | robin_z, the CLI output showing its reaction to trying to play the files you're expecting and proof that they're really there, proper and accessable of course :) |
| 17:52.20 | robin_z | -rwxrwxrwx 1 root root 30758 2006-05-22 15:43 vm-rec-name.g711a |
| 17:52.55 | [TK]D-Fender | robin_z, Oh, btw.. thats not the proper extension for that files. should be .alaw / .ulaw |
| 17:52.58 | [TK]D-Fender | :p |
| 17:53.02 | robin_z | ahh |
| 17:53.09 | [TK]D-Fender | BIG PRINT |
| 17:53.28 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:53.33 | robin_z | thats how they came out of the tar.gz from some site on the wiki |
| 17:53.42 | robin_z | http://www.enicomms.com/cutglassivr/ |
| 17:53.46 | robin_z | thanks... |
| 17:57.48 | rue_mohr | [TK]D-Fender, yar, I think I wored out s,... is the start of everything |
| 17:58.46 | rue_mohr | I just need to work out how to seperate sets being picked up from incomming calls |
| 17:58.51 | [TK]D-Fender | rue_mohr, Yeah, NOBODY gets it. |
| 17:58.54 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 17:59.24 | [TK]D-Fender | rue_mohr, seperate wht sets? |
| 17:59.53 | rue_mohr | :) bbl |
| 18:06.22 | killown | what hardware I need to learn to configure asterisk? |
| 18:06.24 | robin_z | hmmm |
| 18:06.33 | kombi | mvanbaak: are you still there? |
| 18:06.42 | mvanbaak | kindda |
| 18:06.46 | kombi | lol.. |
| 18:07.07 | robin_z | It seems to not play "the person at extewnsion .." although, at least it doesn not complain anymore about not being able to find the vm-*.alaw file ... |
| 18:07.34 | kombi | mvanbaak: do you recall how you went about tftp and the cisco phone? |
| 18:08.11 | kombi | got atftpd running, but then.. |
| 18:09.03 | robin_z | weird .. it just plays. beep.gsm and then records ... |
| 18:09.34 | mvanbaak | kombi: I put some lines in my dhcp server so the phone will know what the tftp server is |
| 18:10.05 | kombi | mvanbaak: ohoh, don't even have dhcp here.. |
| 18:10.12 | |Tiger| | is there any web-based system for adding user for asterisk? if the install have one i cant get the index |
| 18:10.47 | kombi | mvanbaak: I'll get that done first.. |
| 18:11.37 | mvanbaak | good idea |
| 18:13.22 | *** join/#asterisk crimethinker (i=ircuser@legacy.diamond.org) |
| 18:17.26 | [TK]D-Fender | killown, hardware = your eyes |
| 18:21.35 | robin_z | ok, so I used extenstion s5001 in my voicemail instead of 5000 |
| 18:22.41 | kombi | mvanbaak: got it running, could you maybe do a quick grep tftp /etc/dhcpd.conf? |
| 18:23.41 | kombi | ..because I can only find a host specific entry in the man pages.. |
| 18:24.05 | mvanbaak | hang on |
| 18:25.30 | mvanbaak | http://pastebin.three-dimensional.net/index.php?action=view&id=5275befa |
| 18:25.38 | mvanbaak | there's my subnet in dhcpd |
| 18:25.50 | mvanbaak | 192.168.2.4 is my tftp/asterisk box |
| 18:25.52 | kombi | thanks! |
| 18:26.20 | marc\cba | SNAP |
| 18:26.44 | kombi | got your own pastebin, that IS kewl! |
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| 18:29.54 | *** mode/#asterisk [+o Qwell] by irc.freenode.net |
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| 18:32.38 | *** mode/#asterisk [+o Qwell] by irc.freenode.net |
| 18:34.00 | *** join/#asterisk BS2 (n=BS@2001:960:68f:201:211:d8ff:fe9a:6b0b) |
| 18:35.47 | Greek-Boy | any telco guys around? |
| 18:46.35 | *** join/#asterisk _Raptor_ (i=sirasenn@faui08r.informatik.uni-erlangen.de) |
| 18:46.59 | _Raptor_ | hello, how can i deal in extensions.conf when my sip peer tells me "Busy here"? |
| 19:10.15 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
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| 19:14.27 | kombi | mvanbaak: I am not quite sure the phone found the tftp, but anyway, what should be next? |
| 19:23.01 | shido6 | Zzzz |
| 19:23.11 | shido6 | busy here |
| 19:23.35 | shido6 | hrmmm codec's match? do u have any extra settings in the peer name stanza in sip.conf? |
| 19:24.24 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 19:25.24 | robin_z | so, whats a good codec to use for low bandwidth and reasnoable quality? alaw is fine internally, but the external offices are struggling with it over adsl |
| 19:27.39 | shido6 | gsm |
| 19:28.27 | robin_z | whats that one that digiium sell? |
| 19:29.57 | robin_z | G723? |
| 19:30.56 | shido6 | g729 |
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| 19:42.09 | jangell_ | Anyone in here know anything about E&M Wink? I can't get Asterisk to recognize that a call was answered on outbound calling....so asterisk hangs up after 5 minutes |
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| 19:48.47 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 19:49.32 | mvanbaak | heya Qwell_ |
| 19:51.32 | [hC] | Does the meetme/conference stuff in 1.4 still transcode everything to slin? |
| 19:51.44 | Qwell_ | yes, it has to |
| 19:52.08 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 19:52.41 | robin_z | [hC], you can recompile meetme to use any native format, |
| 19:53.33 | robin_z | we recompiled to use gsm I think, to save oodles of transcodings from our gsm based web clients |
| 19:53.43 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 19:53.46 | [hC] | I'm trying to do entirely g729 installs, and would love to eliminate the need for slin in meetme. |
| 19:53.54 | robin_z | this was 1.2.x, 1.4 may not be the same |
| 19:54.01 | Qwell_ | [hC]: not possible.. you need slin for a bunch of the features |
| 19:54.12 | robin_z | Qwell, such as? |
| 19:54.13 | [hC] | Qwell: what does slin provide exactly, that other codecs cannot? |
| 19:54.26 | Qwell_ | dtmf stuff, talker detection |
| 19:55.00 | Qwell_ | and...the most important - the mixing of all the channels |
| 19:55.13 | mvanbaak | indeed |
| 19:55.16 | robin_z | we only have one talker in our rooms, and no dtmf foo, so perhaps its not an issue for us |
| 19:55.24 | [hC] | Ah, that would do it. |
| 19:55.28 | [hC] | I have more than one talker. :) |
| 19:55.38 | mvanbaak | you will have to mod the code for that |
| 19:55.53 | Qwell_ | well, you can turn off talk detection |
| 19:55.56 | [hC] | Well.. That kinda bites, the soekris net4801's dont do so well with transcoding meetme's. I guess ill have to host those offsite. |
| 19:55.58 | robin_z | it was a teeny weeny mod IIRC, just commented out an includem, put in another |
| 19:56.19 | Qwell_ | but you can't turn off the audio mixing |
| 19:56.22 | mvanbaak | I'm used to mod code |
| 19:56.27 | mvanbaak | had a good example today |
| 19:56.35 | mvanbaak | wanted to do dundi with skinny phones |
| 19:56.47 | mvanbaak | ;) |
| 19:56.49 | mvanbaak | sorry Qwell_ |
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| 20:30.34 | lee_is_me | Anyone know how to get into the phone's config on an Aastra phone? |
| 20:32.44 | lee_is_me | nevermind... |
| 20:33.02 | *** join/#asterisk killown (n=killown@unaffiliated/killown) |
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| 20:41.37 | mvanbaak | lee_is_me: use my fireaxe ;) |
| 20:43.09 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 20:43.44 | lee_is_me | that's a little on the hard side for me. I'm more of a Rare Earth/Grand Funk Railroad kinda guy |
| 20:44.08 | *** join/#asterisk poohbah431111 (n=rmillis@S010600183950760b.cg.shawcable.net) |
| 20:44.43 | poohbah431111 | Can anyone go over a few steps i took to compile asterisk and help me see if I overlooked anything, please? |
| 20:45.16 | poohbah431111 | I am running Red Hat Enterprise Linux AS release 4 (Nahant Update 5) |
| 20:45.35 | pipwerk | * 1.4? |
| 20:45.59 | pipwerk | ./configure; make install |
| 20:46.27 | poohbah431111 | On VMware Server 1.0.3 build-44356, would that cause any problems with me not heaing audio when I try to dial the echo test? |
| 20:46.54 | pipwerk | firewall? |
| 20:46.59 | pipwerk | nat? |
| 20:47.08 | poohbah431111 | Asterisk 1.2.18 |
| 20:47.19 | poohbah431111 | I'm on my lan trying this - the is no firewall. |
| 20:47.27 | poohbah431111 | networking is set to bridged |
| 20:47.32 | poohbah431111 | for vmware |
| 20:48.21 | *** part/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com) |
| 20:48.33 | poohbah431111 | I just insatlled and compiled asterisk-1.2.18 asterisk-addons-1.2.6 |
| 20:48.50 | poohbah431111 | oh and asterisk-sounds-1.2.1 |
| 20:49.15 | poohbah431111 | X-lite can conenct, and I can dial my local, X-lite line 2 rings, I answer but cant hear any sound |
| 20:49.32 | poohbah431111 | I'm not sure if i put the right settigns in the correct files though. |
| 20:49.59 | poohbah431111 | I followed the simple setup steps in AsteriskTFOT.pdf |
| 20:50.18 | pipwerk | have you tried a simple speaking clock? |
| 20:50.27 | poohbah431111 | no, how do I do that? |
| 20:52.21 | pipwerk | an exten with something like 'exten => 102,1,SayUnixTime(,,AdBY R) |
| 20:52.22 | poohbah431111 | can I paste my sip.conf to a paste bin someplace? |
| 20:52.50 | pipwerk | if things ring, sip works, it's rtp that is causing trouble |
| 20:52.59 | pipwerk | in one direction or both |
| 20:53.00 | poohbah431111 | Yes it rings |
| 20:53.30 | poohbah431111 | sip.conf has my context in it, right? |
| 20:53.39 | pipwerk | is your client on windows or linux? |
| 20:53.46 | poohbah431111 | Win32 |
| 20:53.53 | poohbah431111 | Xlite 3 |
| 20:53.55 | pipwerk | with firewall? |
| 20:54.08 | poohbah431111 | Windows firewallis on yes |
| 20:54.21 | poohbah431111 | But client was working with external sip orvider |
| 20:54.23 | pipwerk | try to disable the windows firewall, just as a test |
| 20:54.26 | poohbah431111 | ok |
| 20:54.36 | pipwerk | ok, that shouldn't help then |
| 20:54.47 | pipwerk | but please try |
| 20:56.17 | poohbah431111 | No does not seem to |
| 20:56.59 | poohbah431111 | i have created simple sip.conf and extensions.conf |
| 20:57.42 | poohbah431111 | extesnions.conf justhas [internal] and then exten =>100,1,dial (SIP/user1) |
| 20:58.04 | poohbah431111 | And tehn a simolar entry for user2 but 200 for the local |
| 20:58.12 | poohbah431111 | and then a line for 611 to echo |
| 20:58.19 | poohbah431111 | Is that the right place for this stuff? |
| 20:58.46 | pipwerk | any extension context is the right place :) |
| 20:58.52 | poohbah431111 | the pdf document seemed a bit unclear on that |
| 20:59.08 | poohbah431111 | i dont understand sorry.:-( |
| 20:59.15 | pipwerk | a context is exactly that, a context |
| 20:59.31 | poohbah431111 | if i am just calling on my lan from soft phone to soft phone i need 2 extensions? |
| 20:59.53 | poohbah431111 | and these go in /etc/asterisk/extensions.conf? |
| 21:00.03 | pipwerk | correct |
| 21:00.12 | poohbah431111 | ok |
| 21:01.12 | poohbah431111 | and my /etc/asterisk/sip.conf contains [general] |
| 21:01.22 | poohbah431111 | externip=x.x.x.x |
| 21:01.29 | poohbah431111 | context=default |
| 21:01.36 | poohbah431111 | srvlookup=yes |
| 21:01.38 | pipwerk | better paste it to pastebin.ca |
| 21:01.43 | poohbah431111 | ok |
| 21:01.48 | poohbah431111 | how can I do that? |
| 21:02.03 | john-eman0n | i have a vitelity account and sip client idefisk and it registers but i cannot set callerid via idefisk anyone have any ideas? |
| 21:02.28 | poohbah431111 | join pastebin.ca? |
| 21:02.36 | john-eman0n | what's that? |
| 21:02.58 | poohbah431111 | Sorry im using ircII on BSD box as I dont have a win32 IRC client and im a bit confused. :-) |
| 21:03.16 | poohbah431111 | How do i pastre to the pastebin? |
| 21:03.18 | pipwerk | poohbah431111: visit www.pastbin.ca, paste in the paste window and gimme the url it returns |
| 21:03.28 | pipwerk | +e |
| 21:03.31 | poohbah431111 | ok |
| 21:03.42 | robin_z | and shorten your username |
| 21:03.59 | pipwerk | robin_z: tab completion :) |
| 21:04.12 | robin_z | it just makes it look such a mess |
| 21:04.43 | poohbah431111 | http://www.pastebin.ca/499055 |
| 21:05.14 | poohbah431111 | ya i tried but forgot my password for the short version so could not register. |
| 21:06.05 | pipwerk | poohbah431111: youe [user] entries are missing an username statement |
| 21:06.24 | poohbah431111 | oh? |
| 21:06.31 | poohbah431111 | would that cause no sound? |
| 21:06.52 | pipwerk | http://www.pastebin.ca/499060 |
| 21:07.27 | pipwerk | I hope it won't make any difference, but you never know |
| 21:07.28 | poohbah431111 | http://www.pastebin.ca/499063 |
| 21:07.39 | poohbah431111 | And thats all I have set |
| 21:07.45 | robin_z | and no rtp port range? |
| 21:08.02 | poohbah431111 | no rtp |
| 21:08.06 | poohbah431111 | where does that go? |
| 21:08.18 | pipwerk | rtp.conf |
| 21:08.25 | poohbah431111 | its default |
| 21:08.31 | robin_z | k |
| 21:08.40 | pipwerk | and on the lan, without firewalls, it doesn't matter |
| 21:08.46 | robin_z | and you are in the same subnet as the * box with the phones? |
| 21:08.59 | poohbah431111 | let me add teh username |
| 21:09.08 | robin_z | and you are in the same subnet as the * box with the phones? |
| 21:10.11 | poohbah431111 | yes same subnet |
| 21:10.37 | poohbah431111 | or doing a reload from the CLI |
| 21:11.03 | poohbah431111 | i call 611 and it says calling.... |
| 21:11.06 | poohbah431111 | never connects |
| 21:11.19 | pipwerk | never connects? ok |
| 21:11.22 | robin_z | ahh |
| 21:11.33 | robin_z | no ringing tone? |
| 21:11.41 | poohbah431111 | no ring tone |
| 21:11.44 | robin_z | ahh. |
| 21:11.46 | robin_z | ok, |
| 21:11.54 | poohbah431111 | if i call my local 100 it rings and i can answer |
| 21:12.03 | poohbah431111 | but no sounds |
| 21:12.15 | robin_z | local 100? whats that? |
| 21:12.21 | pipwerk | http://www.pastebin.ca/499072 |
| 21:12.29 | poohbah431111 | i just defined a local for my self |
| 21:12.33 | pipwerk | try that for tou echotest |
| 21:12.43 | poohbah431111 | looking... |
| 21:13.06 | pipwerk | first you need to 'answer' before * can echo ;-) |
| 21:13.27 | robin_z | yip |
| 21:13.36 | poohbah431111 | <PROTECTED> |
| 21:13.41 | poohbah431111 | And add that? |
| 21:14.33 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:14.39 | poohbah431111 | Do I need {internal] in extensions.conf? |
| 21:14.46 | poohbah431111 | reload |
| 21:14.47 | poohbah431111 | oosp |
| 21:14.47 | pipwerk | also, with this dialplan curious as to whet dialing 612 would do |
| 21:14.53 | fujin | hey, can anyone tell me what record_in and record_out do? |
| 21:15.29 | pipwerk | poohbah431111: yes, you need [internal] at the top of this all |
| 21:15.56 | poohbah431111 | well its sorta working now |
| 21:16.00 | poohbah431111 | I have sound |
| 21:16.04 | poohbah431111 | I can hear myself |
| 21:16.07 | pipwerk | cool |
| 21:16.15 | poohbah431111 | but I cant hear the echo test at 611 |
| 21:16.22 | poohbah431111 | i called 611 and it connected me to me? |
| 21:16.26 | fujin | I can't seem to get any audio at all unless I specify record_in=Adhoc |
| 21:16.30 | pipwerk | yes |
| 21:16.35 | poohbah431111 | Oh thats what is does, but I thought it said teh time? |
| 21:16.40 | pipwerk | you should hear yourself |
| 21:16.47 | pipwerk | 612 will tell the time |
| 21:17.35 | poohbah431111 | yes that worked |
| 21:17.40 | pipwerk | cool |
| 21:17.41 | poohbah431111 | So what was wrong? |
| 21:17.48 | poohbah431111 | just the username? |
| 21:17.54 | pipwerk | I guess so |
| 21:18.11 | poohbah431111 | ok, so my friend is at his house - I have a local for him |
| 21:18.19 | poohbah431111 | he can conenct but got no audio |
| 21:18.26 | poohbah431111 | maybe teh same issue? |
| 21:18.31 | pipwerk | maybe |
| 21:18.39 | poohbah431111 | Im nated, so is he, so I set the locals to nat? |
| 21:19.18 | poohbah431111 | And i give him my external IP and he enters that for the proxy and domain? |
| 21:19.35 | poohbah431111 | but appends 5060 on for the proxy? |
| 21:20.09 | poohbah431111 | And I forward the ports from my router to my asterisk box? |
| 21:21.00 | poohbah431111 | RTP 10000 to 20000 both tcp and udp? and sip 5060 to 5060 tcp and udp? |
| 21:21.19 | pipwerk | rtp is udp only |
| 21:21.30 | pipwerk | sip for * is udp only too |
| 21:22.10 | pipwerk | and with rtp.conf you can set trpstart and rtpend to limit the rtp range |
| 21:22.23 | pipwerk | rtpstart |
| 21:22.23 | robin_z | and then forward all those ports |
| 21:22.39 | robin_z | and then disable any "sip helper applications" on your nat thing |
| 21:22.47 | robin_z | as they hinder rather than help |
| 21:23.21 | poohbah431111 | i dont think there are any sip helpers |
| 21:24.07 | poohbah431111 | trpstart and rtpend are default |
| 21:26.34 | poohbah431111 | got my buddy on MSn and will try it again. |
| 21:26.42 | poohbah431111 | Thannks for the help so far. |
| 21:32.36 | poohbah431111 | not working |
| 21:32.42 | poohbah431111 | messing a bit with the router now. |
| 21:33.55 | pipwerk | rtp and nat are _not_ friends |
| 21:34.28 | poohbah431111 | Ok I put asterisk in a DMZ |
| 21:34.46 | poohbah431111 | Now its on my side of the conenction so I can say NAT no? |
| 21:34.52 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 21:34.56 | pipwerk | correct |
| 21:35.42 | poohbah431111 | if hes is natted i say nat yes? |
| 21:36.01 | poohbah431111 | if he moves his pc to a dmz I can say nat no? |
| 21:36.48 | poohbah431111 | fell like im in a varizon comercial |
| 21:36.51 | poohbah431111 | can ya here me now... |
| 21:37.21 | pipwerk | don't know :) |
| 21:38.19 | poohbah431111 | hmm.. he can dial 611 and hear himself |
| 21:38.24 | poohbah431111 | 612 he hears nothing |
| 21:38.37 | poohbah431111 | we can establish a call and then nothing |
| 21:40.18 | poohbah431111 | he may have some local lookback thing going. |
| 21:41.03 | pipwerk | hmmm, true, with reinvites you can have a working echo and no working talking clock |
| 21:41.16 | poohbah431111 | reinvites? |
| 21:42.17 | poohbah431111 | would reinvites yes change anything? |
| 21:42.33 | pipwerk | no |
| 21:43.57 | poohbah431111 | ahhhh. frustrating.... |
| 21:44.12 | *** join/#asterisk marcan (i=1337@198.Red-83-54-248.dynamicIP.rima-tde.net) |
| 21:44.34 | pipwerk | yes |
| 21:44.50 | pipwerk | I run * on my firewall at home |
| 21:50.56 | *** join/#asterisk brussel_ (n=brussel@cpe-72-130-172-213.san.res.rr.com) |
| 21:51.52 | *** join/#asterisk |Tiger| (n=Tiger@213.201.58.8) |
| 21:54.28 | *** join/#asterisk zoinho (i=Oakenfol@zoinho.eh.o.cara.dipelnet.com.br) |
| 22:01.37 | rue_mohr | haha |
| 22:02.08 | rue_mohr | I thought recognized the number in http://bugs.digium.com/view.php?id=9099 its written by my buddy, haha |
| 22:03.00 | Nivex | what's the recommendation for an ATA these days? |
| 22:05.21 | [TK]D-Fender | Nivex, Besic use? Linksys SPA-2102 |
| 22:05.25 | [TK]D-Fender | basic* |
| 22:05.43 | *** join/#asterisk keulin (n=cray@AMontpellier-152-1-20-28.w81-251.abo.wanadoo.fr) |
| 22:06.18 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 22:06.18 | Nivex | ah ok. I was looking at the PAP2T-NA (no need for a router) |
| 22:08.26 | [TK]D-Fender | Nivex, Same with the SPA-2102. Except the PAP2 has a weaker processor, fewer calling features, not T.38 support. |
| 22:08.47 | [TK]D-Fender | Nivex, I'd recommend spending a FEW extra bucks on the 2102 over the PAP2 |
| 22:09.41 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 22:14.32 | rue_mohr | how do I distinguish an fxo call from a fxs call in extentions.conf? |
| 22:15.16 | Hymie | by what context they start in, I suppose |
| 22:15.21 | Hymie | I've never had to worry about that |
| 22:15.21 | crimethinker | don't associate them with the same context? |
| 22:15.28 | rue_mohr | well |
| 22:15.49 | rue_mohr | what I know so far is when I pick up the phone, asterisk starts at the first s line |
| 22:16.04 | rue_mohr | I presume its going to do the same thing for both types of lines right now |
| 22:16.07 | rue_mohr | I suspect |
| 22:16.08 | Hymie | in that context, for that line, yes |
| 22:16.15 | Hymie | read up on context |
| 22:16.25 | anonymouz666 | [TK]D-Fender: Did you see the new version of PAP2? I think it's called PAP2 TA |
| 22:16.28 | rue_mohr | "exten => s,1,Dial,Zap/g2/96944569" group 2 in that line is part of the answer |
| 22:20.13 | [TK]D-Fender | rue_extensions.conf has nothing to dow ith the techt he call comes in on |
| 22:20.22 | [TK]D-Fender | rue_mohr, rather |
| 22:20.27 | rue_mohr | ah |
| 22:20.33 | rue_mohr | ok |
| 22:20.44 | [TK]D-Fender | rue_mohr, you should configure your CHANNELS to go where you want them to. |
| 22:20.56 | rue_mohr | cause there needs to be something to distinguish the fxo and fxs calls |
| 22:21.15 | [TK]D-Fender | rue_mohr, you should point each to an appropriate context based on what you wantit to do. |
| 22:21.29 | [TK]D-Fender | rue_mohr, Again, these would be CONTEXTS |
| 22:21.34 | rue_mohr | I dont know how to pick out a channel yet |
| 22:21.49 | rue_mohr | http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction |
| 22:21.55 | rue_mohr | that dosn't seem to have helped |
| 22:22.11 | [TK]D-Fender | rue_mohr, Your FXS & FXO channels all have a CONTEXT. set them into DIFFERENT ones to differntiate between them |
| 22:23.08 | *** join/#asterisk Cabal (n=Cabal@unaffiliated/cabal) |
| 22:23.16 | rue_mohr | your implying that each call already comes in on a unique context, and the modded demo code I have is putting them togethor because its using a default context |
| 22:24.08 | [TK]D-Fender | rue_mohr, Its not that each call comes intoa different context, but rather that each port(/channel) you defined should go to its own. "default" is a term you should do away with FAST. |
| 22:24.18 | rue_mohr | signalling = fxo_ls |
| 22:24.18 | rue_mohr | channel=>1-6 |
| 22:24.29 | [TK]D-Fender | rue_mohr, So you would make a context for your FXO ports to use, andother for your FXS, etc. |
| 22:24.37 | rue_mohr | I think thats from th wrong file |
| 22:24.57 | [TK]D-Fender | rue_mohr, No, that is zapata.conf and exactly where you would set the channel for each of your ports |
| 22:25.07 | [TK]D-Fender | s/channel/context |
| 22:25.09 | [TK]D-Fender | s/channel/context/ |
| 22:25.12 | [TK]D-Fender | ugh |
| 22:25.16 | rue_mohr | :) |
| 22:25.29 | [TK]D-Fender | rue_mohr, No, that is zapata.conf and exactly where you would set the *context* for each of your ports |
| 22:25.52 | rue_mohr | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 22:26.00 | rue_mohr | I see |
| 22:26.29 | [TK]D-Fender | rue_mohr, "channel=>1-6" using "fxo_ls" tells me that you are sending incoming calls from 6 FXS (phone station) ports to a single context |
| 22:26.47 | *** join/#asterisk ohadz (n=ohad@cpe-69-203-27-50.nyc.res.rr.com) |
| 22:26.52 | [TK]D-Fender | rue_mohr, So those 6 phones have the same access as one another. |
| 22:27.15 | rue_mohr | oh no, the hard thing, ohh man, the most difficult task I have to face again and again, I need to choose a name for the context... |
| 22:28.17 | [TK]D-Fender | rue_mohr, What is the contect you hanve in mid give the channel using it access to? That should help with naming it. |
| 22:28.34 | rue_mohr | ok, I'll call them house and telus |
| 22:28.50 | [TK]D-Fender | rue_mohr, I would think a little more abstract than that if I were you |
| 22:29.01 | rue_mohr | maybe later |
| 22:29.18 | rue_mohr | for now I'd like it to be clear |
| 22:29.27 | rue_mohr | and I only have 1 pbx :) |
| 22:30.41 | [TK]D-Fender | rue_mohr, heres a thought : [full-access] . This context implies that any device (zap port, sip phone, etc) you point to it will be able to dial out of any resource in your system. |
| 22:30.54 | [TK]D-Fender | rue_mohr, You could them make more restrictive ones that might limit LD, etc. |
| 22:31.06 | [TK]D-Fender | rue_mohr, but that doesn't have to be now. |
| 22:31.08 | rue_mohr | oo |
| 22:31.43 | rue_mohr | till I'm used to it, think I'll go with what I have there, this is going to change a lot |
| 22:32.11 | [TK]D-Fender | rue_mohr, You would them make another context like [in-from-zap-fxo] for your analog zaptel FXO channels to land in. Here you would normally set up an IVR or have it ring a bunch of phones, etc. |
| 22:32.22 | rue_mohr | that I like more |
| 22:32.40 | killown | anyone know a asterisk module for webmin? |
| 22:32.41 | [TK]D-Fender | rue_mohr, These are 2 completely seperate things which your dialplan should have. |
| 22:33.00 | [TK]D-Fender | killown, Yes, but it SUCKS. Less than useless for the most-part. |
| 22:33.20 | rue_mohr | for now I'm kinda just hacking, my only immediate goal is for us to get an answering machine, but no hurry |
| 22:33.38 | [TK]D-Fender | killown, soryy but you're going to have to actually LEARN * or surrender yourself to one of those GUI's like FreePBX (after which we really won't want to hear from you) |
| 22:33.56 | killown | ok |
| 22:33.58 | *** join/#asterisk saftsack (n=saftsack@pD9E04734.dip.t-dialin.net) |
| 22:34.18 | [TK]D-Fender | rue_mohr, Then why 6 FXS ports? :) You'd only need 1 FXo and just plg it in-line with your phones and answer after 6 rigs or so. |
| 22:34.39 | rue_mohr | heh, I'm gonna do up the whole house |
| 22:34.46 | rue_mohr | we have about 5 phones right now |
| 22:35.05 | rue_mohr | we have 1 line, but another one comming in a few months |
| 22:35.09 | Hymie | rue_mohr: SIP phones would be cooler, and just as cheap as all those zap interfaces |
| 22:35.14 | rue_mohr | hah |
| 22:35.29 | rue_mohr | unless your guru just handed you a channelbank and a t1 card |
| 22:35.31 | rue_mohr | :) |
| 22:35.40 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 22:35.44 | Hymie | at home?! |
| 22:35.48 | rue_mohr | ohyea |
| 22:35.56 | rue_mohr | }:] |
| 22:35.59 | Hymie | how much are you paying for that, btw? |
| 22:36.10 | Hymie | or, nothing |
| 22:36.13 | Hymie | because you are a bastard? |
| 22:36.14 | rue_mohr | no, the T1 is to the channelbank |
| 22:36.27 | Hymie | a T1 is pricy here, for what you get |
| 22:36.30 | rue_mohr | no way I'm paying $1500/mo for 2 lines |
| 22:36.52 | Hymie | two lines is $60 per month here ;P |
| 22:36.54 | rue_mohr | the T1 goes to the channelbank, to the phones and lines |
| 22:37.22 | rue_mohr | tt-weasels? man I need more consoles |
| 22:37.34 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:37.35 | rue_mohr | how do you mean? |
| 22:37.50 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:38.08 | rue_mohr | I'm putting a channelbank in the house, to operate the 2 lines and 6 phones with asterisk, so we can have al sorts of cool phone features |
| 22:38.51 | rue_mohr | ok the contexts worked |
| 22:38.57 | rue_mohr | :) I'm happy |
| 22:39.11 | rue_mohr | now that their seperate, I can start the real fun |
| 22:39.17 | Hymie | yes, and as I said, sip phones are cooler than analog phones and zap interfaces, and the same price |
| 22:39.28 | rue_mohr | but this is all free |
| 22:39.39 | Hymie | is your T1 free? that's what I don't get |
| 22:39.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 22:39.42 | rue_mohr | yes |
| 22:39.48 | Hymie | why? do you work for satan? |
| 22:39.49 | rue_mohr | the T1 is 2 feet long |
| 22:39.53 | Hymie | how do you get free things in your house? |
| 22:39.59 | rue_mohr | it goes from the T1 card to the channelbank |
| 22:40.03 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:40.07 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:40.13 | rue_mohr | I know people who work for places that get rid of things |
| 22:40.15 | *** join/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:40.28 | Hymie | hmm |
| 22:40.34 | rue_mohr | I dont have a T1 to hte house |
| 22:40.36 | *** part/#asterisk funkmaster (n=funky1@vhe-364501.sshn.net) |
| 22:40.36 | Hymie | like I said, satan people ;) |
| 22:40.38 | rue_mohr | that would be insane |
| 22:40.45 | Hymie | channel banks cost 5 BILLION dollars |
| 22:40.51 | Hymie | a country is starving somewhere |
| 22:40.53 | rue_mohr | no! |
| 22:40.56 | rue_mohr | ebay! |
| 22:40.59 | Hymie | while you, rue_mohr, sit on a house of GOLD |
| 22:41.01 | rue_mohr | mainstreet 3624 |
| 22:41.02 | Hymie | how does that make you feel? ;) |
| 22:41.15 | rue_mohr | well, its still half trailer |
| 22:41.27 | rue_mohr | and to me, that makes it cr** |
| 22:41.31 | rue_mohr | but anyhow |
| 22:41.39 | Hymie | heh |
| 22:41.40 | rue_mohr | I got the place cause it was 1/2 acre |
| 22:42.04 | rue_mohr | ignoring that fact it was hells 1/2 acre |
| 22:42.17 | Hymie | well, let me know if you like the sound quality, zap cards suck donkey balls ;D |
| 22:42.26 | Hymie | channel bank used might be ok |
| 22:42.31 | rue_mohr | its ok with this channelbank |
| 22:42.40 | rue_mohr | Hymie, you can get a channelbank |
| 22:42.40 | *** join/#asterisk psi0n (n=123@109.80-202-238.nextgentel.com) |
| 22:42.43 | rue_mohr | from ebay |
| 22:42.44 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 22:42.53 | rue_mohr | not too expensive, maybe $60 or less |
| 22:42.58 | rue_mohr | the T1 card is the killer |
| 22:43.01 | Hymie | rue_mohr: yeah.. I'm thinking of it... do you know if it has built in echo cancellation? |
| 22:43.09 | rue_mohr | you should need it |
| 22:43.13 | rue_mohr | er sholdn't |
| 22:43.25 | rue_mohr | but echocans aren't to bad on ebay either |
| 22:43.27 | rue_mohr | I have two |
| 22:43.34 | rue_mohr | dont think I'll need them |
| 22:43.53 | rue_mohr | k, I |
| 22:44.01 | rue_mohr | m gonna go do yardwork for a while |
| 22:44.19 | rue_mohr | I got a new angle grinder and I'm dying to try it out |
| 22:44.44 | *** join/#asterisk zoa (i=zoa@69-94-204-177.biltmorecomm.com) |
| 22:44.47 | zoa | yo yo |
| 22:44.51 | psi0n | i've been bashing my brains out trying to figure out why cell phones get messed up MoH and ringing tones when calling my PBX |
| 22:44.58 | Hymie | ankle grinder! |
| 22:45.00 | Hymie | wtf! |
| 22:45.13 | JT | Hymie: he said angle grinder, not ankle grinder |
| 22:45.18 | JT | Hymie: need new glasses? :) |
| 22:45.38 | psi0n | lol |
| 22:46.01 | [AST_LANTAzoa] | anybody else arrived alrady |
| 22:46.02 | [AST_LANTAzoa] | ? |
| 22:46.07 | JT | [AST_LANTAzoa]: ? |
| 22:46.16 | [AST_LANTAzoa] | astricon devcon @ atlanta |
| 22:46.22 | Qwell | astlanta...nice |
| 22:46.49 | [AST_LANTAzoa] | qwell you here yet ? |
| 22:46.51 | Qwell | [AST_LANTAzoa]: the Digium folks arrive tomorrow |
| 22:47.10 | [AST_LANTAzoa] | do you know of anyone who comes early ? |
| 22:47.15 | [AST_LANTAzoa] | i think olle should be around |
| 22:52.03 | [AST_LANTAzoa] | omg |
| 22:52.10 | [AST_LANTAzoa] | this is going to be hell |
| 22:52.19 | [AST_LANTAzoa] | my collegue has a little snoring problem |
| 22:52.26 | [AST_LANTAzoa] | im now with earplugs |
| 22:52.31 | [AST_LANTAzoa] | active noise cancelling headsets |
| 22:52.37 | [AST_LANTAzoa] | and music |
| 22:52.41 | [AST_LANTAzoa] | and i can still hear him |
| 22:52.43 | [AST_LANTAzoa] | omfg |
| 22:55.14 | *** join/#asterisk kusznir (n=kusznir@66-233-138-60.lew.clearwire-dns.net) |
| 22:55.27 | JT | [AST_LANTAzoa]: please /join #omg |
| 22:55.30 | JT | :P |
| 22:56.33 | kusznir | Hi all: I've got some protocall-related questions. I've been using asterisk with IAX to a few providers, but recently tried to add SIP. My phone calls via SIP sound like garbage (I get a second or two that is almost clear, then "mr. roboto". The same call to the same provider using IAX2 works great, though. |
| 22:56.49 | JT | prococall? ;) |
| 22:57.11 | kusznir | Sorry...I'm really bad at spelling; I'll work on that. |
| 22:57.11 | JT | maybe it's using a bad codec |
| 22:57.19 | kusznir | I'm using ulaw for both calls. |
| 22:57.38 | JT | is the sip call definitely negotiating ulaw? |
| 22:57.50 | kusznir | Yep.. |
| 22:58.03 | kusznir | Watched the call progress on asterisk. |
| 22:58.16 | JT | with sip debug? |
| 22:58.22 | kusznir | (and if my device was transcoding, the cpu power consumed would be noticable) |
| 22:58.55 | kusznir | I was doing verbose 4 or 5, and I see "attempting native bridge", and my phone (granstream gxp2000) shows ulaw. |
| 22:59.00 | *** join/#asterisk Mad|Cow (n=thirt@74.92.109.205) |
| 22:59.05 | JT | perhaps you have some sort of packet shaping scheme happening on RTP ports between you and your ITSP |
| 22:59.37 | JT | so you didn't actually debug the sip call to the provider? the leg to the grandstream could be a different codec |
| 23:00.25 | kusznir | I run a router (openwrt), and I've tried both with qos on (and trying to help rtp) and qos off, so it would be the provider. Of course, they disavow doing any such thing.... |
| 23:00.54 | JT | tried running a softphone direct to the provider? |
| 23:01.09 | kusznir | I didn't actually debug. But asterisk is running on my OpenWRT, and I don't have any trancoding support available on it. I can do a sip debug, though. |
| 23:01.28 | kusznir | Oh, I've also used a budget tone direct to a provider, and had the same issue. |
| 23:01.40 | JT | isn't that saying something? :) |
| 23:01.42 | kusznir | ulaw was the preferred codec on the HT config. |
| 23:01.49 | JT | althought a budget tone is a poor testing tiool |
| 23:02.17 | *** join/#asterisk DrukenLPY (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:02.28 | kusznir | Yea, I knew the problem wasn't with asterisk, I was just trying to figure out what the problem could be; e.g., does IAX use less bandwidth, or is more tolerant of jitter, etc. |
| 23:02.49 | *** join/#asterisk oej (n=olle@65-182-39-213.cre.bil.biltmorecommunications.net) |
| 23:02.52 | kusznir | Yea...It wasn't really used directly as a testing tool, it was more like the first one to discover the problem :) |
| 23:03.12 | *** join/#asterisk cspot (i=cspot@ip68-1-63-100.pn.at.cox.net) |
| 23:03.14 | kusznir | (until that, I ran only IAX out of my network) |
| 23:03.25 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 23:04.14 | kusznir | Well, if anyone here is considering using ClearWire as an ISP and wants to do any VoIP over it, find another provider.... |
| 23:04.28 | kusznir | Thanks for your help JT. |
| 23:04.34 | JT | try xlite |
| 23:04.41 | JT | budget tones are bad phones |
| 23:05.20 | kusznir | Is the gxp2000 considered "reasonable"? |
| 23:05.25 | JT | iax2 uses a single udp port for signalling and media, sip uses a port for signalling, and chooses 2 ports from a range of udp ports for media over rtp |
| 23:05.28 | JT | no |
| 23:05.30 | JT | ~gs |
| 23:05.31 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 23:05.41 | JT | ~phones |
| 23:05.43 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 23:06.32 | fujin | some idio chose Mitel phones here |
| 23:06.35 | fujin | they're terrible so far ;| |
| 23:07.53 | JT | yeah? |
| 23:07.57 | [TK]D-Fender | fujin, I've heard somre really good things about a number of Mitel models, but hav no personal experience. They seem pricey and the brand makes me nervous though. |
| 23:08.01 | JT | heard someone saying they were good |
| 23:08.05 | JT | but not many people use them |
| 23:08.19 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:09.22 | RypPn | I'm using an atcom 520 here, sounds pretty good |
| 23:09.27 | RypPn | 530* |
| 23:10.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:11.18 | [TK]D-Fender | atcom = *shudder*. PA166(8) = POS. |
| 23:14.26 | RypPn | hmm, this one has Infineon chipset, is that different? |
| 23:16.44 | fujin | o0oh, the mitel just got surprisingly better |
| 23:16.49 | fujin | so, does anyone know about record_in/out? |
| 23:17.53 | fujin | basically the system I've inherited requries record_in=Adhoc, record_out=Adhoc to get anything to work - I can't find any docs on it |
| 23:17.57 | fujin | it's not in tfot |
| 23:17.59 | fujin | ~record_in |
| 23:18.50 | psi0n | does anyone know if the ringtone on asterisk is a soundfile or if it is generated? |
| 23:19.49 | JT | generated by a tone generator |
| 23:20.05 | psi0n | thanks |
| 23:21.37 | RypPn | anyone tried the utstarcom f3000 at all? I'd appreciate some feedback before I buy as it's quite expensive. |
| 23:21.56 | RypPn | reviews on the net have been a bit mixed |
| 23:22.03 | JT | wireless? |
| 23:22.09 | RypPn | yeah, its wifi |
| 23:22.10 | JT | all wifi voip phones suck |
| 23:22.13 | JT | to varying degrees |
| 23:22.33 | remmo | why use wifi voip when one can use a wireless DECT handset ? |
| 23:23.36 | RypPn | can you give me a couple of examples please? so I can check them out |
| 23:24.06 | [TK]D-Fender | RypPn, |
| 23:24.09 | [TK]D-Fender | ~wifisip |
| 23:24.11 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 23:24.47 | RypPn | k, thats saved me 140 quid then |
| 23:24.59 | RypPn | cheers :) |
| 23:25.07 | [TK]D-Fender | fujin, Go read up on "monitor", and features.conf on the WIKI for recording. |
| 23:25.10 | [TK]D-Fender | ~wikis |
| 23:25.13 | jbot | i guess wikis is http://www.voip-info.org |
| 23:25.25 | psi0n | when i call in to my pbx, i get an IVR, which forwards me to the extension of my choice. but when im calling in using a cell phone, when it forwards me i hear the first ringtone chopped into two bits, then complete silence. |
| 23:25.39 | JT | RypPn: better off with a dect phone or similar |
| 23:25.57 | *** join/#asterisk znoG (n=gs@235-180-235-201.fibertel.com.ar) |
| 23:25.58 | RypPn | can you recommend any? or are they all much the same? |
| 23:26.02 | kusznir | JT, remmo: what is a dECT phone? |
| 23:26.20 | JT | a cordless phone complying with the dect standard |
| 23:26.20 | kusznir | ~dect |
| 23:26.22 | jbot | Digital European Cordless Telecommunications (telecommunication) [source: V.E.R.A.] |
| 23:26.45 | kusznir | So is that the same as a regular cordless phone in the US? |
| 23:26.45 | JT | there are some sip phones that have dect handset options, otherwise just connect them with an fxs port |
| 23:27.08 | JT | a regular cordless phone... that is dect compliant |
| 23:27.17 | remmo | correct. |
| 23:27.18 | JT | it specifies a digital transmission standard |
| 23:27.34 | JT | i never recommend old shitty analogue modulation cordless phones |
| 23:27.37 | JT | no over the air security |
| 23:27.43 | remmo | JT: do you know if DECT is secure? |
| 23:27.53 | JT | relatively |
| 23:27.57 | kusznir | Does the DECT compliance give me the ability to use some form of AP or repeater network to cover entire buildings and allow a bunch of them to run in the same space? |
| 23:28.04 | JT | i wouldn't use it for national secrets |
| 23:28.29 | JT | yes, commercial dect phones can have multiple base stations and do handover i believe |
| 23:28.39 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:28.44 | fujin | I've got two basestations in my house, DECT |
| 23:28.50 | JT | you can get dect options for proprietary pbxes |
| 23:36.55 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 23:36.55 | [hC] | yeah, they do seamless handoff |
| 23:37.45 | *** join/#asterisk DrukenHME (n=jdumais@CPE001346f4961f-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 23:37.45 | kombi | been on this for hours now, just can't get this 7941 to work, what might be the pitfall? |
| 23:38.06 | kombi | is dhcp needed? |
| 23:38.21 | [hC] | where are you stuck? |
| 23:38.34 | *** join/#asterisk burt75 (n=burt@189.157.128.236) |
| 23:39.49 | kombi | hc: quite the beginning actually, phone plugged in, skinny.conf tweaked but the phone just show either "configuring CM-List" or "Registering" |
| 23:40.08 | kombi | ..and never gets anywhere |
| 23:41.31 | [hC] | kombi: you'll want to use SIP firmware. |
| 23:41.41 | [hC] | I dont have a copy, and its not a free download unfortunately |
| 23:41.51 | [hC] | but if you search google or find someone maybe they can give you a copy |
| 23:42.00 | [hC] | Anyone know if the polycom phones support LLDP? |
| 23:42.02 | kombi | [hC] so I hear.. |
| 23:42.29 | *** join/#asterisk b1shop (n=b1shop@c-76-16-224-140.hsd1.il.comcast.net) |
| 23:42.56 | kombi | [hC]: would .. maybe even you have one? |
| 23:43.13 | kombi | [hC]: googled my eyes out already.. |
| 23:44.09 | [hC] | kombi: i do, but not here... unfortunately... sorry |
| 23:44.29 | kombi | [hC]: no problem.. sigh.. |
| 23:44.31 | b1shop | <-- moving office in a month and i wanted to test asterisk to replace my nortel pbx. whats the simplest way to run some tests? trixbox, asterisk now? |
| 23:44.45 | b1shop | i am assuming that standard modems do not work. |
| 23:45.52 | [TK]D-Fender | b1shop, What's to "test"? *'s capabilities are well known. What is it you wish to know to help you make up your mind? |
| 23:46.45 | b1shop | [TK]D-Fender: i've never used it and i am no phone guru. want to replace the nortel system cause i'm sick of paying some dude $100/hr to add an extension or make a change |
| 23:47.37 | JT | i think he wants to learn |
| 23:47.40 | b1shop | [TK]D-Fender: it'd be nice if i could set up a test on my fax line (only line not behind the pbx) and run some tests, create IVR prompts, test vm-->email etc |
| 23:48.44 | b1shop | i'd hate to buy 10 phones and a card and run into issues. |
| 23:49.12 | [TK]D-Fender | b1shop, well in short, IVR's, VM (to e-mail, tec), dialplans, queues, all that "work"'s and you just need to see if it fits your budget and time frame (for learning). |
| 23:49.57 | [TK]D-Fender | b1shop, Getting good hardware is the key to avoiding issues. No cheap card you use in tests will make you happy in production and would likely jsut dissapoint you in testing whereas the proper product wouldn't |
| 23:50.05 | [TK]D-Fender | b1shop, Bit of a catch-22 |
| 23:50.08 | b1shop | plus. what's the best distro. start from a base centos (or similiar distro) and build it out or use one of the other distros |
| 23:50.14 | [hC] | b1shop: I am really happy with the Sangoma A200d cards |
| 23:50.25 | [hC] | they expand up to 24 channels, too |
| 23:50.31 | [hC] | er |
| 23:50.34 | [hC] | 16 |
| 23:50.39 | [TK]D-Fender | b1shop, best distr is whichever you're most comfortable administering where you can satisfy *'s pagage dependencies. |
| 23:50.50 | JT | [hC]: 24 |
| 23:51.07 | b1shop | [TK]D-Fender: understood.. if i went full scale i would of course buy good hardware.. i'm more interested in testing out the administration |
| 23:51.12 | [TK]D-Fender | b1shop, I personally use CentOS & Slackware in my installs and both work wonderfully straight out. |
| 23:51.13 | [hC] | JT: I thought it only did 3 extra daughterboards? |
| 23:51.36 | JT | [hC]: that's 24 chans, is it not? |
| 23:51.45 | [hC] | JT: at 4 lines per card, 4 cards total. 4x4 = 16 |
| 23:51.51 | [TK]D-Fender | b1shop, I am presuming you know nothing about * setup so far. The first hurdle is the learning curve. What is your deadline? |
| 23:52.10 | b1shop | [TK]D-Fender: moving in about a month! ;-) |
| 23:52.33 | Swat2 | b1shop: theres a good document called "trixbox without tears" that might be able to help you |
| 23:52.42 | Swat2 | it's a little outdated |
| 23:52.46 | Swat2 | but covers most things |
| 23:52.51 | [hC] | theres a new one out for 2.x i saw.. |
| 23:52.51 | [TK]D-Fender | b1shop, How many phones / lines in your projected setup? |
| 23:52.54 | JT | As you need them, additional REMORA. cards can be added to the base four-port A200 card. A single PCI or PCI Express slot hosts connection for up to 24 ports and ensures common synchronous clocking for all channels. |
| 23:52.55 | b1shop | googleing now |
| 23:52.56 | burt75 | someone know something in chan_cellphone or chan_mobile ? |
| 23:53.04 | burt75 | I cant pair BT phone |
| 23:53.07 | [hC] | I find it pretty amusing that trixbox is created to be "Asterisk with out tears" then someone had to go make a "trixbox without tears" |
| 23:53.29 | b1shop | [TK]D-Fender: 8-10 phone + reception and 5 lines + fax i think would be enough to get us started |
| 23:53.36 | JT | [hC]: so it is expandable to 24 ports |
| 23:53.42 | Swat2 | [hC] irony ;) |
| 23:53.42 | JT | the A400 is up to 48 |
| 23:53.52 | [hC] | JT: where am i losing my mind then... there must be the ability to add more than 3 remora cards |
| 23:54.06 | [hC] | JT: it must take up to 7 additional |
| 23:54.10 | JT | [hC]: max config of 6 cards |
| 23:54.21 | JT | 24/6=4 |
| 23:54.27 | [hC] | yeah |
| 23:54.30 | [hC] | ok |
| 23:54.39 | [hC] | thought it only took up to 3 remora's |
| 23:54.40 | b1shop | [TK]D-Fender: i'm really just looking for a small scale test now. soft phone or cheap card to do some sandboxing |
| 23:54.41 | [hC] | nevermind! :) |
| 23:54.44 | JT | :) |
| 23:54.46 | [TK]D-Fender | b1shop, that in mind, getting * setup and ready is NOT a big issue. Its that you'll either spend a good long while seriously concentrating on learning *, or pay a consultant to set up your initial system, pay for some basic training, and then pick up your education on your own after tha. |
| 23:55.00 | Swat2 | b1shop: try a TDM400P, i personally have had no troubles |
| 23:55.11 | [TK]D-Fender | b1shop, No need ofr any hardware jsut to get to learn *. *nix box to install on, and a softphone will do. |
| 23:55.16 | [hC] | I have :( however, that was a year ago |
| 23:55.35 | [hC] | Have any of you guys had experience installing on Dell hardware? |
| 23:55.47 | b1shop | [TK]D-Fender: i've allocated a P4 3ghz, 1GB ram for testing. that should be plenty |
| 23:55.47 | JT | yes |
| 23:55.53 | [hC] | I have an interesting amount of quality issues even w/ sangoma on dell hardware, and im trying to figure out if its the dell's fault. |
| 23:55.55 | [TK]D-Fender | b1shop, indeed plenty |
| 23:56.08 | Swat2 | Anyone know how to force tha callerid Name to be the same as the callerid number? |
| 23:56.15 | [TK]D-Fender | [hC], where I presume an analog phone right on the line sounds perfect? |
| 23:56.20 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:56.23 | [hC] | [TK]D-Fender: yep. |
| 23:56.26 | b1shop | [TK]D-Fender: standard modem cards do not work correct? basically anything will do for a small test |
| 23:56.35 | [hC] | [TK]D-Fender: things like tromboning, echo, dropped calls, calls that dont get answered... |
| 23:56.40 | [TK]D-Fender | Swat2, Set(CALLERID(name)=${CALLERID(num)}) |
| 23:56.43 | [hC] | staticy, etc.. |
| 23:56.48 | [hC] | seems totally random. |
| 23:56.57 | kusznir | b1shop: nope, most modem cards won't work, and the few that are compatable have a lot of issues. |
| 23:57.03 | lee_is_me | Hey these Aastra's are pretty nice... |
| 23:57.06 | Swat2 | [TK]D-Fender: i'm having troubles finding where i should put that... |
| 23:57.06 | [hC] | I am actually wondering if the onboard nic has something to do with it, ive read a bunch of bad forum posts on the onboard dell broadcom nic. |
| 23:57.13 | [TK]D-Fender | [hC], OUCH. Tried different wanpipe / zaptel revisions? What card exactly? |
| 23:57.28 | [TK]D-Fender | [hC], Also is it sharing an interrupt? |
| 23:57.32 | [hC] | [TK]D-Fender: The a200d. Tried multiple asterisk/wanpipe/zaptel over the past year |
| 23:57.39 | [hC] | [TK]D-Fender: really hard to tell, cause the problems seem to come and go. |
| 23:57.50 | [TK]D-Fender | [hC], EEK |
| 23:57.52 | [hC] | [TK]D-Fender: I also have 3 boxes that like to kernel panic every 2 weeks because of the sangoma and no idea why. |
| 23:58.06 | [hC] | [TK]D-Fender: yeah, analog installs are my worst nightmare. they never seem to go smooth, when you'd expect they would. |
| 23:58.09 | [TK]D-Fender | [hC], Ok, go work it out with their tech support.... |
| 23:58.24 | [TK]D-Fender | [hC], You could jsut have a flakey card... |
| 23:58.32 | [hC] | [TK]D-Fender: yup. Ive seen this in about 6 installs now. |
| 23:58.37 | [hC] | all different hardware. |
| 23:58.43 | [hC] | other than they are running in a dell. |
| 23:58.43 | kusznir | I'd recommend testing in isolation in the beginning, then possibly adding an external IP-based provider (asterlink is a good deal for a testing setup; I also use vitelity.net |
| 23:59.06 | [hC] | [TK]D-Fender: probably gonna have to gather a ton of data and have a serious talk w/ sangoma. |
| 23:59.07 | kusznir | Its a lot cheaper and 95% of the learning will work well with them and some soft phones. |
| 23:59.13 | lee_is_me | [hC]: I had problems with a Dell onboard nic recently, not sure if it was broadcom |
| 23:59.24 | [hC] | lee_is_me: what was the symptom? |