| 00:01.07 | *** join/#asterisk RoyK (~roy@110.80-203-29.nextgentel.com) |
| 00:03.13 | harryvetch | well, found the last error now pstn to xlite is working. yea..and i have some echo |
| 00:04.10 | harryvetch | I guess its not unusuall to have echo for these cards? |
| 00:04.30 | _Vile | use mark3 echo cancellation |
| 00:04.34 | _Vile | or mark2 |
| 00:04.45 | *** join/#asterisk VoIPMasta (~John@201.133.111.242) |
| 00:04.50 | VoIPMasta | Hi |
| 00:04.59 | _Vile | ~google ECHO_MARK3 |
| 00:05.04 | VoIPMasta | Has anyone set up a Cisco ATA-186 (SIP) with Asterisk? |
| 00:05.08 | _Vile | ~google ECHO_MARK2 |
| 00:05.15 | _Vile | shit |
| 00:05.20 | *** join/#asterisk scubasteve (~steve@rdu88-248-113.nc.rr.com) |
| 00:05.35 | _Vile | scuba, what's the config file for setting mark2 echo cancellation? |
| 00:05.43 | scubasteve | huh? |
| 00:05.48 | stevekstevek | ECHO_CAN_MARK2 |
| 00:05.52 | _Vile | yeah |
| 00:05.56 | stevekstevek | ~google ECHO_CAN_MARK2 |
| 00:05.59 | _Vile | ~google ECHO_CAN |
| 00:06.01 | _Vile | u got it |
| 00:06.18 | _Vile | check out http://www.voip-info.org/wiki-Asterisk+echo+cancellation |
| 00:06.20 | stevekstevek | Makefile:# Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :) |
| 00:06.44 | _Vile | mark2 worked for me in the past but i'm on PRI now, so I dont bother |
| 00:06.44 | stevekstevek | that would be Steve U. |
| 00:06.55 | _Vile | underwood :) |
| 00:07.32 | zoa | hey krambo!!!! |
| 00:07.47 | scubasteve | burrrrrp!!! |
| 00:07.54 | harryvetch | Thanks for the echo information. |
| 00:08.54 | *** join/#asterisk rustyb (~rustyb@adsl-8-237-29.mia.bellsouth.net) |
| 00:09.03 | _Vile | VoIPMasta, yes |
| 00:09.58 | VoIPMasta | _Vile I'm getting a lot of error msgs, do you have a couple of minutes? |
| 00:10.21 | _Vile | not really but ask your question about a particular message |
| 00:10.30 | zoa | what does bbiab mean? |
| 00:10.37 | _Vile | be back in a bit |
| 00:10.39 | zoa | be back in a bbbbbb ? |
| 00:10.39 | VoIPMasta | The message is file.c:548 ast_readaudio_callback: Failed to write frame |
| 00:11.02 | _Vile | ~google ast_readaudio_callback: Failed to write frame |
| 00:11.10 | VoIPMasta | already did that |
| 00:11.11 | zoa | aaah |
| 00:11.31 | _Vile | kk what phone? |
| 00:11.37 | _Vile | err ata-186 k |
| 00:11.47 | VoIPMasta | yup, ata 186 |
| 00:12.06 | _Vile | k what's the next message |
| 00:12.31 | VoIPMasta | chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 1050153554@201.133.111.242 for seqno 2 (Non-critical Response) |
| 00:12.50 | zoa | aaah my favorite error :) |
| 00:12.52 | _Vile | Maximum retries is an OK message, expect it... next? |
| 00:13.11 | _Vile | I wouldn't worry about Failed to write frame either |
| 00:13.14 | VoIPMasta | the call ends |
| 00:13.29 | _Vile | no more? |
| 00:13.30 | VoIPMasta | it really ends after the first message |
| 00:13.31 | VoIPMasta | nope |
| 00:13.32 | VoIPMasta | no more |
| 00:13.36 | _Vile | dont worry about it |
| 00:14.09 | _Vile | it shouldn't be affecting * or the ata |
| 00:14.21 | _Vile | so I wouldn't have even bothered to spend the time asking the question |
| 00:14.28 | _Vile | much less researching it |
| 00:14.37 | VoIPMasta | http://pastebin.ca/5184 |
| 00:14.52 | VoIPMasta | but the call stops there |
| 00:14.53 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 00:14.58 | VoIPMasta | I'm trying to dial the echo-test |
| 00:15.33 | _Vile | not working? |
| 00:15.39 | VoIPMasta | nope |
| 00:15.58 | _Vile | can you dial anything else from the ata? |
| 00:16.32 | rustyb | goodevening ;-) |
| 00:16.47 | DaLion | ~jbot PRI |
| 00:16.48 | jbot | i heard pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 00:17.56 | VoIPMasta | _Vile when I dial a phone number * starts ringing but as soon as I pick up the other phone (cell phone) the call ends |
| 00:18.24 | _Vile | type sip debug into the cli and post that to pastebin.ca |
| 00:18.30 | *** join/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
| 00:18.30 | _Vile | and make a couple of calls |
| 00:18.32 | david | hello |
| 00:18.39 | david | is there a version of zaprtc which builds on 2.6? |
| 00:18.42 | BoRiS | Codec translation problems VoIPMasta? |
| 00:18.48 | _Vile | sounds like it |
| 00:18.50 | BoRiS | david: Yup....Its called ztdummy |
| 00:19.13 | VoIPMasta | BoRiS but I don't get any codec-related messages |
| 00:19.22 | _Vile | have you debugged sip? |
| 00:19.23 | david | BoRiS, doesn't that need a uhci usb controller? |
| 00:19.23 | BoRiS | VoIP: Hmmm |
| 00:20.02 | BoRiS | david: Nope, that was a 2.4 thing. Simply edit Makefile and uncomment "# ztdummy".. then make clean; make linux26; depmod -a; modprobe ztdummy |
| 00:20.12 | rustyb | is anyone familiar with the timing config when using multiple t100 spans |
| 00:20.18 | tzanger | rustyb: yup |
| 00:20.29 | tzanger | choose your telco as timing of '1' and everything else to '0' |
| 00:20.35 | tzanger | oh wait |
| 00:20.39 | david | BoRiS, hrm, okay - I need to kill off RTC in my kernel though, right? |
| 00:20.40 | tzanger | multiple t1000s? |
| 00:20.43 | tzanger | they each have their own timer |
| 00:20.50 | david | BoRiS, I've got ztdummy loaded, but music on hold still isn't working |
| 00:20.53 | david | BoRiS, but con calling is |
| 00:20.53 | rustyb | thats what I'm not sure of |
| 00:21.03 | VoIPMasta | now I'm getting rtp.c:1114 ast_rtp_raw_write: RTP Transmission error to 192.168.2.188:16384 |
| 00:21.09 | rustyb | I want to slave from the 1st span telco |
| 00:21.17 | tzanger | rustyb: so do it then |
| 00:21.20 | VoIPMasta | I removed the NATIP param from my ATA |
| 00:21.31 | BoRiS | david: rtc is not needed for moh..... mpg123 or the mp3 codec in the addons will allow moh |
| 00:21.39 | _Vile | tzang, speaking of which, I have 2 pri's on a t400 from two different providers.. a setting of 1 each should work, or should I just choose one timing source and allow the provider to use me as the timing source? |
| 00:21.41 | rustyb | and be the reference clock for the 2nd span a tie to an older pbx |
| 00:21.41 | VoIPMasta | because I saw a line in the debug indicating that there was no nat to that IP |
| 00:22.13 | tzanger | rustyb: so have it set to 0 and have the older pbx provide clock |
| 00:22.17 | david | BoRiS, okay, well, I don't get 'starting moh on iax2/blahblah' |
| 00:22.33 | david | BoRiS, how do I begin to debug that? |
| 00:22.56 | BoRiS | hmmm, do you have mpg123 (not mpg321) installed? |
| 00:23.15 | rustyb | when it is set to 0 are we providing clock or slaving to the far end? |
| 00:23.30 | _Vile | VoIP, firewall? |
| 00:23.46 | david | BoRiS, yup |
| 00:23.50 | _Vile | and nat=yes in SIP? |
| 00:23.51 | VoIPMasta | _Vile yes, but I have 3 siphones running without a prob |
| 00:23.54 | _Vile | or nat=no? |
| 00:23.55 | BoRiS | What version david? |
| 00:23.55 | VoIPMasta | yes, nat=yes |
| 00:23.57 | harryvetch | man this echo is bad heheh |
| 00:23.57 | david | BoRiS, I just figured it out - Needed to pass the 'M' option to Meetme() in my extensions.cofn |
| 00:24.05 | ManxPower | If anyone has recommendations for a SMALL and CHEAP web store service, please /msg me |
| 00:24.12 | david | ..................................duh |
| 00:24.30 | BoRiS | lol, ok |
| 00:24.52 | _Vile | canreinvite=no? |
| 00:24.59 | david | BoRiS, easy fix, I guess :-) |
| 00:25.01 | _Vile | is your phone external? |
| 00:25.10 | VoIPMasta | yup, canreinvite=no |
| 00:25.21 | VoIPMasta | _Vile what do you mean by "external"? |
| 00:25.29 | _Vile | external IP address |
| 00:25.39 | _Vile | traversing the public ip network |
| 00:25.51 | _Vile | into an internal asterisk IP address |
| 00:25.58 | VoIPMasta | the other way |
| 00:26.00 | _Vile | or vice versa |
| 00:26.04 | VoIPMasta | the * is in the public internet |
| 00:26.08 | VoIPMasta | the ATA is in my LAN |
| 00:26.09 | _Vile | ok |
| 00:26.13 | _Vile | you have nat problems then |
| 00:26.16 | BoRiS | david: I wasn't sure you had it working in the first place. :-p |
| 00:26.26 | david | BoRiS, me neither |
| 00:26.28 | BoRiS | david: had to start at square one :-p |
| 00:26.53 | rustyb | Notice[1147] chan_zap.c:7388 pri_dchannel: Pri got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 00:26.56 | _Vile | nat=yes, how are you handling the nat settings on your ata? |
| 00:27.22 | rustyb | <tzanger> this is the error that's been scrolling |
| 00:27.36 | VoIPMasta | _Vile: I'm trying using a STUN server |
| 00:27.49 | _Vile | and can you please do a sip debug and dump the results of a phone call to pastebin.ca |
| 00:27.51 | _Vile | ? |
| 00:28.03 | VoIPMasta | sure |
| 00:28.09 | VoIPMasta | but I won't be able to get it complete |
| 00:28.13 | _Vile | i don't use stun, so someone else may have to help you there |
| 00:28.15 | Mavvie | rustyb: I see one of them all the time. |
| 00:28.16 | VoIPMasta | because the screen scrolls too fast |
| 00:28.23 | _Vile | ahh |
| 00:28.24 | rustyb | <tzanger> at the same time voice on the 2nd span goes bad |
| 00:28.33 | _Vile | too many of those frame errors? |
| 00:28.36 | david | BoRiS, thanks for your help anyway - That'll teach me to check my configs before trying to fix a problem that doesn't exist |
| 00:29.02 | Mavvie | rustyb: no idea why, it happens during day and night (idle and busy) |
| 00:29.36 | _Vile | :) someone can help you w/ stun, but those errors don't help me at all.. and shouldn't hurt your ability to place a call, seems like a nat problem w/ returning packets back to your ata or vice versa |
| 00:29.55 | rustyb | mavvie: yes fairly regular about every 10 min for about 15 sec. |
| 00:30.03 | _Vile | probably a problem w/ what ports you have open |
| 00:30.15 | VoIPMasta | however there are other sip phones in my lan |
| 00:30.18 | _Vile | or, with the return ip address you have set in your ata for nat |
| 00:30.19 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 00:30.20 | VoIPMasta | and all of them are working fine |
| 00:30.28 | _Vile | any other ata's though? |
| 00:30.50 | rustyb | does it seem like T1 timing or some other resource problem? |
| 00:31.06 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 00:31.19 | _Vile | rusty, I'm having similar problems -- it's probably timing |
| 00:31.30 | _Vile | or signalling |
| 00:31.31 | DaLion | ~jbot pri |
| 00:31.32 | jbot | pri is, like, Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 00:31.33 | VoIPMasta | I don't understand... |
| 00:31.37 | _Vile | if it was signalling |
| 00:31.38 | VoIPMasta | now it's working... |
| 00:31.48 | _Vile | then you may not be able to start * |
| 00:31.49 | _Vile | so |
| 00:31.51 | _Vile | *shrug* |
| 00:31.53 | _Vile | probably timing |
| 00:32.12 | _Vile | voip, you're welcome, have a nice day. :) |
| 00:32.20 | VoIPMasta | I did get some "RTP Transmission error to" messages |
| 00:32.23 | VoIPMasta | but the call got connected |
| 00:32.25 | VoIPMasta | :s |
| 00:32.25 | rustyb | i thought timing cause both spans go to shit at the same time |
| 00:32.40 | _Vile | i wouldn't worry much about the rtp messages. |
| 00:32.41 | rustyb | and figured 1 was the clock reference for the other |
| 00:32.55 | VoIPMasta | _Vile: thank you very much, I will try to just add some echo cancelation as there is too much of it |
| 00:32.56 | _Vile | im curious about that too, rusty |
| 00:33.16 | rustyb | but could be something else common |
| 00:33.22 | _Vile | check timing, use one source, have your provider prefer you as the timing source, set your timing to 0 on that span |
| 00:33.50 | *** join/#asterisk porkchop_ (~porkchop@porkchop.nat.cccp.porkchop.net) |
| 00:33.55 | *** join/#asterisk wm501 (~wm501@wsip-70-182-68-99.ok.ok.cox.net) |
| 00:34.04 | rustyb | i'm unclear what the 1,2,3 timing options really mean |
| 00:34.11 | _Vile | 3 is secondary |
| 00:34.21 | _Vile | 0 is master I think |
| 00:34.28 | _Vile | 1 is use this as a primary clocking source |
| 00:34.35 | _Vile | or 2 may be secondary |
| 00:34.40 | wm501 | Hello, I have a radio show that goes on the air in 30 minutes and we are trying to get asterisk to put 4 people on hold that call in and then we can pull them into our meetme conference 101, 1 at a time, any suggestions on how i could go about this? |
| 00:34.42 | _Vile | I only know of 0, 1, 2? |
| 00:34.46 | rustyb | my provider says they have to be the timing source. and are asking me to use them as external reference or slave |
| 00:34.54 | porkchop_ | Lo, all. I was trying to get voicemail trees working... I have dmtf set to inband (what my SIP provider wants), and my codec does support inband DTMF (ulaw), but it dosnt work. It does work locally (another voip phone). Any ideas? |
| 00:35.03 | rustyb | so is that done with a 1 or 0? |
| 00:35.05 | _Vile | two providers, rusty? |
| 00:35.08 | _Vile | 1 |
| 00:35.18 | _Vile | use a 1 |
| 00:35.20 | rustyb | oops sorry yes 0,1,2 |
| 00:36.25 | rustyb | yes 2 spans. 1 is my provider the 2nd is a tie to another pbx |
| 00:36.44 | rustyb | the other pbx is expecting the span to provide clock |
| 00:36.50 | _Vile | rusty, use 1 for your provider, 0 for your pbx |
| 00:36.55 | *** join/#asterisk DaGrim (~jason@dagrim.user) |
| 00:37.07 | rustyb | so i need to slave from by provider on 1 and source to the 2nd span |
| 00:37.13 | _Vile | yep |
| 00:37.24 | rustyb | -Vile: ok thanks |
| 00:39.31 | wm501 | can anyone help me with my question, if not that is cool, but I am literally lost. |
| 00:40.48 | ManxPower | wm501, Not in 30 mins we can't. |
| 00:40.48 | eKo1 | You won't be doing that in thirty minutes. I can guarantee that. |
| 00:40.54 | _Vile | wm501, you |
| 00:41.03 | DaLion | Anyone got pri experience in canada ? |
| 00:41.08 | _Vile | ll be looking at a day or two of help and development |
| 00:41.39 | Nukemizer | Can anone help me in setting up the S100I there were no instructions and I can not find info where i have been googling |
| 00:42.07 | ariel_ | wm501, park the calls then pick each one up and transfer them to the meetme you want. |
| 00:43.05 | _Vile | yep |
| 00:43.08 | ariel_ | pri in canada is about the same as far as I know like the states. |
| 00:43.08 | DaGrim | well Im done trying to use * to make money.. aint gonna happen.. anything i try ill end up loosing money on.. .. isnt worth it. |
| 00:43.11 | VoIPMasta | _Vile: do you know the parameter for GSM on ATA 186? |
| 00:43.39 | DaGrim | Its an awesome toy .. beyond that tho. |
| 00:43.40 | wm501 | okay |
| 00:43.43 | ariel_ | DaGrim, sorry to hear it. (I am making a living with asterisk). |
| 00:43.56 | *** join/#asterisk imcdona (imcdona@49-139.175-24.bham.rr.com) |
| 00:44.04 | _Vile | voip, should be on the codec list amongst ulaw, alaw, 729 etc |
| 00:44.05 | imcdona | Anyone need a gmail account? |
| 00:44.06 | wm501 | ariel_: okay, i'm looking at the asterisk wiki. i don't see how to park the calls like that. |
| 00:44.35 | VoIPMasta | _vile: I know how to set it in asterisk, but I need to set it in the ATA |
| 00:44.36 | DaGrim | ariel_: well the only setup I could even consider attempting to start with would be prepaid.. and there competition is just to overwhelming.. I cant beat anybodys rate.. |
| 00:44.43 | DaGrim | *the |
| 00:44.46 | ariel_ | wm501, when a call comes in setup parking on the system and park the call Then just pick up the call and transfer it to a meeting you setup. |
| 00:44.56 | _Vile | voip, I've never logged into an ATA web interface, only pap |
| 00:45.15 | _Vile | but I'd suggest looking in the area in the web interface where you set the codec |
| 00:45.23 | _Vile | such as 729, alaw, ulaw etc |
| 00:45.24 | ariel_ | DaGrim, I don't setup voip service as my biz. I setup customers with asterisk server for ther biz. |
| 00:45.35 | DaGrim | ariel_: ahh right on.. |
| 00:45.47 | ariel_ | I sell it on the fact that it can save them money and pay for it's self in less then a year. |
| 00:46.01 | DaGrim | i see.. |
| 00:46.06 | rustyb | _Vile: ok just discovered whenever I switch alt-F2 to another session the dchannel errors roll. |
| 00:46.14 | brc__ | _Vile~ |
| 00:46.23 | brc__ | how was the smoke? |
| 00:46.30 | _Vile | brc, absolutely beautiful |
| 00:46.32 | ariel_ | I let the vonage, nufone vpc and broadvoice be the voip provider. I setup the asterisk systems. |
| 00:46.45 | *** join/#asterisk MrEntropy (~entropy@ppp136-158.lns1.adl2.internode.on.net) |
| 00:46.52 | _Vile | rustyb, can you do a pri intense debug on the cli and post it to pastebin.ca? |
| 00:46.52 | wm501 | thanks ariel_: i will do my best, don't think i can do it in 15 though |
| 00:46.53 | wm501 | heh |
| 00:47.15 | ariel_ | In fact I know a new voip provider that will be out in a week with great service. ( am helping them set there servers up). |
| 00:47.16 | rustyb | _Vile: ok |
| 00:47.26 | DaGrim | hmm |
| 00:47.38 | zoa | brc, go check out the http://www.astertest.com/forum/viewtopic.php?t=4 !!! |
| 00:47.52 | zoa | i need some comments! |
| 00:48.01 | _Vile | rusty, also are they (and you) in esf, b8zs mode? |
| 00:48.09 | DaGrim | Im not even trying to like get rich or anything.. I just wish I could like.. survive anymore.. ugh |
| 00:48.20 | ariel_ | wm501, it can be done it's very basic to setup meetme's and parking. it is another thing to get your dialing rules correct. |
| 00:48.37 | _Vile | make sure both widebanks are in b8zs mode, make sure zaptel.conf reflects b8zs, esf, make sure they're providing esf |
| 00:48.48 | rustyb | _Vile: yes esf,b8zs for the PRI and d4,b8zs for the em span |
| 00:48.58 | _Vile | em going to pbx? |
| 00:49.05 | ariel_ | DaGrim, where do you live? |
| 00:49.06 | rustyb | yes em to pbx |
| 00:49.12 | _Vile | zapata.conf reflects the same? |
| 00:49.13 | DaGrim | illinois |
| 00:49.17 | _Vile | ignoring the d channel? |
| 00:49.30 | rustyb | _Vile yes and signalling works well |
| 00:49.35 | _Vile | ok |
| 00:49.41 | _Vile | post me pri debug thx |
| 00:49.46 | rustyb | just freaks out every so often |
| 00:49.55 | ariel_ | well setup flyers for small biz see what the phone line rates are and start selling a small pbx to them. |
| 00:50.00 | _Vile | have you taken a telescout to it yet? |
| 00:50.00 | Duckbizkit | hey.....let's say you want to play an intro message while you dial the extension, but you want the intro message to stop just as soon as the extension picks up. what would be best for this, playback, background, moh, or what |
| 00:50.16 | _Vile | bridge taps, etc can interfere w/ a pri |
| 00:50.31 | _Vile | and they may even want to move you to a different isdn module |
| 00:50.43 | _Vile | to make sure their module isn't causing the problem |
| 00:50.49 | DaLion | montreal |
| 00:51.02 | _Vile | they'll scoff at moving you to a different module, but you're the customer. |
| 00:51.28 | _Vile | what switch type btw? |
| 00:51.49 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca) |
| 00:52.06 | *** join/#asterisk jterrero (~some@ool-43576e0d.dyn.optonline.net) |
| 00:52.16 | rustyb | i think they have a DMS, the pbx is a Harris |
| 00:52.22 | *** join/#asterisk abcbooze (abcbooze@marooned.us) |
| 00:52.31 | _Vile | harris sucks... dms, what flavor? |
| 00:52.45 | rustyb | 200 i think |
| 00:52.51 | _Vile | harris is mid 80s early 90s architecture, I'd throw it away |
| 00:52.51 | _Vile | :) |
| 00:52.52 | _Vile | k |
| 00:53.00 | _Vile | I'm having similar problems on a dms-200 |
| 00:53.08 | rustyb | right now its a big channel bannk |
| 00:53.16 | porkchop_ | I was trying to get voicemail trees working... I have dmtf set to inband (what my SIP provider wants), and my codec does support inband DTMF (ulaw), but it dosnt work. It does work locally (another voip phone). Any ideas? |
| 00:53.20 | _Vile | i'd leave it as a big channel bank :) |
| 00:53.46 | _Vile | can they do any d channel tracing for you? |
| 00:53.46 | abcbooze | hi |
| 00:53.55 | _Vile | btw who is your provider? |
| 00:54.05 | rustyb | when i switch to another session the errors jump back at me |
| 00:54.26 | rustyb | what are min server requirements fro 2 spans? |
| 00:54.30 | _Vile | can you post the error & some pri debugging info? |
| 00:54.37 | _Vile | 2 spans? |
| 00:54.49 | _Vile | what kind of a box do you have? |
| 00:54.53 | rustyb | 2 spans yes |
| 00:55.01 | _Vile | unless they're bouncing the t, you should be fine |
| 00:55.12 | ariel_ | porkchop_, what do you call voicemail trees? |
| 00:55.26 | _Vile | unless you have an amd-400 etc |
| 00:55.38 | _Vile | at which point, I'd call interrupt problems |
| 00:55.45 | _Vile | but you don't, do you? |
| 00:55.47 | *** join/#asterisk canerabbioso (~presnitz@host26-172.pool8252.interbusiness.it) |
| 00:56.22 | _Vile | besides, few calls are happening when you jump sessions |
| 00:56.26 | porkchop_ | ariel_: well I guess its just extensions... the context Background()s a audio file and if they hit #, it goes to ext # which is a Goto() |
| 00:56.40 | rustyb | _Vile: supermicro 1.6G P4 1G RAM |
| 00:56.53 | canerabbioso | any italians here??? |
| 00:57.05 | _Vile | you're fine.. the ram is over-done, I'd invest more into the cpu power.. but you're ok |
| 00:57.22 | zoa | ram is heavily overdone |
| 00:57.22 | _Vile | is it locking up any of your sessions? |
| 00:57.32 | porkchop_ | so I guess the better way to say it is... * dosnt seem to be recognising inband dtmf when coming from (in this case) santaphone, which says "use inband dtmf" and transports to me in ulaw |
| 00:57.48 | ariel_ | porkchop_, you lost me. Are you trying to make an inbound menu or when the line is busy send it to voicemail. |
| 00:57.55 | *** join/#asterisk firestrm (F1r3570rm@S010600047577bccd.gv.shawcable.net) |
| 00:58.03 | porkchop_ | ariel_: inbound menu |
| 00:58.04 | rustyb | no but when I go to the other session to check configs the spans freak |
| 00:58.18 | porkchop_ | ariel_: If you know your parties extension, dial it now. |
| 00:58.23 | _Vile | i don't understand that |
| 00:58.25 | ariel_ | ok first thing ulaw always gets inband for dtmf. |
| 00:58.44 | ariel_ | 2nd have you tested dtmf on your system to see if it works from your phones? |
| 00:58.50 | _Vile | jumping a session shouldn't cause a span to error |
| 00:58.59 | _Vile | anyone else have input? |
| 00:59.11 | porkchop_ | ariel_: local phones...err... hang on :) |
| 00:59.31 | _Vile | rusty, if it wasn't consistent, then I'd call it a coincidence, and a bouncing T |
| 00:59.57 | _Vile | *shrug* |
| 01:00.03 | rustyb | my zapata timing is backwards 0 for provider 1 to pbx. I'll fix & restart 1st |
| 01:00.04 | _Vile | any B channel restarts? |
| 01:00.05 | porkchop_ | ariel_: you're right. my local phone was not set to inband dtmf |
| 01:00.13 | _Vile | nonono |
| 01:00.15 | _Vile | 1 for provider |
| 01:00.17 | _Vile | 0 for pbx |
| 01:00.18 | _Vile | ! |
| 01:00.25 | _Vile | k |
| 01:00.28 | rustyb | _Vile: yes been having B chan restarts too but not as often |
| 01:00.28 | porkchop_ | ariel_: is there anything I need to set to have * listen for inband dtmf or is it just the obvious? |
| 01:00.33 | _Vile | ok |
| 01:00.38 | _Vile | 1 for provider |
| 01:00.41 | _Vile | 0 for pbx |
| 01:03.06 | _Vile | bbiam |
| 01:04.44 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps No such file or directory errors for all modules.... I have Fedore Core 1; any idea ???? thanks Dave |
| 01:04.49 | terrapen | fucking polycom phone |
| 01:04.50 | terrapen | 0101000519|cfg |3|00|Removed all log files due to limited space during file update. |
| 01:04.50 | terrapen | 0101000520|app1 |6|00|Error, not enough space for configuration. |
| 01:04.52 | terrapen | what the hell! |
| 01:06.23 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 01:07.07 | rustyb | _Vile: ok restarted and still have the errors |
| 01:08.17 | porkchop_ | ariel_: thanks for the idea, but no joy. Set the phone (and * on my phone's section of sip.conf) to listen for inband and it worked properly. The problem still only happens on the santaphone inbound. |
| 01:08.35 | rustyb | one t100p is sharing interrupt 5 with usb-uhci |
| 01:08.47 | dan2 | kram: ping |
| 01:08.50 | rustyb | _Vile is this a problem if there are no usb devices? |
| 01:10.49 | MrEntropy | if without answering, asterisk Hangs up the call, does the telco charge a phonecall, in other words, does asterisk first need to pick up the call to hang it up or is there a different procedure? |
| 01:11.57 | _Vile | remove IRQ sharing w/ your card in bios |
| 01:12.21 | ManxPower | If Asterisk doesn't pick up the call then Asterisk cannot hang up the call. |
| 01:13.26 | terrapen | i'm not real sure but i don't think this polycom phone is supposed to reboot over and over |
| 01:13.39 | firestrm | irq sharing with any digium card = problems.. they by their nature very irq intensive, not much left for anything else |
| 01:13.53 | terrapen | oh wait, here we go...something different |
| 01:13.54 | MrEntropy | what about if the context a call arrives on just has s,1,Hangup ? |
| 01:13.56 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps No such file or directory errors for all modules.... I have Fedore Core 1; any idea ???? thanks Dave |
| 01:14.12 | terrapen | i had to downgrade the version of the SIP firmware |
| 01:14.22 | ManxPower | MrEntropy, pretty much nothing is ACTUALLY going to happen. Hangup() should be smart enough to just not do anything. |
| 01:15.20 | *** join/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 01:17.04 | Defraz | I have an spa 2000 and I can't see to call from extention to extention but I can call my WIP-5000 that is registered from the extentions on the spa 2000 |
| 01:17.12 | Defraz | is there some setting in the spa 2000 its self. |
| 01:17.14 | Defraz | ? |
| 01:19.04 | *** join/#asterisk Rick_Hunter (~rhunter@05-138.008.popsite.net) |
| 01:19.13 | *** join/#asterisk CletusColeman (~CletusCol@c-24-0-179-254.client.comcast.net) |
| 01:19.41 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps No such file or directory errors for all modules.... I have Fedore Core 1; any idea ???? thanks Dave |
| 01:20.48 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 01:23.22 | *** join/#asterisk ionix (ionix@MTL-HSE-ppp201195.qc.sympatico.ca) |
| 01:25.52 | porkchop_ | Figured out my problem of inband dtmf not working thru santaphone. Their documentation lies. They use rfc2833, not inband. |
| 01:26.03 | tessier_ | Wow. Fax through VOIP works. |
| 01:26.04 | *** join/#asterisk lilneon (~tj_r3@200.108.22.87) |
| 01:26.10 | lilneon | hi everyone |
| 01:26.11 | tessier_ | As long as there are no errors, I'm sure. |
| 01:26.12 | tessier_ | Hi lilneon |
| 01:26.15 | JunK-Y | tessier: T38? |
| 01:26.25 | tessier_ | JunK-Y: No. Just ulaw over SIP. |
| 01:26.30 | tessier_ | asterisk doesn't do T38 anyhow. |
| 01:26.31 | lilneon | hi tessier |
| 01:26.39 | tessier_ | I just stuck an ATA on our office fax machine and voila... |
| 01:26.43 | JunK-Y | tessier: that's a start. |
| 01:26.52 | JunK-Y | it will one day. |
| 01:27.58 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 01:28.30 | canerabbioso | ?? |
| 01:28.49 | lilneon | hey i need someone to test an iaxwebfon for me |
| 01:28.52 | lilneon | anyone?? |
| 01:29.16 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps 'No such file or directory' error for all modules when compiling.... I have Fedora Core 1; any idea ???? thanks Dave |
| 01:29.19 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps 'No such file or directory' error for all modules when compiling.... I have Fedora Core 1; any idea ???? thanks Dave |
| 01:29.25 | wm501 | How do i hang up a call from CLI> prompt? |
| 01:29.39 | lilneon | wm501: type hangup |
| 01:29.47 | wm501 | won't that disconnect all calls? |
| 01:29.48 | lilneon | ? |
| 01:29.53 | lilneon | oh yeah |
| 01:29.58 | harryvv | error for all modules means you dont have possibly all the compile tools and devs installed |
| 01:30.00 | lilneon | don't u have the manager running? |
| 01:30.10 | ManxPower | lilneon, "show application softhangup" |
| 01:30.18 | *** join/#asterisk |Vulture| (~Vulture@109.238.204.68.cfl.rr.com) |
| 01:30.24 | lilneon | thanx Manxpower :) |
| 01:32.59 | harryvv | I dont think voip is protected against wiretapping is it? |
| 01:33.24 | JunK-Y | harryvetch: ur wire can be tapped too. |
| 01:34.36 | harryvv | junk, when I took a telecom course a few years ago in Seattle our instructor said that any and all channels on telcos |
| 01:34.38 | *** join/#asterisk ArkyLady (ArkyLady@93-95.hspg-ubr2-blk1.cablelynx.com) |
| 01:34.51 | *** join/#asterisk toddf (lzfqc0d59n@default.fries.net) |
| 01:35.58 | lilneon | brb |
| 01:36.00 | *** part/#asterisk lilneon (~tj_r3@200.108.22.87) |
| 01:36.48 | harryvv | cannot be used to the benifit of the telco technican if he/she is listening to the convo. It was at one time thay had to at 2 am to wait for the conversation to end then disconect the line. He said you wont believe the amount of booring conversations we have to listen to in our job ;) |
| 01:46.04 | *** join/#asterisk jpablo (~jpablo@host-148-244-137-95.block.alestra.net.mx) |
| 01:46.48 | jpablo | hi, im having trouble making xorcom detect a wtdm4 with 4 fxo modules ... it wants to set fxo signaling for it, any one knows if that can be fixed ? |
| 01:48.51 | wm501 | how can i disconnect this call. |
| 01:48.52 | wm501 | IAX2/voicepulse-in-0 66.234.228.144 voicepulse 00003/00034 00032/00037 00000ms 0005ms 0000ms ulaw |
| 01:49.01 | wm501 | the id i guess is 00003 |
| 01:49.19 | hermie | 'soft hangup' |
| 01:49.34 | cypromis | soft hangup IAX2/voicepulse-in-0 |
| 01:49.46 | wm501 | the problem with that is |
| 01:49.56 | wm501 | i have another call i can't disconnect |
| 01:50.04 | wm501 | Channel Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format |
| 01:50.04 | wm501 | IAX2/voicepulse-in-0 66.234.228.144 voicepulse 00002/00068 00084/00089 00000ms 0004ms 0000ms ulaw |
| 01:50.04 | wm501 | IAX2/voicepulse-in-0 66.234.228.144 voicepulse 00003/00034 00061/00065 00000ms 0003ms 0000ms ulaw |
| 01:50.11 | wm501 | i need to only disconnect ID 00003 |
| 01:50.14 | wm501 | not 00002 |
| 01:51.01 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps 'No such file or directory' error for all modules when compiling.... I have Fedora Core 1; any idea ???? thanks Dave |
| 01:51.14 | ManxPower | wm501, "show channels" to get the channel ID. Then see "show application softhangup" |
| 01:51.26 | wm501 | thanks |
| 01:51.33 | MrEntropy | regex Q: zero-or-more times is * or +? |
| 01:52.07 | srt | * |
| 01:52.17 | srt | + is at least once |
| 01:52.32 | MrEntropy | srt: thought so, lost my damn bookmark to a really good regex site. thanks |
| 01:53.45 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-242-cust.telepacific.net) |
| 01:54.00 | ta[i]nted | how do i add wait and pause into my dial application |
| 01:54.10 | canerabbioso | Hi All, After patching the Zaptel drivers with bristuff-0.2.0-RC5 it dumps 'No such file or directory' error for all modules when compiling.... Note that without patching all works fine! I have Fedora Core 1; any idea ???? thanks Dave |
| 01:55.15 | ta[i]nted | say i want to DIAL(SIP/Provider/1234567890), have it wait 2 seconds, and then press '123' |
| 01:55.25 | srt | canerabbioso: what do u mean by "patching"? |
| 01:55.42 | srt | bristuff downloads the stuff it needs and patches itself... |
| 01:56.03 | Atacomm | ~seen moc_ |
| 01:56.05 | jbot | moc_ <~mochouina@modemcable212.49-80-70.mc.videotron.ca> was last seen on IRC in channel #asterisk, 4d 8h 15m 10s ago, saying: 'I dont see any phone beeting polycom feature any time soon..'. |
| 01:56.25 | *** join/#asterisk file (~symlink@mctnnbsah25-142166093009.nb.aliant.net) |
| 01:56.53 | Atacomm | anyone want a IP 3000 conference phone? looking to replace ours with a IP 4000 model. Barely been used, in great condition.... looking for around $500 |
| 01:58.10 | canerabbioso | srt: yes in fact, after bristuff patched zaptel I can't compile 'couse those erros! |
| 01:58.37 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 01:58.47 | srt | canerabbioso: you just did a ./compile.sh ? |
| 02:00.48 | srt | canerabbioso: ls /lib/modules/`uname -r`/build |
| 02:03.05 | marl | ok i know this isnt the rit eplace to ask this, but does anyone know of a linux based program that will allow usage of a standard voice modem and convert to voip? |
| 02:03.22 | marl | or preverable is there any chance of this apearing in * ? |
| 02:05.06 | *** join/#asterisk canerabbioso (~presnitz@host26-172.pool8252.interbusiness.it) |
| 02:08.01 | hermie | fearnor, you around? |
| 02:08.21 | porkchop_ | anyone know of config options for passing inbound callerid? |
| 02:08.30 | *** part/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 02:09.05 | *** join/#asterisk Cresl1n (~matt@216.207.244.186) |
| 02:09.21 | wm501 | if i do soft hangup IAX2/voicepulse-in-01@66.234.228.170:4569/6 |
| 02:09.30 | wm501 | will that only disconnect PRI 6 |
| 02:09.37 | *** join/#asterisk loko (rbrown@c-67-171-69-120.client.comcast.net) |
| 02:12.49 | wm501 | any of you guys have that pulver innovations wisip phone? |
| 02:12.53 | wm501 | i got one and it is awesome |
| 02:16.21 | Chuji | Atacomm : You have the 4000's yet? |
| 02:17.14 | NormAst | I give up... Is there a linux distro that asterisk zaptel will install without any problems.. Debian .. has so many problems... |
| 02:17.30 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 02:17.51 | Chuji | NormAst : Just have to make sure your kernel sources are available |
| 02:18.00 | NormAst | I do... have themm.. |
| 02:18.01 | Chuji | NormAst : I never had any problems in Deb |
| 02:18.24 | Chuji | Try fedora code 2 of 3 |
| 02:18.44 | Chuji | That is what it gets installed on the most |
| 02:18.47 | NormAst | I do apt-get install kernel-source-2.4.7 |
| 02:18.53 | Chuji | 2 or 3 I mean |
| 02:19.14 | Nukemizer | I am trying to make an incoming line (FXO) Zap/3-1 ring to all phones in my house. Can this be done ? |
| 02:19.49 | Chuji | Nukemizer |
| 02:19.51 | Chuji | sure |
| 02:20.05 | Chuji | What are the phones plugged into? |
| 02:20.20 | Nukemizer | well, I have to FXS and 4 SIP |
| 02:20.43 | Cresl1n | Ok, second round :-) Calling all 5ESS switches... Does anybody have a 5ESS compatible switch that I could test something really neat on? |
| 02:20.59 | NormAst | I get things like unresolved symbols in /lib/modules/2.4.27-1-386/misc/ztdynamic.o for example..... |
| 02:21.03 | Chuji | Dial(Zap/1&Sip/1&Sip/2) etc |
| 02:21.35 | Chuji | Cresl1n : Ohh yeah, I have one in the spare room. I just keep it off because it drains too much electricity |
| 02:21.46 | Nukemizer | Chuli, thanks testing |
| 02:21.46 | Chuji | NormAst : Yup, that's a kernel thang |
| 02:21.55 | Cresl1n | Chuji: no joke? :-) |
| 02:22.14 | NormAst | How do I fix it? It's driving me to drink...Red wine.. |
| 02:22.15 | Chuji | Cresl1n : Uhh yeah, it's a joke |
| 02:22.31 | JunK-Y | mouahhaha |
| 02:22.38 | Cresl1n | Chuji: Heh, I wish I had one in the back room, lol |
| 02:22.58 | *** join/#asterisk lilneon (~tj_r3@200.108.22.87) |
| 02:23.05 | lilneon | hi again all |
| 02:24.09 | NormAst | Anyone a debian pro? Wanna make some $$$ |
| 02:24.20 | NormAst | ;) |
| 02:24.47 | NormAst | Would it be becuase I am trying Sarge? |
| 02:24.52 | Chuji | NormAst : I'd give it a shot if I wasn't watch'n my kid |
| 02:26.02 | Chuji | NormAst : Where are the kernel sources? |
| 02:26.58 | NormAst | On the machine.. |
| 02:27.01 | NormAst | :) |
| 02:27.08 | Chuji | bleh, what dir? |
| 02:27.41 | NormAst | cd /usr/src |
| 02:28.03 | Chuji | what is the full path? |
| 02:28.32 | Chuji | like on my RHEL it's /usr/src/linux-2.4.21-15.EL |
| 02:29.14 | NormAst | oh.. Sorry.. /usr/src/kernel-source-2.4.27 |
| 02:29.24 | NormAst | i have linked it to /usr/usr/linux |
| 02:29.45 | Chuji | link it to linux-2.4.27 too |
| 02:31.38 | NormAst | trying it.. |
| 02:32.32 | *** part/#asterisk lilneon (~tj_r3@200.108.22.87) |
| 02:32.34 | NormAst | I give up.. time to format and start over... hmmm...Red hat version 9.0... |
| 02:32.41 | Chuji | haha |
| 02:32.44 | Chuji | I've been there |
| 02:32.47 | Chuji | what card? |
| 02:32.50 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 02:32.55 | NormAst | I just want to play with the pri card... |
| 02:33.03 | Chuji | t100p? |
| 02:33.03 | NormAst | it's a dual port.. |
| 02:33.11 | Chuji | Dual? |
| 02:33.12 | NormAst | Sangoma A102 |
| 02:33.14 | Chuji | Ohh |
| 02:33.29 | Chuji | Yeah, I need to check those out too |
| 02:33.51 | NormAst | They are just down the street from me... Really nice guy.. |
| 02:33.56 | *** join/#asterisk three55ml (~who@cs666898-229.austin.rr.com) |
| 02:33.59 | Chuji | Well, I was going to recommend you call Digium, but not with that card |
| 02:34.17 | hermie | Great quote: "I would feel better robbing banks for a living than helping spammers/faxblasters/telemarketers." |
| 02:34.25 | NormAst | I do own two Digium Quad cards.. :) |
| 02:34.48 | Chuji | Both in production? |
| 02:34.52 | three55ml | I've been trying to figure this out for two days. I cannot register my IAX clients to my Asterisk boxes (tried 2). IAX debug shows info going back and forth, but no registration. I know it supports NAT, but where should I begin looking? |
| 02:35.13 | *** join/#asterisk WGFreewill (~chatzilla@24-75-221-174.miamfl.adelphia.net) |
| 02:35.14 | Nukemizer | Chuji ! Man that works Sweet :) thank you so much for your help ! |
| 02:35.26 | NormAst | One is ... the other one was DOA. |
| 02:35.32 | Chuji | yuck |
| 02:35.34 | Darwin35 | are the password correct and the user names |
| 02:35.43 | three55ml | Yes |
| 02:35.45 | Darwin35 | did you setup both parts needed |
| 02:35.49 | NormAst | chuji: I was a two minute phone call... |
| 02:35.57 | Chuji | three55ml : post your iax.conf to pastebin.ca editing out anything personal |
| 02:36.02 | Chuji | three55ml : We'll take a look |
| 02:36.06 | Darwin35 | did you tell it how to pass the password |
| 02:36.09 | NormAst | is the loop back plug in.. ... yup.....pattest... hmmm okay dead. |
| 02:36.30 | three55ml | I'm led to believe it's a network issue, because (at least as of 2 days ago) it worked OK from my girlfriends - but she has the same provider and the same router as I have here. |
| 02:36.30 | three55ml | OK |
| 02:36.32 | Chuji | three55ml : post your Dial line if there is one too |
| 02:36.32 | NormAst | chuji: No show us all the passwords.. :) |
| 02:36.36 | NormAst | I like free LD. |
| 02:36.56 | three55ml | Chuji: Do you want any of the IAX debug info? |
| 02:37.02 | three55ml | haha |
| 02:37.11 | Chuji | three55ml : So it used to work? And quit? |
| 02:37.30 | Chuji | three55ml : You sure your incoming contexts didn't change |
| 02:37.32 | NormAst | Norm cry's on his keyboard... :( |
| 02:37.45 | tzanger | don't cry norm |
| 02:37.47 | three55ml | Chuji: Let me look at that, I thought I checked that though. |
| 02:37.52 | NormAst | need more red wine. |
| 02:39.26 | three55ml | Haha, I have about 30 bottles less than 3 feet away at the bar in my living room. |
| 02:40.15 | NormAst | yea.. my wife just picked up ours.. :) |
| 02:40.27 | Cresl1n | my stomach keeps on trying to talk to me |
| 02:40.35 | Cresl1n | I think maybe it's food time ;-) |
| 02:41.27 | JunK-Y | Cresl1n: ur stomach is running *? |
| 02:41.27 | JunK-Y | :P |
| 02:41.55 | Cresl1n | that's funny |
| 02:42.02 | Cresl1n | it probably is |
| 02:42.08 | Cresl1n | it's running CVS head right now |
| 02:42.30 | JunK-Y | communicating via IAX2 or SIP? |
| 02:42.33 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:42.39 | Chuji | ~head |
| 02:42.41 | jbot | from memory, head is HEAD |
| 02:42.55 | JunK-Y | until ya've fixed chan_zap, everything's fine :) |
| 02:44.11 | three55ml | Chuji: Here's my iax.conf : http://www.spirestudios.com/misc/asterisk/iax.conf |
| 02:45.08 | three55ml | On a sidenote, I'm working on a GUI. Anyone who has some comments feel free to fire away. It's in final testing right now, just has the basic features. Tried to make it as non-technical as possible. http://www.spirestudios.com/misc/pbx_screenshot.jpg |
| 02:47.46 | NormAst | three55ml: Pretty.. |
| 02:49.00 | three55ml | Thanks |
| 02:49.08 | three55ml | Made a few updates since that screenshot |
| 02:49.16 | BoRiS | three: Is there a beta? |
| 02:49.29 | three55ml | There will be in the next week or so |
| 02:49.46 | three55ml | Shoot me an email at andym@spirestudios.com if you're interested |
| 02:50.05 | BoRiS | does it hook to a database? PostGresSQL, MySQL? |
| 02:50.09 | freat | anyone ever see this behavior? It's an IAXy that was registered fine the other day, now it doesn't work |
| 02:50.10 | freat | http://pastebin.ca/5189 |
| 02:50.15 | three55ml | MySQL |
| 02:50.33 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 02:50.39 | BoRiS | ugh... ok :-p |
| 02:50.47 | three55ml | It's probably going to be strictly hosted at first though. |
| 02:51.01 | freat | it's a Linksys router he's behind. Block WAN requests is turned OFF |
| 02:51.16 | freat | the pastebin is from IAX2 debug |
| 02:51.30 | NormAst | normast pulls out the Red hat 9.0 disks. |
| 02:51.30 | three55ml | In theory PostgreSQL should work fine, I just don't use it. The only custom modules are a modified app_meetme that supports a better web interace. |
| 02:51.33 | PakiPenguin | hello everyone , can i have an ip based context ? like i have calls coming in from a certain ip like this 0092XXXXXXX , i want them to be handled in a different way from others , how to do that , i cannot register to the sender as it will just be sending out the calls to me ( Incoming only ) |
| 02:51.52 | *** join/#asterisk dontmsgme (~none@adsl-68-121-22-193.dsl.irvnca.pacbell.net) |
| 02:51.57 | NormAst | drinks another glass of red wine.. |
| 02:52.35 | Nukemizer | I hope someone can help me out. I get my softphones to dial our using "9NXXXXXX" but when i try to dial out from an FXS port and an intercept at the first number after dialing 9 |
| 02:52.53 | three55ml | freat: I'm actually having the exact same issue right now, but with a D-Link router and a softphone |
| 02:52.59 | Nukemizer | [outgoing] |
| 02:52.59 | Nukemizer | exten => _9NXXXXXX,1,Dial,Zap/3-1 <-- this is what I was using |
| 02:53.28 | mrproper_ | anyone know why id be getting: res_musiconhold.c:306 monmp3thread: Request to schedule in the past?!?! when trying to do musiconhold? |
| 02:53.53 | freat | three55ml: rebooted the router and then the IAXy and now it works |
| 02:54.10 | freat | difficult to troubleshoot as it's remote |
| 02:54.14 | freat | thx though |
| 02:55.21 | *** part/#asterisk david (~dcoulson@muffin.davidcoulson.net) |
| 02:57.51 | three55ml | mrproper_: Search voip-info.org, lots of info about it on there I believe |
| 03:00.29 | JunK-Y | mrproper_: not enuf cpu. |
| 03:00.30 | shido6 | um |
| 03:00.39 | shido6 | what the heck is that nukemizer? |
| 03:00.49 | mrproper_ | JunK-Y: machine is running at .1% cpu usage |
| 03:01.50 | WGFreewill | three55ml: ignorepat => |
| 03:03.13 | three55ml | I think that's for Nukemizer, right? |
| 03:05.12 | Nukemizer | sorry was on other screen |
| 03:05.32 | Nukemizer | you mean my dial string ? |
| 03:06.00 | three55ml | In the extensions.conf, add ignorepat => 9 in the context |
| 03:06.17 | Nukemizer | ahh |
| 03:07.38 | Nukemizer | three55ml: you mean only put it in my "outgoing" context right ? |
| 03:07.53 | three55ml | Yeah, if that's where you extension is at |
| 03:09.52 | *** join/#asterisk Frantic (~ab@68-175-94-179.nyc.rr.com) |
| 03:10.55 | WGFreewill | anybody running bi -driectional h323 trunks between * and cisco devices? |
| 03:11.05 | WGFreewill | (call manager, 5850, rotuer) |
| 03:11.45 | bjohnson | is G711u on the sipuras the same as ulaw on *? |
| 03:11.51 | *** join/#asterisk DaLion (anon@HSE-QuebecCity-ppp3497095.sympatico.ca) |
| 03:12.35 | *** join/#asterisk andrew` (~andrew@adsl-67-119-26-96.dsl.snfc21.pacbell.net) |
| 03:12.55 | WGFreewill | yes g711u = ulaw |
| 03:12.59 | Nukemizer | three55ml - yes it looks like that statement was in ther already. i guess i just get confused by the flow of how extensions pick a context flow. |
| 03:13.26 | three55ml | Cool, so did you get it working now? |
| 03:13.31 | Nukemizer | no.. :( still looking though |
| 03:13.59 | Nukemizer | i wonder if the "conference bridge being 91 has anything to do with brokenness ? |
| 03:15.13 | modulus_ | touch my nuts |
| 03:16.00 | hermie | modulus_ wrong kinda chat :) |
| 03:16.40 | hermie | exten => 8675309,1,Dial(SIP/Jenny) |
| 03:17.08 | *** join/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
| 03:17.53 | *** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net) |
| 03:18.08 | WGFreewill | telephony / teleporno |
| 03:18.10 | WGFreewill | its confusing |
| 03:19.14 | bjohnson | anyone familiar with playing with line voltages on SPA 3ks? |
| 03:19.32 | modulus_ | oops |
| 03:19.57 | WGFreewill | to interface with non-US telecom equipment? |
| 03:21.59 | Nukemizer | Note ** should play with Voltage ! |
| 03:22.25 | nestAr | hahah |
| 03:22.28 | nestAr | ENGRISH |
| 03:22.29 | nestAr | No D-channels available! Using Primary on channel anyway 24! |
| 03:22.42 | *** join/#asterisk dca (~teliax@c-67-166-37-218.client.comcast.net) |
| 03:23.59 | bjohnson | WGFreewill: no .. I have a fax data switch and it shows different voltages that my regular pstn line |
| 03:25.39 | WGFreewill | co voltage should be -48v |
| 03:25.50 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:25.58 | WGFreewill | you'll always have drop |
| 03:26.02 | WGFreewill | long cable runs |
| 03:26.11 | WGFreewill | is it higher or lower |
| 03:26.15 | WGFreewill | than your house like |
| 03:26.24 | MrEntropy | anywhere i can pick up an educational version of codec_g7231? |
| 03:26.36 | *** join/#asterisk alakdan (~alakdan@210.213.185.59) |
| 03:26.47 | nestAr | i think the license is only like $10 from digium |
| 03:27.15 | *** part/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:27.25 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
| 03:27.36 | MrEntropy | nestAr: and when i buy the licence they give me the lib? |
| 03:27.50 | nestAr | i assume so, but i dunno |
| 03:28.21 | bjohnson | WGFreewill: I get +49 according to my SPA |
| 03:28.24 | MrEntropy | nestAr: because i've seen those "we sell you the licence, but you get the lib from somewhere else" deals |
| 03:28.38 | nestAr | ask them |
| 03:28.44 | Grooby | hallo people |
| 03:28.48 | Grooby | what's happening tonight |
| 03:28.49 | nestAr | hi |
| 03:28.55 | nestAr | law and order |
| 03:28.56 | alakdan | anyone subscribed to nufone? Im just wondering given a single subscription (business) how many simultaneous calls (incoming/inbound) can it handle? |
| 03:29.03 | WGFreewill | possible mis-wire |
| 03:29.15 | WGFreewill | make sure your have a normal rj-11 cable |
| 03:29.28 | bjohnson | normal? |
| 03:29.35 | bjohnson | what would be abnormal? |
| 03:30.06 | nestAr | hahah.. i'm up to 50 gmail invites |
| 03:30.12 | nestAr | yesterday i think i had 10 |
| 03:30.31 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 03:30.33 | nestAr | i thought by now everyone who wanted a gmail had one.. |
| 03:30.45 | three55ml | nestAr: Feel free to send one my way :) |
| 03:30.53 | Qwell | My mother-in-law has gmail, and I didn't give it to her. |
| 03:31.00 | Qwell | So, yes...everybody has gmail. heh |
| 03:31.01 | nestAr | three55ml: give me a email address to send the invite too |
| 03:31.09 | three55ml | andym@spirestudios.com |
| 03:31.22 | freat | anything you guys think I should be looking for that would degrade ulaw quality? is there anything to do to help with ulaw... possibly for Polycom phones? |
| 03:31.26 | three55ml | Need another account to send junkmail too |
| 03:31.28 | nestAr | three55ml: done. |
| 03:31.33 | three55ml | nestAr: Thanks |
| 03:31.55 | nestAr | np |
| 03:32.08 | Grooby | i collect spams |
| 03:32.09 | Grooby | j/k |
| 03:32.44 | nestAr | we just created a wedding registry, and they wanted a email address.. i used "spam-makes-the-baby-jesus-cry@wewt.net" |
| 03:33.11 | harryvetch | are there any major/minor voip services carriers that end with the suffix of .net? |
| 03:33.25 | nestAr | nufone.net, i think |
| 03:33.56 | harryvetch | Seems like every fricken word in the dictionry releated to the telco biz been used up for .com |
| 03:34.14 | bjohnson | WGFreewill: I have a second one that is behind a fax switch .. -23V .. I can't get it to dial out |
| 03:35.32 | MrEntropy | nestAr: they only sell 729 licence |
| 03:36.07 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 03:36.19 | DaLion | harryvetch: why ? |
| 03:36.50 | harryvetch | Da, was wondering. |
| 03:36.52 | WGFreewill | yeah thats not gonna do it |
| 03:36.55 | DaLion | anyone know if sveasoft goes into WRT54GPTA |
| 03:36.57 | WGFreewill | and the sipura isnt gonna make it |
| 03:37.15 | bjohnson | WGFreewill: is there a way to config the SPA to compensate? |
| 03:37.20 | harryvetch | If I wanted to create a company with any known acronym in this biz seems most are used up. |
| 03:37.42 | bjohnson | WGFreewill: I need a fax/data switch to use a dial in modem |
| 03:38.46 | WGFreewill | not compensate that much |
| 03:38.52 | WGFreewill | I think they are thinking a few volts |
| 03:40.16 | harryvetch | It would be nice if there was a option to increase the font size on xlite |
| 03:40.31 | file[laptop] | why do you build me up, build me up, buttercup baby just to let me down |
| 03:41.07 | bjohnson | WGFreewill: do you know of another way to deal with incoming data modem calls? |
| 03:41.32 | WGFreewill | having it dial an extension |
| 03:41.41 | WGFreewill | to get to the right FXS port |
| 03:42.07 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 03:42.21 | bjohnson | WGFreewill: I thought you couldn't do that with data modem calls |
| 03:42.37 | WGFreewill | yeah |
| 03:42.44 | bjohnson | yes you can? |
| 03:42.47 | WGFreewill | xxx-xxx-xxxx,,,,,,,,<modem extension> |
| 03:42.51 | WGFreewill | or look for caller ID |
| 03:42.57 | WGFreewill | of the calling modem |
| 03:43.01 | WGFreewill | handoff |
| 03:43.04 | WGFreewill | to extension |
| 03:43.10 | bjohnson | so * can pass through modem calls? |
| 03:43.15 | *** part/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 03:43.17 | bjohnson | which codec should I use |
| 03:43.31 | mikegrb | ulaw |
| 03:43.33 | mikegrb | alaw |
| 03:43.59 | bjohnson | everything I've read says modem calls can't go through voip |
| 03:44.00 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 03:45.16 | mikegrb | jitter can be a problem |
| 03:45.20 | mikegrb | but on lan is fine |
| 03:46.22 | bjohnson | so just send out a fxs to a data modem? |
| 03:47.54 | bjohnson | any other hardware restrictions for the fxo to detect a fax call? |
| 03:48.55 | mikegrb | auto fax detection only works with digium hardware |
| 03:49.09 | alakdan | anyone subscribed to nufone? Im just wondering given a single subscription (business) how many simultaneous calls (incoming/inbound) can it handle? |
| 03:49.35 | Qwell | alakdan: support@nufone.net |
| 03:49.42 | Qwell | your best option |
| 03:49.45 | mikegrb | alakdan: unlimited |
| 03:50.17 | MrEntropy | is it possible to add a directory to the 'search path' asterisk uses for it's ivrs? |
| 03:50.53 | bjohnson | mikegrb: ok .. I have 2 digium X100Ps .. should they work for fax detection? |
| 03:51.07 | mikegrb | yes |
| 03:51.15 | mikegrb | as far as I know |
| 03:51.28 | mikegrb | just add an extention named fax |
| 03:51.30 | alakdan | mikegrb: so if I route the incoming calls to the 'demo' context all of them would be able to hear the demo provided in asterisk? and basically the only limitation is the bandwidht of my broadband? |
| 03:51.49 | mikegrb | alakdan: basically, yeah |
| 03:53.42 | alakdan | mikegrb: one more question, do you happen to know how much bandwidth would iax protocol use up per incoming call? |
| 03:54.00 | Qwell | alakdan: thats mostly codec dependent |
| 03:54.34 | *** join/#asterisk [Bernard] (~N4SH@c-67-180-105-69.client.comcast.net) |
| 03:56.02 | *** join/#asterisk PMantis (~PrayingMa@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 03:57.01 | PMantis | GotoIf help? the following is not working to test for blank variable: GotoIf($[${ARG1} = ""]? |
| 03:58.02 | alakdan | Qwell: I see. so for using a gsm codec it would require about 8kb ? |
| 03:58.17 | Qwell | I'm the wrong person to ask |
| 03:58.32 | [Bernard] | hi i'm new in this technology. i'm trying to implement a Phone-Phone solution where a User calls a local number then forwarded for authentication and the number dialed. is this possible with asterisk? i found a link is this the same thing (http://areski.net/asterisk-stat-v1_3/about.php) |
| 03:58.35 | Qwell | but, I would imagine so(plus overhead, blah) |
| 03:59.39 | PMantis | [Bernard], You want someone to dial a number, authenticate, then dial the real number, and have Asterisk "proxy" the phone call? |
| 03:59.57 | [Bernard] | <PROTECTED> |
| 04:00.39 | PMantis | [Bernard], Yes, this can be done easily. Look at DISA and Authenticate/// actually in reverse order. lol. |
| 04:01.06 | [Bernard] | <PROTECTED> |
| 04:01.07 | *** part/#asterisk cbachman (~cbachman@129.105.7.250) |
| 04:01.26 | alakdan | Qwell: Ok thanks a lot :) |
| 04:01.45 | mikegrb | PMantis: DISA has it's own authentication built in |
| 04:02.13 | PMantis | [Bernard], The asterisk Wiki will tell you how to use both of those applications.. basically, you want astereisk to run answer, play a message, run authenticate, then disa |
| 04:02.36 | mikegrb | PMantis: DISA has it's own authentication built in, Authenticate is unnecessary |
| 04:02.59 | [Bernard] | <PROTECTED> |
| 04:03.04 | PMantis | mikegrb, I've had problems with disa's authentication.. don't remember if it was a functionality problem, or if it simply didn't fit the way i wanted ti to work. |
| 04:03.21 | bjohnson | or you could authenticate based on cid |
| 04:03.29 | mikegrb | I'd guess two ;) havent had any authentication problems |
| 04:03.34 | PMantis | mikegrb, but yes, [Bernard] could use the built-in authentication. :) |
| 04:03.36 | mikegrb | indeed, cid is another option |
| 04:03.40 | bjohnson | authenticate has a couple of more options |
| 04:03.54 | bjohnson | cid can be spoofed .. but is suitable in some cases |
| 04:03.58 | PMantis | bjohnson, of course, callerid can be spoofed. |
| 04:04.00 | bjohnson | makes it much more user friendly |
| 04:04.02 | PMantis | Haha |
| 04:04.46 | PMantis | Now that I got everyone talking, can someone help me rewrite this?? GotoIf($[${ARG1} = ""]?blah:) |
| 04:04.57 | PMantis | i want to test for a blank variable |
| 04:05.10 | PMantis | the above doenes't work... |
| 04:05.40 | mikegrb | PMantis: then do the bash trick, set another var to ${ARG1 |
| 04:05.44 | mikegrb | er |
| 04:05.56 | [Bernard] | <PROTECTED> |
| 04:05.59 | mikegrb | PMantis: then do the bash trick, set another var to ${ARG1}z and then test if your new var is set to z |
| 04:06.04 | mikegrb | if so, arg1 was empty |
| 04:07.30 | PMantis | [Bernard], if you want to terminate the call on the PSTN, you'll have to pay a provider to take the voip call and pass it back to the PSTN... |
| 04:07.40 | PMantis | mikegrb, That'll work. :) |
| 04:07.47 | mikegrb | PMantis: :D |
| 04:08.06 | mikegrb | PMantis: old tricks are good to have in your bag of tricks, every now and then still good to use |
| 04:08.10 | *** join/#asterisk WizardWlf (~shawn@wrt54g.djernes.org) |
| 04:08.28 | PMantis | mikegrb, Yup. Thanks! |
| 04:08.29 | [Bernard] | <PROTECTED> |
| 04:08.45 | PMantis | [Bernard], NP! |
| 04:08.58 | WizardWlf | anyone know how to make th tdm400 fxo wait a few seconds before it dials. |
| 04:09.42 | mrproper_ | do i need to have libpri and zaptel compiled to use music on hold with a BRI CAPI card? |
| 04:10.47 | *** join/#asterisk jcollie (~jcollie@dsl-ppp239.isunet.net) |
| 04:12.10 | simon_ca | wizard: w in the dialstring... |
| 04:14.06 | *** join/#asterisk Inv_arp (junya@adsl-8-230-175.mia.bellsouth.net) |
| 04:14.46 | Inv_arp | whats a good site i can use to make sip calls where i dont have to suscribe like vonage? usa to usa |
| 04:15.08 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 04:15.59 | alakdan | Inv_arp: free world dialup? |
| 04:16.36 | Inv_arp | Qwell: heh |
| 04:16.54 | alakdan | www.freeworlddialup.com |
| 04:17.06 | Inv_arp | Qwell: we'll pay by min not by monthyly or somethin |
| 04:17.25 | Qwell | oh, there are a bunch of those |
| 04:17.43 | Qwell | Wait around, I guarantee you'll get a message or two in a moment :p |
| 04:17.48 | Inv_arp | heh |
| 04:18.22 | Qwell | I'll save dca the trouble. teliax.com |
| 04:18.30 | Inv_arp | ahh thx |
| 04:18.34 | Qwell | shop around though, of course |
| 04:18.45 | Inv_arp | was looking at iax.cc also |
| 04:20.12 | DaLion | Now that I got everyone talking, can someone help me rewrite this?? GotoIf($[${ARG1} = ""]?blah:) |
| 04:20.16 | DaLion | try |
| 04:20.36 | DaLion | GotoIf($["${ARG1}" =="] |
| 04:20.40 | Qwell | is "blah(funky char):)" part of it? |
| 04:20.46 | DaLion | making arg in quotes comapres with string |
| 04:21.35 | DaLion | exten => s,10,GotoIf($["${CALLERIDNUM}" = ""]?s|1000) |
| 04:21.48 | DaLion | that would wotk |
| 04:22.00 | netsurfer | or GotoIf($[${foundRow} = 1]?13:3) |
| 04:22.39 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 04:26.51 | brc__ | uh |
| 04:26.56 | brc__ | that don't make no sense at all |
| 04:31.10 | datareactor | is there log file in grandstream budgetone 100 |
| 04:33.32 | Grooby | doo dee dooo |
| 04:35.24 | simon_ca | anyone use simpletelecom.com? work well? |
| 04:37.52 | `Sauron | Sigh. |
| 04:40.09 | `Sauron | simon_ca: Looks too expensive |
| 04:41.26 | simon_ca | looking for one that's usage only w/ no min billing. anyone have any suggestions? |
| 04:41.42 | Inv_arp | silik0n: heh im looking for a provider myself with min billing |
| 04:42.10 | `Sauron | simon_ca: Good luck. :) |
| 04:42.26 | Inv_arp | simon_ca: look at these Broadvoice, Stanaphone, Sipphone, Iaxtel |
| 04:42.35 | Inv_arp | oh and iax.cc |
| 04:43.24 | `Sauron | you could get everybody to switch to free world dialup |
| 04:43.27 | `Sauron | It's completely free |
| 04:44.55 | mrproper_ | anyone know why only when i put another extension on hold i get: res_musiconhold.c:306 monmp3thread: Request to schedule in the past?!?! |
| 04:45.25 | mrproper_ | i've checked the wiki and the answers there are related to cpu usage, which is not the case here |
| 04:45.34 | `Sauron | simon: You won't find someone who'll only bill you for usage. They'll all have a monthly service charge. |
| 04:45.46 | `Sauron | Unless they're voip-only providers, like fwd |
| 04:46.41 | Nugget | voicepulse and nufone both just charge usage with no monthly charge. |
| 04:47.00 | Nugget | nufone doesn't even charge monthly for inbound DIDs |
| 04:48.22 | `Sauron | Nugget: Voicepulse still charges you $7/month |
| 04:48.31 | Nugget | no, only if you have a phone number. |
| 04:48.46 | Nugget | outbound calls are usage-billed |
| 04:48.50 | *** join/#asterisk zotz (~zotz@24.231.32.191) [NETSPLIT VICTIM] |
| 04:48.50 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) [NETSPLIT VICTIM] |
| 04:48.50 | *** join/#asterisk forty2 (fortytwo@secure.netbsd.se) [NETSPLIT VICTIM] |
| 04:49.06 | `Sauron | which brings us back to the voip-only providers being the only free ones |
| 04:49.17 | Nugget | well, I also mentioned nufone. |
| 04:49.22 | Nugget | which brings us back to you being wrong. :) |
| 04:49.48 | `Sauron | now if their website worked... |
| 04:49.53 | `Sauron | at least it didn't the other day |
| 04:54.08 | simon_ca | nugget: thanks |
| 04:56.55 | Inv_arp | 86ms is that good response time to for voip thru internet? |
| 04:58.41 | Nugget | 86ms is a bit rough, but if it's stable it's within reason. |
| 04:59.04 | Nugget | 250ms is where things start to go nutty, imho. |
| 04:59.12 | Grooby | hehehehe |
| 04:59.20 | Grooby | voip over your GPRS or something? |
| 05:01.44 | harryvetch | gprs? |
| 05:02.05 | `Sauron | Huh? |
| 05:02.12 | `Sauron | That'd be evil. |
| 05:04.07 | Wi_Fi | any one with budge tone 100 |
| 05:04.35 | simon_ca | wi_fi: couple of them... |
| 05:04.49 | Grooby | voip over the cellphone data |
| 05:04.55 | Wi_Fi | how the hell do i get it to stop flashing |
| 05:04.55 | Wi_Fi | hehe |
| 05:05.13 | Qwell | Wi_Fi: bang it on a table a couple times. :) |
| 05:05.17 | Wi_Fi | hehe |
| 05:05.20 | simon_ca | flashes when it has a voicemail indicator |
| 05:05.27 | Wi_Fi | yes |
| 05:05.37 | Wi_Fi | but wont let me get to voicemail |
| 05:06.05 | simon_ca | voicemail button only works if you program speed dial with a dial string that maps to your vmail access... |
| 05:06.20 | Wi_Fi | how do i do that |
| 05:06.29 | Wi_Fi | can it be done via web admin |
| 05:06.30 | simon_ca | wi_fi: take the mailbox= out of sip.cong :) |
| 05:06.36 | simon_ca | s/cong/conf |
| 05:07.00 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 05:07.16 | Wi_Fi | i want it to flash but i also want to be able to get to messages |
| 05:08.01 | *** join/#asterisk chipig (~chip@constant.northnitch.com) |
| 05:08.32 | simon_ca | do you have an extensions that calls voicemailmain? |
| 05:08.39 | Wi_Fi | yes |
| 05:08.43 | Wi_Fi | SUBSCRIBE for MWI: No, do not send SUBSCRIBE for Message Waiting Indication |
| 05:08.43 | Wi_Fi | Yes, send periodical SUBSCRIBE for Message Waiting Indication |
| 05:08.54 | Wi_Fi | i have set yes |
| 05:10.45 | simon_ca | wi_fi: you can just call the extension manually to check msgs... |
| 05:11.10 | simon_ca | wifi: i think it is the voicmail userid field in the config where you prog the extension for the button.. |
| 05:11.52 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 05:12.18 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 05:12.25 | Wi_Fi | ahh |
| 05:12.27 | Wi_Fi | ill check |
| 05:14.07 | Wi_Fi | and i have that set |
| 05:14.17 | Wi_Fi | what do you have dtmf set at |
| 05:14.46 | *** join/#asterisk nestAr (nester@makes.all.the.girlies.go.wewt.wewt.net) |
| 05:14.51 | nestAr | freenode rocks lately |
| 05:16.14 | *** join/#asterisk pranav (dawda_pran@203.115.89.185) |
| 05:16.54 | Wi_Fi | simon_ca it wont accept voicemail password |
| 05:19.08 | Nugget | the freenode budget graph is insulting. |
| 05:19.18 | Nugget | it's embarassing to me that digium has financially supported this network. |
| 05:20.17 | `Sauron | Hum. Why? |
| 05:20.36 | Nugget | because it's all going straight to lilo. |
| 05:20.48 | `Sauron | elaborate |
| 05:20.50 | Nugget | and he's not offering anything that other networks don't offer for free. |
| 05:21.07 | `Sauron | Ah. |
| 05:21.53 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-58-62.dsl.pltn13.pacbell.net) |
| 05:22.46 | Nugget | nearly all the donated money just goes straight to lilo. |
| 05:23.01 | *** join/#asterisk mitcheloc_ (~mitcheloc@67.153.163.202) |
| 05:23.29 | Nugget | the 2004 budget can be restated as "$16,000 for lilo, $2000 for lilo, $2250 for lilo, $750 for lilo, and $1000 for legal and accounting fees." |
| 05:23.54 | *** part/#asterisk mitcheloc_ (~mitcheloc@67.153.163.202) |
| 05:23.58 | *** join/#asterisk mitcheloc_ (~mitcheloc@67.153.163.202) |
| 05:23.59 | Nugget | and freenode doesn't do anything that you can't get for free on another network that doesn't have the unrealistic expectation of earning a living by being an ircop. |
| 05:25.12 | mitcheloc_ | anyone know a command to "refresh" the user list in a chat room? trilian isn't loading it automatically for some reason |
| 05:26.31 | `Sauron | Ah. |
| 05:26.51 | niZon | mitcheloc_: /names #channel |
| 05:27.12 | brc__ | .2 |
| 05:27.13 | mitcheloc_ | niZon: thanks, that did the trick |
| 05:27.15 | brc__ | 45 |
| 05:27.17 | niZon | np |
| 05:29.41 | *** join/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 05:31.52 | Inv_arp | hmm just tried voipjet works nice |
| 05:32.28 | *** join/#asterisk dizzydiffi (~diseyi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 05:34.25 | wolfson | inv_arp: just don't even complain, might get your account canceled like mine |
| 05:34.42 | wolfson | even=ever |
| 05:38.14 | Inv_arp | wolfson: ahh ok thx for the tip |
| 05:38.18 | moonwick | omg, getting thrown off of a PSTN gateway? wherever would you go then! |
| 05:38.20 | `Sauron | kram, know if Theo hangs out here? |
| 05:38.29 | moonwick | not like they're a dime a dozen, or anything. |
| 05:39.21 | `Sauron | moonwick: Good PSTN gateways aren't a dime a dozen |
| 05:39.23 | EvlHimeko | from the cosby show? i think he is on jeremiah now |
| 05:40.09 | `Sauron | theo the greek |
| 05:40.21 | `Sauron | I can't even begin to try to write his last name |
| 05:41.32 | `Sauron | and the chan_bluetooth pages said to ask mark questions about it |
| 05:41.44 | `Sauron | s/mark/kram |
| 05:41.45 | `Sauron | :p |
| 05:42.08 | kram | to ask me about it? |
| 05:42.10 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 05:42.17 | kram | i only made chan_btp not chan_bluetooth |
| 05:42.22 | `Sauron | Hum. |
| 05:42.40 | `Sauron | I coulda sworn the chan_bluetooth page said for questions to contact kram |
| 05:42.47 | letherglov | eh? mark, kram? one in the same? |
| 05:42.59 | letherglov | obviously it's backwards |
| 05:43.34 | `Sauron | I could be wrong. :) |
| 05:44.12 | letherglov | I suppose it's just to confuse the dislexic people ;-) |
| 05:44.30 | *** join/#asterisk adnans (~adnans@linuxgoeroe.demon.nl) |
| 05:45.32 | `Sauron | Hum, then it's off to emailing Theo |
| 05:45.49 | Qwell | letherglov: or perhaps the nick "mark" was taken on freenode :p |
| 05:47.34 | `Sauron | he could pay off lilo to get mark ;) |
| 05:48.32 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-85-97.sentechsa.net) |
| 05:49.47 | Qwell | hmm, whats the going rate for "north"? :p |
| 05:50.15 | `Sauron | Dunno, find out. :) |
| 05:50.16 | niZon | I'm curious as to why mettme needs a hardware timer, anyone care to shed some light or urls? :P |
| 05:50.25 | niZon | *meetme |
| 05:50.43 | `Sauron | it does? |
| 05:51.32 | kram | you have to have a reliable source of timing to do the mixing |
| 05:51.41 | `Sauron | Hum, ah. |
| 05:51.42 | niZon | ah |
| 05:52.01 | Qwell | kram: Is the address for the disclaimer still the same one as on bugs.digium.com? |
| 05:52.02 | `Sauron | Although I suspect that's not accurate enough :) |
| 05:52.14 | kram | qwell: yah, but you can just fax it if you want :) |
| 05:52.24 | Qwell | h.232...no thanks :p |
| 05:52.25 | *** join/#asterisk Moc (~Moc@modemcable212.49-80-70.mc.videotron.ca) |
| 05:52.30 | Qwell | kram: "Checks in the mail" |
| 05:52.46 | Qwell | I went ahead and did the longer one |
| 05:52.59 | Qwell | and added an amendment |
| 05:53.30 | Qwell | a Rider, as you will. |
| 05:53.47 | mitcheloc_ | qwell: what kind of code patch are you donating? |
| 05:53.54 | Qwell | mitcheloc_: None yet |
| 05:54.31 | `Sauron | You ahve to sign a disclaimer to submit a patch? |
| 05:54.36 | `Sauron | have |
| 05:54.40 | mitcheloc_ | for it to be included, yes |
| 05:55.11 | `Sauron | Hum. |
| 05:55.35 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 05:57.39 | *** part/#asterisk WizardWlf (~shawn@wrt54g.djernes.org) |
| 05:58.26 | alakdan | hello, anyone willing to help me out understanding John Todd's experiment (particularly on the GSM result part). http://www.voip-info.org/tiki-print.php?page=Asterisk+bandwidth+iax2 . Is my understanding correct on using GSM codecs, Assuming I set trunk=no, then 35.4 kbps for each call + 20.7 kbps IP/IAX2 overhead? |
| 05:59.06 | `Sauron | hmm. |
| 05:59.16 | `Sauron | This is because of the dual licensing? |
| 06:01.12 | kram | anybody know a URI that has an SRV record? |
| 06:01.20 | kram | for _sip._udp i mean |
| 06:01.22 | kram | or whatever |
| 06:01.55 | `Sauron | Hum, nope. |
| 06:03.08 | mitcheloc_ | kram: i think that broadvoice does ... sip.broadvoice.com |
| 06:03.16 | mitcheloc_ | kram: if not i can set one up for you to test with |
| 06:03.33 | `Sauron | I just looked |
| 06:03.47 | `Sauron | no SRV records for sip.broadvoice.com |
| 06:03.49 | `Sauron | according to dig |
| 06:04.00 | kram | yah i'm just trying ot track one of these down |
| 06:04.25 | mitcheloc_ | kram: ok, then, do you want me to set one up? its just a quick insert into my mysql database (i think) |
| 06:06.00 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 06:09.02 | *** join/#asterisk sjaak538 (~sjaaknabu@d5c53145.dsl.concepts.nl) |
| 06:10.01 | *** join/#asterisk dizzydiffi (~diseyi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 06:10.16 | dizzydiffi | hello room |
| 06:10.19 | dizzydiffi | whats 411 |
| 06:10.39 | `Sauron | directory lookup |
| 06:10.44 | Wi_Fi | any budge tone 100 gurus |
| 06:11.14 | dizzydiffi | anyone used Porta |
| 06:11.19 | dizzydiffi | One |
| 06:11.47 | *** join/#asterisk wasim (~wasim@203.81.200.8) |
| 06:13.03 | sjaak538 | Hello does anyone know how to be sure/test if a ISDN wire is connected to my ISDN card my server is remote !! |
| 06:13.50 | sjaak538 | I think everything is well compiled and installed but not any connection is possible |
| 06:14.11 | sjaak538 | cat /proc/interupts gives hfc |
| 06:14.28 | sjaak538 | zttool is okay and ztcfg is okay |
| 06:15.51 | alakdan | hello, anyone willing to help me out understanding John Todd's experiment (particularly on the GSM result part). http://www.voip-info.org/tiki-print.php?page=Asterisk+bandwidth+iax2 . Is my understanding correct on using GSM codecs, Assuming I set trunk=no, then 35.4 kbps for each call + 20.7 kbps IP/IAX2 overhead? |
| 06:16.44 | alakdan | which means around 56kbps per call? |
| 06:17.08 | *** join/#asterisk djin (~marius@gridfox.xs4all.nl) |
| 06:19.05 | sjaak538 | No ISDN experts here |
| 06:20.06 | modulus_ | ISDN == it still does nothing |
| 06:20.33 | sjaak538 | Yes in my case it's true ;-) |
| 06:22.16 | sjaak538 | My ISP told me it's pluged into 2 ISDN cards is that allowed !! (I have 2 servers) |
| 06:22.33 | *** part/#asterisk RyanE (~ryan@rberick.dsl.xmission.com) |
| 06:22.42 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 06:35.38 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 06:37.17 | *** part/#asterisk dizzydiffi (~diseyi@adsl-70-240-241-42.dsl.hstntx.swbell.net) |
| 06:39.26 | wasim | all ISDN experts are currently drunk or sleeping ... please wait, your help request will be answered in the order it was received |
| 06:44.10 | Qwell | There are currently - 43 - callers ahead of you. |
| 06:46.23 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 06:46.23 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:49.27 | djin | Logged in after the question. I only have CAPI experience, so I'm not sure I can help. |
| 06:49.39 | *** join/#asterisk [Bernard] (~N4SH@c-67-180-105-69.client.comcast.net) |
| 06:53.30 | djin | is everybody quiet for the reboot, or is it plain quiet here? |
| 06:55.56 | freat | so is TDMoE a good way to go if you want to have redundant asterisk boxes? I'm still having trouble figuring out good topologies for making asterisk more highly available |
| 06:56.07 | denon | no .. use IAX2 |
| 06:56.30 | denon | TDMoE is very cool, but not really useful in the real world |
| 06:56.38 | denon | nor is there a huge advantage to using it |
| 06:56.50 | denon | some would argue latency, but the difference is extremely trivial imho |
| 06:57.03 | freat | I'd like to just have 2 boxes at colo that I split our offices up among |
| 06:57.20 | freat | then if one goes down, somehow get all traffic to go to the remaining box |
| 06:57.25 | denon | yeah .. |
| 06:58.03 | freat | seems like a sensible approach to me... |
| 06:59.02 | freat | can you assign multiple IPs to an * box, and if it binds to 0.0.0.0 will it handle it all? |
| 06:59.28 | freat | I'm thinking... if box1 dies, assign box2 both addresses... |
| 07:00.39 | freat | of course dial plan would be an issue too if box2 tries to send calls off to box one |
| 07:37.00 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 07:37.00 | *** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released |
| 07:37.23 | *** join/#asterisk Elshar (~Elshar@ip206-91.oregonfast.net) |
| 07:40.24 | *** join/#asterisk Chotaire (chotaire@chotaire.net) |
| 07:40.51 | *** join/#asterisk cbachman (~cbachman@victory.ece.northwestern.edu) |
| 07:41.48 | tzafrir | qwerp, what do you mean? |
| 07:47.42 | *** join/#asterisk tessier__ (~treed@wsip-68-224-172-77.sd.sd.cox.net) |
| 07:48.28 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
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| 07:51.45 | qwerp | tzafrir: wakeup script like suggested in voip-info ? |
| 07:52.33 | qwerp | http://www.voip-info.org/wiki-Asterisk+tips+Wake-Up+Call+PHP |
| 07:52.36 | *** join/#asterisk Primer (~primus@sh.nu) |
| 07:52.39 | Primer | stupid nickserv |
| 07:52.43 | qwerp | which is not working... |
| 07:53.09 | Beirdo | Oh, I'm a happy man |
| 07:53.24 | Beirdo | I just found BBC7's archives :) |
| 07:55.36 | *** join/#asterisk DaGrim (~junglesto@dagrim.user) |
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| 08:02.55 | *** join/#asterisk Martohtar (Martohtar@194.19.32.100) |
| 08:06.29 | *** join/#asterisk wm501 (wm501@ip68-97-54-71.ok.ok.cox.net) |
| 08:06.58 | wm501 | I can't find any examples on getting call parking to work, I know so much as to add include => parkedcalls to extensions.conf |
| 08:07.01 | wm501 | then i'm lost |
| 08:07.48 | wm501 | i looked at features.conf and it shows parkext => 700 and parkpos => 701-720 and context => parkedcalls |
| 08:07.49 | MrEntropy | can i get asterisk to dump it's converted IVRs? for example, if i have my ivr's in .gsm form, but when someone calls in asterisk converts them to g729 for transmission to the phone. |
| 08:10.10 | wm501 | my first guess is that i need to add a [parkedcalls] context to extensions.conf? |
| 08:10.11 | wm501 | anyone/ |
| 08:12.11 | *** join/#asterisk mitcheloc_ (~mitcheloc@67.153.163.202) |
| 08:12.22 | mitcheloc_ | freat: did you get the answer you were looking for? |
| 08:15.56 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 08:18.23 | *** join/#asterisk multrix (~chatzilla@ALyon-110-1-12-50.w81-48.abo.wanadoo.fr) |
| 08:18.26 | wm501 | does anyone that isn't afk have experience in call parking? |
| 08:18.35 | mitcheloc_ | yea, whats up? |
| 08:18.56 | wm501 | I have no idea how to do this and I can't find any good examples on the wiki |
| 08:19.07 | wm501 | i have gotten as far as looking at the features.conf |
| 08:19.13 | wm501 | and i am satisfied with the default settings |
| 08:19.16 | mitcheloc_ | its' just a simple include => parking (or similar) |
| 08:19.23 | mitcheloc_ | what happens is this |
| 08:19.33 | wm501 | okay, i did that and restarted, so how to i park a call when they dial for example extension 1 |
| 08:19.36 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 08:19.37 | mitcheloc_ | the call parking module generates a dialplan when it's loaded by asterisk |
| 08:19.45 | wm501 | okay |
| 08:19.51 | mitcheloc_ | no, parking is different then what your thinking |
| 08:19.56 | wm501 | oh |
| 08:19.57 | mitcheloc_ | if your on a call you press # then 700 |
| 08:20.01 | mitcheloc_ | and it transfers it there |
| 08:20.06 | wm501 | okay... |
| 08:20.10 | mitcheloc_ | then you listen, and asterisk tells you which extension the call is at |
| 08:20.11 | wm501 | let me tell you what i need to do. |
| 08:20.13 | mitcheloc_ | then you hang up |
| 08:20.17 | mitcheloc_ | go to another phone and dial that extension |
| 08:20.21 | wm501 | ohhh |
| 08:20.33 | mitcheloc_ | not call queues, thats what you were thinking right? |
| 08:20.58 | wm501 | basically i need it to do this. they call our telephone number, hit extension 1, it puts them on hold and we can join them into a meetme conference 1 at a time. |
| 08:21.51 | mitcheloc_ | well...i don't see why you would do that, but you'll need to do something different, it'll take a bit of scripting |
| 08:22.17 | mitcheloc_ | basically park them, then have a webpage use the manager api to transfer them to the conference when your ready |
| 08:22.25 | wm501 | exactly |
| 08:22.28 | wm501 | that is what i want to do |
| 08:22.31 | wm501 | park them and use PHP |
| 08:22.46 | wm501 | to put them into our conference |
| 08:23.00 | wm501 | i can do the php, i just don't know how to park a call when they hit 1# |
| 08:23.18 | mitcheloc_ | why 1 #? |
| 08:23.25 | mitcheloc_ | just create an extension "1" |
| 08:23.36 | mitcheloc_ | and then use the park command |
| 08:23.45 | wm501 | okay |
| 08:23.52 | mitcheloc_ | i think "parkandannounce", i can look it up, but i'd just check the wiki, info is there |
| 08:24.05 | wm501 | this is what i have |
| 08:24.30 | wm501 | exten => 1,1,Answer |
| 08:24.30 | wm501 | exten => 1,2,Wait(1) |
| 08:24.32 | wm501 | then, i need |
| 08:24.57 | wm501 | exten => 1,3,parkandannounce(.....) |
| 08:25.12 | mitcheloc_ | why the wait(1)? |
| 08:25.22 | mitcheloc_ | but yea just use parkandannounce |
| 08:25.33 | wm501 | one more thing. |
| 08:26.10 | wm501 | when they first call in, we have an mp3 that plays and says "Thanks for calling .... press 1 to go on the air and 2 for the hotline." |
| 08:26.33 | wm501 | then i use read(${ext}) |
| 08:26.34 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) |
| 08:26.42 | wm501 | er |
| 08:26.44 | wm501 | read(ext) |
| 08:27.04 | wm501 | the problem is, if they hit 1# before the message gets done playing it disconnects the call |
| 08:27.23 | mitcheloc_ | oh cool, what radio station is this? |
| 08:27.24 | wm501 | sorry, i'm new to asterisk and probably havn't read all that i should |
| 08:27.29 | wm501 | naiasportstalk.com |
| 08:27.55 | wm501 | we just started it up tonight, and we have hotline calls working, but i can't get call parking for listeners to call in. |
| 08:28.16 | mitcheloc_ | hmm ok, hold on let me read that all again |
| 08:28.28 | wm501 | sorry, i didn't make that clear. |
| 08:28.34 | mitcheloc_ | oh your problem is read(${ext}) |
| 08:28.38 | wm501 | yeah, i thought so. |
| 08:28.40 | mitcheloc_ | just do a playback |
| 08:28.44 | mitcheloc_ | and wait for their input |
| 08:28.55 | mitcheloc_ | before the plabyack you need an answer, and some settimeout things |
| 08:29.07 | mitcheloc_ | hold on, i'm looking @ my config to get you the right ones |
| 08:29.29 | fa | mitcheloc_ give me to.. some example |
| 08:29.43 | wm501 | okay |
| 08:29.51 | mitcheloc_ | <PROTECTED> |
| 08:29.51 | mitcheloc_ | <PROTECTED> |
| 08:29.51 | mitcheloc_ | <PROTECTED> |
| 08:30.10 | fa | DigitTimeout is timeout for user press the key |
| 08:30.11 | mitcheloc_ | thats my first three, then priority 4 is a playback using "Background" |
| 08:30.22 | mitcheloc_ | yes, then make another extensiont hat is 1 and another that is 2 |
| 08:30.31 | mitcheloc_ | thats the best practice |
| 08:30.36 | wm501 | ohhh |
| 08:31.00 | wm501 | i was doing this |
| 08:31.03 | wm501 | [voicepulse-incoming] |
| 08:31.03 | wm501 | exten => _NXXNXXXXXX,1,Goto(outgoing,1000,1) |
| 08:31.17 | wm501 | then, 1000 plays the mp3, and read(ext) |
| 08:31.23 | wm501 | i don't know why i did it that way. |
| 08:31.52 | mitcheloc_ | your thinking of a scripting language i think, you know, step by step, top down format |
| 08:31.56 | fa | what is the best practice for open, close and holidays |
| 08:32.02 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 08:32.21 | fa | and maybye anybody now, is it module PGSQL to make SELECT in extensions |
| 08:33.13 | mitcheloc_ | fa: there is a way to include certain contexts based on the time, it's on the wiki somewhere |
| 08:33.41 | wm501 | okay, so i need to accept the call, play my mp3 using playback, set a digittimeout, (then they will be able to call, hit 1 right away [without pound]) and it will move them to extension 1) |
| 08:33.47 | fa | mitcheloc_ what about PGSQL - like this http://lists.digium.com/pipermail/asterisk-dev/2003-July/001052.html |
| 08:33.57 | wm501 | or do i need to convert my mp3 to gsm |
| 08:33.59 | *** join/#asterisk meppl (~mephisto@pD9542339.dip.t-dialin.net) |
| 08:34.07 | fa | wm501 if you want, yes |
| 08:34.31 | meppl | guten morgen |
| 08:34.42 | fa | meppl hi, engilsh please. thanks. |
| 08:35.02 | meppl | good morning fa |
| 08:35.23 | meppl | fa, so i make "/amsg guten morgen" you know |
| 08:35.34 | fa | nice |
| 08:35.38 | meppl | ("guten morgen"="good morning") |
| 08:36.21 | meppl | all other sentences from me are english here ;) |
| 08:37.00 | mitcheloc_ | fa:i don't know anything about postgres |
| 08:37.00 | fa | meppl do you know how to enable PGSQL in asterisk. to make query in extensions without AGI |
| 08:37.00 | meppl | almost all ("gute nacht"="good night") |
| 08:37.11 | wm501 | do i HAVE to convert my mp3 to .gsm to make playback work? |
| 08:37.15 | wm501 | havn't checked the wiki sorry |
| 08:37.32 | mitcheloc_ | wm501: yes |
| 08:37.43 | mitcheloc_ | at least to wav, but gsm is better |
| 08:37.46 | meppl | fa, i dont know it |
| 08:37.50 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 08:38.21 | wm501 | okay |
| 08:38.39 | wm501 | what is a good way to convert mp3 to gsm... or mp3 to wav then wav to gsm i guess |
| 08:39.31 | fa | wm501 soq |
| 08:39.34 | fa | sox |
| 08:39.34 | sjaak538 | I don't know if sox can do converting to mp3 but sox is mostly used for converting |
| 08:39.34 | mitcheloc_ | it's on the wiki, look up sox |
| 08:40.12 | wm501 | fa: thanks. |
| 08:40.14 | wm501 | thanks sjaak538 |
| 08:40.25 | Zeeek | I only recently learned that sox can do mp3 if you have the right version |
| 08:40.50 | *** join/#asterisk otiske (~otiske@kauai.sys.pas.earthlink.net) |
| 08:40.51 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 08:40.56 | Zeeek | I needed to cvt gsm and mp3 to wav and it worked well |
| 08:43.10 | *** join/#asterisk datareactor (datareacto@203.81.192.33) |
| 08:43.34 | fa | Zeeek where can i get application PGSQL |
| 08:44.13 | *** join/#asterisk pointer-gaim (~pointer@router.cathey.us) |
| 08:45.15 | *** join/#asterisk naturalvoice (naturalvoi@node-40247a6a.ewr.onnet.us.uu.net) |
| 08:45.21 | multrix | booo : "GOT SIP response 481 "Call Leg Does Not Exist"" what to do ? |
| 08:45.40 | wm501 | exten => 3,1,Answer |
| 08:45.40 | wm501 | exten => 3,2,Wait(1) |
| 08:45.40 | wm501 | exten => 3,3,Authenticate(56) |
| 08:45.40 | wm501 | exten => 3,4,MeetMe(101) |
| 08:45.40 | wm501 | exten => 3,5,Playback(vm-goodbye) |
| 08:45.41 | wm501 | exten => 3,6,Goto(outgoing,1000,1) |
| 08:45.43 | wm501 | oh SHIT |
| 08:45.45 | wm501 | i am so sorry |
| 08:45.55 | multrix | my SIP peer is authenticated when I do a SIP show peers |
| 08:46.01 | multrix | :( |
| 08:46.49 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 08:46.54 | mitcheloc_ | wm501: lets watch the language more then the flooding ;) |
| 08:46.55 | libpcp | hi guys |
| 08:46.58 | naturalvoice | Hi Gurus! I have my new IAXy here, how to create a provisioning system for it if I don't have a linux installed in this network ??? My asterisk server is in another public IP. (to anyone) |
| 08:48.17 | wm501 | sorry ;[ |
| 08:48.25 | wm501 | anyway, i found the syntax |
| 08:48.41 | libpcp | what should be the possible reason if the phone doesnt ring if someone call? does it have an issue with the firewall? |
| 08:48.43 | *** join/#asterisk three55ml (~who@cs666898-229.austin.rr.com) |
| 08:49.17 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 08:49.36 | wm501 | ParkAndAnnounce(pbx-transfer:PARKED|7200|SIP/${EXTEN:1}|default,${EXTEN:1},1) |
| 08:49.45 | *** join/#asterisk yxa (~void@203.118.40.42) |
| 08:50.15 | *** join/#asterisk HjemmeRoyK (~roy@110.80-203-29.nextgentel.com) |
| 08:50.19 | wm501 | so, my 1 extension should be |
| 08:50.21 | *** part/#asterisk mitcheloc_ (~mitcheloc@67.153.163.202) |
| 08:50.22 | wm501 | exten => 1,1,Answer |
| 08:50.46 | HjemmeRoyK | wm501: if people dial 1? |
| 08:51.12 | fa | wm501 use exten => s,1,Answer if it's start context |
| 08:51.19 | wm501 | okay |
| 08:51.28 | wm501 | then i want to park them in order starting at 700 |
| 08:52.02 | fa | so use goto |
| 08:52.06 | Mother_ | naturalvoice: how does it say it has to be provisioned? DHCP and TFTP me guesses? |
| 08:52.06 | *** join/#asterisk pranav (dawda_pran@203.115.89.185) |
| 08:52.13 | wm501 | what? |
| 08:52.21 | wm501 | now i'm really confuses |
| 08:52.28 | wm501 | i thought you parked them by using ParkAndAnnounce |
| 08:53.37 | wm501 | confused* |
| 08:54.14 | HjemmeRoyK | CrashAndPark works better :P |
| 08:54.28 | Zeeek | multrix |
| 08:54.35 | Zeeek | Case 2: Unreliable transport was used for REGISTER/200 OK transaction, |
| 08:54.35 | Zeeek | <PROTECTED> |
| 08:54.35 | Zeeek | <PROTECTED> |
| 08:54.35 | Zeeek | <PROTECTED> |
| 08:54.35 | Zeeek | <PROTECTED> |
| 08:54.35 | Zeeek | Case 2: Unreliable transport was used for REGISTER/200 OK transaction, |
| 08:54.53 | Zeeek | Hej! |
| 08:55.01 | *** join/#asterisk Tornad (~Tornad@81.255.65.249) |
| 08:55.08 | multrix | Zeeek: and so what is the problem ? :s |
| 08:55.12 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 08:55.12 | *** mode/#asterisk [+o denon] by ChanServ |
| 08:55.38 | Zeeek | multrix look here : http://lists.cs.columbia.edu/pipermail/sip-implementors/2001-June/001354.html |
| 08:55.44 | Zeeek | Interesting |
| 08:55.59 | *** join/#asterisk phoenix__ (~jeff@155.245.2.46) |
| 08:56.10 | Zeeek | naturalvoice - I have an IAXy what are you trying to do? |
| 08:56.17 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 08:56.25 | mAsH` | morning all |
| 08:56.27 | Zeeek | it is provisioned from the same server as the asterisk |
| 08:56.31 | phoenix__ | ello all |
| 08:56.39 | multrix | Zeeek: I don't see any relation with my problem on this site :s |
| 08:56.58 | Zeeek | 481 Leg does not exist multrix |
| 08:57.05 | *** join/#asterisk Nix (~Nix@dsl81-214-9283.adsl.ttnet.net.tr) |
| 08:57.27 | Zeeek | the answer is that someone didn't answer a in 40 seconds |
| 08:57.58 | phoenix__ | Zeeek: I fixed the problems from yesterday :D |
| 08:58.16 | Zeeek | great newsz! you should publish as the pbx problem comes up from time to time |
| 08:58.28 | Zeeek | put a note on the wiki or something! |
| 08:58.50 | multrix | Zeeek: but the 481 is provoqued by the fac I get a Forbidden - wrong password on auth. for INVITE TO "XXX <sip:xxx@a.b.c.d>,tag=as123345 |
| 08:59.14 | phoenix__ | it wasn't the pbx issue |
| 08:59.29 | phoenix__ | my desk-phone uses a bloody non-standard cable!!! |
| 09:00.05 | mAsH` | anyone used * with 2 hfc cards ? |
| 09:00.25 | Zeeek | that's even weirder! |
| 09:00.34 | phoenix__ | yup |
| 09:00.37 | phoenix__ | but it works now!! |
| 09:00.38 | HjemmeRoyK | mAsH`: I think that works |
| 09:00.42 | naturalvoice | <Zeeek> Sorry Zeeek, I'm trying to configure my IAXy ... |
| 09:00.50 | HjemmeRoyK | mAsH`: perhaps a little irq storm, but then |
| 09:00.55 | HjemmeRoyK | you get what you're paying for |
| 09:01.02 | datareactor | mAsh which mode both cards are running |
| 09:01.02 | *** join/#asterisk djin (~marius@62.58.40.196) |
| 09:01.13 | fa | HjemmeRoyK do you know how to install application PGSQL fot asterisk ? |
| 09:01.36 | HjemmeRoyK | fot? |
| 09:01.59 | Zeeek | naturalvoice what I do is use the provision program on my server but asterisk can also do it in the newer versions |
| 09:02.07 | *** join/#asterisk jluk (~jluk@pl6.lawrence.org.uk) |
| 09:02.09 | Makenshi | morning |
| 09:02.19 | Zeeek | the program is in /usr/src/iaxyprov IIRC |
| 09:02.55 | datareactor | fa do you read http://www.voip-info.org/wiki-Asterisk+cdr+pgsql |
| 09:03.02 | wm501 | how can i use the Dial() function to call an extension in my [incoming] context? |
| 09:03.21 | wm501 | i can't use goto in this situation. |
| 09:03.40 | pashah | morning! |
| 09:04.00 | naturalvoice | <Zeeek> Yes, but how to configure the IAXy to go my asterisk ??? |
| 09:04.18 | fa | datareactor i have postgrew for cdr, now i want for select in extensions |
| 09:04.24 | *** join/#asterisk wolfson (~hehe@bcp-68-187-181-159.man.nc.charter.com) |
| 09:04.34 | jointe_79 | is there anyone who have Packegen and packetscan from GL. The dowloadlink by them doesnt work so... |
| 09:04.37 | djin | naturalvoice, there is a little piece of software for to push the config to the IAXy. |
| 09:05.00 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 09:05.02 | Zeeek | the opposite the asterisk talks to IAXy - you should forward port 4569 on local router to the IAXy then run provision 123.123.123 iaxy.conf |
| 09:05.03 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@d01m-6-3.d4.club-internet.fr) |
| 09:05.24 | Zeeek | this is one little gotcha of the IAXy if it is behind NAT |
| 09:05.31 | fa | Feb 3 10:05:12 WARNING[732]: pbx.c:1390 pbx_extension_helper: No application |
| 09:05.32 | fa | GSQL' for extension (outgoing, 80691761693, 1) |
| 09:05.41 | naturalvoice | <Zeeek> Great !!! I'll try this. |
| 09:06.09 | Zeeek | make up the iaxy.conf from the sample changing all the stuff obviously |
| 09:06.29 | Zeeek | I think you have to download the provi=sion prog - it's a simple make of a couple files |
| 09:08.14 | *** join/#asterisk Tough_Nuts (~chatzilla@26.137.202.68.cfl.rr.com) |
| 09:08.19 | Jas_Williams | You can provision the iaxy directly from asterisk |
| 09:08.34 | Jas_Williams | iax2 provision plus options |
| 09:08.40 | Zeeek | I've never tried it |
| 09:08.59 | Zeeek | because the server is on a dynamic ip I have to jump thru a few extra hoops |
| 09:09.13 | *** join/#asterisk tty74 (~tiziano@151.11.170.20) |
| 09:09.13 | Zeeek | like generating the conf file when the ip changes :) |
| 09:09.28 | pranav | hello zeeek |
| 09:09.37 | Zeeek | I hope to get a static ip RealSoonNow |
| 09:09.43 | Zeeek | lo pranav |
| 09:10.18 | alakdan | Zeeek: I also recently got iaxy, I can make outbound calls aixy -> my * box -> voip provider. But I can not make 'local' calls to the iaxy (it does not ring). any sample config for the iaxy on the asterisk box part? |
| 09:10.38 | alakdan | that you can share? :) |
| 09:11.00 | Zeeek | alkadan - I have the same problem |
| 09:11.11 | Zeeek | It won't ring some non-US phones |
| 09:11.25 | Zeeek | If someone has a fix I'm all ears! |
| 09:11.37 | *** join/#asterisk naif (~User@host250-27.pool62110.interbusiness.it) |
| 09:11.53 | naif | hi! anyone have ever hear of IAX2e or IAXe as an encrypted IAX protocol? |
| 09:11.55 | pranav | i have a pstn line connected to digium card and i have two internal lines sipura spa 2000 |
| 09:12.02 | Tough_Nuts | good morning all.... my 1st time in the * irc. thought I would check it out, see if maybe anyone here could help me with this brand new TDM11B I just got and trying to make it work under CentOS v3.3 which is to my understanding a RHEL3 knock off.. |
| 09:12.18 | pranav | but i cant make calls between the external and the internal lines |
| 09:12.43 | Zeeek | Tough_Nuts where are you in the install? |
| 09:12.45 | tty74 | Hi, how i can change in the cisco 7940 phone (SIP image) the label of menu button, because when i answer i can't view immediately transfer anb blind transfer (without press more button) ? Thank's Tiziano |
| 09:12.46 | Jas_Williams | pranav: sounds like an extensions.conf issue |
| 09:12.54 | alakdan | Zeeek: oh sorry, just local extensions I mean. say I have to extensions 1234 routes to the iaxy, and 2468 routes to my x-lite. when I dial 1234 from my x-lite seems to have some errors |
| 09:13.21 | Zeeek | post the errors to pastebin |
| 09:13.24 | pranav | hello mr jas. i made the chaneges in the extensions.conf |
| 09:13.33 | Tough_Nuts | trying to get the zaptel modules to load.. keep getting some errors when modprobe runs.. even tho I checked it out from CVS.. |
| 09:13.47 | pranav | should i paste my extensions.conf |
| 09:14.02 | Jas_Williams | tty74: You cannot change the lables on a sip image |
| 09:14.09 | naturalvoice | WORKED !!! <Thanks guys!> Thanks Zeeek, Jas_Williams. |
| 09:14.10 | fa | where to take pgsql addon? |
| 09:14.17 | Zeeek | good news! |
| 09:14.32 | Zeeek | theoretically you can now close the port forwarding |
| 09:15.10 | naturalvoice | Upsssssss.... I got an error in my *. |
| 09:15.14 | naturalvoice | chan_iax2.c:5725 socket_read: Rejected connect attempt from 64.36.122.106, requested/capability 0x4/0x24 incompatible with our capability 0xff03. |
| 09:15.24 | Zeeek | codec error |
| 09:15.30 | Tough_Nuts | error message I get when loading modprobe wcfxs is this: |
| 09:15.32 | Tough_Nuts | [root@asterisk1 etc]# modprobe wcfxs |
| 09:15.33 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: init_module: No such device |
| 09:15.35 | Tough_Nuts | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 09:15.36 | Tough_Nuts | <PROTECTED> |
| 09:15.36 | Zeeek | you used ulaw in the iaxy ? |
| 09:15.38 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: insmod /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o failed |
| 09:15.39 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: insmod wcfxs failed |
| 09:15.44 | pranav | Jas_Williams:i have put the commands for the external line in the context internal |
| 09:15.58 | pranav | do i need to change that |
| 09:16.17 | naturalvoice | Zeeek: Yes. |
| 09:16.41 | Zeeek | and in the iax.conf entry? disallow=all ? |
| 09:16.46 | Jas_Williams | pranav: Can you post your extensions.conf to pastebin.cs |
| 09:16.49 | Zeeek | then allow=ulaw |
| 09:16.55 | Jas_Williams | opps pastebin.ca |
| 09:17.18 | pranav | yes certainly, just a minute |
| 09:17.31 | Zeeek | Tough_Nuts tell us what steps you have performed to get where you are? |
| 09:18.25 | naturalvoice | Zeeek: WORKED ! Thanks. |
| 09:18.42 | Zeeek | cool! |
| 09:19.00 | Zeeek | watch out for edgy power supplies on the IAXy - it's very sensitive |
| 09:19.06 | Zeeek | needs 1200ma |
| 09:19.52 | tty74 | Mmmmmmmm there is a shortcut for cisco 7940 for transfer or blind transfer immediately when i answer a call (is difficult evrery time presse more an then transfer) |
| 09:20.08 | naturalvoice | Zeeek: In this case I "HAVE" to delete the port forwarding in my router. If not anyone can re-provisioning my adapter to another place... Right ? |
| 09:20.15 | alakdan | Zeeek: here is the error http://rafb.net/paste/results/Onbk3x16.html |
| 09:20.17 | *** join/#asterisk Darwin35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 09:20.26 | Darwin35 | good morning |
| 09:20.35 | Zeeek | naturalvoice - right |
| 09:20.58 | naturalvoice | Thanks guys! Time to sleep ! |
| 09:21.58 | Zeeek | alkadan looks like codec error again |
| 09:22.06 | fa | Darwin35 hi. do you know how to get and install PQSQL extensions? for postgresql ? |
| 09:22.29 | Darwin35 | not on linux |
| 09:22.31 | Jas_Williams | alakdan: What codecs are you running on the sip device ? |
| 09:22.34 | Darwin35 | I use fbsd |
| 09:22.47 | soundguy | whats the command to show codecs during a call in the asterisk console? |
| 09:22.57 | alakdan | Jas_Williams: x-lite seems to use gsm |
| 09:23.00 | Jas_Williams | sip show channels |
| 09:23.01 | Darwin35 | sip show channel |
| 09:23.02 | Zeeek | it's in the show channels stuff |
| 09:23.11 | Zeeek | wow three on one! |
| 09:23.27 | wasim | ast_orgy |
| 09:23.28 | Zeeek | talk about wearing the right aftershave! :) |
| 09:23.29 | Tough_Nuts | ok... |
| 09:23.30 | Tough_Nuts | 1. Checkout Zaptel Drivers from CVS |
| 09:23.32 | Tough_Nuts | |
| 09:23.33 | Tough_Nuts | cd /usr/src/ |
| 09:23.35 | Tough_Nuts | |
| 09:23.37 | Tough_Nuts | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 09:23.38 | Tough_Nuts | |
| 09:23.38 | Zeeek | wasim... what's erm, NEW? |
| 09:23.40 | Tough_Nuts | cvs login |
| 09:23.41 | Tough_Nuts | password is anoncvs |
| 09:23.43 | Tough_Nuts | |
| 09:23.44 | Tough_Nuts | cvs checkout zaptel |
| 09:23.46 | Tough_Nuts | 2. Compiling Zaptel Drivers |
| 09:23.48 | Tough_Nuts | |
| 09:23.48 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:23.49 | Tough_Nuts | cd /usr/src/zaptel/ |
| 09:23.51 | Tough_Nuts | |
| 09:23.52 | Tough_Nuts | make clean |
| 09:23.54 | Tough_Nuts | |
| 09:23.55 | Tough_Nuts | make install |
| 09:23.56 | Zeeek | use the pastebin Tough |
| 09:23.57 | Tough_Nuts | 3. Configuring zaptel.conf for use with a TDM11B |
| 09:23.59 | Tough_Nuts | # |
| 09:24.00 | Tough_Nuts | # Zaptel Configuration File |
| 09:24.02 | Tough_Nuts | # |
| 09:24.02 | wasim | Zeeek: trying to get your mailing address to my german warehouse |
| 09:24.04 | Tough_Nuts | fxoks=1 |
| 09:24.07 | Tough_Nuts | fxsks=4 |
| 09:24.10 | Tough_Nuts | loadzone=us |
| 09:24.11 | Tough_Nuts | defaultzone=us |
| 09:24.13 | HjemmeRoyK | ~pastebin? |
| 09:24.14 | jbot | hmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 09:24.14 | Tough_Nuts | |
| 09:24.15 | Tough_Nuts | 4. modprobe wcfxs |
| 09:24.16 | Zeeek | wasim I have my cust here newt week |
| 09:24.17 | Tough_Nuts | then BOOM ! and this: |
| 09:24.18 | Darwin35 | tough pastebin.ca |
| 09:24.19 | Tough_Nuts | [root@asterisk1 etc]# modprobe wcfxs |
| 09:24.22 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: init_module: No such device |
| 09:24.22 | libpcp | sip show channels shows alot of entries, what does this channels do? |
| 09:24.23 | Tough_Nuts | Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. |
| 09:24.25 | Tough_Nuts | <PROTECTED> |
| 09:24.25 | Darwin35 | stop flooding |
| 09:24.27 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: insmod /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o failed |
| 09:24.29 | HjemmeRoyK | /kick Tough_Nuts |
| 09:24.30 | Tough_Nuts | /lib/modules/2.4.21-20.EL.c0/misc/wctdm.o: insmod wcfxs failed |
| 09:24.31 | Tough_Nuts | thats it.. I checked dmesg and /var/log/messages.. they just said that it loaded then unloaded.. |
| 09:24.33 | Tough_Nuts | :) |
| 09:24.35 | Tough_Nuts | sorry for the spam.. |
| 09:24.37 | Tough_Nuts | its a long process.. |
| 09:24.39 | Tough_Nuts | pastebin ? |
| 09:24.41 | HjemmeRoyK | ~lart Tough_Nuts |
| 09:24.52 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 09:24.56 | Zeeek | http://pastebin.ca |
| 09:24.57 | HjemmeRoyK | jbot: tell Tough_Nuts about pastebin |
| 09:24.58 | soundguy | hmm..soemthing is weird |
| 09:25.02 | Zeeek | dump it all in there and post the link |
| 09:25.03 | Darwin35 | learn to use pastebin.ca |
| 09:25.11 | Jas_Williams | ~pastebin |
| 09:25.13 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 09:25.14 | Zeeek | ok this is his first time |
| 09:25.20 | Tough_Nuts | ok.. cool |
| 09:25.22 | Tough_Nuts | <PROTECTED> |
| 09:25.23 | Tough_Nuts | didnt know about that.. |
| 09:25.36 | soundguy | when I get a phonecall from an external source and I picked up the phone I cant hear any audio, however if I leave it go to message bank audio can be heard (at each end) |
| 09:25.37 | soundguy | Any ideas? |
| 09:25.55 | *** join/#asterisk Darwin_35 (~darwin35@c-24-3-241-22.client.comcast.net) |
| 09:26.16 | Darwin35 | wheres my boots |
| 09:26.21 | libpcp | Jas_Williams: what does it mean by the entries of "sip show channels" |
| 09:26.28 | libpcp | is that a call channels? |
| 09:26.36 | *** join/#asterisk Dibbler_ (~Dibbler@zidane.pi-net.net) |
| 09:26.39 | Darwin35 | help show |
| 09:26.44 | Darwin35 | help channels |
| 09:26.47 | Darwin35 | help sip |
| 09:26.51 | Darwin35 | read them |
| 09:27.33 | pranav | Jas_Williams: I have pasted it in pastebin.ca/5197 |
| 09:27.55 | *** join/#asterisk zeedo (~notroot@www.bsrf.org.uk) |
| 09:28.12 | Jas_Williams | libpcp: show codecs |
| 09:28.56 | *** join/#asterisk meppl (~mephisto@pD9542339.dip.t-dialin.net) |
| 09:29.17 | Darwin35 | remote updates |
| 09:29.28 | Darwin35 | gotta love em |
| 09:30.34 | Tough_Nuts | ok... I did the pastbin, thats SO cool.. Never knew it was out there.. Here is the link: |
| 09:30.36 | Tough_Nuts | http://pastebin.ca/5198 |
| 09:30.37 | Tough_Nuts | Thanks. |
| 09:32.05 | Darwin35 | ok 3 questions |
| 09:32.12 | Darwin35 | what flavor of linux |
| 09:32.28 | pranav | Mr.Jas_Williams:i have made context=from-pstn in the zapata.conf and context=internal in the sip.conf |
| 09:32.32 | Darwin35 | did you get all the updated 1.0.5 ver of the src |
| 09:33.04 | Darwin35 | and what you make ing me for breakfast for being here so early to help you out |
| 09:33.09 | *** join/#asterisk muchtodo (~jack@82-32-5-69.cable.ubr01.azte.blueyonder.co.uk) |
| 09:33.10 | datareactor | pranav the line exten =>_.,1,($phone1,30) |
| 09:33.13 | Jas_Williams | pranav: Which phone is trying to call what |
| 09:33.24 | datareactor | is not correct |
| 09:33.59 | Darwin35 | ${phone1} |
| 09:34.00 | pranav | i am trying external phone to connect to the internal |
| 09:34.04 | Tough_Nuts | 1. CentOS v3.3, which is a knock off RHEL3.. |
| 09:34.05 | Tough_Nuts | 2. Well the problem is in the zaptel, and should have nothing to do with asterisk. But its all comming from the current CVS. |
| 09:34.07 | Tough_Nuts | 3. French Toast, with Bacon... |
| 09:34.21 | pranav | then what should be the change |
| 09:35.06 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 09:35.20 | alakdan | Zeeek, Jas_Williams: Thanks a lot. |
| 09:35.25 | alakdan | :) |
| 09:35.30 | alakdan | its now working |
| 09:35.32 | Darwin35 | tough hold a min |
| 09:35.44 | libpcp | Jas_Williams: i got UNK of all the channels in the format field |
| 09:35.49 | Tough_Nuts | ok.. no prb.. :) |
| 09:35.53 | datareactor | pranav where you are trying to call |
| 09:35.55 | slePP | anyone have modems behind asterisk/channel bank? |
| 09:36.33 | Darwin35 | you hsve to lod modeprobe zaptel first |
| 09:36.38 | HjemmeRoyK | slePP: should work.... |
| 09:36.43 | *** join/#asterisk sean33 (~yaknow@gw.neurometrics.net) |
| 09:36.45 | slePP | HjemmeRoyK: heh. that's the theory... |
| 09:36.45 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 09:36.47 | slePP | but does it? :> |
| 09:36.56 | pranav | I am first trying to make a call to the internal phone i.e with extension 2000 |
| 09:36.58 | HjemmeRoyK | Darwin35: modprobe autoloads dependant modules |
| 09:37.02 | slePP | <PROTECTED> |
| 09:37.02 | slePP | <PROTECTED> |
| 09:37.02 | slePP | <PROTECTED> |
| 09:37.07 | HjemmeRoyK | slePP: in theory, yes :P |
| 09:37.08 | slePP | the modem is just barely hanging on |
| 09:37.14 | slePP | about 70% packet loss |
| 09:37.22 | HjemmeRoyK | shite |
| 09:37.34 | sean33 | hi im a newbie to asterisk and im tring to install under fedora core2 is there a rpm or yum install ? |
| 09:37.38 | pranav | but when i dial 2000 from the external phone the call does not go |
| 09:37.46 | HjemmeRoyK | sean33: no, no, no, no, NO |
| 09:37.52 | HjemmeRoyK | sean33: use the source, luke |
| 09:37.59 | HjemmeRoyK | ~lart sean33 |
| 09:38.00 | sjaak538 | Can I use telnet ipAddress 5036 to test firewall open on IAX2 port for my customers ?? |
| 09:38.12 | Delvar | morning |
| 09:38.12 | Darwin35 | you did not install libpri did you |
| 09:38.20 | HjemmeRoyK | sjaak538: heh. no. |
| 09:38.24 | Zeeek | French toast, like French fries isn't French. Bacon isn't bacon. But ulaw is ulaw |
| 09:38.27 | HjemmeRoyK | sjaak538: IAX2 is UDP |
| 09:38.32 | Tough_Nuts | nope.. dont need it.. libpri.. |
| 09:38.34 | sjaak538 | Argh |
| 09:38.41 | sean33 | ok so what do i do then people ? |
| 09:38.48 | sjaak538 | is there a way to test it on another way |
| 09:38.57 | Jas_Williams | IAX2 is port 4569 |
| 09:39.01 | Jas_Williams | UDP |
| 09:39.02 | HjemmeRoyK | sjaak538: see http://voip-info.org/tiki-index.php?page=Asterisk%20monitoring |
| 09:39.46 | pranav | what is the mistake |
| 09:40.02 | sean33 | anyone want to help me get started ? |
| 09:40.43 | Jas_Williams | pranav: Sipura dialplan ? |
| 09:40.44 | Darwin35 | you have to edit your zapata file also |
| 09:41.02 | Darwin35 | zapata.cooonf |
| 09:41.16 | Darwin35 | sticky keys |
| 09:41.22 | wm501 | if i am talking to someone |
| 09:41.28 | wm501 | how can i transfer them to a different extension? |
| 09:41.32 | datareactor | pranav make only one context to make to things easy for you |
| 09:41.33 | wm501 | *extension? |
| 09:41.46 | datareactor | pranav what context users have in sip.conf |
| 09:42.05 | pranav | Jas_Williams:i have put everything in the extensions.conf |
| 09:42.05 | Tough_Nuts | thats not needed for the driver just to load is it ? |
| 09:42.07 | Tough_Nuts | I dont care about asterisk yet, just want all the modprobes to work and ztcfg. |
| 09:42.37 | Darwin35 | yes I am going threw mine and compaiiring |
| 09:42.41 | *** join/#asterisk Othello (Othello@nusnet-156-21.dynip.nus.edu.sg) |
| 09:42.46 | pranav | datareactor:i have used context=incoming in the sip.conf |
| 09:43.13 | Darwin35 | if you read you have to modeprobe zaptel before you mode probe the wcfxs |
| 09:43.24 | pranav | datareactor: do you want me to paste the sip.conf |
| 09:44.00 | Tough_Nuts | did that, forgot to post it... still same msg... |
| 09:45.47 | pranav | should i put context=default everywhere |
| 09:46.00 | Zeeek | wm501 what kind of phone are you talking on? |
| 09:46.18 | pranav | bcos in my zapata.conf i have put context=from-pstn |
| 09:46.29 | Zeeek | if connected to ZAP just flash dial the extension and announce |
| 09:46.49 | tzafrir | pranav, the context tells asterisk where to start in the dialplan when a call coms from that channel |
| 09:47.00 | Zeeek | if the phone can't do attended xfers you have to just flash, dial and hangup |
| 09:47.02 | sean33 | thanks for the help 'NOT!' |
| 09:47.10 | wm501 | okay basically i have callers in a queue, then i answer a call, then i need to transfer them, right now when i hit *, it just disconnects the other person and puts me (an agent) on musiconhold |
| 09:47.14 | wm501 | zeeek: ^ |
| 09:47.20 | Zeeek | sean33 you have a question? |
| 09:47.25 | djin | bye bye, sean33. |
| 09:47.37 | wm501 | it is iax2 |
| 09:47.49 | Zeeek | ah queue, that may be different. |
| 09:47.58 | Zeeek | not sure since I odn't use queues |
| 09:48.07 | wm501 | okay, thanks anyway |
| 09:48.08 | wm501 | ;) |
| 09:48.23 | fa | djin are you? |
| 09:48.26 | pranav | ok |
| 09:48.41 | Darwin35 | hmmm |
| 09:48.43 | djin | fa, I am? |
| 09:48.52 | datareactor | pranv paste your sip.conf do you want to first dial internally |
| 09:49.00 | Zeeek | quiet day today djin... |
| 09:49.22 | fa | djin I am looking for PGSQL (like MYSQL) addons to make simple select from extensions. do you know where i can find it.. i was on google, on digium, on wiki.. and i can't find. |
| 09:49.30 | djin | is it? Just started looking ;) |
| 09:49.34 | jointe_79 | is there anyone who have Packegen and packetscan from GL. The dowloadlink by them doesnt work so... |
| 09:49.42 | pranav | i have two internal lines and 1 external , i an succesfully able to dial internally , but not from internal to external |
| 09:50.01 | djin | fa, Asterisk Realtime? |
| 09:50.22 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 09:50.29 | fa | djin no. o ast_data. and no realtime. i want make s,1,PGSQL(SELECT blablabla) |
| 09:50.47 | cjk | hi, is there an embedded asterisk linux OS out somewhere on the net? |
| 09:50.55 | djin | cjk, yes. |
| 09:51.03 | fa | djin like this http://lists.digium.com/pipermail/asterisk-users/2003-May/012632.html |
| 09:51.05 | Darwin35 | tooughnuts |
| 09:51.11 | djin | ther is a Gentoo <256Mb somewhere. |
| 09:51.21 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:51.38 | Darwin35 | did you by chance list the fxs and fxo in the wrong order in the conf file |
| 09:52.23 | *** part/#asterisk alakdan (~alakdan@210.213.185.59) |
| 09:53.11 | fa | djin you know? |
| 09:53.19 | soundguy | Anyone see my problem above and have any ideas? |
| 09:53.38 | soundguy | I did a "sip show channels" and when the phone is used it uses ulaw and message bank uses ulaw too |
| 09:53.39 | soundguy | I hav eno idea |
| 09:54.02 | djin | fa, don't have PostgreSQL experience. |
| 09:55.17 | wm501 | If I am an agent and i get sent a caller using Queue(myQueue|t) how do I as an agent transfer that caller to an extension? |
| 09:55.51 | *** part/#asterisk sean33 (~yaknow@gw.neurometrics.net) |
| 09:56.16 | wm501 | The wiki says Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call. |
| 09:56.24 | wm501 | but i don't know what that means |
| 09:57.20 | fa | djin i found that name is Simple PostgreSQL Interface - Do several SQLy things (http://asterisk.drunkcoder.com/apps.html) but i can't find that anywhere |
| 09:58.38 | Darwin35 | tes festive 1.95 works with asterisk on fbsd |
| 09:59.10 | Darwin35 | now to get sphiinx 4 working |
| 09:59.40 | Zeeek | wm501 you want to start looking here |
| 09:59.42 | Zeeek | http://www.voip-info.org/tiki-index.php |
| 09:59.51 | Zeeek | for transfer and parking |
| 09:59.57 | *** join/#asterisk forrestc{hm} (~forrestc@iMach.com) |
| 09:59.58 | wm501 | I looked there zeeek, I just don't see examples of # transfering |
| 10:00.10 | forrestc{hm} | Hello everyone. |
| 10:00.13 | wm501 | let me look in parking, |
| 10:00.21 | Zeeek | I think parking will tell you |
| 10:00.38 | Zeeek | but like I said I haven't used queues (other than just to test a queue) |
| 10:01.23 | Darwin35 | man this rocks . I have everything working on fbd/asterisk execp g729 |
| 10:01.40 | Darwin35 | and waiting to hear back from digiuum on that one |
| 10:02.28 | soundguy | Anyone? |
| 10:03.22 | forrestc{hm} | soundguy: what was your question.. Just came on |
| 10:03.25 | Darwin35 | thats called a native connection soundguy |
| 10:04.11 | wm501 | zeeek: no parkedcall parkandannounce and call parking don't talk about # transfer |
| 10:04.18 | soundguy | When I get external calls and I pick up either end can not hear audio, however if I leave it ring out and it go to voicemail you can hear it and leave a message fine |
| 10:04.43 | soundguy | When I pickup I do a "sip show channels" and it says it is using ulaw, same when message bank gets it |
| 10:04.45 | soundguy | Any ideas? |
| 10:05.21 | Darwin35 | how is he call comming in to the system ? |
| 10:05.30 | fa | anyone use PGSQL apps? |
| 10:05.36 | Darwin35 | from sip or zap |
| 10:05.44 | tzafrir | There's a debian bug about * I don't sure I understand: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=293124 |
| 10:05.45 | soundguy | sip |
| 10:05.57 | soundguy | oh wait, I have narrowed it down. It must be something to do with the Grandstream phone |
| 10:06.08 | soundguy | I just made all incoming calls to go to my softphone (xlite) and it wokred |
| 10:06.41 | soundguy | Now this is confusing |
| 10:06.47 | tzafrir | If I understand this correctly, the user wants to drop privilges (-U, -G) but to run the Asterisk user with all the groups of the user asterisk |
| 10:07.07 | tzafrir | e.g: add it to the group "sound" as well |
| 10:07.33 | soundguy | That to me tzafrir? |
| 10:07.44 | tzafrir | no |
| 10:07.51 | soundguy | ok |
| 10:07.58 | tzafrir | It just happens to be on that group in my settings |
| 10:08.25 | slePP | this whole modem via asterisk thing is so very close to working, it's pissing me off :P |
| 10:09.03 | Darwin35 | modem to * ? |
| 10:09.15 | HjemmeRoyK | http://blink.dagbladet.no/user/files/c/ch/choclad/Artig_baby.jpg |
| 10:09.25 | slePP | pri -> t1 card -> asterisk -> t1 card -> channel bank -> modem bank |
| 10:09.25 | forrestc{hm} | Is there a command that I can run to verify that asterisk actually knows about a PRI and thinks the channels are active? |
| 10:09.33 | HjemmeRoyK | slePP: running asterisk with -p ? |
| 10:09.36 | forrestc{hm} | by active I mean like working... |
| 10:09.41 | Darwin35 | ahhh ok |
| 10:09.43 | forrestc{hm} | as opposed to in a call. |
| 10:09.46 | mAsH` | anyone can help me to connect * to an old PABX? |
| 10:09.57 | forrestc{hm} | mAsH: Depends |
| 10:10.04 | djin | mAsH, how old? |
| 10:10.05 | slePP | HjemmeRoyK: i don't know that that'll make much difference, since it is currently connected to the modem and the CPU load is. uh. 0. :> |
| 10:10.08 | forrestc{hm} | mAsH: what do you want to do? |
| 10:10.14 | slePP | zeon 3.2ghz w/ 2mb cache |
| 10:10.18 | slePP | xeon |
| 10:10.34 | djin | mAsH, As long as it can simulate lines. |
| 10:10.38 | slePP | i'll try for fun, thoug |
| 10:10.54 | soundguy | Darwin35 any ideas? |
| 10:10.56 | mAsH` | no no, just 2 years :) |
| 10:11.13 | mAsH` | i would call my extension with asterisk can i do it? |
| 10:11.26 | Darwin35 | sound I shot my gs phone aweek ago |
| 10:11.38 | forrestc{hm} | mAsH: depends more on the PBX than on asterisk. |
| 10:11.40 | Darwin35 | it pissed me off |
| 10:12.07 | Darwin35 | 9mil whas fun for o 2 sec |
| 10:12.28 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
| 10:12.32 | mAsH` | i knwon :/ |
| 10:12.39 | Darwin35 | update your flash on the gs and check your configs |
| 10:12.48 | Darwin35 | in the phone |
| 10:12.56 | HjemmeRoyK | slePP: try it |
| 10:13.02 | slePP | HjemmeRoyK: no change |
| 10:13.04 | HjemmeRoyK | ok |
| 10:13.05 | forrestc{hm} | mAsh: asterisk looks like either a standard FXO or FXS (phone line or phone set), or a Channelized T1/PRI (either CO or PBX) |
| 10:13.15 | slePP | bing-bong-bing-bong.... shhhhhhhhhhhhhhhhhhhh |
| 10:13.20 | slePP | Connected, 26.4 |
| 10:13.21 | HjemmeRoyK | slePP: file a bug report :P |
| 10:13.25 | slePP | 80% packet loss :> |
| 10:13.27 | forrestc{hm} | mAsH: so it really depends on where you hook it to the PBX and what you can do on that PBX port. |
| 10:13.47 | *** join/#asterisk nazgool (~nazgoool@gatekeeper-e0.twc.de) |
| 10:13.48 | nazgool | hi |
| 10:13.55 | slePP | you'd think the zaptel driver could be told 'copy these channels identically to one another, asap' |
| 10:14.25 | Darwin35 | bbiab shower time |
| 10:15.02 | forrestc{hm} | mAsh: if you use a FXS card and hook it into an Analog extension you should be able to do anything you can do from an extension. I.E. dial other extensions, etc. |
| 10:16.38 | cjk | djin, sorry fot the late answer, but thanks... |
| 10:18.55 | slePP | that sucked |
| 10:18.59 | slePP | i think i _just_ got it working |
| 10:19.02 | slePP | and my laptop battery died |
| 10:19.20 | forrestc{hm} | Let me repeat my earlier question.. I've got a PRI plugged into an asterisk box. I *think* everything is ok (ztcfg looks fine, etc. etc.), but I can't seem to find any command within asterisk to verify that the PRI is actually working. |
| 10:19.43 | forrestc{hm} | Like, it thinks it's got 23 channels live and everything. |
| 10:20.12 | slePP | zttool |
| 10:21.37 | slePP | if windows goes into standby during modem dialup connecting phase |
| 10:21.39 | slePP | it gets stuck |
| 10:21.39 | slePP | hah |
| 10:22.09 | forrestc{hm} | Does anyone know how to get asterisk to tell you what channels it knows about as opposed to which ones have an active call on it? |
| 10:22.11 | multrix | how to deal with this : if my phone calls 0012345678, it actually calls 3312345678@sipprovider but if we call 0..004412345678 it calls international code 44 with folowing code, so : 4412345678 |
| 10:22.15 | multrix | a sort of "default international code" actually |
| 10:22.20 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 10:22.20 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:22.59 | fa | cypromis hey |
| 10:24.31 | forrestc{hm} | multrix: have you dug through your dial plan? |
| 10:26.19 | fa | sf.net cvs is dead. anyone have a iax library wiax.dll in sources from cvs? |
| 10:28.40 | multrix | I did this : exten => _0XXXXXXXXXX,1,Dial(SIP/*833${EXTEN:1}@sipprovider) |
| 10:29.12 | multrix | this works, when I do 0012345678 it calls *83312345678@sipprovider (*8 is the IAC for my provider) |
| 10:29.39 | multrix | but I want to go farther : deal with 000 at the beginning |
| 10:29.55 | naif | iaxclient library are a dead project? |
| 10:29.56 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 10:29.57 | multrix | but what is the rule for orders in exten ? |
| 10:30.49 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 10:31.46 | forrestc{hm} | multrix: It does some wierd things. |
| 10:32.37 | multrix | forrestc{hm}: huh ? |
| 10:32.49 | forrestc{hm} | multrix: I think the command show dialplan [contextname] will show you the sort order |
| 10:33.15 | forrestc{hm} | multrix: asterisk sorts the dialplan - but I haven't ever figured out what rules it uses to do so. |
| 10:34.01 | forrestc{hm} | multrix: the best way to force it is to include another context. |
| 10:34.29 | forrestc{hm} | multrix: thus if you have something like: |
| 10:34.31 | multrix | forrestc{hm}: My aim is to deal phone calls like a normal PBX, people do 0 for going out, then 00 for IAC, then prefix and number, if local country, only one 0 |
| 10:34.50 | Zeeek | multrix - the wiki has a page that gives the details of how to force the *order* of evaluation of contexts |
| 10:35.10 | forrestc{hm} | multrix: Zeek's got a point. |
| 10:35.24 | forrestc{hm} | multrix: let me see here... |
| 10:35.29 | Zeeek | I have no direct link though :( |
| 10:35.31 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 10:35.35 | forrestc{hm} | multrix... see http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 10:35.44 | Zeeek | impressive! |
| 10:36.43 | forrestc{hm} | multrix: for what you are doing, through, you may not need to force it if you can come up with a list of rules which only have one match each. |
| 10:36.51 | multrix | forrestc{hm}: and after dealing this, I want to have scripts who redirect phone calls to different providers (analog, SIP, etc... for best price) |
| 10:37.07 | multrix | for exemple : if country code is germany, use X provider because calls are cheap to germany |
| 10:37.31 | forrestc{hm} | multrix: probably the best way is for you write down all the numbers, giving each a range and figure out how to write rules for them. |
| 10:38.33 | multrix | exten => _000.,1,Dial(SIP/*8${EXTEN:4}@sipprovider) |
| 10:38.34 | forrestc{hm} | For example, all us-dialed numbers start with a 1, then have a digit which isn't a zero or a one, followed by 9 digits. |
| 10:38.34 | multrix | exten => _0XXXXXXXXXX,1,Dial(SIP/*833${EXTEN:1}@sipprovider) |
| 10:38.42 | multrix | this is incorrect... |
| 10:39.26 | forrestc{hm} | the first exten, needs to be changed, since it matches the exact same thing as the second one. |
| 10:39.50 | forrestc{hm} | What makes _0XXXX different from _000... ? |
| 10:39.59 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
| 10:40.02 | forrestc{hm} | Does the first X need to be NOT zero? |
| 10:40.41 | multrix | yes, how to say "number not beeing a zero" ? I saw that on a doc but I couldn't find this back ! |
| 10:40.45 | multrix | N ? |
| 10:40.55 | forrestc{hm} | I.E. _0ZNNNNNNNNN? |
| 10:41.03 | HjemmeRoyK | ZZZZZZZzzzzzzzzz......... |
| 10:41.12 | multrix | lol |
| 10:41.13 | forrestc{hm} | Which would match 01234234, but not 00230948290384 |
| 10:41.14 | multrix | :p |
| 10:41.19 | multrix | I'm getting crazy with asterisk :) |
| 10:41.27 | forrestc{hm} | Z matches 1...9, not Zero. |
| 10:41.37 | HjemmeRoyK | [1-9] matches 1-9 |
| 10:41.40 | HjemmeRoyK | :P |
| 10:41.49 | multrix | ok very good it's what I need :) |
| 10:41.53 | multrix | let me try... |
| 10:41.58 | forrestc{hm} | N matches 2..9 |
| 10:42.04 | HjemmeRoyK | as in [246] matches 2,4 and 6 |
| 10:42.12 | Zeeek | multrix do you need to route different country codes differently? |
| 10:42.19 | forrestc{hm} | see http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns |
| 10:42.31 | Zeeek | or |
| 10:42.33 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 10:42.33 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 10:42.34 | Makenshi | if ${A} = 44 and ${B} = 1234, how could i split off 5556666 from 4412345556666 using the variables? |
| 10:42.59 | Makenshi | (${A} being local country code, ${B} being local area code) |
| 10:43.02 | fa | DO anobody knopw anything about PGSQL addons? |
| 10:43.18 | HjemmeRoyK | Makenshi: agi? |
| 10:43.20 | multrix | Zeeek: you're obsessed of this doc ! are you the authos ? ;) |
| 10:43.32 | Makenshi | i could write an agi for it i guess, didn't think of that :) |
| 10:43.39 | Zeeek | no but I wish you'd read it since it does explain what you've been asjing for three days :) |
| 10:43.47 | Makenshi | thanks |
| 10:45.01 | forrestc{hm} | multrix: You probably really need _0NXXXXXXXX (didn't count X'es) |
| 10:45.28 | Makenshi | probably worth writing an agi to do all the outbound toll calling |
| 10:46.30 | HjemmeRoyK | Makenshi: I've done that |
| 10:46.32 | multrix | exten => _000Z.,1,Dial(SIP/*8${EXTEN:3}@sipprovider) |
| 10:46.33 | multrix | exten => _00ZXXXXXXXX,1,Dial(SIP/*833${EXTEN:2}@sipprovider) |
| 10:46.35 | multrix | it works :) |
| 10:46.38 | HjemmeRoyK | 1000 line perl agi script :P |
| 10:46.53 | Makenshi | HjemmeRoyK, do you use a lookup table? |
| 10:47.03 | HjemmeRoyK | database based |
| 10:47.30 | HjemmeRoyK | with sip uid/cid mapping and acl checks |
| 10:47.42 | multrix | Zeeek: if this doc was good, I wouldn't need to ask... but it's shity I think, not clear at all |
| 10:48.09 | Zeeek | it isn't shitty doc, saying that irritates people - it's just that for some reason you don't understand it |
| 10:48.30 | Zeeek | the aurthors are on the channel by the way |
| 10:48.45 | Zeeek | in case you want to keep saying how shitty it is |
| 10:49.28 | multrix | Zeeek: Maybe it's a good doc to remind thinks to somebody who knows asterisk, but I keep saying that it's shitty for begginners, there are no good docs for beginners on asterisk docs ! |
| 10:49.46 | Zeeek | You'll have to write it! :) |
| 10:50.16 | forrestc{hm} | In all fairness to multrix (and others). I haven't seen *any* docs which adequately describe how to write a dialplan to do something. |
| 10:50.20 | multrix | Zeeek: actually, this is in my projects :) "beginner how-to in asterisk" |
| 10:50.35 | multrix | forrestc{hm}: thanks ! |
| 10:51.02 | Zeeek | I think there is no doc that takes you from 0 to 60 painlessly but there is enough doc out there to know how to write a dialplan |
| 10:51.37 | Zeeek | the fact is, it's awfully hard to explain a dialplan to someone until they know how they work and then it's too late :) |
| 10:51.37 | forrestc{hm} | What I mean is that I haven't found one which actually describes how to set up a set of pattern maches which actually work. |
| 10:52.00 | Makenshi | sounds like some example dialplans would be useful |
| 10:52.19 | Zeeek | there are a few hundred on the web |
| 10:52.22 | forrestc{hm} | I.E. "In the us, phone numbers are like X" |
| 10:52.23 | Makenshi | i forget, did someone write a tool to visualize them with graphviz? |
| 10:52.38 | forrestc{hm} | International ones look like X |
| 10:52.49 | Makenshi | british dial plan is ****ing stupid |
| 10:52.59 | forrestc{hm} | And so on. |
| 10:53.07 | forrestc{hm} | And then go through the steps of deriving each pattern. |
| 10:53.31 | Makenshi | for some reason they call 01234 the area code, instead of just "1234" |
| 10:53.58 | Zeeek | if you think about it, we don't even have ONE standard way to write international numbers |
| 10:54.03 | Makenshi | outlook doesnt work properly if you add the 0 in the uk locale |
| 10:54.14 | multrix | Zeeek: I think this type of doc is quietly possible to do, and won't take me lots of time when I will understand this ! :) |
| 10:54.28 | Zeeek | even in the US, when people submit their phone to abusiness database, it comes with several variations |
| 10:54.42 | multrix | Makenshi: there are exemple dialplan on the wiki, it helped me a lot ! |
| 10:54.47 | Makenshi | i think the most usual way is +(country code)(area code)(number) |
| 10:54.55 | forrestc{hm} | Yes, but the digits in a standard US dialplan are all the same. |
| 10:55.08 | forrestc{hm} | Ranges that is. |
| 10:55.09 | Zeeek | +33(1) +33(0)1 |
| 10:55.17 | Zeeek | +33145423425 |
| 10:55.29 | *** join/#asterisk jas_williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 10:55.33 | forrestc{hm} | I.E. To make a local call, you dial 7 digits, the first of which cannot be a zero or a 1 |
| 10:55.36 | Zeeek | +33 (0)1 45 42 39 85 |
| 10:55.37 | Makenshi | i live in a town which has one of the last 20 or so remaining exchanges that still has 5 digit local numbers |
| 10:55.54 | Zeeek | My first phone number was Aldrich 3462 |
| 10:55.57 | Makenshi | usually it's 6 or 7 digits |
| 10:56.00 | Zeeek | and I'm OLD! |
| 10:56.04 | forrestc{hm} | Alternatively, you can dial 10 digits, the first of which cannot be a zero or a 1 |
| 10:56.05 | Makenshi | but could be 8 to in london |
| 10:56.06 | Makenshi | argh |
| 10:56.11 | multrix | forrestc{hm}: I totally agree with you, and If you'd like to work with me on a tuto, it could be kewl |
| 10:56.42 | Makenshi | eg london - 020 xxxx xxxx... redditch - 01527 xxxxx, birmingham - 0121 xxx xxxx |
| 10:57.04 | Makenshi | i prefer the us dial plan |
| 10:57.06 | forrestc{hm} | For direct dial LD, it is always 1 followed by 10 digits, the first of which cannot be a zero or a one. |
| 10:57.11 | forrestc{hm} | And so on and so on. |
| 10:57.21 | Makenshi | 0 is national access code, 00 is internation access code |
| 10:57.31 | Makenshi | dial 0 for ld, 00 for int |
| 10:57.54 | Makenshi | though the way ofcom describes it, you can just direct dial long distance |
| 10:58.00 | Makenshi | as 0 is part of the area code |
| 10:58.04 | Makenshi | stupid >.< |
| 10:58.23 | Makenshi | yet when you call from abroad, you dont use the 0 |
| 10:59.35 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 11:00.05 | forrestc{hm} | Every dialplan example I've every seen just throws out the wildcards and then throws a sample dialplan, and assumes everyone knows the details of how a "non-asterisk" dialplan is built. |
| 11:00.54 | Makenshi | well, i think its fair to say that if you should have prior knowledge of dialplans in order to set up a pbx ;) |
| 11:01.01 | Makenshi | s/if// |
| 11:01.33 | forrestc{hm} | I've seen phone installers with absolutely no clue what a dialplan *IS* |
| 11:02.01 | Makenshi | sure, installing phones is a different kettle of fish |
| 11:02.12 | Makenshi | it's the MACs that require dp knowledge |
| 11:03.00 | multrix | Makenshi: 00 is not allway the IAC, that's a fucking think :p |
| 11:03.18 | Makenshi | multrix, yeah, in au it's 0011 |
| 11:03.49 | Makenshi | so i guess you have two options... limit your pbx to users in one country, or set up different contexts for them |
| 11:04.00 | Makenshi | oh, also you could define different access codes for each user |
| 11:04.16 | Makenshi | but you'd probably need an agi to do that |
| 11:04.38 | Makenshi | i just hope one day there's one big enum tree that everyone uses |
| 11:04.47 | Makenshi | then theres no access code bollocks :) |
| 11:04.59 | multrix | Makenshi: I must explore every possibilities, actually I just begin a study on asterisk for making my company become an Asterisk installer |
| 11:06.53 | forrestc{hm} | AHA!!! Found it... zap show channels. |
| 11:09.11 | *** part/#asterisk aggelos (~aggelos@egate.eleven.de) |
| 11:10.13 | forrestc{hm} | Well, I guess I'm off to bed now since it looks like my new asterisk box has all the ZAP Pri channels configured correctly... |
| 11:10.17 | forrestc{hm} | L8r all. |
| 11:10.21 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
| 11:10.39 | *** join/#asterisk florz (nobody@odnb-d9baa442.pool.mediaWays.net) |
| 11:10.44 | Makenshi | cya |
| 11:16.41 | *** join/#asterisk Tornad (~Tornad@81.255.65.249) |
| 11:16.49 | *** join/#asterisk Silensius (~si@21-154.241.81.adsl.skynet.be) |
| 11:25.34 | *** join/#asterisk _Omer (~rsdf@202.147.174.177) |
| 11:27.15 | _Omer | asterisk -rw "show queues" <------is that correct? |
| 11:30.16 | *** join/#asterisk zoa (~zoa@ip-212-239-162-34.dsl.scarlet.be) |
| 11:30.47 | HjemmeRoyK | ~lart zoa |
| 11:30.52 | HjemmeRoyK | hm |
| 11:30.54 | HjemmeRoyK | hmmmmmmm |
| 11:31.06 | HjemmeRoyK | is there a way to probe PRIs from asterisk? |
| 11:31.34 | _Omer | yes..by using T1 modules from Digium..... |
| 11:32.00 | HjemmeRoyK | _Omer: wot? |
| 11:32.11 | _Omer | I mean PRI cards..:D ... |
| 11:32.20 | HjemmeRoyK | _Omer: I have that |
| 11:32.22 | HjemmeRoyK | te410p |
| 11:32.45 | _Omer | ok then whats the problem? |
| 11:33.01 | HjemmeRoyK | it doesn't work :( |
| 11:33.41 | _Omer | hmmmmm |
| 11:34.16 | _Omer | I haven.t worked at digium modules. |
| 11:34.44 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 11:38.55 | *** join/#asterisk meppl (~mephisto@pD9E6871B.dip.t-dialin.net) |
| 11:39.17 | *** join/#asterisk Andrey_Kirov (~Andrey_Ki@16-51-customer.kirov.mtsnet.ru) |
| 11:39.49 | Andrey_Kirov | hello! |
| 11:40.32 | DrPete | I get this since I upgraded, do i need to create a dir ??? res_musiconhold.c:124 spawn_mp3: /usr/share/asterisk/mohmp3 is not a valid directory |
| 11:40.38 | Andrey_Kirov | anybody can help me with compile asteriks-h323? |
| 11:42.58 | Andrey_Kirov | <PROTECTED> |
| 11:42.58 | Andrey_Kirov | patched). Both libraries compile fine, but I get following errors on asterisk-oh323-0.6.5 |
| 11:43.11 | HjemmeRoyK | ~h323 |
| 11:43.12 | jbot | h323 is, like, evil! Don't ask about h323 here. Ask JerJer if you need to bug someone. He says it works, but others don't. |
| 11:45.21 | *** join/#asterisk konkyz0rk (~malevolen@rc01.peri.dk02605vstg018.arrownet.dk) |
| 11:46.49 | Andrey_Kirov | mm |
| 11:51.15 | fa | anybody use PGSQL addons? to make simple query like PGSQL(SELECT....) |
| 11:51.28 | fa | ~iax2 |
| 11:51.30 | fa | ~iax |
| 11:51.31 | jbot | extra, extra, read all about it, iax is 4569 and 5036, or pronounces "Eeks" |
| 11:51.39 | fa | ~sip |
| 11:51.40 | jbot | X11 PPP dialer interface written in gtk+. URL: http://www.geocities.com/SiliconValley/Campus/3104/sip/ Session Initiation Protocol (see RFC 3261) |
| 11:52.17 | Andrey_Kirov | h323 is a hard problem :) |
| 11:57.14 | *** join/#asterisk drray (~drray@dsl254-011-243.sea1.dsl.speakeasy.net) |
| 11:57.21 | *** join/#asterisk w0w0 (~w0w0@80.26.164.244) |
| 12:04.25 | _Omer | how do I stop my asterisk and then reload it? |
| 12:05.43 | fa | stop? and reload? stop and then start |
| 12:06.31 | _Omer | wow its a simple english :D |
| 12:06.34 | _Omer | thanks fa |
| 12:06.49 | fa | np. heh |
| 12:06.56 | xpasha | can anybody tell why app_queue records messages bad? |
| 12:07.08 | xpasha | it's just a header and nothing more |
| 12:07.19 | xpasha | it the conversation was too small |
| 12:07.34 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 12:07.34 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 12:07.35 | _Omer | fa: show agents <-----doesnt show anything? |
| 12:07.37 | bkw_ | yo yo yo |
| 12:07.40 | bkw_ | bkw in da house |
| 12:07.51 | bkw_ | just a quick stop by to say Hi... (can't sleep tonight) |
| 12:07.52 | zoa | yo yo yo |
| 12:07.52 | tzafrir | _Omer, asterisk -rx restart |
| 12:07.52 | zoa | bkw |
| 12:07.54 | zoa | heyaaa |
| 12:07.56 | zoa | brian |
| 12:07.58 | bkw_ | hey zoa quick question |
| 12:07.59 | zoa | check out astertest |
| 12:08.00 | zoa | :) |
| 12:08.03 | zoa | im coming to von bkw |
| 12:08.06 | zoa | i hope you are too |
| 12:08.10 | zoa | a mac ? |
| 12:08.11 | bkw_ | I'll be there |
| 12:08.17 | bkw_ | yep |
| 12:08.17 | bkw_ | a mac |
| 12:08.18 | bkw_ | now |
| 12:08.19 | _Omer | show agents <-----doesnt show anything? |
| 12:08.23 | zoa | im thinking of buying a G5 for testing with te410ps |
| 12:08.27 | bkw_ | _Omer then you don't have any mappings |
| 12:08.27 | zoa | should be better for the io |
| 12:08.30 | fa | _Omer what you mean? |
| 12:08.46 | fa | bkw_ do you know how to compile asterisk with adds_sql_pgsql ? |
| 12:08.47 | _Omer | I want to see ......who is online and who is offline...I mean to say AGENTS |
| 12:08.52 | fa | for simple PGSQL selects do datbase |
| 12:08.56 | bkw_ | well here is my two options... ibook+imac or powerbook |
| 12:08.58 | bkw_ | which would you do? |
| 12:09.09 | zoa | hmm |
| 12:09.13 | fa | _Omer iax2 show peers or sip show peers |
| 12:09.19 | zoa | so basically you have a budget |
| 12:09.24 | zoa | and you want to spend all of it ? |
| 12:09.26 | zoa | :p |
| 12:09.26 | bkw_ | 2200 bucks |
| 12:09.29 | bkw_ | :P |
| 12:09.30 | bkw_ | or so |
| 12:09.50 | zoa | so thats is company budget i suppose ? |
| 12:09.55 | zoa | otherwise id say buy an imac |
| 12:09.55 | bkw_ | no mine |
| 12:10.06 | bkw_ | well I can get both an imac and an ibook |
| 12:10.13 | zoa | why not just spend some money on an imac and see if you like it first |
| 12:10.15 | bkw_ | (company is gonna buy me ram upgrades fer it) |
| 12:10.16 | zoa | maybe you will just hate it |
| 12:10.21 | bkw_ | oh I know I like it |
| 12:10.23 | bkw_ | haha |
| 12:10.30 | zoa | hehe |
| 12:10.34 | fa | bkw_ all likes macs |
| 12:10.41 | bkw_ | just never had the cash |
| 12:12.29 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 12:14.14 | *** join/#asterisk pranav (dawda_pran@203.115.89.185) |
| 12:14.45 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 12:15.11 | *** join/#asterisk Voicelynx (~rda@rrcs-24-97-233-114.nys.biz.rr.com) |
| 12:15.26 | zoa | i dont know really |
| 12:15.31 | zoa | never played on one before |
| 12:16.18 | Andrey_Kirov | hey!!! Somebody compiled h332 addon? |
| 12:16.24 | Andrey_Kirov | 323 sorry |
| 12:16.33 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 12:16.44 | HjemmeRoyK | VoRS232 |
| 12:16.45 | pranav | what is the meaning of astguiclient |
| 12:16.46 | HjemmeRoyK | :P |
| 12:17.10 | pranav | is it neccesary in the asterisk |
| 12:17.50 | pranav | does it have anything to do with graphical interface |
| 12:17.50 | Zeeek | gui = graphioc (windows-like) interface |
| 12:17.54 | Zeeek | yes |
| 12:18.04 | Zeeek | it's supposed to be friendly like Microsoft :) |
| 12:18.29 | HjemmeRoyK | hm |
| 12:18.38 | HjemmeRoyK | I have this little problem with an asterisk box |
| 12:18.44 | Zeeek | I want a Mac |
| 12:18.49 | Andrey_Kirov | :) good description |
| 12:18.56 | pranav | ok thats grt, actually i am installing from http://astguiclientsourceforge.net/ |
| 12:19.13 | Zeeek | I'm thinking of buying one of theose $500 cigar box MAcs |
| 12:19.15 | HjemmeRoyK | it used to run nicely, but after an upgrade, it can nolonger see it's PRIs |
| 12:19.22 | HjemmeRoyK | Zeeek: hehe |
| 12:19.30 | pranav | but i am stuck, at a place it says insert phone values |
| 12:19.49 | HjemmeRoyK | pranav: 42 |
| 12:20.07 | HjemmeRoyK | ~42? |
| 12:20.09 | jbot | i guess 42 is the answer to life the universe and everything, see also http://en.wikipedia.org/wiki/the_answer_to_life,_the_universe,_and_everything |
| 12:20.18 | pranav | sorry what is 42 |
| 12:21.22 | *** join/#asterisk loick (~loick@APuteaux-151-1-50-158.w82-124.abo.wanadoo.fr) |
| 12:22.14 | tzafrir | pranav, that is: what is the question? |
| 12:22.23 | Zeeek | the answer is 42 |
| 12:22.36 | zoa | yeah |
| 12:22.39 | zoa | its 42 |
| 12:22.48 | zoa | fourtytwo=42 |
| 12:23.08 | pranav | ok the question is that in the site they have inserted values for the phone grandstream 102 |
| 12:23.15 | *** join/#asterisk Bonbon (~bonbon@83.146.53.34) |
| 12:23.22 | pranav | in the mysql database |
| 12:23.36 | pranav | how should i enter with my sipura phones |
| 12:25.55 | pranav | they have used insert into phones values('gs102','102','102','10.10.10.16','10.10.9.16','10.10.10.15','gs102','test', 'ADMIN','Y','Grandstream BT 102','Test Admin Phone','TEST','','0','0'); |
| 12:26.25 | pranav | for their grandstream 102 phone what should i enter here |
| 12:26.25 | DrPete | <PROTECTED> |
| 12:26.36 | pranav | i have sipura phone |
| 12:27.06 | pranav | with 2 channels |
| 12:27.51 | *** join/#asterisk LarsAC (~chatzilla@134.130.124.227) |
| 12:28.12 | pranav | so what should i enter |
| 12:29.56 | pranav | hello plz somebody tell me |
| 12:31.41 | *** join/#asterisk mAsH` (~mAsH@host46-29.pool8173.interbusiness.it) |
| 12:31.57 | adrianhensler | If trying to call out via FWD returns "Rejected connect attempt from 192.168.1.10, request 'xxxxxx@default' does not exist"; where does the 'default' come from in this example? |
| 12:36.15 | djin | Is the context |
| 12:37.20 | djin | There is no dial-out rule for xxxxxx in context default in your extensions.conf |
| 12:39.29 | DrPete | what dirs moved to /usr/share in the CVS? just mohmp3? |
| 12:42.22 | pranav | hello |
| 12:42.49 | *** join/#asterisk ZX81 (~ZX81@222-153-18-2.jetstream.xtra.co.nz) |
| 12:44.59 | ZX81 | ~ping |
| 12:45.17 | jbot | pong |
| 12:45.17 | Andrey_Kirov | pong :) |
| 12:45.25 | ZX81 | :) |
| 12:45.31 | datareactor | ~pong |
| 12:45.36 | jbot | wheeeeeeeeeeeeeeeeeeeeeeeeee! |
| 12:45.36 | ZX81 | pang |
| 12:45.39 | ZX81 | oh |
| 12:45.42 | ZX81 | :) |
| 12:45.58 | *** part/#asterisk loick (~loick@APuteaux-151-1-50-158.w82-124.abo.wanadoo.fr) |
| 12:46.36 | Andrey_Kirov | 2 ZX81: I now, you is an expert in h323 addon for asterix :) |
| 12:46.44 | ZX81 | heh |
| 12:46.45 | ZX81 | sif |
| 12:46.47 | ZX81 | :) |
| 12:46.51 | ZX81 | IAX man! |
| 12:46.52 | ZX81 | :) |
| 12:47.00 | ZX81 | ~zx81 |
| 12:47.01 | jbot | rumour has it, zx81 is the creater of the Daily Asterisk News (see ~adn) |
| 12:47.13 | ZX81 | :) |
| 12:47.28 | Andrey_Kirov | :) |
| 12:49.44 | fa | but who is exper in PGSQL addon for astris, hu? |
| 12:50.10 | Andrey_Kirov | Why here nobody knows about h323? :( |
| 12:50.38 | zoa | i know about h323 |
| 12:50.41 | zoa | stay away for it |
| 12:50.51 | zoa | fa i know about pgsql and asterisk |
| 12:50.53 | zoa | we use it |
| 12:50.59 | zoa | but not any standard addons |
| 12:51.00 | adrianhensler | djin - tks; looking at that now |
| 12:53.18 | fa | zoa Can I ask on priv? |
| 12:53.22 | *** join/#asterisk mgeorge (~george@216.157.203.105) |
| 12:53.59 | zoa | sure |
| 12:54.05 | zoa | dont think i can help you though |
| 13:04.16 | *** join/#asterisk HjemmeRoyK (~roy@83.80-203-29.nextgentel.com) |
| 13:05.55 | *** part/#asterisk JunK-C (~junky@modemcable144.95-37-24.mc.videotron.ca) |
| 13:12.07 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:12.39 | datareactor | how can i see logs granstream phone i am unable to reg to go2call |
| 13:13.28 | adrianhensler | I am still getting "xxxx@default' does not exist'; I have tried following the directions on this link but I must be not understanding someething http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD#comments |
| 13:14.17 | multrix | zoa: is h323 so shitty ? :)= |
| 13:14.36 | *** join/#asterisk meppl (~mephisto@pD9E6871B.dip.t-dialin.net) |
| 13:17.27 | Zeeek | muntrix can you perhaps expand your vocabulary a little? |
| 13:22.38 | tzafrir | anybody encountered problems with spaces in the history of the asterisk command: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=281690 |
| 13:22.42 | tzafrir | I know I do |
| 13:22.47 | *** join/#asterisk pbxjunkie (~stormtroo@videocomputer.gr) |
| 13:23.27 | pbxjunkie | guys, SIP is killing me. How do I tell my dialplan, if busy, do that, otherwise, if no response in ..some seconds do something else |
| 13:23.32 | adrianhensler | I think I get it - FWD needs to be assigned to an extension, is that correct? |
| 13:24.20 | tzafrir | pbxjunkie, basically read 'show application dial' |
| 13:25.01 | pbxjunkie | whoa:) nice :] |
| 13:25.04 | pbxjunkie | thanks :D |
| 13:25.32 | Silensius | zoa ? |
| 13:25.57 | HjemmeRoyK | <PROTECTED> |
| 13:27.46 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 13:28.45 | pbxjunkie | how do I need to configure sip.conf and my grandstream bt102's so that they return busy signals when they are busy |
| 13:29.04 | pbxjunkie | anything to do with incominglimit? |
| 13:31.19 | pashah | what rtpchecksums means? found in rtp.conf anyone? |
| 13:31.27 | _Brian | anyone know if the new prompts were uploaded to CVS (the ones mentioned on 1/16/05 in the mailing list?) |
| 13:40.01 | *** join/#asterisk _Omer (~rsdf@202.147.174.177) |
| 13:41.47 | *** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 13:42.17 | _Omer | how to decrease the rap up time of agents? |
| 13:42.24 | *** join/#asterisk MatsK (~mk@8.80-202-60.nextgentel.com) |
| 13:44.44 | fa | fucking pgsql adds ;] |
| 13:45.47 | _Omer | hey fa |
| 13:46.06 | _Omer | How to check the RAPUP time of the agents/extensions? |
| 13:46.09 | _Omer | :) |
| 13:46.19 | fa | i don't know what is RAPUP time ;[[[ |
| 13:46.49 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 13:47.16 | _Omer | aaah!! so I found someone like me ..;) |
| 13:47.40 | _Omer | Time or delay in sending the call to agent again.... |
| 13:47.56 | _Omer | wrapup time |
| 13:48.18 | *** join/#asterisk Cherebrum (~jgarland@216.32.77.10) |
| 13:48.31 | fa | a this.. it's cool. but i find on bugs. that you must have some patch to use that. |
| 13:48.35 | Cherebrum | Morning |
| 13:48.38 | fa | _Omer do you use PGSQL? |
| 13:48.40 | fa | Cherebrum hi |
| 13:48.47 | Cherebrum | Anyone here using Level3's SIP service? |
| 13:49.46 | *** join/#asterisk bowman (~bowman@snert3.tal.de) |
| 13:50.33 | Cherebrum | Guess not. |
| 13:50.34 | _Omer | fa: no |
| 13:50.47 | Cherebrum | They just started offering it |
| 13:50.50 | bowman | what do I need to convert to GSM with sox? |
| 13:51.54 | vaewynAFK | sox blah.wav -r 8000 -c 1 blah.gsm |
| 13:52.06 | *** join/#asterisk zoa (~zoa@ip-212-239-162-34.dsl.scarlet.be) |
| 13:52.18 | vaewyn | PROTO! |
| 13:52.36 | vaewyn | sorry... couldn't resist |
| 13:53.36 | zoa | hey you mr vaewyn |
| 13:53.53 | vaewyn | :} |
| 13:54.34 | *** join/#asterisk mutilator (~animenodv@65.111.201.79) |
| 13:57.14 | Zeeek | what is AFK ? |
| 13:57.21 | zoa | away from keyboard |
| 13:57.25 | zoa | using the guitar |
| 13:57.28 | netsurfer | away from kentucky |
| 13:57.29 | Zeeek | haha |
| 13:57.36 | netsurfer | :P |
| 13:57.42 | *** join/#asterisk imperfect- (~tbw@216.32.77.10) |
| 13:57.46 | netsurfer | hey Zeeek |
| 13:57.49 | Zeeek | All For Kraftwerk |
| 13:58.06 | netsurfer | Zeeek - get ur isp fixed ? |
| 13:58.14 | Zeeek | no getting new isp |
| 13:58.21 | netsurfer | good choice :) |
| 13:58.24 | Zeeek | actually waking up old one to new connex |
| 13:58.54 | Zeeek | they're all the same, but we had a backup 512/128 for years, upgrading now to 2048/256 - good enough |
| 13:59.10 | DrPete | any idea what this is all about?? ast_unregister_indication_country: Removed default indication country 'uk' |
| 13:59.12 | Zeeek | these guys have given us a real static ip from the gitgo |
| 13:59.14 | bowman | vaewyn: I don't have a GSM lib installed, hence the question :) libgsm or what? |
| 13:59.38 | netsurfer | DrPete - the default indication in indications.conf |
| 13:59.43 | Zeeek | asterisk hates British telephone sys |
| 13:59.57 | DrPete | lol |
| 14:00.08 | DrPete | netsurfer: do i need to fix something? |
| 14:00.10 | *** join/#asterisk tty74 (~tiziano@151.11.170.2) |
| 14:00.11 | Zeeek | so are the faxes running hot and heavy DrPete? |
| 14:00.33 | *** join/#asterisk liversmudge (~liversmud@217-14-176-201.as25582.net) |
| 14:00.49 | netsurfer | DrPete - I see that error on my * box, only it said "us" til I changed us to uk (im in uk) |
| 14:01.07 | netsurfer | now it says Feb 3 13:57:11 NOTICE[4542]: indications.c:401 ast_unregister_indication_country: Removed default indication country 'uk' |
| 14:01.07 | liversmudge | hello my little schneebles |
| 14:01.24 | netsurfer | hehe not sure why, it dosent seem to cause any problems though |
| 14:01.26 | DrPete | netsurfer: I get the same |
| 14:01.35 | liversmudge | whos up for 100 brownie points |
| 14:01.44 | DrPete | netsurfer: heh ok fair enough, I will leave it then |
| 14:01.47 | mAsH` | how can i test a crossover T1 cable? |
| 14:01.49 | DrPete | netsurfer: thanks |
| 14:01.55 | netsurfer | liversmudge - i'll just make do with 100 brownies ;) |
| 14:02.01 | liversmudge | whats this meen |
| 14:02.02 | liversmudge | Got SIP response 488 "Not Acceptable Here" |
| 14:02.43 | tty74 | Hi i need help , i'm trying use diax to asterisk, i'm able to register an in debug mode all work, but i've this message " chan_iax2.c:5413 socket_read: Rejected connect attempt from ........" astersik server is behind a cisco 1750 with static nat on iax2 . Where is the problem? |
| 14:03.15 | netsurfer | DrPete - it may be something else in the indications.conf file - im gonna take a look, been too busy to fix errors that dont actually break anything |
| 14:03.36 | liversmudge | come on who wants the bornie points then? |
| 14:03.38 | liversmudge | Got SIP response 488 "Not Acceptable Here" |
| 14:03.44 | liversmudge | whats this then? |
| 14:04.26 | DrPete | netsurfer: yeah I know the feeling, I get I couple of errors like this, can i leave these too?? res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified# |
| 14:05.22 | netsurfer | DrPete - u can, its SQL type stuff, wont do any harm |
| 14:05.41 | netsurfer | unless ur using SQL |
| 14:05.55 | liversmudge | lol |
| 14:06.16 | DrPete | netsurfer: nah, i am not using it, yet, lol. Is the a way I can stop it, I dont like warnings heh |
| 14:06.28 | liversmudge | yes mr bloggs having your legs cut off wont effect you at all .. unless you want to walk that is |
| 14:06.41 | _Omer | How do I increase of decrease WRAP UP TIME? |
| 14:06.46 | DrPete | lol |
| 14:06.47 | _Omer | How do I increase or decrease WRAP UP TIME? |
| 14:07.03 | netsurfer | DrPete - could try commenting out stuff in res_odbc.conf |
| 14:07.39 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 14:07.40 | tty74 | The complete error is "chan_iax2.c:5413 socket_read: Rejected connect attempt from ....my remote ip.... |
| 14:07.55 | DrPete | netsurfer: yeah oki. is it because of a missing package? I have deb so its easy to install |
| 14:09.14 | netsurfer | DrPete - http://www.voip-info.org/tiki-index.php?page=Asterisk res_config |
| 14:09.27 | DrPete | netsurfer: oooh thanks |
| 14:10.11 | liversmudge | aint res_config out of date now |
| 14:10.23 | liversmudge | res_odbc.conf me things is inn vogue this year |
| 14:10.45 | liversmudge | you wear it your your neck .. like an olf fox fur |
| 14:10.49 | Zeeek | Anyone have a Siemens cordless phone? Like a Gigaset C200? |
| 14:11.01 | liversmudge | anyone know what this is Got SIP response 488 "Not Acceptable Here" |
| 14:11.10 | liversmudge | only 50 brownie point now |
| 14:11.12 | blitzrage | morning Zeeek |
| 14:11.13 | Zeeek | pr0n |
| 14:11.18 | Zeeek | hi blitz |
| 14:11.22 | blitzrage | how goes today? |
| 14:11.36 | DrPete | netsurfer: I am just learning asterisk myself, is there any advice you can give me, stuff i need to do speific to the UK># |
| 14:11.38 | Zeeek | pr0n audio : "Not acceptable here" |
| 14:12.07 | liversmudge | was that for me ?? Got SIP response 488 "Not Acceptable Here" |
| 14:12.16 | netsurfer | DrPete - yup... http://www.voip-info.org/ use the wiki :D |
| 14:12.24 | Zeeek | I have my new sooper siemens c200 that does callerid and name, and I see nothing from asterisk |
| 14:12.45 | DrPete | netsurfer: heh, oki |
| 14:13.59 | *** join/#asterisk nazgool (~nazgoool@port-83-236-180-106.static.qsc.de) |
| 14:14.02 | nazgool | hi |
| 14:14.03 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 14:14.16 | nazgool | obviously i somehow got disconnected |
| 14:14.29 | *** join/#asterisk mtqh (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 14:14.59 | nazgool | so i didn't see any answers (i don't even know if my question got through). the question is/was: |
| 14:15.04 | nazgool | is there any reason why i would want to give different names to one same user for different channels, i.e joe_capi, joe_sip, joe_iax instead of just putting "joe" everywhere and just making the difference in the channel type in the Dial command? |
| 14:15.06 | liversmudge | and while I at it |
| 14:15.15 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
| 14:15.16 | W1thdraw | anyone know of a simple solution (with out asterisk) and with only one pstn line to forward calls to my cell phone? |
| 14:15.18 | liversmudge | I have terrible trouble with realtime and loosing phones |
| 14:15.26 | liversmudge | untill the reseed |
| 14:15.33 | *** join/#asterisk fafnir (~hello@tdds-gw.Moscow.gldn.net) |
| 14:15.33 | marlowe | W1thdraw: It's called call forwarding |
| 14:15.53 | W1thdraw | yeah but i want to control when it forwards on a computer |
| 14:16.06 | liversmudge | asterisk if good at that |
| 14:16.07 | marlowe | Some telcos offer that |
| 14:16.10 | marlowe | Some dont |
| 14:16.12 | *** join/#asterisk thieumS (~darkmind@nanterre-7-82-229-210-142.fbx.proxad.net) |
| 14:16.34 | thieumS | what's the best calling card addon actually ? |
| 14:16.40 | liversmudge | who uses realtime here from sipfriends |
| 14:16.46 | W1thdraw | marlowe, i dont want it to show up on my phone bill |
| 14:16.54 | marlowe | W1thdraw: Umm |
| 14:17.07 | marlowe | You can't forward phone calls - to a long distance number especially w/o getting charged. |
| 14:17.09 | W1thdraw | i just want to keep it simple |
| 14:17.13 | marlowe | You're not exactly making sense. |
| 14:17.18 | marlowe | Simple is not what you want |
| 14:17.20 | Nivex | W1thdraw: Try a Sipura SPA-3000 |
| 14:17.27 | marlowe | You're saying you have 1 PSTN, you want calls forwarded to your cell phone. |
| 14:17.35 | W1thdraw | yeah |
| 14:17.38 | W1thdraw | is that possible? |
| 14:17.41 | liversmudge | : keep it simple .. what like a piece of string and 2 paper cups? |
| 14:17.41 | marlowe | Why the need to control it from a computer and why the need to not show up on a phone bill? |
| 14:17.50 | mAsH` | how can i test a crossover T1 cable? |
| 14:17.54 | marlowe | You dont need asterisk, you dont need anything' |
| 14:17.58 | fa | Who use PGSQL adds? |
| 14:18.00 | liversmudge | : and you dont get these calls showing up on your phone bill |
| 14:18.04 | marlowe | Call your telco and tell them you want to forward calls. |
| 14:18.23 | marlowe | I think you're the one being difficult. |
| 14:18.24 | liversmudge | : but even then the calls WILL show up on your bill |
| 14:18.31 | Nivex | oh yeah, with only 1 PSTN line the only calls you'll be able to forward to your cell are VoIP calls. |
| 14:18.32 | W1thdraw | yeah but then its gonna show on my bill |
| 14:18.42 | marlowe | So who cares? |
| 14:18.51 | marlowe | Are you trying to get free phone service or what? |
| 14:18.56 | marlowe | What are you trying to do? |
| 14:19.08 | W1thdraw | no i just need to make it seem like im at home when im not |
| 14:19.14 | liversmudge | some come on .... realtime use .... who uses it? |
| 14:19.16 | marlowe | SO there you go |
| 14:19.20 | marlowe | CALL your telco |
| 14:19.24 | marlowe | Tell them to forward calls. |
| 14:19.26 | marlowe | end of story. |
| 14:19.31 | marlowe | Makes it seem like your home, when your not. |
| 14:19.34 | W1thdraw | no but it cant show on the phone bill |
| 14:19.37 | marlowe | why not? |
| 14:19.37 | Makenshi | W1thdraw, do what marlowe says |
| 14:19.54 | marlowe | Answer the questions of why it can't show up on the phone bill? |
| 14:19.54 | Nivex | W1thdraw: you are screwed then. |
| 14:19.58 | liversmudge | its gonna show on your phone bill ... as a redirected call .. and thats the way the cookie crumbles |
| 14:20.00 | W1thdraw | cuz it wont look like im home it i says my calls are forwarded |
| 14:20.07 | marlowe | omg |
| 14:20.14 | Makenshi | so youre trying to defraud someone? |
| 14:20.15 | liversmudge | lol |
| 14:20.19 | Makenshi | noone is giong to help you with that |
| 14:20.25 | W1thdraw | Makenshi, yeah my mom |
| 14:20.31 | Makenshi | then stay at home! |
| 14:20.33 | marlowe | On your phone bill |
| 14:20.37 | W1thdraw | wheres the fun in that |
| 14:20.41 | liversmudge | I would help him .. If I though I was getting summit outta it |
| 14:20.43 | liversmudge | but Im not |
| 14:20.43 | marlowe | It'll look like someone from your house |
| 14:20.45 | marlowe | called your cell phone |
| 14:20.47 | marlowe | So.. |
| 14:20.52 | marlowe | You juts tell your mom |
| 14:20.57 | Nivex | /ignore W1thdraw |
| 14:20.57 | marlowe | you like to call yourself a lot |
| 14:21.03 | marlowe | She might commit you to a mental institution |
| 14:21.07 | nazgool | hehe |
| 14:21.17 | marlowe | But, you do look in fact like you were home |
| 14:21.24 | liversmudge | after reading this channel they are already on the way |
| 14:21.25 | *** join/#asterisk stevekstevek (~stevekste@slim-eth0.horizonlive.net) |
| 14:21.26 | marlowe | Looks like you were home and you call yourself a lot, end of story. |
| 14:21.40 | W1thdraw | yeah but if she calls the telco and finds that im forwarding the calls ill get fucked |
| 14:21.46 | nazgool | marlowe: for some people, home is where the walls are padded .... ;-) |
| 14:21.50 | liversmudge | realtime .. realtime ... realtime |
| 14:21.55 | marlowe | So your screwed. period. |
| 14:22.13 | liversmudge | seeding ... seeding .. seeding |
| 14:22.23 | Makenshi | W1thdraw, if you're still a minor, you'd better do what your parents say |
| 14:22.24 | liversmudge | come on someone apart from me must use realtime |
| 14:22.27 | marlowe | Espo |
| 14:22.27 | marlowe | Espeiclally since you dont want to use asterisk |
| 14:22.38 | marlowe | Cant type this early |
| 14:22.50 | W1thdraw | Makenshi, nope not a minor anymore take pitty on me!!!!!! |
| 14:22.53 | liversmudge | mid afternoon in the UK |
| 14:23.04 | liversmudge | my fingers a re limbered up and ready to go |
| 14:23.05 | marlowe | W1thdraw: How old are you? |
| 14:23.12 | liversmudge | my mistakes are due to the fact I cant spell |
| 14:23.14 | W1thdraw | 46 |
| 14:23.20 | nazgool | make that 16 |
| 14:23.21 | W1thdraw | jk 18 |
| 14:23.23 | Makenshi | W1thdraw, then leave home and be done with it |
| 14:23.24 | marlowe | Your 46, and your trying |
| 14:23.25 | marlowe | ok |
| 14:23.27 | marlowe | So your 1 |
| 14:23.28 | marlowe | 18 |
| 14:23.32 | marlowe | Your an adult |
| 14:23.36 | marlowe | Stop being a kid |
| 14:23.40 | marlowe | Dont fuck with your mom |
| 14:23.47 | marlowe | moms find out and will hunt you down |
| 14:23.48 | W1thdraw | no must keep leeching |
| 14:24.02 | marlowe | So what, she calls... |
| 14:24.05 | marlowe | Dont pick up |
| 14:24.09 | marlowe | "why didnt you pick up?" |
| 14:24.11 | marlowe | i was sleeping. |
| 14:24.12 | Makenshi | is this where society is headed |
| 14:24.23 | liversmudge | lol |
| 14:24.24 | Nivex | Makenshi: sadly, so it seems |
| 14:24.26 | marlowe | Thats what I say when my girlfriend calls. |
| 14:24.28 | marlowe | :) |
| 14:24.31 | W1thdraw | marlowe, yeah i tried that like 50 times |
| 14:24.44 | Nivex | marlowe: apparently this guy doesn't learn. |
| 14:24.48 | W1thdraw | i think she knows what im doing |
| 14:24.53 | marlowe | Nivex: I know, it's a bit entertaining though. |
| 14:24.58 | marlowe | W1thdraw: What ARE you doing? |
| 14:25.33 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 14:25.33 | *** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released |
| 14:25.42 | Zeeek | your activities with tailpipes have been widely spread around :) |
| 14:25.58 | liversmudge | jesus .. your 46 and your not allowed out on your own |
| 14:26.00 | Zeeek | and your blackened unit shows it |
| 14:26.04 | Nivex | W1thdraw: just out of morbid curiosity, did you even graduate high school? |
| 14:26.13 | W1thdraw | in a few months |
| 14:26.36 | W1thdraw | so close..... |
| 14:26.59 | liversmudge | ok ... W1thdrawn you manage to fool your mom .. with the phone calls .. how you gonna fool the police with your leg tag ??? |
| 14:27.14 | liversmudge | you cant go more than 25 yards from your home |
| 14:27.47 | Nivex | liversmudge: meh... eventually he's gonna piss of his mom and she'll just give him the boot. |
| 14:27.47 | W1thdraw | no im only on limited probation |
| 14:28.01 | nazgool | W1thdraw, try "load => app_foolmom" in your modules.conf |
| 14:28.06 | liversmudge | lol |
| 14:28.07 | vaewyn | "Friends come and go... but you have to stay.. that's why they call it house arrest" |
| 14:28.35 | W1thdraw | im not under house arrest |
| 14:28.43 | liversmudge | come on gurus awake and spill the beans whats Got SIP response 488 "Not Acceptable Here" back from meen |
| 14:28.49 | liversmudge | you should be |
| 14:28.54 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 14:29.07 | marlowe | wait what id miss |
| 14:29.11 | marlowe | your on probation? |
| 14:29.21 | Zeeek | liver would this be of any help? |
| 14:29.22 | W1thdraw | yeah i was only a minor |
| 14:29.22 | Zeeek | http://lists.digium.com/pipermail/asterisk-users/2004-June/049035.html |
| 14:29.30 | liversmudge | : the web weaves itself thicker |
| 14:29.33 | nazgool | actually "home" is a federal prison, "mom" are the guards and he somehow managed to hijack a pc and a phone line and wants to set up a way for the mafia to organize by phone without the guards taking notice |
| 14:29.49 | *** join/#asterisk CletusColeman (~CletusCol@c-24-0-179-254.client.comcast.net) |
| 14:30.11 | Zeeek | livers ? did you see? |
| 14:30.16 | liversmudge | I read that zeeky boy .. but .... the phone dont have annon call block |
| 14:30.16 | vaewyn | PC? nah.. .he's 1337 and just whistles into the phone |
| 14:30.27 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:30.29 | thieumS | what's the better calling card application addon for asterisk please? |
| 14:30.29 | liversmudge | so its summit else |
| 14:30.53 | liversmudge | these are new phones I have imported from TW ... as testers |
| 14:31.12 | Zeeek | not acceptable media? |
| 14:31.37 | vaewyn | bwahaha... Hitachi IP-5000 is on it's way |
| 14:31.50 | liversmudge | okeeeee |
| 14:31.52 | liversmudge | mmm |
| 14:32.02 | liversmudge | that may be .. but I can make calls out of the phone to * |
| 14:32.07 | liversmudge | no prooooblem |
| 14:32.14 | Zeeek | ~seen wasim |
| 14:32.15 | jbot | wasim is currently on #asterisk |
| 14:32.16 | liversmudge | its only inbound calls to the phone |
| 14:32.25 | Zeeek | wasim wasim wasim |
| 14:32.55 | Zeeek | so wasim... the guy is coming next week to um, "get his phone" |
| 14:33.03 | Zeeek | is there any chance... ? |
| 14:33.07 | liversmudge | too much emoting for your own good |
| 14:33.23 | Zeeek | livers that's what the message not acceptable means! |
| 14:33.54 | Zeeek | you're too involved sexually with inanimate objects |
| 14:34.00 | Zeeek | Not Acceptable! |
| 14:34.03 | Zeeek | oh oh |
| 14:34.12 | liversmudge | that may be so .. but still I cant get an incomming phone call |
| 14:34.13 | Zeeek | drums stop. not good. |
| 14:34.21 | Zeeek | what phone? |
| 14:34.27 | *** join/#asterisk cbachman (~cbachman@129.105.7.250) |
| 14:34.28 | Zeeek | the cheazpies? |
| 14:34.29 | liversmudge | what media ... erm like the cd wont fit in the phone ..... nahhhhhh!!! |
| 14:34.51 | Zeeek | diskette you fool, diskette! |
| 14:35.06 | Zeeek | -- Got SIP response 488 "Not Acceptable Here" |
| 14:35.08 | liversmudge | yeah I have a problem cramming it down a cat 5 cable |
| 14:35.12 | *** join/#asterisk NormAst (HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca) |
| 14:35.15 | liversmudge | thats it |
| 14:35.18 | NormAst | Hi all. |
| 14:35.19 | liversmudge | you see you do knoe |
| 14:35.26 | toddf | unlike others that I can find, at least voicepulse allows me to be notified when they have service in my area ;-) |
| 14:35.27 | liversmudge | and you know as well |
| 14:35.31 | Zeeek | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg70432.html |
| 14:35.54 | liversmudge | I know this |
| 14:36.01 | NormAst | What provides the '/lib/modules/2.6.7/build' directory? Anyone? |
| 14:36.07 | liversmudge | and do indead have such desirable licencual device |
| 14:36.15 | Zeeek | using 729? |
| 14:36.16 | Nivex | NormAst: on which distro? |
| 14:36.16 | *** join/#asterisk channan (~channan9@66.180.121.185) |
| 14:36.26 | liversmudge | and as I say I can make such desired calls from phone to * on g729 |
| 14:36.48 | NormAst | debain. |
| 14:36.57 | Zeeek | but as they say some 729 don't work apparently |
| 14:36.58 | liversmudge | so I presume that da phone know da codec |
| 14:37.03 | *** join/#asterisk _-Jon-_ (~jon@CPE000d8861e8f7-CM00080d290642.cpe.net.cable.rogers.com) |
| 14:37.13 | liversmudge | yeeees but it do going out bound |
| 14:37.14 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 14:37.22 | liversmudge | phony to * works |
| 14:37.24 | _-Jon-_ | Hey I'm wondering if someone can offer an opinion on something |
| 14:37.25 | Zeeek | whatevah |
| 14:37.33 | liversmudge | like yeah dude |
| 14:37.37 | liversmudge | W |
| 14:37.47 | NormAst | Nivex: I guess it's them same on most distro however.. |
| 14:37.50 | pbxjunkie | is there ANY way to set Grandstream BT102 to DND mode?:) |
| 14:38.10 | Nivex | NormAst: well in Fedora it's kernel-devel. In debian it's probably something like kernheaders or something. |
| 14:38.14 | Zeeek | absolutely, press the down arrow about 5 times |
| 14:38.15 | liversmudge | oh so childish ... Im off to do knock and run on W1thdraw's moms house |
| 14:38.31 | Zeeek | on the BT102 |
| 14:39.00 | pbxjunkie | Lool Zeeek :) |
| 14:39.04 | liversmudge | BT102 .... ach |
| 14:39.04 | _-Jon-_ | Here's my problem: Call comes into my * box and if I don't answer my phone then it dials out to reach me on my cell, however the audio always breaks up and it's very difficult to have a conversation. Yet 2 seperate calls work perfectly |
| 14:39.04 | Zeeek | works! |
| 14:39.14 | Zeeek | that's what I do |
| 14:39.16 | liversmudge | send thoes barbie phones back save touself some headache |
| 14:39.18 | pbxjunkie | I want asterisk to know it so that it will handle the call accordingly |
| 14:39.19 | Zeeek | but there is a DND, no? |
| 14:39.29 | pbxjunkie | you mean button? nope. |
| 14:39.30 | Zeeek | I have an extension for that |
| 14:39.37 | Zeeek | ah, maybe not |
| 14:40.07 | Zeeek | I wrote a little ext that when I dial it I neter a time and calls are diverted to vmail until that time |
| 14:40.08 | mAsH` | anyone known how can i test a crossover T1 cable? |
| 14:40.23 | Zeeek | that way when a girl comes over I dial DID 3 minutes :) |
| 14:40.25 | _-Jon-_ | Can anyone off some input as to why that might be? |
| 14:40.39 | pbxjunkie | lol |
| 14:40.47 | Zeeek | but seriously - that works |
| 14:40.47 | Delvar | pbxjunkie: you nead to turn on phone features then dial *70 |
| 14:41.02 | Zeeek | I'm sure there's something that works |
| 14:41.12 | Zeeek | the big challenge is remembering to turn off DND |
| 14:41.15 | pbxjunkie | Delvar: hmm.. oh phone features. Yea. |
| 14:41.21 | Zeeek | which is why I use a timed version |
| 14:41.41 | *** join/#asterisk simon_ca (sedgett@S01060004e23873e1.vc.shawcable.net) |
| 14:42.58 | liversmudge | so who uses realtime???? |
| 14:43.54 | Cherebrum | yey! |
| 14:44.33 | thieumS | what's realtime |
| 14:44.50 | liversmudge | realtime config files on * |
| 14:44.59 | Delvar | database addon for asterisk to keep account sin DB, saves on realoads and helps clustering |
| 14:45.02 | *** join/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 14:45.09 | pbxjunkie | Delvar: I've enabled call features, did a *70 but it doesn't work :/, u sure it's supported? |
| 14:45.25 | Delvar | what firmware version you on? |
| 14:45.44 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 14:45.51 | PakiPenguin | Evening everyone |
| 14:45.52 | vaewyn | Also adds a single point of failure and communication overhead |
| 14:45.56 | vaewyn | :} |
| 14:46.10 | PakiPenguin | can anyone suggest some windows based iax client that has g729 support? |
| 14:46.23 | liversmudge | Xten |
| 14:46.38 | PakiPenguin | IAX :p |
| 14:46.41 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 14:46.42 | Delvar | vaewyn:thats what mysql clusters are for :) |
| 14:47.06 | liversmudge | you can have multiple odbc servers registered |
| 14:47.15 | shido6 | if it has g729 support you're going to have to pay for it |
| 14:47.38 | liversmudge | yup even xten needs to be bought to get g729 |
| 14:47.42 | pbxjunkie | Delvar: Program-- 1.0.5.18 Bootloader-- 1.0.0.21 HTML-- 1.0.0.42 VOC-- 1.0.0.7 |
| 14:47.44 | liversmudge | anyone use realtime then? |
| 14:48.02 | djin | tried Realtime, didn't work for me. |
| 14:48.07 | liversmudge | right |
| 14:48.39 | PakiPenguin | shido6: suggest now , i didnt say i cannot pay for it |
| 14:49.06 | Cherebrum | Is anyone having problems with Broadvoice right now? |
| 14:49.13 | Cherebrum | My sip registration keeps timing out |
| 14:49.16 | vaewyn | realtime is kindof a ironic feature... it works great for large amounts of users on a small setup... :} |
| 14:49.20 | Delvar | pbxjunkie: hmm u sure your phone features are on?, just dial *70 for DND and *71 to turn it off |
| 14:49.24 | Cherebrum | It has worked fine for weeks |
| 14:49.58 | pbxjunkie | Delvar: hmm.. ok thanks :) |
| 14:50.24 | Delvar | vaewyn: we jsut set it up on a couple boxes with a lot of accounts, seems to be working solid. |
| 14:50.31 | liversmudge | vaewyn: why ironic? |
| 14:50.59 | liversmudge | do you get sip timeouts then? |
| 14:51.55 | Delvar | the only problem we keep getting is when it losing the DB conenction and refuses a conenction, it reconects tho |
| 14:51.57 | vaewyn | liversmudge: not many people have a "large" user set on a "small" system |
| 14:52.10 | *** join/#asterisk mhnoyes (~mhnoyes@user-38lc19a.dialup.mindspring.com) |
| 14:52.44 | liversmudge | I have it running fine .. authenticating sip clients .. and IF within say 5 mins no outbound call is made on client .. we loose the client can cant make calls to them |
| 14:52.52 | *** join/#asterisk bprice20 (~brandon@Dynamic-216.120.224.151.hrnoc.net) |
| 14:53.01 | *** join/#asterisk zno (~chatzilla@160.79.174.99) |
| 14:53.12 | liversmudge | each time the client makes a call I get 5 mins of ability to call the client |
| 14:53.25 | liversmudge | I dont get this with a flat file sip config |
| 14:53.33 | Delvar | what is your registration timeout? |
| 14:53.36 | Delvar | ok |
| 14:53.40 | Delvar | odd |
| 14:53.40 | liversmudge | on the phone |
| 14:53.50 | Delvar | does it lose conenction to teh database? |
| 14:53.58 | liversmudge | I have used different phones each seems to have sip reg set to 60 mins |
| 14:54.34 | liversmudge | no databas is there and you can query it to get the regseconds etc |
| 14:54.46 | Delvar | very odd |
| 14:54.52 | liversmudge | IF I change the reg timeout to 5 mins they stay there |
| 14:55.21 | liversmudge | but I dont wanna have to reset every phone used .. especially IF I dont have access to them |
| 14:55.38 | pbxjunkie | guys in the 'sip show peer' screen, what 's the last column, that says Status: unmonitored? Is there any way for asterisk to monitor the status of sip peers? |
| 14:55.52 | Zeeek | qualify=300 |
| 14:55.59 | bprice20 | I am trying to troubleshoot a problem with sound quality, its not echo, more of a cracking sound, and it doesn't occur when a call is placed from one iaxy to another via the asterisk box, only out of network calls, or when calling the asterisk box for checking voicemail |
| 14:56.04 | liversmudge | you type quicker than me |
| 14:56.15 | *** join/#asterisk maruz (~maumar@adsl-123-3.38-151.net24.it) |
| 14:56.51 | Zeeek | has anyone ever looked to see if all those comments in the sample configs are worth reading? |
| 14:57.00 | liversmudge | lol |
| 14:57.06 | liversmudge | rtfm ... never |
| 14:57.09 | Zeeek | or are they just there to pad out the file size? |
| 14:57.20 | Zeeek | yea, that must be it |
| 14:57.54 | liversmudge | delvar: what phones do you use with realtime? |
| 14:57.56 | *** join/#asterisk zoa (~zoa@ip-212-239-162-34.dsl.scarlet.be) |
| 14:58.11 | liversmudge | delvar : do your phones gothrough a natted router say for remote workers? |
| 15:00.09 | ManxPower | liversmudge, You are not on the mailing list or did not read my e-mail on the list detailing what things you have to deal with to make NAT work with Asterisk |
| 15:00.44 | Zeeek | NAT doesn't work with asterisk... except when it does :) |
| 15:00.58 | liversmudge | MaNxPower : Im not on the list, but have many phones working through a natted router.. just stops working with realtime |
| 15:01.18 | Zeeek | oops, after 4PM - gotta run |
| 15:01.25 | liversmudge | imodium |
| 15:01.29 | liversmudge | try that :) |
| 15:01.34 | ManxPower | liversmudge, I don't normally help people that are not on the mailing list (if they are too lazy to help themselves, then I'm too lazy to help them) |
| 15:01.42 | Zeeek | no I *need* to run |
| 15:01.48 | liversmudge | oh right |
| 15:01.53 | ManxPower | HOWEVER, I take pitty on your: My notes on SIP w/Asterisk: SIP w/NAT works just fine if: |
| 15:01.53 | ManxPower | <PROTECTED> |
| 15:01.53 | ManxPower | <PROTECTED> |
| 15:01.53 | ManxPower | <PROTECTED> |
| 15:01.53 | ManxPower | <PROTECTED> |
| 15:01.54 | ManxPower | <PROTECTED> |
| 15:01.54 | liversmudge | nice attitude. |
| 15:01.56 | ManxPower | If your NAT router is SIP aware then you can 1) turn off it's SIP awareness and treat it like a dumb NAT router or 2) enable it's SIP awareness and turn off nat=yes in sip.conf. A SIP aware router might make reinvites work of both SIP clients have a SIP aware router. |
| 15:02.02 | ManxPower | * You can keep your NAT alive by using a registration of 60 seconds on the NAT device, or use qualify=yes in sip.conf, or use the NAT Keepalive features of your SIP device. |
| 15:02.05 | ManxPower | EEEK!!!!! |
| 15:02.11 | ManxPower | Here's the pastebin: http://pastebin.ca/5207 |
| 15:02.32 | Zeeek | another take |
| 15:02.37 | vaewyn | ~lart ManxPower |
| 15:02.37 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 15:02.42 | vaewyn | hehehe |
| 15:02.42 | liversmudge | thats great .. but under realtime there aint a keepalive I think |
| 15:02.46 | AgiNamu | How much is the LERG? |
| 15:03.04 | vaewyn | ManxPower: had too... not often you slip up with the paste :P |
| 15:03.23 | ManxPower | liversmudge, qualify= is the only asterisk specific thing. The rest are specific to the phone. |
| 15:03.27 | ManxPower | vaewyn, Yeah, I know. |
| 15:03.43 | liversmudge | It would appear that from googling (self help) that I am going to have to make each ohone reregister each 60 seconds .. not waht I wanted |
| 15:04.03 | ManxPower | vaewyn, My linux box's mother board blew up. I'm using my laptop (only Win32 box I have) and the whole "some apps copy on select like unix |
| 15:04.06 | vaewyn | rereg once a minute... OUCH |
| 15:04.08 | Zeeek | http://willypick.mindsay.com/?entry=10 |
| 15:04.13 | liversmudge | exactly |
| 15:04.19 | ManxPower | and "some apps copy on select and ctrl-C" can be annoying. |
| 15:04.20 | Zeeek | The asterisk config that dare not speak its name: Double NAT! |
| 15:04.32 | vaewyn | ManxPower: agrred |
| 15:05.01 | AgiNamu | Or... does anyone else have a way of getting "nice names" for an areacode + exchange? |
| 15:05.51 | AgiNamu | My provider gives me stuff like "CA LSAN DA 01" |
| 15:05.57 | AgiNamu | or "CA SNFC CNTRL" |
| 15:06.32 | ManxPower | AgiNamu, But those ARE the "nice names" |
| 15:06.51 | AgiNamu | ok... I should say "average customer of mine friendly names" |
| 15:07.22 | PakiPenguin | can have a context for calls coming in from a specific ip only , like a gateway ( far away , not registered to my * ) sends me calls like 009X300XXXXX@myip , i need to handle these calls in a different way then anyother incoming call , how do i do it ( the sending gateway ip addy is static ) |
| 15:07.26 | AgiNamu | SIP sucks balls. |
| 15:08.40 | AgiNamu | theoretically you could have any number of NAT Zeek |
| 15:08.54 | AgiNamu | so long you keep forwarding shit around. but that's hardly a maintainable or in many cases, possible, setup. |
| 15:09.01 | liversmudge | NAT and relatime ... heartache |
| 15:09.39 | ManxPower | liversmudge, So fix it. you ARE using a developement version of Asterisk where things change on a daily basis. |
| 15:09.42 | toddf | anyone laugh like me at someone wanting to hook a fax machine up to an ATA device and fax through an asterisk <-> asterisk -> another ata + fax machine setup? |
| 15:10.10 | *** join/#asterisk matty__ (~matty@cp34900-c.gelen1.lb.home.nl) |
| 15:10.14 | matty__ | hi |
| 15:10.23 | matty__ | could someone help me with the extesions ? |
| 15:10.25 | vaewyn | toddf: I do it... but I control the network between :} |
| 15:10.32 | Makenshi | toddf, if the ata does t38, fine |
| 15:10.37 | toddf | I thought so. I have a client that seems to insist on trying it .. |
| 15:10.40 | toddf | t38 ? |
| 15:10.48 | matty__ | i have 2 isdn fritz cards running with chan_capi |
| 15:10.51 | vaewyn | FoIP |
| 15:11.00 | Makenshi | yes, if the two atas support t38 and reinvite, it should work just fine |
| 15:11.03 | matty__ | and i want to dail-out with 4 channels at the same time |
| 15:11.05 | toddf | hmmm there are ata's that do it? |
| 15:11.09 | toddf | wow |
| 15:11.10 | Makenshi | toddf, yes |
| 15:11.24 | toddf | this is inter-company-over-internet-vpn-long-distance-avoidance .. wow, thanks .. |
| 15:11.34 | Makenshi | ata186 does |
| 15:11.37 | PakiPenguin | toddf: it actually works :) |
| 15:11.38 | *** join/#asterisk Casper_UA (~casper@as-2-22.ar43-2x.kharkov.ukrtel.net) |
| 15:11.39 | _-Jon-_ | Does anyone know why if I receive a call in through VoicePulse and then it calls out to my cell phone using BroadVoice is sounds horrible? |
| 15:11.52 | toddf | makenshi: iaxy ? |
| 15:11.57 | _-Jon-_ | Yet individual calls on VP and BV sound fine |
| 15:12.04 | Makenshi | toddf, i doubt it |
| 15:12.06 | PakiPenguin | toddf: i mean the linux->ata -> * -> VOIP |
| 15:12.06 | toddf | heh |
| 15:12.27 | PakiPenguin | FoIP :) |
| 15:12.58 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 15:13.10 | vaewyn | If you know you are faxing between sites only then use hylafax or spandsp to receive the fax locally and then email it to the other machine and use hyla/span to send it again :P |
| 15:13.25 | matty__ | can someone help me with the extension for dailing out with 4 isdn chans over 2 msn's ? |
| 15:13.49 | *** join/#asterisk stickynomore (~jeff@nsc66.147.11-46.newsouth.net) |
| 15:14.36 | *** join/#asterisk malcolmd (~malcolmd@malcolmd.digium.sponsor.pdpc) |
| 15:15.32 | toddf | vaewyn: yeah, I guess that's another option ... ;-) |
| 15:16.56 | vaewyn | or heck... just make it fax->email and leave it as a pdf |
| 15:16.59 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:17.01 | vaewyn | :} |
| 15:17.09 | _-Jon-_ | Hey does Asterisk support faxing yet?> |
| 15:17.44 | ManxPower | _-Jon-_, Only if you have the Magical Fairy Dust add-on. |
| 15:18.08 | AgiNamu | So, is there an easy way to get City + detail given NPA-NXX? |
| 15:18.13 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:18.33 | ManxPower | AgiNamu, not automagically. Well, yes, but not cheap and automatic. |
| 15:18.41 | mAsH` | anyone known how can i test a crossover T1 cable? |
| 15:18.45 | AgiNamu | Not cheap meaning? |
| 15:18.55 | _-Jon-_ | Hmm |
| 15:19.06 | HjemmeRoyK | _-Jon-_: not unless you have very, very low latency so you can run T.30 fax over alaw or ulaw. Otherwice, it's http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38%20Bounty |
| 15:19.06 | ManxPower | AgiNamu, I think the LERG is several thousand per year. |
| 15:20.02 | _-Jon-_ | How low of a latency? |
| 15:20.21 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 15:20.21 | AgiNamu | fuck... and it's not even localized |
| 15:20.25 | *** join/#asterisk CBAsteriskUK (~cblunt@208.51.30.218) |
| 15:21.18 | AgiNamu | Manx,a nd the LERG has nicer records ? My provider said that "CA LSAN DA 01" is the best it has |
| 15:21.29 | *** join/#asterisk pulu (~chatzilla@65.77.78.3) |
| 15:21.37 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 15:22.35 | ManxPower | AgiNamu, Ask your provider for the "rate center" not the CLLI. |
| 15:22.42 | AgiNamu | ok |
| 15:22.53 | CBAsteriskUK | Any one in the uk with Digium TFM400P FXO? |
| 15:23.07 | ManxPower | CLLI is formatted like "CA LSAN DA 01". Rate center is formatted like "NEW ORLEANS, LA" |
| 15:23.13 | AgiNamu | OK |
| 15:24.03 | ManxPower | CLLI specifies the actual switch (I think), but rate center just provides the general area. |
| 15:24.24 | HjemmeRoyK | _-Jon-_: 10ms or so |
| 15:24.29 | HjemmeRoyK | _-Jon-_: at least <20ms |
| 15:24.34 | AgiNamu | cool |
| 15:25.08 | greg_work | what's the state of dialtone detection on zap channels? |
| 15:25.23 | ManxPower | greg_work, It does not exist. No matter how many times people ask. |
| 15:25.24 | AgiNamu | sigh.. usa database copyright stuff :\ |
| 15:26.15 | greg_work | ManxPower: is it not possible or something? |
| 15:26.17 | ManxPower | My carrier gave me just the Louisiana and Mississipi parts of the LERG in spreadsheet format. Very useful. |
| 15:26.27 | ManxPower | greg_work, nobody has written support for it. |
| 15:26.53 | *** join/#asterisk freat (~freat@node-40242662.mdw.onnet.us.uu.net) |
| 15:26.55 | *** join/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 15:27.11 | AgiNamu | I just want it when users request DIDs |
| 15:27.27 | *** join/#asterisk Juggie (agony@24.114.136.55) |
| 15:27.30 | AgiNamu | So they can search for "los angeles" ors omething |
| 15:27.37 | AgiNamu | somes times they do not know the area code |
| 15:27.50 | ManxPower | The problem with things loike T.38 and dialtone detection is that the only people that are qualified to write it are the same people that don't need it. |
| 15:28.10 | tzanger | ManxPower: :-) |
| 15:28.11 | AgiNamu | no, that's the problem with community-driven dev :) |
| 15:28.20 | tzanger | AgiNamu: that's the *power* of community-driven dev |
| 15:28.24 | tzanger | that's what I love about open source |
| 15:28.29 | tzanger | if you have an itch, you scratch it |
| 15:28.39 | AgiNamu | That exists in both models, btw. |
| 15:28.39 | tzanger | if you have an itch and can't code, you get money to bribe someone who can |
| 15:28.50 | AgiNamu | so long as you have a damn API that's stable and an extensibility model. |
| 15:28.54 | tzanger | AgiNamu: not really -- if you have an itch and don't have access to the code, you're screwed |
| 15:28.59 | AgiNamu | Not at all |
| 15:29.03 | tzanger | AgiNamu: this is true |
| 15:29.24 | AgiNamu | I can modify Windows enough to suit my needs, I can modify Office too |
| 15:29.34 | AgiNamu | I dont need access to Windows source, nor Office source. |
| 15:29.37 | tzanger | but it has been my experience that closed-source's APIs move not only as the software grows, but also politically to prevent competition |
| 15:29.45 | tzanger | AgiNamu: office's api is a perfect example |
| 15:29.49 | AgiNamu | And the reason is that the core product is designed to cover a lot of stuff |
| 15:29.57 | AgiNamu | and the extensibility model makes sense for a wide range of people |
| 15:30.19 | ManxPower | DSP stuff is pretty weird stuff and not a lot of people are qualified to work with it. |
| 15:30.38 | AgiNamu | Sure, but if it was a simple extensibility model, having source to the CORE product is unneeded. |
| 15:30.48 | AgiNamu | So long as the core product uses settings and likewise, and not build flags |
| 15:30.51 | tzanger | AgiNamu: agreed |
| 15:30.53 | AgiNamu | and dumbass things like that :P |
| 15:30.54 | mAsH` | WARNING[3393]: chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 |
| 15:31.02 | mAsH` | what does it mean ? :/ |
| 15:31.03 | tzanger | ManxPower: yes. it is fun to work on though |
| 15:31.10 | tzanger | mAsH`: check the source and see |
| 15:31.20 | AgiNamu | saying "oh, if you want a differnet log file, recompile changing this field" shows a completely lack of design or customer understanding. |
| 15:31.21 | tzanger | chan_zap.c line 7411 prints that -- see what it's looking at |
| 15:31.27 | tzanger | AgiNamu: yes |
| 15:31.54 | tzanger | AgiNamu: but having a nice logging app that works with Office97 and having ot rewrite it for office2k and again for office2k3 shows a complete lack of resepect on the part of the office devs |
| 15:32.00 | AgiNamu | One thing I wanna do when I get serious funding is "pull a RedHat" with Asterisk |
| 15:32.10 | tzanger | although it does, in fact, allow the plugin dev more money for upgrading his clients |
| 15:32.10 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-70-18-115-202.buff.east.verizon.net) |
| 15:32.16 | tzanger | AgiNamu: fork it? |
| 15:32.18 | AgiNamu | tzanger, I'd be surprised if EVERY app breaks for a new version of office |
| 15:32.25 | AgiNamu | tzanger, damn straight. fork it, design it, test it, etc. |
| 15:32.26 | tzanger | AgiNamu: every app does not |
| 15:32.35 | fa | hm |
| 15:32.36 | tzanger | but the little inconsistencies drive you right up the wall |
| 15:32.45 | AgiNamu | MS has huge compat... VB has the damn error codes from the original version Bill wrote |
| 15:33.42 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 15:34.05 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 15:34.28 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 15:35.56 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 15:37.42 | fa | ;] |
| 15:38.04 | CBAsteriskUK | Hi All TDM400P FXO not detecting hangup any ideas? (UK) |
| 15:38.06 | *** join/#asterisk kletter-matze (~kletter-m@212.126.219.82) |
| 15:40.42 | pulu | does anyone know how to quickly clear the ip4 connection tracking table? (what shows up in /proc/net/ip_conntrack)? |
| 15:40.56 | rontecxt44 | hullo all... |
| 15:41.22 | rontecxt44 | quick question...I've just set up my first asterisk...and am running into a snafoo... |
| 15:41.35 | rontecxt44 | i'm very cozy in linux....just now with the pbx |
| 15:41.46 | zno | MS tries to be backwards compat, they've kept all their APIs and made new ones with ...Ex like LogonUser and LogonUserEx |
| 15:41.56 | rontecxt44 | i can recieve incoming calls...but not place outgoing |
| 15:41.59 | zno | it's just that they rewrite their software every few years |
| 15:42.09 | ManxPower | pulu, unload iptables/ipchains, then reload them |
| 15:42.12 | rontecxt44 | i get a fast busy |
| 15:42.26 | rontecxt44 | this is a single analog line setup |
| 15:42.28 | AgiNamu | zno, well, now Windows NT 6 will have a 100% managed API |
| 15:42.32 | pulu | my bad, they're compiled in |
| 15:42.38 | AgiNamu | hell, a lot of the core OS code is managed |
| 15:42.47 | AgiNamu | so, Win32 API is finally being "deprecated" in a way :D |
| 15:43.47 | *** join/#asterisk doughecka_ (~dheckaman@doughecka.user) |
| 15:43.52 | zno | I don't think they will deprecate them, they will have a complete managed API for everything you need, right now you still have to do win32api calls even if you're developing in a managed world |
| 15:44.03 | jas_williams | rontecxt44: What does ztcfg -vvv give from the shell |
| 15:44.14 | pulu | if my asterisk box comes up before the gateway on a power loss situation (which it always does because it's like 6 times faster), all the iax connections are conntracked before they get natted, and get stuck that way |
| 15:44.19 | AgiNamu | zno, I doubt they will continue recommending calling win32 in longhorn |
| 15:44.36 | rontecxt44 | one moment folks |
| 15:44.41 | zno | we'll see |
| 15:44.42 | rontecxt44 | someone lending a hand |
| 15:44.57 | jas_williams | rontecxt44: Post your extensions.conf to pastebin.ca |
| 15:45.07 | AgiNamu | zno, well, for the visuals... Avalon HAS no Win32 API :) |
| 15:45.11 | rontecxt44 | k |
| 15:45.49 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 15:46.16 | rontecxt44 | jas_williams |
| 15:46.21 | *** join/#asterisk jcims (~jcims@dsl093-208-067.clb1.dsl.speakeasy.net) |
| 15:46.24 | rontecxt44 | jas_williams, thanks...I will |
| 15:46.33 | rontecxt44 | it is a mutation of John Todd's config |
| 15:47.39 | *** part/#asterisk jcaustin (~jcaustin@206.127.19.236) |
| 15:47.45 | jcims | can * see the called number on an inbound call to a PRI? i.e. in an analog hunt group you have no clue |
| 15:48.24 | jcims | i.e. can i get say 50 DIDs into a single PRI? |
| 15:49.08 | *** join/#asterisk tty74 (~tiziano@151.11.170.20) |
| 15:50.16 | tty74 | I can't use diax to connect via iax2 to asterisk My error is:Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
| 15:50.17 | tty74 | <PROTECTED> |
| 15:50.17 | tty74 | <PROTECTED> |
| 15:50.24 | JerJer | add some authority |
| 15:50.39 | zoa | jj, going to von ? |
| 15:50.52 | tty74 | How i add some autority? |
| 15:51.23 | JerJer | zoa: we are exhibiting there :) |
| 15:51.57 | zoa | aha super |
| 15:52.01 | zoa | ill see you there then |
| 15:52.02 | zoa | :) |
| 15:52.21 | ManxPower | tty74, That means the username/secret/context/extension or section name do not match what the client is sending |
| 15:52.44 | ManxPower | jcims, Yes. |
| 15:52.53 | ManxPower | We put 100 numbers on each of our PRIs |
| 15:53.01 | JerJer | ManxPower: a peer should never request a specific context |
| 15:53.12 | jcims | thanx manx :) |
| 15:53.17 | zno | how much does it cost for DiDs? |
| 15:53.22 | JerJer | tty74: you need a [blah] type=user with a valid secret and context entry |
| 15:53.34 | ManxPower | zno, on my PRIs? US$20/month/per hundred. |
| 15:53.43 | bprice20 | ManxPower |
| 15:53.46 | bprice20 | how are ya man |
| 15:53.50 | JerJer | I've noticed quite a few of these kinds of floods, since upgrading to newest cvs -head: |
| 15:53.51 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 15:53.51 | ManxPower | JerJer, There's really no reason to use user/peer for things like DIAX |
| 15:53.52 | zno | wow, that's pretty good |
| 15:53.53 | JerJer | Dropping incompatible voice frame on IAX2/<censored>@<censored> of format gsm since our native format has changed to speex |
| 15:53.58 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 15:54.10 | JerJer | ManxPower: eh? |
| 15:54.22 | JerJer | you have to have a type=user in iax.conf for anyone to call in |
| 15:54.25 | ManxPower | JerJer, or at least I can't think of any. |
| 15:54.35 | ManxPower | JerJer, I meant user/peer .vs. friend |
| 15:54.50 | netsurfer | Guys, whats a general price for a T1 in north america? |
| 15:54.59 | tzanger | netsurfer: varies wildly by region |
| 15:55.02 | JerJer | ManxPower: then you haven't deployed asterisk big enough yet |
| 15:55.02 | tty74 | User is authent ok because i can show ixa2 show peers registred |
| 15:55.10 | ManxPower | netsurfer, between $300/month and $2,500/month. No way to tell. |
| 15:55.10 | JerJer | type=friend WILL bite you, someday |
| 15:55.34 | ManxPower | JerJer, Actually I just don't have IAX phones. I use user/peer for any server/server communications. |
| 15:55.49 | JerJer | this is not just an iax issue |
| 15:55.50 | ManxPower | The ONLY time I use type=friend is for PHONES, not servers. |
| 15:55.57 | netsurfer | ManxPower - just wondering, trying to compare to our E1's |
| 15:56.05 | JerJer | trust me |
| 15:56.08 | JerJer | type=friend is evil |
| 15:56.16 | ManxPower | netsurfer, price varies depending on where you are in the USA |
| 15:56.17 | JerJer | and will bite you, eventually |
| 15:56.21 | jcims | with friends like these.... ta dum! |
| 15:56.51 | netsurfer | ManxPower - let me guess... east or west coasts being the cheapest ? |
| 15:56.57 | ManxPower | netsurfer, no. |
| 15:57.23 | ManxPower | netsurfer, More competition in a market can help, but it's really the regulators for each state that approve telco pricing. |
| 15:57.25 | jcims | netsurfer, i just got a quote for $350/mo for a PRI in an existing datacenter (colo for my machine) |
| 15:57.39 | jcims | PRI == ISDN t1 |
| 15:57.47 | netsurfer | jeez! |
| 15:58.00 | ManxPower | i.e Tennesee regulators are very hostile to the phone company and they have some of the cheapest prices on ISDN BRI in the entire country. |
| 15:58.00 | netsurfer | thats dirt cheap! |
| 15:58.08 | jcims | of course i have to pay colo fees...it's no good for internet access |
| 15:58.09 | djin | that's a lot of money, jcims. |
| 15:58.21 | jcims | for a PRI? |
| 15:58.26 | ManxPower | djin, That's practically free for a PRI. |
| 15:58.29 | jcims | seems fairly cheap to me |
| 15:58.29 | djin | I have 4 PRI's (30channels) for Euro 0,00 |
| 15:58.32 | jcims | doh |
| 15:58.45 | jcims | loopback? |
| 15:58.46 | *** part/#asterisk CBAsteriskUK (~cblunt@208.51.30.218) |
| 15:58.53 | jcims | :P |
| 15:58.55 | djin | hehe |
| 15:59.03 | ManxPower | In Louisiana a 0 mile T-1 is about $350/month |
| 15:59.12 | tzanger | jcims: not quite but close enough |
| 15:59.14 | jcims | this is in columbus, ohio |
| 15:59.20 | djin | Normal price is around Euro 180 here |
| 15:59.23 | ManxPower | i.e. both ends of the T-1 are served out of the same central office. |
| 15:59.54 | *** join/#asterisk powD (~atob100@83.146.53.34) |
| 16:00.01 | jcims | anyone ever try playing with alarm circuits? |
| 16:00.06 | tzanger | jcims: yup |
| 16:00.07 | Darwin35 | anyone here done a park and announce on pa extension ? |
| 16:00.15 | tzanger | works well so long as the telco cuts out the bridge taps |
| 16:00.17 | jcims | can you drop a fxs on one end and a phone on the other? |
| 16:00.18 | shido6 | oooh |
| 16:00.20 | shido6 | home automation |
| 16:00.21 | shido6 | an alarms |
| 16:00.25 | powD | Can anyone tell me how to get debug for an X100P card from asterisk CLI |
| 16:00.27 | tzanger | it pays to get in good with your line techs :-) |
| 16:00.30 | jcims | haha |
| 16:00.32 | tzanger | jcims: oh for FXS it works just fine |
| 16:00.32 | jcims | my brother is one |
| 16:00.35 | *** join/#asterisk jets (~jetsn@guardian.pmt.org) |
| 16:00.43 | jcims | you thinking dsl/t1 tzanger? |
| 16:00.46 | ManxPower | jcims, We ran almost 100 customers on alarm circuits many years ago using some of the first DSL in Louisiana |
| 16:00.48 | Delvar | me |
| 16:01.04 | tzanger | jcims: SDSL yeah... you should be abel to do HDSL2 as well but it's powered and that's not nice |
| 16:01.18 | jcims | i just want dialtone :) |
| 16:01.33 | netsurfer | ManxPower - thx for that, just curious :) |
| 16:01.37 | tzanger | jcims: yeah alarm circuit will work just fine |
| 16:01.45 | Darwin35 | I need a exten sith autopark and then annouce on the pa that there is a call for this person on exten of the parked call |
| 16:02.01 | ManxPower | We pay about $600/month for a T-1 PRI w/4 channels |
| 16:02.05 | ManxPower | and 100 numbers |
| 16:02.06 | jcims | do they need to be 0-mile per manx's definition above? can i get an alarm circuit through to co's? |
| 16:02.09 | shido6 | corndelacob |
| 16:02.32 | AgiNamu | Manx: My provider says "The listed Rate Center for 212-660 is "NWYRCYZN01". The listed Locality for 212-660 is "NYNWYRCYZN01Z"" |
| 16:02.36 | ManxPower | jcims, For the most part you need your alarm circuit to be "0 mile". |
| 16:02.38 | netsurfer | 4 channels, that gives 24 voice circuits ? |
| 16:02.48 | ManxPower | AgiNamu, That's the CLLI |
| 16:02.48 | shido6 | make them cut 200 more bucks off or threaten to disrupt their 99.999 percent uptime by jumping rope with random cables |
| 16:03.05 | AgiNamu | so ... my clec doesnt know how to read the lerg. sig. |
| 16:03.06 | ManxPower | netsurfer, no, 4 channels gives us 4 voice circuits |
| 16:03.16 | shido6 | tie all of their rj45 cables into a giant spaghetti not |
| 16:03.21 | jcims | ManxPower |
| 16:03.23 | jcims | :oop |
| 16:03.37 | jcims | how much do typical alarm circuits run /month? |
| 16:03.38 | greg_work | ManxPower: ouch, that's pretty high, isn't it? |
| 16:03.42 | Cherebrum | only 4 channels? |
| 16:03.45 | Cherebrum | for $600 |
| 16:03.46 | Cherebrum | damn |
| 16:03.48 | ManxPower | MOST of that cost is for the T-1, not for the channels. |
| 16:03.49 | AgiNamu | Note to all devs: Memory is slow as shit. Just because you have 2GB of ram in each machine doens't mean you should use ram left and right. CPU cycles are CHEAP (perf wise) compared to ram |
| 16:03.58 | Cherebrum | did they use lube? |
| 16:04.00 | ManxPower | Cherebrum, The same line from bellsouth is $1,200/month |
| 16:04.10 | vaewyn | AgiNamu: amen! |
| 16:04.16 | netsurfer | lol Cherebrum |
| 16:04.23 | *** part/#asterisk bowman (~bowman@snert3.tal.de) |
| 16:04.33 | AgiNamu | Blow the cache and take the fucking perf ugly stick to your head |
| 16:04.43 | shido6 | get a long hose from the outside and ride the elevator to the puter room and put a sprinkler on the other end. go outside and turn it on if they dont cut your t1 price down |
| 16:04.45 | ManxPower | Sorry! I just checked. it's SIX channels |
| 16:05.05 | netsurfer | ManxPower - that mean u have 6 voice channels and the rest is data ? |
| 16:05.13 | ManxPower | netsurfer, no, the rest is unused. |
| 16:05.21 | ManxPower | We have frame relay for data. |
| 16:05.23 | `Sauron | AgiNamu: http://puck.nether.net/npa-nxx/lookup.cgi?npa=212&nxx=660 |
| 16:05.38 | `Sauron | You probably know that site already, though. |
| 16:05.38 | *** part/#asterisk djin (~marius@62.58.40.196) |
| 16:05.41 | AgiNamu | and secondary freaking note: elegance is more important than perf until tests show otherwise |
| 16:05.42 | netsurfer | ManxPower - would it not be cheaper to use 3 isdn2 lines ? |
| 16:05.51 | shido6 | AgiNamu how fast is your ram? |
| 16:05.59 | ManxPower | netsurfer, No. We are in the USA. |
| 16:06.09 | eKo1 | Does anyone have * working with SER? I want to call from a phone connected to * to a phone behind the SER. Can this be done? |
| 16:06.11 | AgiNamu | shido6, um, 400 or 800MHz? Compared to a 3.8GHz proc.... |
| 16:06.24 | ManxPower | ISDN BRI is about $100/ea here, our provider does not offer BRI service and Asterisk does not really support USA BRI. |
| 16:06.27 | AgiNamu | plus, retrieve time |
| 16:06.27 | shido6 | u have 800mhz ram? |
| 16:06.29 | AgiNamu | and so on |
| 16:06.35 | netsurfer | holy crap |
| 16:06.37 | AgiNamu | well, 400MHz doubled |
| 16:06.38 | ManxPower | Our provider also gives us unlimited free calling in Louisiana and Mississippi |
| 16:06.40 | netsurfer | $100/mo lmao |
| 16:07.41 | ManxPower | So if we went with 3xBRI we would no longer have free calling within Louisiana and Mississippi, we would be using hardware that's not supported by Asterisk, and we would have to change carriers. |
| 16:08.23 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 16:09.01 | pashah | if I need only 1 FXO port which device should I get? I think I need the cheapest in that case. |
| 16:09.16 | jets | x100p |
| 16:09.18 | ManxPower | The office keeps saying that they will be expanding with many more employees, but that's happening slowly. |
| 16:09.32 | ManxPower | Once we get 10 or more channels the price will be about the same as POTS service. |
| 16:09.33 | AgiNamu | well, cheapest is a cheap modem pos |
| 16:09.36 | eKo1 | Hmm...guess not. |
| 16:09.47 | ManxPower | The X100P is no longer sold by Digium. |
| 16:09.49 | AgiNamu | but the TDM card is a more decent investment |
| 16:10.08 | pashah | TDM card with one FXO module? |
| 16:10.12 | AgiNamu | yea, that'd what I'd do |
| 16:10.19 | AgiNamu | or a sipura |
| 16:10.24 | AgiNamu | or any ATA :) |
| 16:10.30 | pashah | sipura sounds evil |
| 16:10.33 | jets | If digium no longer sells a cheaper one port FXO -- you could buy a $10 modem with a specific chipset. |
| 16:10.38 | jets | It would be unsupported tho |
| 16:10.59 | ManxPower | jets, Yes, but those modems will become harder and harder to find soon. |
| 16:11.03 | pashah | jets: unsupported - no use |
| 16:11.22 | jets | ManxPower: Are they stopping that chipset i assume |
| 16:11.29 | ManxPower | jets, That's what I hear |
| 16:11.38 | jets | Good -- that means digium employee's get to eat.... |
| 16:11.47 | AgiNamu | pashah, you will be happer with an ATA or the TDM card. ATA will be cheaper. |
| 16:12.02 | AgiNamu | puck.nether.net -- is that a reliable service? |
| 16:12.27 | netsurfer | jets - the X100P generic clone is on sale here in the UK, a lot of ppl are very happy with them, however obviously they aren't modular like the TDM |
| 16:12.31 | pashah | AgiNamu: thanks |
| 16:12.45 | pashah | netsurfer: url? |
| 16:12.53 | *** join/#asterisk calvinhp (~calvinhp@rrcs-24-123-25-236.central.biz.rr.com) |
| 16:13.03 | *** part/#asterisk jcims (~jcims@dsl093-208-067.clb1.dsl.speakeasy.net) |
| 16:13.28 | netsurfer | pashah - i'd prefer not to promote the company yet, I ordered a card last friday and its still not here |
| 16:13.56 | netsurfer | they haven't emailed to confirm either.. however another guy I spoke to said he got his in 3 days |
| 16:14.19 | pashah | =0 |
| 16:14.20 | AgiNamu | how can nether distribute that NPA NXX info? isn't it all copyrighted and expensive? |
| 16:14.55 | jets | I don't believe so. |
| 16:15.11 | jets | The phone company I work for receive NPA NXX updates all the time so they can place it in there switches. |
| 16:15.12 | AgiNamu | so... i can get a db for free? i dont need to buy the LERG? |
| 16:15.24 | *** join/#asterisk bannerman (~bannerman@dpc6682105089.direcpc.com) |
| 16:15.27 | wolfson | npa pxx data and whats in the lerg are not the same |
| 16:15.44 | wolfson | its got much more data as I understand it, like whats actually local, extended local, etc... |
| 16:20.15 | wankel | the lerg has lots of local routing information, as its name implies. pricing for LD is broken up in a lot of strange ways, with LATAs and rate centers and bands and all sorts of things you'd rather not know about. if you do want to know, though, the LERG will tell you. |
| 16:21.58 | bannerman | Hi. I've got some newb questions, but I've read everything I can find, and can't solve my problem on my own. I'm looking at purchasing VoIP via Covad, but I'd like to use their basic $19.95 voip service with my own pbx. Is there a way to route calls through these Covad "lines" via Asterisk? |
| 16:22.56 | toddf | you can get $6/mo through libretel.com |
| 16:23.01 | *** join/#asterisk Zaw (zaw@zaw.subneural.net) |
| 16:23.01 | toddf | if all you want is inbound |
| 16:23.17 | bannerman | need outbound too, basically just replacing our POTS lines with VoIP if we can |
| 16:23.38 | bannerman | I don't want to pay $69.95/mo for 15 lines for our 15 extensions using their vPBX, since we only use 4-5 lines at once |
| 16:24.11 | bannerman | But regarding libretel, or any other VoIP service-- how does it work, to route calls from that through an Asterisk system? |
| 16:24.14 | jets | voicepulse will allow you to have up to 6 recurring inbound/outbound connections |
| 16:24.15 | wankel | they use mgcp. i haven't heard of anyone tying it in with asterisk, but it ought to work, i would think. |
| 16:24.36 | wankel | bannerman: if you're using an IAX voip provider, it's trivial to set up asterisk to route the calls from local phones |
| 16:25.09 | bannerman | jets: thanks, I'll check into that |
| 16:25.34 | bannerman | wankel: so it's a common thing to use a provider like that to route calls to regular phones, via my own asterisk system |
| 16:25.44 | *** join/#asterisk sivana (~richard@209.91.159.221) |
| 16:25.51 | bannerman | Is there another provider I should look into? It sounds like Covad does some odd stuff |
| 16:25.57 | wankel | very common |
| 16:26.01 | AgiNamu | Wolfson, yea, I dont need ht lerg. I just want to have the NPA-NXX names |
| 16:26.02 | jets | bannerman: fairly -- most people use other PBX's like a mitel or a panasonic but if its a compliant protocol asterisk will work too. |
| 16:26.11 | wankel | there are lots. take a look at www.voip-info.org |
| 16:26.25 | `Sauron | AgiNamu: http://puck.nether.net/npa-nxx/ |
| 16:26.26 | wankel | it has a lot of great information about setting up asterisk as well as information about asterisk-compatible providers |
| 16:26.36 | BoRiS | bannerman: $69.96 per each line? |
| 16:26.56 | jets | These companies terminate with IAX for Asterisk -- and there may be cheaper and better providers available here: http://www.iaxprovider.net/index.php?module=pnAddressBook&func=main |
| 16:27.06 | bannerman | BoRiS: they provide full PBX services for $69.95 per extension |
| 16:27.20 | wankel | _per extension_? |
| 16:27.24 | wankel | holy shit. |
| 16:27.38 | bannerman | wankel: yeah, that's what I said |
| 16:27.45 | wankel | you can get traditional centrex for less than that |
| 16:28.04 | bannerman | it's a good deal if you have two people making calls all over the US and Canada, and need multiple lines and such |
| 16:28.05 | wankel | i was thinking $70 for 15 extensions was a pretty good deal :) |
| 16:28.28 | bannerman | but a regular office with 15 people and minimal phone usage for the most part, it's a horrible ripoff |
| 16:28.55 | AgiNamu | Sauron, but where do they get their DB? |
| 16:29.15 | `Sauron | AgiNamu: http://www.nanpa.com/area_codes/index.html |
| 16:29.42 | `Sauron | Or, email Jared and ask him |
| 16:29.43 | `Sauron | duh |
| 16:30.01 | AgiNamu | I went to that nanpa site, but only got the area code db |
| 16:30.07 | AgiNamu | didnt have the exchanges listed with names |
| 16:30.30 | `Sauron | <`Sauron> Or, email Jared and ask him |
| 16:30.30 | `Sauron | <`Sauron> duh |
| 16:30.36 | AgiNamu | yea ... who's jared? :) |
| 16:30.49 | cypromis | smith |
| 16:30.57 | `Sauron | http://puck.nether.net/ |
| 16:31.01 | `Sauron | Jared Mauch |
| 16:31.02 | tzanger | AgiNamu: he's the guy who ate nothing but subway subs and lost tons of weight :-) |
| 16:31.03 | AgiNamu | yea... looking around now :) |
| 16:31.05 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 16:31.08 | AgiNamu | tzanger... oh him |
| 16:31.10 | `Sauron | Need me to hold your hand for anything else? |
| 16:31.59 | `Sauron | ManxPower: See what you've done to me? :( |
| 16:32.46 | AgiNamu | OK.... what about this idea guys |
| 16:33.03 | AgiNamu | In addition to the bug tracker (which desperately needs to be overhauled to reflect a real community driven dev model) |
| 16:33.18 | AgiNamu | People could report success or failure on testing a patch |
| 16:33.23 | bannerman | You folks seem like a knowledgeable lot, I know this is OT, but I'm setting up a new office in a town with no DSL, cable or wireless internet. I'm looking offset the high cost of a T1 by using VoIP instead of POTS. Any suggestions for a T1 provider? Covad so far has the best offer, at $350/mo for 1.54 |
| 16:33.41 | `Sauron | 350/mo is fairly cheap |
| 16:34.44 | dan2 | bannerman: is that a full t1? |
| 16:34.57 | tzanger | bannerman: you need to phone all the LECs and get pricing |
| 16:34.59 | *** join/#asterisk lohelle (slamm@213.161.252.253) |
| 16:35.29 | bannerman | dan2: that's what the sales guy said |
| 16:35.39 | bannerman | dan2: next best offer was $375/mo from a local ISP |
| 16:35.51 | *** join/#asterisk jaiger (~jaiger@205.139.10.95) |
| 16:35.52 | bannerman | tzanger: LECs? |
| 16:36.04 | tzanger | people who can provide you with a T1 to the PSTN |
| 16:36.27 | jaiger | tzanger: I wired my echo canceller yesterday and racked it just now |
| 16:36.38 | bannerman | tzanger: Is there a better place to get that, or is dslreports.com the best? |
| 16:36.52 | tzanger | jaiger: awesome, how's it working |
| 16:36.52 | dan2 | bannerman: yes, look at CLECs, Competitive Local Exchange Carriers |
| 16:37.01 | bannerman | dan2: ok, thanks |
| 16:37.03 | wolfson | bandwidth.com will also price shop for you. pretty nice people |
| 16:37.08 | jaiger | tzanger: ok |
| 16:37.11 | lohelle | when I try to dial out (zaphfc/isdn) I always get "Zap/1-1 is circuit-busy" and "Everyone is busy/congested at this time" . Dial in is OK. ISDN card connected to s0 adapter to pbx (alcatel 4400).. Any idea? |
| 16:37.32 | jaiger | my partner is complaining about some static but he complained about that before we used the echo canceller |
| 16:38.04 | jaiger | there is some echo at the beginning of calls but within a short period of speaking it completely disappears |
| 16:38.06 | Zeeek | anyone have callerid issues with siemens phone connected to TDM400 FXS ? |
| 16:38.22 | bannerman | wolfson: thanks |
| 16:38.45 | jaiger | tzanger: I found some standard molex connectors that fit the wire wrap posts |
| 16:38.50 | Zeeek | wasim |
| 16:38.55 | tzanger | excellent |
| 16:39.00 | *** join/#asterisk jpayne (~jpayne@baconhouse.sackheads.org) |
| 16:39.08 | Mother_ | anyone can recommend a good SIP or IAX provider towards Italy mainly? |
| 16:39.09 | tzanger | jaiger: hmm okay |
| 16:39.12 | jaiger | I had to trim some plastic but that was no big deal |
| 16:39.30 | Mother_ | calls originating from Europe |
| 16:39.49 | Zeeek | I have something in Italy |
| 16:39.54 | jaiger | my setup still needs tweaking for sound quality (I think tx/rx gain) but is functional |
| 16:40.02 | Zeeek | I'd have to log in to my gmail to remember where though |
| 16:40.03 | AgiNamu | whats up with this "go back to iraq" shit?" |
| 16:40.15 | Mother_ | Zeeek: no worries |
| 16:40.41 | Mother_ | I just want a provider that has good rates & good service towards Italy, or Europe for that matter |
| 16:40.55 | Zeeek | wait for it.... |
| 16:40.58 | Mother_ | with calls originating from Europe too |
| 16:41.00 | Mother_ | thanks :) |
| 16:41.07 | Zeeek | http://mytcom.it/ |
| 16:41.21 | Zeeek | No idea how good, I just use a free SIP acct |
| 16:41.41 | Mother_ | heh OK, thanks for the info |
| 16:41.45 | Mother_ | will try that out |
| 16:41.47 | Zeeek | they are good about one thing though, they always email before maintenance downtime |
| 16:41.53 | Zeeek | in Italian :) |
| 16:42.05 | Zeeek | Si informano gli iscritti a myTCom.it che nella giornata di giovedi 27/01/2005 verrà effettuato un intervento di manutenzione sui dispositivi adibiti al servizio dalle ore 20.00 alle ore 21.30. |
| 16:42.06 | Mother_ | hahaha nice of them, very nice |
| 16:42.18 | Zeeek | they are the only provider I have that has done this |
| 16:42.19 | jaiger | tzanger: and inbound fax still works fine |
| 16:42.26 | *** join/#asterisk dalabera (~Dalabera@146.82.190.162) |
| 16:42.34 | tzanger | jaiger: perfect :-) |
| 16:42.37 | dalabera | Hello everybody!!! |
| 16:42.38 | Zeeek | http://www.mytcom.it/icms_eng/RunScript.asp?Article_type=news&p=ASP\Pg0.asp |
| 16:43.13 | Mother_ | excellent |
| 16:43.13 | Zeeek | asterisk logo at the bottom :) :) :) |
| 16:43.21 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 16:43.24 | Darwin35 | ok anyone having problems with inbound on broadvoice |
| 16:43.26 | AgiNamu | the first italian telephon company? |
| 16:43.28 | Zeeek | and an HTML tag... |
| 16:43.41 | Zeeek | <TABLE> |
| 16:43.53 | *** part/#asterisk yaout (eric@CPE-65-30-220-56.wi.rr.com) |
| 16:44.13 | AgiNamu | Who's a good originator for Europe and Asia? I'm looking to pay a few $ a month for the DIDs, plus per minute |
| 16:44.31 | Mother_ | hahaha nice |
| 16:44.37 | Zeeek | check the names in the wiki - I've met at least one iof those huys |
| 16:45.05 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 16:45.06 | Zeeek | http://www.voip-info.org/wiki-Acropolis+Telecom |
| 16:45.06 | bprice20 | does anone have any advice for choppy audio |
| 16:45.20 | shido6 | dont run x windows on that box? |
| 16:45.23 | AgiNamu | Does anyone have good info on how #includes are handled? |
| 16:45.24 | Zeeek | heh |
| 16:45.26 | shido6 | stop using celerons? |
| 16:45.35 | Zeeek | bprice what is the lag in ms on the channel? |
| 16:45.39 | AgiNamu | is it shoved inline? i.e., if I #include in a context, the included data will be in context? |
| 16:45.41 | channan | hi Darwin35, I've used Broadvoice for a month now and did not have any problem yet. |
| 16:45.44 | AgiNamu | (talking config files) |
| 16:45.44 | shido6 | if it aint 256k cache u can kick it to the curb? |
| 16:45.50 | bprice20 | i have tried setting jitter buffer, echo cancel (not echo) free up irq's, enable alsa and using alsa driver, disabling zaptel and removing modules |
| 16:45.57 | Darwin35 | i can dial out from my pbx threw broadvoice but when they cll inbound they get a fast busy |
| 16:45.57 | shido6 | stop with the jitter |
| 16:45.57 | bprice20 | i have rtc in kernel |
| 16:46.00 | shido6 | are you running xwindows? |
| 16:46.09 | shido6 | bprice20, are you running x windows? |
| 16:46.13 | bprice20 | shido6 no gentoo linux |
| 16:46.19 | Darwin35 | hmmm |
| 16:46.24 | shido6 | so no GUI no x windows? |
| 16:46.25 | bprice20 | no x, haven't compiled it |
| 16:46.27 | ManxPower | AgiNamu, #includes are handled eactly the same way as C #includes |
| 16:46.30 | channan | darwin35 - it probably because your box is not registered |
| 16:46.31 | Darwin35 | I am not getting inbound calls |
| 16:46.32 | AgiNamu | ok |
| 16:46.35 | shido6 | bprice20, no celeron, right? |
| 16:46.39 | Darwin35 | its registerd |
| 16:46.42 | Zeeek | AgiNamu : "We also have our own DNS with ENUM standards operational connected in Paris and connected in Europe " |
| 16:46.43 | bprice20 | nope a dual xeon box |
| 16:46.43 | Darwin35 | I checked |
| 16:46.45 | AgiNamu | so it doesnt help at all with pattern matching |
| 16:46.58 | shido6 | brpice20 , using trunking? |
| 16:47.06 | bprice20 | shido6 its a dual xeon box |
| 16:47.15 | bprice20 | shido6, no trunking |
| 16:47.18 | *** join/#asterisk calisto (~sod@82-38-193-149.cable.ubr04.shef.blueyonder.co.uk) |
| 16:47.20 | JerJer | no framebuffer? |
| 16:47.20 | sivana | ~seen normast |
| 16:47.22 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca> was last seen on IRC in channel #asterisk, 2h 9m 35s ago, saying: 'Nivex: I guess it's them same on most distro however..'. |
| 16:47.25 | *** join/#asterisk sabre (~urfos@69.149.209.81) |
| 16:47.33 | shido6 | bprice20, whats the traceroute look like from your box to where you are calling? |
| 16:47.37 | channan | darwin35 - did you have problem with inbound sip call or pstn? |
| 16:47.38 | *** part/#asterisk calisto (~sod@82-38-193-149.cable.ubr04.shef.blueyonder.co.uk) |
| 16:47.41 | *** join/#asterisk calisto (~sod@82-38-193-149.cable.ubr04.shef.blueyonder.co.uk) |
| 16:47.59 | ManxPower | There have been extensive talks on the mailing lists about poor incoming voice quality from several providers. |
| 16:48.01 | calisto | anyone here from the uk use a ht286 ata |
| 16:48.09 | bprice20 | shido6 its 3 hops away under 6 ms |
| 16:48.19 | Darwin35 | sip inbound from broadvoice |
| 16:48.31 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 16:48.32 | *** join/#asterisk Bentley (~bentley@S01060080c8135e6a.cg.shawcable.net) |
| 16:48.37 | ariel_ | morning all |
| 16:48.46 | bprice20 | shido6, and i get the choppiness when just calling the box for instance when using voicemail |
| 16:48.56 | bprice20 | shido6 i'm using yesterdays cvs |
| 16:49.08 | channan | darwin35 - did it ever work before? |
| 16:49.12 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 16:49.18 | Zeeek | bprice what is your lag in ms to the asterisk server? |
| 16:49.35 | Zeeek | oops |
| 16:49.48 | channan | darwin35 - did you turn on debug and check? |
| 16:50.01 | AgiNamu | colocation is fun... now I've got .7ms ping times to my provider |
| 16:50.05 | Darwin35 | yes it worked before and debuging shows nothing |
| 16:50.46 | channan | dawrin35 - hmm... do you go thru vpn tunnel or direct public ip? |
| 16:52.15 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 16:52.28 | Darwin35 | direct ip |
| 16:52.31 | Darwin35 | right now |
| 16:52.37 | Darwin35 | I need to setup a stun |
| 16:53.02 | JerJer | stun is unnecessary |
| 16:53.36 | Zeeek | set phasers to kill, screw stun |
| 16:53.42 | ManxPower | JerJer, Amazing how many people what to make things as complicated as they can, huh? |
| 16:53.56 | _Brian | Has anyone had problems when using a T100P in which it would recieve the inbound PRI call, and then disconnect the call within 5-10 seconds of connection. The unit reports back a Error code 16 (normal Call clearing) |
| 16:54.11 | JerJer | some peoples children |
| 16:54.12 | Darwin35 | well if I use a sip phone setup the same way it all works |
| 16:54.18 | ManxPower | _Brian, only when you set callprogress=yes or busydetect=yes |
| 16:54.32 | Darwin35 | but when I use the asterisk box outbound works and inbound does not |
| 16:54.36 | channan | darwin35 - hmm... what kind of SIP phones r u using? |
| 16:55.04 | Darwin35 | zyxel wireless a gs 101 and a ata 286 |
| 16:55.22 | *** join/#asterisk twisted[work] (~twisted@twisted.active.supporter.pdpc) |
| 16:55.23 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 16:55.25 | channan | drwin35- that's why |
| 16:55.29 | _Brian | nope..both of them are not enabled |
| 16:55.35 | Darwin35 | ? |
| 16:55.42 | channan | darwin35 - just kidding.... I have not used zyxel |
| 16:56.05 | channan | darwin35 - I do have gs100 and it worked fine |
| 16:56.13 | Darwin35 | its nice when I go to the local cyber cafe and can take it with me and still get calls |
| 16:56.26 | JerJer | HA! i knew you were going to say barbietone |
| 16:56.27 | channan | darwin35- sure |
| 16:56.33 | JerJer | but i was going to be nice and not say anything |
| 16:56.48 | JerJer | so much for that |
| 16:57.03 | JerJer | enable NAT processing on the device |
| 16:57.09 | JerJer | then nat=yes on your entries in sip.conf |
| 16:57.12 | JerJer | problem solved |
| 16:57.21 | Darwin35 | it is enabled |
| 16:57.29 | JerJer | then STUN won't help you |
| 16:57.37 | JerJer | someone is blocking udp then |
| 16:57.42 | ManxPower | STUN only MAY be useful for a small number of people. |
| 16:58.02 | AgiNamu | for everyone else, there's IAX2. |
| 16:58.10 | JerJer | right on |
| 16:58.13 | *** join/#asterisk Yoda-BZH (~yoda-bzh@213.223.52.79) |
| 16:58.23 | calisto | anyone here from the uk use a ht286 ata |
| 16:58.28 | *** join/#asterisk dolson (~dana@Sudbury-HSE-ppp3979976.sympatico.ca) |
| 16:58.29 | AgiNamu | We'll probably be the first VSP to use IAX2 CPE |
| 16:58.39 | Delvar | calisto: iv got one here |
| 16:58.48 | JerJer | AgiNamu: i wouldn't go that far |
| 16:58.57 | calisto | delvar: you from the uk as I think its uk related |
| 16:58.58 | Darwin35 | broadvoice doesnot support iax2 right now |
| 16:59.03 | Darwin35 | I wish they would |
| 16:59.06 | Delvar | calisto: yep |
| 16:59.06 | AgiNamu | Know of any residential services shipping IAX2 devices? |
| 16:59.10 | channan | darwin35- can you explain what you meant by setting all SIP phones the same way and it worked (ie. same configure except extension?) |
| 16:59.38 | AgiNamu | And we're got the drop on all the other providers looking at say, guatemala. cause they are gonna run into double NAT |
| 16:59.47 | Darwin35 | I tested them all with same nmbr/passwd |
| 16:59.51 | *** join/#asterisk Yoda-BZH (~yoda-bzh@213.223.52.79) |
| 16:59.57 | calisto | delvar: any idea why it won't ring a phone attached even with the adaptor with a ring capacitor. Works fine with cheap phone borrowed from work but not with my vtech cordless |
| 16:59.59 | dolson | does anyone have any experience with snom's 4S proxy? my boss wants to use it, but I would rather learn SER... he wants snom because it runs on Windows. the question is, does it work as well as SER with Asterisk? |
| 17:00.07 | Delvar | just use an outbound proxy and nat isnt an issue anymore |
| 17:00.43 | Yoda-BZH | le bonsoir je vous souhaite / Hi ppl |
| 17:00.47 | bannerman | I hear a lot about echo, both here and on the different VoIP sites that I've been browsing around. What causes echo? If I purchase service through an IAX provider and use SIP phones connected via an asterisk pbx box, is this an issue I'm going to face at some point? |
| 17:01.01 | Delvar | calisto: no, some phones work ok but other dont, fortionatly all the ones iv tested seem to work fine. |
| 17:01.46 | calisto | delvar: it seems to not be giving up enough juice as the work phone doesn't seem to ring as loud as it does at work |
| 17:02.00 | Delvar | bannerman: you realy only get echo with pstn problems, iv nver got echo off a pure VOIP call |
| 17:02.01 | Darwin35 | brb door |
| 17:02.11 | bannerman | Delvar: That's what I wanted to hear, thanks |
| 17:02.44 | AgiNamu | I get echo on VoIP calls... when I use a softphone and no headset |
| 17:02.48 | AgiNamu | or speakerphones |
| 17:03.08 | Delvar | AgiNamu: thats not a voip isue, thats a client issue... :) |
| 17:03.11 | jets | are there any good IAX hardware phones yet |
| 17:03.15 | AgiNamu | jets yea |
| 17:03.16 | calisto | delvar: what setting do you use for fxs impedance |
| 17:03.18 | AgiNamu | the PA168 ones |
| 17:03.22 | AgiNamu | they're quite nice |
| 17:03.52 | Delvar | calisto: US, i leave on default, spoke to grandstream support and they said it shouldnt affect us in the uk |
| 17:04.07 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 17:04.29 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 17:04.33 | bannerman | My Covad sales rep is cautioning me about delay and fadeout using competing VoIP services. His argument is that if I'm using a Covad T1 and their hardware, I'm getting a certain amount of bandwith set aside for my VoIP calls and won't have those issues. Is that something that I should be concerned about? |
| 17:05.02 | AgiNamu | banner, "bandwidth set aside" |
| 17:05.35 | Delvar | bannerman: erm a T1 is ISDN? or using it as pure data? |
| 17:05.38 | ariel_ | bannerman, he is a sales man. Wants your biz they say anthing to get your biz. |
| 17:05.45 | AgiNamu | he's trying to sell you his own VoIP serviceS? |
| 17:05.54 | AgiNamu | of course it'll be "Better"... you'll be closer to them |
| 17:05.58 | jets | Ya $69.95 per line is pure rape. |
| 17:06.10 | bannerman | Right, not paying $69.95 per line. |
| 17:06.14 | Delvar | sounds leik bolox to me |
| 17:06.18 | AgiNamu | banner what? |
| 17:06.20 | AgiNamu | $70 for what? |
| 17:06.28 | calisto | delvar: do you know what ren ht286 supports cos i think that might be the issue |
| 17:06.30 | ariel_ | If I get that from my T1/pri provider I will go with another one due to they need to get it correct regarless on who's voip service you get. |
| 17:06.44 | bannerman | $69.95 for their vPBX system, which provides voicemail and all that good stuff on their side. It's pretty nice, but not for me. |
| 17:06.45 | *** join/#asterisk cervajs (~cervajs@cervajs.fpf.slu.cz) |
| 17:06.58 | AgiNamu | So do they sell just DIDs and termination? |
| 17:07.13 | AgiNamu | and what's wrong with $70 for hosted pbx? |
| 17:07.18 | bannerman | AgiNamu: I think so, but the sales rep doesn't know enough, waiting for a call back from someone who does. |
| 17:07.31 | AgiNamu | cause if they terminate for you, at a relatively good price |
| 17:07.37 | Delvar | calisto: iv got a bunch of cheap phones here and they all work fine, some ppl have said they dont seem to work well with higher end phones, id say get a cheepo phone. |
| 17:07.38 | AgiNamu | it's nice being only 1 ms or so away :) |
| 17:08.08 | AgiNamu | and that if their connections get screwed, your phone will work |
| 17:08.12 | bannerman | AgiNamu: I only need 4-5 lines at any given time, I can't afford to pay $1000/mo for something I can get from the telco for hakf that,. |
| 17:08.15 | AgiNamu | that said, if their connections are any good, any service should be ok. |
| 17:08.26 | AgiNamu | Whaer's the $100 from? |
| 17:08.37 | AgiNamu | er, $1000 |
| 17:08.39 | bannerman | I need 15 extensions. |
| 17:08.48 | AgiNamu | iut's $70 PER EXTENSION |
| 17:08.49 | AgiNamu | ? |
| 17:08.55 | AgiNamu | I'd laugh so hard at this guy |
| 17:08.58 | bannerman | According to the rep, yaeh, $70 per did |
| 17:09.01 | AgiNamu | I'd just not stop until he hung up |
| 17:09.04 | AgiNamu | hahahahah |
| 17:09.22 | AgiNamu | tell him if he even TRIES to sell you that again, you'll buy the T1 somewhere else |
| 17:09.26 | bannerman | $69.95 is great if you have 2-3 people in the office and do a ton of calling |
| 17:09.27 | ariel_ | bannerman, that is allot of money per did. |
| 17:09.27 | AgiNamu | cause $70 is farking retarded. |
| 17:09.34 | AgiNamu | I can sell em for $7 a month |
| 17:09.47 | *** join/#asterisk Tiranad (~tiranad@w034.z064000138.lax-ca.dsl.cnc.net) |
| 17:09.48 | netsurfer | wow BT want £2145 to connect 24 channels |
| 17:09.48 | AgiNamu | they pay like $0.50 a month or something |
| 17:10.03 | bannerman | it's not just for a DiD, they have a nifty software interface that allows you to route calls, music on hold, all the advanced PBX features |
| 17:10.09 | AgiNamu | even so |
| 17:10.17 | bannerman | I agree, it's too much |
| 17:10.19 | AgiNamu | I'd pay like $100 for software, then a decent price |
| 17:10.21 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 17:10.22 | AgiNamu | for DIDs |
| 17:10.30 | AgiNamu | well, I'd just insult them seriously |
| 17:10.30 | ariel_ | bannerman, use asterisk at that rate it will pay for it's self in no time. |
| 17:10.34 | Tiranad | Good Morning All |
| 17:10.43 | AgiNamu | Hi Tiranad! |
| 17:10.45 | bannerman | ariel_: That's my plan. |
| 17:11.01 | Tiranad | Anyone feeling expert on app_queue today? May have found a bug. |
| 17:11.10 | ariel_ | Besides I know that covad t1 are more expensive down here the X/O or others. |
| 17:11.19 | Delvar | aye get asterisk and an 8channel isdn |
| 17:11.41 | HjemmeRoyK | wtf is an 8chan isdn? |
| 17:11.52 | HjemmeRoyK | reduced PRI? |
| 17:11.55 | Delvar | something i just made up :) |
| 17:12.04 | Delvar | if thats what its called |
| 17:12.06 | bannerman | Delvar: never heard of such a thing, do you ... lol, you dork! |
| 17:12.25 | Delvar | hehe |
| 17:12.56 | Delvar | as long as it made you laugh its ok |
| 17:13.00 | bannerman | ariel_: I'll check into other providers, but so far, Covad is definitely the lowest. |
| 17:13.15 | ariel_ | bannerman, where are you located? |
| 17:13.22 | bannerman | North Bonneville, WA |
| 17:13.27 | bannerman | the sticks.. qq. |
| 17:13.36 | ariel_ | Ok |
| 17:13.42 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 17:13.46 | ariel_ | Your at almost the other end of the map to me. |
| 17:13.49 | calisto | delvar: if i'd wanted to buy a new phone i'd have probably gone with an IP one anyway. Guess i'll just use my current method of ring both upstairs sip and downstairs ht286 and if I hear it ring upstairs pickup phone downstairs |
| 17:14.29 | Delvar | calisto: what model phone is it you are using? |
| 17:14.48 | calisto | delvar:vtech vt650 |
| 17:14.52 | *** join/#asterisk PTG123 (Preston@ip68-106-19-249.ph.ph.cox.net) |
| 17:16.33 | rontecxt44 | hi all...thought i was on the right track with fixing my outbound calling issue...but am still missing something |
| 17:16.39 | rontecxt44 | anyone up for looking at a config? |
| 17:16.40 | Delvar | calisto: not used one, your probably right about the voltage being too low for it to detect |
| 17:16.51 | freat | good morning... ulaw, as implemented by Asterisk, has Packet Loss Concealment enabled, right? |
| 17:17.04 | AgiNamu | rontecxt44, post it and they'll come |
| 17:17.08 | ManxPower | freakin, no. |
| 17:17.12 | calisto | delvar: wonder if theres any way to boost it somehow |
| 17:17.15 | Delvar | dont post your config! lol |
| 17:17.18 | rontecxt44 | lol |
| 17:17.21 | AgiNamu | IN PASTEBIN.CA |
| 17:17.23 | rontecxt44 | i wasn't going to |
| 17:17.26 | rontecxt44 | thnx |
| 17:17.29 | AgiNamu | :) |
| 17:17.34 | blitzrage | zup |
| 17:17.40 | Delvar | calistoL like an amp? i wouldnt want to try it lol |
| 17:17.58 | Delvar | oh yes spam pastebin :) |
| 17:18.07 | freat | good morning... ulaw, as implemented by Asterisk, has Packet Loss Concealment enabled, right? |
| 17:18.09 | `Sauron | muffins! |
| 17:18.35 | *** join/#asterisk reseaux (~reseaux@host9-132.pool82105.interbusiness.it) |
| 17:18.39 | blitzrage | wish the muffins weren't virtual, I'm starving |
| 17:18.41 | Delvar | freat: duno |
| 17:18.42 | reseaux | dear all list |
| 17:18.46 | calisto | delvar: guess not should be something available though... something for people with 10 phone at home etc. |
| 17:18.53 | `Sauron | I always said the way to a man's heart went through his stomach |
| 17:19.13 | freat | you can discard the muffin stumps and just eat the tops. there's plenty |
| 17:19.33 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 17:19.38 | AgiNamu | hey... i need help |
| 17:19.44 | Delvar | calisto: if you find anything id be intrested in it. get a few ppl complaining baout it, would be nice to offer somethign to help. |
| 17:19.51 | AgiNamu | I need a name for my production server |
| 17:20.00 | freat | Wintermute |
| 17:20.02 | blitzrage | europa |
| 17:20.07 | AgiNamu | staging was easy... staging = stage = first stage = Emerald |
| 17:20.08 | Delvar | David |
| 17:20.11 | freat | hal |
| 17:20.12 | rontecxt44 | k.. |
| 17:20.12 | AgiNamu | (for those who played sonic) |
| 17:20.13 | reseaux | AgiNamu: me too.. :-) |
| 17:20.13 | rontecxt44 | http://pastebin.ca/5217 |
| 17:20.15 | zoa | freat: no |
| 17:20.22 | freat | zoa: really? |
| 17:20.28 | rontecxt44 | i borrowed various configs |
| 17:20.30 | blitzrage | zoa: hey! |
| 17:20.33 | freat | zoa: is there a way to change that? |
| 17:20.35 | AgiNamu | and something that I can add to... i..e, a set |
| 17:20.42 | zoa | freat: no |
| 17:20.50 | freat | zoa: wow huh |
| 17:20.54 | *** join/#asterisk multrix (~chatzilla@ALyon-252-1-15-228.w82-122.abo.wanadoo.fr) |
| 17:21.03 | blitzrage | zoa: what do you think is the best way to generate about 1000 simultaneous SIP calls? |
| 17:21.03 | freat | zoa: is it too much a CPU hit or what? Any idea? That's interesting |
| 17:21.23 | Delvar | blitzrage: a while bunch of .call files? |
| 17:21.33 | Delvar | whole* |
| 17:21.36 | reseaux | Help on Agi? many thz |
| 17:21.39 | blitzrage | zoa: approximately 20,000 calls an hour with about a 3 minute call time |
| 17:21.45 | freat | blitzrage: how about a really long Dial command? & & & & & & & & & & ... |
| 17:21.47 | AgiNamu | ronte, whats the problem |
| 17:21.51 | blitzrage | Delvar: I don't think that scales over a few calls |
| 17:21.56 | riksta | blitzrage: it begs the question why 1000! |
| 17:21.58 | AgiNamu | freat, but as soon as one answerd |
| 17:22.06 | blitzrage | riksta: load testing a system |
| 17:22.18 | blitzrage | zoa is the man when it comes to this :) |
| 17:22.22 | riksta | i'd say call files rather than manager |
| 17:22.24 | rontecxt44 | i can recieve inbound calls on my analog line...and answer it either with analog handset or sip phone |
| 17:22.30 | rontecxt44 | but I cannot make outgoing calls |
| 17:22.34 | rontecxt44 | i get a fast busy |
| 17:22.42 | rontecxt44 | from either handset or sip |
| 17:22.51 | blitzrage | riksta: zoa tried that, and I don't believe he had good success with over like 40-50 sim calls |
| 17:22.55 | freat | AgiNamu: well, I've been having issues with 'blank spots' in ulaw. We think it's the provider of the PRI that our VoIP provider is using. The VoIP provider is dropping them and going to move to Level 3. |
| 17:22.57 | AgiNamu | What does Asterisk console say |
| 17:23.11 | riksta | blitzrage: nothing you can do really |
| 17:23.28 | Delvar | blitzrage: iv not tried it, but your probably right. |
| 17:23.32 | rontecxt44 | it doesn't say anything when I try with sip |
| 17:23.36 | blitzrage | riksta: I don't care about easy, I can develop a system, just trying to think of a good way of going about it |
| 17:23.36 | rontecxt44 | like it is not even trying |
| 17:23.48 | zoa | blitzrage: www.astertest.com |
| 17:23.51 | zoa | go look at the forum |
| 17:23.56 | zoa | its available :) |
| 17:23.59 | blitzrage | zoa: booyah, thanks! |
| 17:24.10 | blitzrage | zoa: what kind of scaling have you got? |
| 17:24.17 | rontecxt44 | with handset...it just hangs up as soon as I hit any digits |
| 17:24.24 | zoa | depends on the servers you are using |
| 17:24.40 | zoa | and how fast you generate them |
| 17:24.53 | reseaux | dear zoa |
| 17:24.55 | AgiNamu | How can I go about programmatically collecting a dump of asterisk? |
| 17:24.56 | zoa | we are also working on load balanced call generation |
| 17:25.01 | zoa | but thats not the thing that is online |
| 17:25.03 | AgiNamu | i.e., make it so that if asterisk crashes, the crash is sent off |
| 17:25.06 | zoa | so if you need to go very high |
| 17:25.10 | zoa | thousands of channels |
| 17:25.23 | zoa | it wont do it unless you have a very fast pc for the graphs probably :) |
| 17:25.36 | blitzrage | zoa: the forums are not available? |
| 17:25.41 | zoa | sure they are |
| 17:25.42 | zoa | refresh |
| 17:25.43 | zoa | :) |
| 17:25.46 | riksta | zoa: eww its a windows app! |
| 17:25.47 | AgiNamu | zoa, do you have IAX2 testing too?> |
| 17:25.50 | blitzrage | <PROTECTED> |
| 17:25.55 | zoa | yeah its windows only |
| 17:25.59 | zoa | blitzrage: its there |
| 17:26.02 | zoa | should be there at least |
| 17:26.06 | bannerman | Why IAX instead of SIP? |
| 17:26.08 | blitzrage | I hit shift+refresh... nada |
| 17:26.11 | AgiNamu | Why IAX? |
| 17:26.14 | AgiNamu | um,sip sucks! |
| 17:26.16 | AgiNamu | NAT |
| 17:26.24 | AgiNamu | and it should be slower... all those strings |
| 17:26.29 | zoa | http://www.astertest.com/forum/ |
| 17:26.34 | zoa | go here immediately then |
| 17:26.43 | AgiNamu | wow zoa... so ... really... a Xeon machine can run a LOT of simultaneous calls eh? without transcoding? |
| 17:26.51 | AgiNamu | like 300 calls no problem eh? |
| 17:26.54 | blitzrage | ahhh, you need to change the link to that instead of /forum.htm |
| 17:26.55 | zoa | yeah without transcoding |
| 17:26.59 | zoa | 300 calls no problem |
| 17:27.05 | zoa | blitzage: it is changed afaik |
| 17:27.05 | AgiNamu | 500? 1000? |
| 17:27.12 | zoa | depends on what exactly you do |
| 17:27.13 | bannerman | Is there a downside to using an IAX provider with SIP phones? |
| 17:27.15 | zoa | try and find out |
| 17:27.17 | Delvar | depends on the speed/ram of the box |
| 17:27.19 | bannerman | SIP phones = cheap |
| 17:27.21 | zoa | speed only |
| 17:27.27 | zoa | asterisk doesnt use a lot of ram |
| 17:27.32 | AgiNamu | I got a Xeon 3.2GHz , 1MB cache |
| 17:27.35 | AgiNamu | and 2GB of RAM |
| 17:27.39 | blitzrage | zoa: oh there we go |
| 17:27.41 | AgiNamu | just a cheap server |
| 17:27.56 | blitzrage | ok... f00d time! |
| 17:28.07 | tessier_ | <PROTECTED> |
| 17:28.13 | zoa | funky |
| 17:28.25 | zoa | anyway, there is no config file online for sip |
| 17:28.30 | tessier_ | How hard would it be to include the username the person tried to register with? |
| 17:28.32 | zoa | but the iax2 should work with what is provided |
| 17:28.42 | *** join/#asterisk WildPikachu[BAR] (~wildpikac@wildpikachu.user) |
| 17:28.46 | reseaux | plz some help on FastAGI :-) |
| 17:28.46 | AgiNamu | IAX2 is probably a bit faster i'd imagine.. |
| 17:29.02 | zoa | AgiNamu: check the ppt on astertest.com |
| 17:29.09 | reseaux | I have a problem of intense use of CPU.. |
| 17:29.19 | zoa | it will show that trunked iax2 >> sip >> iax2 >> h323 |
| 17:29.24 | zoa | for cpu usage |
| 17:29.42 | AgiNamu | h323 is better? wow.l |
| 17:30.14 | WildPikachu[BAR] | how would i get the called number from zaptel device? (zaphfc) |
| 17:30.18 | WildPikachu[BAR] | < Called Number (len=10) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3498004' ] |
| 17:30.29 | WildPikachu[BAR] | i see that in the debug info, i'm after the 3498004 |
| 17:30.47 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:30.50 | Zeeek | anyone using siemens dect phones with asterisk? |
| 17:30.53 | zoa | h323 is not better |
| 17:30.55 | zoa | its worse!!! |
| 17:30.59 | AgiNamu | so if I have, say, 1000 customers, one Xeon server is more than enough (no transcoding) |
| 17:31.09 | zoa | aginamu, well you will run into other problems |
| 17:31.10 | AgiNamu | you said iax2 >> sip >> iax2 >> h323 |
| 17:31.18 | AgiNamu | IU thought it meant "greater than" CPu usaeg |
| 17:31.21 | zoa | no no |
| 17:31.23 | zoa | is faster i mean |
| 17:31.29 | AgiNamu | SIP is faster than IAX? |
| 17:31.32 | AgiNamu | how the hell is that possible? |
| 17:31.32 | zoa | yes |
| 17:31.33 | rontecxt44 | k...got distracted for a minute... |
| 17:31.37 | zoa | current implementation that is |
| 17:31.47 | rontecxt44 | AgiNamu, did you look at that config? |
| 17:31.49 | denon | course IAX != IAX2 :) |
| 17:31.50 | AgiNamu | but... IAX is all compact messages |
| 17:31.52 | Zeeek | the siemens phone is capable of SMS and alphanumeric calleridnum and name but I see neither on asterisk |
| 17:31.55 | AgiNamu | IAX2 :) |
| 17:32.17 | zoa | have a look at that call generator and find out for yourself |
| 17:32.21 | zoa | doesnt do trunked iax2 yet |
| 17:32.24 | zoa | maybe next week |
| 17:34.09 | netsurfer | Zeeek - there is certain SMS support in asterisk |
| 17:34.34 | Zeeek | I know I have it - but what I'm trying to find is why Callerid sin't showing |
| 17:34.42 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 17:35.12 | netsurfer | silly question I guess, but u tried the wiki ? |
| 17:35.34 | AgiNamu | trunked IAX2... ea... I wonder, will ztdummy be good enough? |
| 17:35.36 | Zeeek | no because it was someone here who suggested siemens |
| 17:35.54 | Zeeek | however, if it were on the wiki I'd have to post this for myself: |
| 17:36.16 | Zeeek | oops I do,'t have that macro here |
| 17:36.39 | netsurfer | Zeeek - the wiki page is quite detailed for the sms cmd |
| 17:36.45 | netsurfer | including config |
| 17:36.46 | terrapen | registration keeps failing |
| 17:36.46 | Zeeek | http://www.voip-info.org/tiki-index.php |
| 17:36.54 | Zeeek | I'm not interested in SMS |
| 17:37.06 | Zeeek | now that I got it working |
| 17:37.10 | wankel | terrapen: set the password right :) |
| 17:37.10 | Zeeek | need CID |
| 17:37.23 | terrapen | wankel: durr :) first thing i checked, mang |
| 17:37.34 | *** join/#asterisk Nix (~Nix@81.213.125.220) |
| 17:37.44 | wankel | terrapen: run ethereal and see what it sends. probably it's not using what you think. |
| 17:37.59 | wankel | i haven't used the polycom, but the cisco has phone name, phone short name, and auth name. |
| 17:38.08 | wankel | good luck guessing which one it registers with |
| 17:38.30 | mikegrb | I would guess auth name |
| 17:38.36 | wankel | i guessed that, too! |
| 17:38.41 | mikegrb | oh |
| 17:38.48 | mikegrb | I bet it does the funkyness |
| 17:38.51 | Mother_ | I think it uses auth name |
| 17:39.00 | reseaux | Im looking for some help on FastAgi plz thz |
| 17:39.02 | mikegrb | auth name is for authentication but registration uses the other name |
| 17:39.11 | wankel | yep! |
| 17:39.29 | mikegrb | what do I win? |
| 17:39.33 | Mother_ | then it must be 'name' |
| 17:39.34 | wankel | thank god for tcpdump and ethereal |
| 17:39.48 | wankel | mike: absolutely nothing, but you get to save an hour if you ever set up 7960s. |
| 17:39.48 | Mother_ | I have the extension # configured in both name and authname |
| 17:39.52 | mikegrb | do I get a cisco phone? |
| 17:40.00 | mikegrb | oh, okay ;) |
| 17:40.44 | labo | most people who configure 7960's take like, 4 5 hours |
| 17:40.49 | zoa | yeah |
| 17:40.52 | zoa | it takes ages |
| 17:40.52 | labo | not to mention the multiple sip lines problem |
| 17:40.55 | labo | yes. |
| 17:41.04 | wankel | which multiple sip lines problem? |
| 17:41.23 | Mother_ | labo: it took me just over 45 minutes (?) |
| 17:41.30 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 17:41.30 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 17:41.36 | labo | ;( |
| 17:41.38 | Mother_ | firmware upgrade and all |
| 17:41.46 | labo | took me a lot more. |
| 17:41.47 | bkw_ | I need someone to fax like 10,25,20 and 25 page documents to 877-2787565 for testing if someone can do that really fast? |
| 17:41.55 | Mother_ | there is a TFTP server for windows that helps a lot, you can see realtime what the phone is doing |
| 17:41.56 | bkw_ | oh btw HI!! |
| 17:41.57 | mikegrb | it takes bkw_ like five minutes |
| 17:41.57 | bkw_ | haha |
| 17:42.02 | Mother_ | haha |
| 17:42.11 | zoa | haha |
| 17:42.27 | mikegrb | bkw_: (time to configure cisco phone, they said like 45 minutes and hours and stuff) |
| 17:42.27 | wankel | mother: yeah, tftpd32 is handy. |
| 17:42.38 | Mother_ | that one it is :) |
| 17:42.41 | wankel | though tftpd32's dhcp server won't set the tftp server DHCP extension :( |
| 17:42.43 | reseaux | hi bk: !! |
| 17:42.50 | reseaux | hi bkw_ !! |
| 17:42.54 | Juggie | theres a cisco tftp server which works ok |
| 17:43.06 | bkw_ | hppa-tftp or some shit like that |
| 17:43.08 | bkw_ | can't stay in here long |
| 17:43.10 | zoa | brian dont forget to check out the astertest stuff |
| 17:43.11 | bkw_ | it drains my life force |
| 17:43.15 | Mother_ | wankel: will it not? I had no problems as I ran both the DHCP and TFTP using tftpd32 |
| 17:43.25 | mikegrb | bkw_: :D |
| 17:43.32 | zoa | haha lol |
| 17:43.33 | bkw_ | but if you can fax some goodies to me |
| 17:43.34 | bkw_ | hehe |
| 17:43.35 | bkw_ | bbl |
| 17:43.36 | wankel | mother: no, it won't set it via dhcp. you can override the tftp server on the phone, though. |
| 17:43.38 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 17:44.04 | wankel | or i think it defaults to the DHCP server, maybe |
| 17:44.16 | wankel | in my case the two were different |
| 17:44.18 | Mother_ | wankel: yes, I think it does |
| 17:44.37 | Mother_ | the 7905 worked fine, but the 7960 was iffy so I did as you say, set the TFTP IP fixed |
| 17:44.48 | anti | Do USB Skype phones work with asterisk? |
| 17:45.08 | Mother_ | anti: Skype is a propietary protocol |
| 17:45.42 | Juggie | the 7960 has problems with ttftp, it wont transverse subnets properly |
| 17:45.51 | Mother_ | yes, that is true |
| 17:45.54 | Juggie | so if you run your tftp on a seperate subnet the phone takes forever to boot |
| 17:45.58 | Juggie | and wont do fw upgrades |
| 17:46.24 | thieumS | is somebody available to provide me some help with sip friends registration ? |
| 17:46.26 | Mother_ | mine had problems releasing some 80.x.x.x IP it had, it kept requesting it |
| 17:46.53 | wankel | heh |
| 17:46.58 | wankel | they're a bit stubborn |
| 17:47.26 | wankel | juggie: i run my tftp server on the other side of the internet. it works fine. |
| 17:47.39 | wankel | your routing may be set up wrong |
| 17:48.22 | Juggie | the mitel works fine.... |
| 17:49.35 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 17:51.40 | postel | Mother_: you dont need a windows tftp to monitor what files the phone asks for, tail -f /var/log/tftpd would do just fine while rebooting the phones, and you dont need 45 minutes either |
| 17:52.07 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 17:52.44 | postel | edit the config with the mac address for specific phones or the global one for the universe, place the files in tftp and reboot, walk in the park |
| 17:53.24 | postel | dont even have to remove power to reboot, *+6+settings would reboot them |
| 17:53.58 | Mother_ | postel: thanks for the tip, it was my first even configuration of a Cisco phone :) |
| 17:54.03 | Mother_ | s/even/ever |
| 17:54.28 | reseaux | some help on AGI? thz |
| 17:56.22 | *** join/#asterisk CBAsteriskUK (~cblunt@208.51.30.218) |
| 17:56.38 | *** join/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 18:00.13 | terrapen | ok, im really dreading the cisco 7960 setup |
| 18:00.19 | terrapen | i can barely get through the polycom |
| 18:01.28 | labo | seems like polycom is changins its policy towards asterisk |
| 18:01.33 | labo | changing* |
| 18:01.37 | Mother_ | really? |
| 18:02.02 | labo | like, not supporting it on their latest firmwares, and not giving support by phone about * |
| 18:02.26 | redder86 | labo: did they ever? |
| 18:02.57 | labo | as soon as you mention * they "mmm" |
| 18:03.04 | redder86 | labo: to my knowledge Polycom has always been antagonistic towards Asterisk. |
| 18:03.09 | terrapen | i couldn't get the latest firmware to load on my brand new IP600 |
| 18:03.12 | terrapen | not sure why... |
| 18:03.19 | terrapen | had to go to an older version |
| 18:03.42 | *** join/#asterisk Syrus_ (~pascal@tahiti.mpl.rullier.net) |
| 18:03.52 | Mother_ | when I called Polycom and mentioned asterisk they went "what is that?" |
| 18:03.54 | terrapen | maybe my phone did not have sufficient flash space for the new image |
| 18:03.55 | redder86 | labo: but SIP is SIP, so if they intend to support SIP then they will work with Asterisk |
| 18:04.09 | Mother_ | and then started the babble about their phones being certified with certain PBX etc. etc. |
| 18:04.16 | ManxPower | Polycom refuses to deal with any PBX except the ones they certify. |
| 18:04.33 | ManxPower | they also don't deal with end users, all support is supposed to go thru your reseller |
| 18:04.39 | Mother_ | ManxPower: exactly what they told me, and also that I had to get certified with them to buy their phones |
| 18:04.53 | Mother_ | so I told them to go someplace not nice |
| 18:04.55 | ManxPower | Mother_, Correct. Makes sense. |
| 18:06.12 | redder86 | Polycom are nice phones, but don't buy Polycom expecting them to help you at all. |
| 18:06.28 | ManxPower | redder86, The same can be said about Cisco |
| 18:06.35 | redder86 | ManxPower: yup |
| 18:06.39 | ManxPower | And to a lesser extent, SIPura |
| 18:07.39 | redder86 | well, in most cases where products are only sold through "authorized" channels, manufacturers generally will expect end-users to turn to the vendors for support. |
| 18:08.54 | Mother_ | redder86: it's all a policy to save money on support infrastructure |
| 18:09.12 | labo | its like those supermarkets that require an id for you to buy from them. |
| 18:09.35 | Mother_ | you sell the phones, earn a meager margin, then have to deal with all the end user's issues |
| 18:10.03 | Mother_ | oh, and you get a nice 'partner' sticker to put on your bussiness front door |
| 18:13.16 | postel | ManxPower: we;;, cisco never said anything bout other pbxs, the sell their phones to work with CCM, and yes you need a support contract |
| 18:13.52 | redder86 | Mother_: the VARs are *supposed* to use the authorization to allow them to "get their foot in the door" for more lucrative sales |
| 18:14.33 | redder86 | Mother_: so your customers may come to you first for Polycom phones, but ultimately you're supposed to make your money in other ways... like servicing their PBX or whatever. |
| 18:15.27 | redder86 | Mother_: but with the Asterisk community, most users are do-it-yourselfers, and so the phones and other hardware turn into a commodity... except for a rare few users |
| 18:16.32 | terrapen | argh |
| 18:17.02 | terrapen | i have asterisk configured to put this phone in the 'outgoing' context: |
| 18:17.02 | terrapen | [poly1] |
| 18:17.02 | terrapen | type=friend |
| 18:17.02 | terrapen | context=outgoing |
| 18:17.18 | terrapen | but the damned thing keeps trying to go to the 'default' context |
| 18:17.22 | terrapen | nothing about this in the wiki |
| 18:17.31 | terrapen | *CLI> Feb 3 18:12:19 NOTICE[28996]: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' |
| 18:18.42 | terrapen | i'd love to see somebody's working polycom configs |
| 18:19.00 | terrapen | and not just the phone configs, the relevant * sip.conf bits |
| 18:19.06 | tessier_ | I'd love to see a hot chicks tits right about now. |
| 18:19.19 | tessier_ | But I don't know if either of us will get our wish. |
| 18:19.31 | tessier_ | Actually, I could just go to the titty bar for lunch I suppose. |
| 18:19.33 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 18:19.39 | tessier_ | You, on the other hand, may be SOL. |
| 18:19.48 | tessier_ | Or you could try the wiki for polycom configs. |
| 18:19.55 | bjohnson | I have developed a problem while playing with spa 3k config options .. I can't dial out. It acts as though it goes off hook .. and then just sits there until I get a pstn error message |
| 18:20.25 | ManxPower | terrapen, That doesn't look like a polycom problem. that looks like a asterisk config problem |
| 18:21.44 | *** join/#asterisk tzafrir_home (~chatzilla@bzq-179-40-134.cust.bezeqint.net) |
| 18:21.47 | bjohnson | to me seems like a codec or a dtmf issue. Do others think same thing? |
| 18:22.57 | terrapen | yep, Manx |
| 18:23.11 | terrapen | in my Line 1 settings on the phone, i have: |
| 18:23.15 | terrapen | Display name: outgoing |
| 18:23.19 | terrapen | Address: outgoing |
| 18:23.23 | terrapen | Label: outgoing |
| 18:23.26 | terrapen | is that wrong? |
| 18:23.37 | terrapen | (if i want line 1 in the outgoing context? |
| 18:23.47 | *** part/#asterisk CBAsteriskUK (~cblunt@208.51.30.218) |
| 18:26.52 | reseaux | :-) some on AGI thanks :-) |
| 18:27.03 | Mother_ | redder86: agreed |
| 18:27.17 | terrapen | holy cow, the damned phone just rebooted spontaneously...grreaatt |
| 18:27.49 | terrapen | ooo ooo |
| 18:27.53 | terrapen | <PROTECTED> |
| 18:27.58 | terrapen | maybe that's a good sign |
| 18:28.02 | Mother_ | postel, you work for Cisco? I have a question if you do |
| 18:28.09 | Mother_ | about something I was told the other day |
| 18:28.19 | *** join/#asterisk BozzaCamilleri (~connect@213.217.225.32) |
| 18:28.34 | labo | i have a person from cisco in front of me, hes ccie, what question |
| 18:28.48 | postel | Mother_: no, i work for Sun as an architect, but i use cisco eq daily |
| 18:29.01 | Mother_ | do they really require you to give them your client's contact info before they sell you VoIP phones? |
| 18:29.12 | Mother_ | like company name, contact name, phone number, etc. |
| 18:29.21 | Mother_ | postel: OK thanks :) |
| 18:29.32 | labo | No, they ask you if you have a PICA account, or a CCO |
| 18:29.45 | labo | Pica is like cco but without having the possbilility to get quotations. |
| 18:29.54 | Mother_ | labo: OK that clears it up |
| 18:30.02 | postel | Mother_: no, you can get partner status and byw ALL the phones they would ever make, no questions asked |
| 18:30.30 | Mother_ | postel: I got this from one of their distributors (Ingram) |
| 18:30.51 | labo | buy them on ebay, no questions :) |
| 18:30.55 | Mother_ | and didn't think them having my client's info would serve any good purpose |
| 18:31.11 | *** join/#asterisk Meznev (~Elshar@ip205-68.oregonfast.net) |
| 18:31.18 | Mother_ | labo: hahaha yes - I got them from another distributor who didn't want me to submit DNA samples too |
| 18:31.30 | labo | heh |
| 18:31.31 | postel | Mother_: well, for the end home user messing around, ebay should be a better answer |
| 18:32.08 | postel | for the corp deploying a solution its a whole different story |
| 18:32.15 | bjohnson | note for future: urinating on the requestor's shoes DOES count as giving a dna sample |
| 18:32.22 | Mother_ | postel: agreed, I want to start building this into a bussiness, so if all goes well I'll be buying quite a few phones soon, so that's what worried me |
| 18:32.30 | *** join/#asterisk t3t (~t3t@cust018.mke.attron.net) |
| 18:33.05 | Mother_ | postel: give them all your client's info, then they come visit them with their good better buddy partners and take your costumer away |
| 18:33.46 | Mother_ | in other countries bussiness is more civilised, but here (Spain) your client's details are your most precious assets |
| 18:33.51 | KalD|Work | is there a meeting thing for linux that works w/ exchange? |
| 18:34.04 | Mother_ | so you do NOT go around giving them to people that easily |
| 18:34.29 | KalD|Work | I'm thinking about writing a meeting plugin for asterisk that would call me based on my meeting schedule and remind me to show up for company meetings =) |
| 18:34.53 | postel | get an alarm clock |
| 18:34.56 | tzafrir_home | KalD|Work, using connector? |
| 18:35.48 | tzafrir_home | I know it has some support outside evolution |
| 18:36.05 | Mother_ | LOL! |
| 18:36.05 | KalD|Work | tzafrir_home, connector? oh yeah - well kinda.. only I'd speak MAPI |
| 18:36.11 | tzafrir_home | I also saw some parsers for the schedualing mail it sends |
| 18:36.22 | *** part/#asterisk rontecxt44 (~rontecxt4@dsl5-66.rb.comporium.net) |
| 18:36.26 | KalD|Work | anyone interested in trying it when I'm done? |
| 18:36.48 | tzafrir_home | You can use them to create callfiles |
| 18:37.03 | Mother_ | are the spandsp pre releases stable enough to use? or just stick with 0.0.1? |
| 18:37.39 | tzafrir_home | Mother_, I saw recomendations to use the pres. our package has pre10 |
| 18:37.52 | Mother_ | tzafrir: thanks a lot |
| 18:39.31 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 18:39.50 | ManxPower | Mother_, the 0.0.2 works with CVS-HEAD, the 0.0.1 works with CVS 1.0.x stable |
| 18:41.07 | mtqh | how stable is spandsp ? |
| 18:41.15 | Mother_ | ManxPower: thanks, I'm using CVS-HEAD so I'll try 0.0.2 and see how it goes |
| 18:41.26 | _Brian | is there a way to display which codec is being utilized on a call (using sip) |
| 18:42.20 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 18:42.57 | WildPikachu[BAR] | how can i get the called number into my dialplan? |
| 18:43.23 | mtqh | ${EXTEN} |
| 18:43.33 | WildPikachu[BAR] | um? |
| 18:43.43 | WildPikachu[BAR] | aha |
| 18:44.11 | greg_work | _Brian: sip show channels in the console |
| 18:44.53 | WildPikachu[BAR] | hrmmm |
| 18:45.05 | WildPikachu[BAR] | in the config examples i think i only see dialing examples (extensions.conf.sample) |
| 18:45.51 | _Brian | greg_work: Doh!! thanks :) |
| 18:46.12 | *** join/#asterisk PBXtech (~nik@67.51.185.20) |
| 18:46.28 | PBXtech | can you not pass CID through broadvoice anymore? |
| 18:46.58 | bjohnson | _Brian: sometimes the device will also display it |
| 18:47.20 | WildPikachu[BAR] | can i use ${EXTEN} in context= ? |
| 18:47.41 | bjohnson | WildPikachu: in sip.conf? I doubt it |
| 18:47.52 | WildPikachu[BAR] | nope... zapata.conf |
| 18:48.08 | bjohnson | I doubt that too |
| 18:48.19 | WildPikachu[BAR] | see... i have 8 numbers mapped to 1 phone line |
| 18:48.27 | WildPikachu[BAR] | if i enable debuggin in pri, i see the called number |
| 18:48.36 | `Sauron | ManxPower: They made me grumpy like you, earlier today. |
| 18:48.40 | WildPikachu[BAR] | now... i want diff voice prompts depending on which number is dialed |
| 18:48.43 | WildPikachu[BAR] | by a client :) |
| 18:49.07 | Delvar | isnt it somethin glike exten => number|_XXX.,1,dial() ? im jsut guessing |
| 18:49.19 | `Sauron | So, I know * has the soft-fax-abillity (through the extra module) |
| 18:49.24 | bjohnson | probably possible .. but not in zapata.conf |
| 18:49.45 | `Sauron | Does anyone know if there's a pty emulator module, so you could route modem calls through * to something that can set up a PPP connection? |
| 18:49.48 | WildPikachu[BAR] | Delvar, talking to me? :) |
| 18:50.26 | bjohnson | `Sauron: I was told last night that * can route a modem call through to a fxs port that has a modem plugged into it |
| 18:50.28 | greg_work | WildPikachu: http://voip-info.org/wiki-Asterisk+tips+DID |
| 18:50.34 | `Sauron | Hrm. |
| 18:50.36 | bjohnson | however .. 1. I haven't tried it yet |
| 18:50.39 | `Sauron | That's not really what I want. |
| 18:50.40 | mtqh | yes It can |
| 18:50.49 | `Sauron | Cuz, there's potential for 23 modem calls |
| 18:50.49 | bjohnson | 2. * cannot auto-sense that it is a data call |
| 18:51.01 | `Sauron | bjohnson: 2) isn't a problem. |
| 18:51.15 | `Sauron | I designed 2) away. |
| 18:51.28 | greg_work | WildPikachu: i believe you do context=from-outside in your zaptel.conf, then in extensions: [from-outside] exten=>5551234,1,Goto(voice-promt-1,s,1) exten=>5551235,1,Goto(voice-promt-1,s,2) |
| 18:51.28 | greg_work | etc |
| 18:51.31 | bjohnson | I don't think that you can do it without a fxs and a hardware modem |
| 18:51.36 | `Sauron | blah |
| 18:51.57 | `Sauron | they have fax receiver crap |
| 18:52.07 | `Sauron | why don't they have it for modem as well |
| 18:52.19 | greg_work | `Sauron: why not just run mgetty? |
| 18:52.20 | PBXtech | can you not pass CID through broadvoice anymore? |
| 18:52.31 | `Sauron | greg: Against what TTY? |
| 18:52.38 | greg_work | oh, to setup ppp |
| 18:52.46 | `Sauron | setup ppp |
| 18:52.47 | greg_work | hm, i dunno. i've never done that before |
| 18:52.49 | `Sauron | yeah |
| 18:53.07 | greg_work | how does pppd normally interface with incoming calls? |
| 18:53.10 | WildPikachu[BAR] | yea!!!! |
| 18:53.18 | `Sauron | you run pppd against /dev/ttyS0 |
| 18:53.18 | WildPikachu[BAR] | overlapdial=yes & immediate=no |
| 18:53.21 | WildPikachu[BAR] | thanks guys! |
| 18:53.24 | bjohnson | mgetty can answer and start ppp commands .. but I haven't seen a config that shares a line with * |
| 18:53.25 | Delvar | http://voip-info.org/wiki-Asterisk+config+extensions.conf |
| 18:53.27 | `Sauron | or any other pty/tty device node |
| 18:53.33 | Delvar | exten => s/9184238080,2,SetCIDName(EVIL BASTARD) |
| 18:53.41 | Delvar | might be what you are looking for? |
| 18:53.55 | `Sauron | asterisk's pppd command only works with isdn/hdlc |
| 18:53.58 | `Sauron | not analog modems |
| 18:54.01 | Delvar | for specific caller id's do something differnt |
| 18:54.45 | bjohnson | looks to me like my spa 3k is picking up the line but the numbers to dial aren't getting to the pstn .. is there a setting for this? |
| 18:54.51 | greg_work | `Sauron: can you just not tell * about the other ports, and not tell pppd/mgetty/whatever about the ports * is using? |
| 18:55.17 | `Sauron | greg: Heh. The point here is to get rid of T1's |
| 18:55.27 | `Sauron | currently, we have a fax/dialup server with a dual port PRI card |
| 18:55.31 | *** join/#asterisk okieplaya (~okieplaya@cdm-208-180-154-4.slsp.cox-internet.com) |
| 18:55.42 | `Sauron | one T1 is for modem dialups, and runs against mgetty/pppd |
| 18:55.55 | `Sauron | the other T1 is for fax calls, and runs against mgetty-sendfax |
| 18:56.20 | `Sauron | using * to route the calls to fax/modem is easy, since we can do it based on DID |
| 18:56.26 | *** join/#asterisk mogorman (~mogorman@dhcp-162.digium.com) |
| 18:56.27 | okieplaya | anyone help me with bellster? |
| 18:56.29 | `Sauron | and * can do the fax -> email stuff easily |
| 18:56.39 | bjohnson | `Sauron: let me know if you find something .. I was surprised to hear a modem would work through * on a fxs port |
| 18:56.44 | `Sauron | but I'm looking to solving the modem dialup side |
| 18:56.59 | `Sauron | bjohnson: I'd be surprised if it wouldn't work on an fxs |
| 18:57.00 | Juggie | bjohnson, why were you suprised? |
| 18:57.05 | greg_work | `Sauron: aside, did you get rxfax() working ok? what version of *? |
| 18:57.08 | `Sauron | okieplaya: It's called FWDout.net |
| 18:57.27 | `Sauron | greg: I haven't played with rxfax yet - once the project gets approved in 2006, I'll play with it. |
| 18:57.28 | Juggie | bjohnson, g711 is the same compression used everytime you call long distance |
| 18:57.32 | Juggie | and probally even locally. |
| 18:57.35 | `Sauron | Well, I'll play with it at home before then |
| 18:57.45 | bjohnson | Juggie: because I read so much about faxes being a problem .. I figured data calls would be worse since lower interest in finding a solution |
| 18:57.57 | greg_work | `Sauron: how many fax/modem ports do you need? |
| 18:57.57 | reseaux | Some help on AGI? |
| 18:58.02 | `Sauron | well, approved "next fiscal year" - so 2nd half 2005 |
| 18:58.08 | `Sauron | greg: 24 |
| 18:58.12 | Juggie | bjohnson, g711 wont loose any of the data for modems or faxes |
| 18:58.15 | Juggie | the problem is latency |
| 18:58.17 | `Sauron | T1 worth |
| 18:58.27 | greg_work | `Sauron: so you're just trying to eliminate one T1 |
| 18:58.32 | bjohnson | Juggie: I read there was a problem .. regardless of cause |
| 18:58.35 | okieplaya | ok can some help me fwdout.net |
| 18:58.37 | okieplaya | ? |
| 18:58.45 | `Sauron | Basically |
| 18:58.55 | Juggie | bjohnson, the problem is that the latency of a ip network is much greater then the latency of the TDM network |
| 18:59.06 | Juggie | fax machines get impatient and time out waiting to hear back from the other end |
| 18:59.16 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.rr.com) |
| 18:59.18 | florz | Juggie: Nope. |
| 18:59.30 | Juggie | florz, then enlighten me. |
| 18:59.45 | florz | Juggie: Latency isn't much of a problem - jitter is the real problem. |
| 19:00.07 | Juggie | florz, and what causes jitter? |
| 19:00.13 | Juggie | (latency) |
| 19:00.14 | *** join/#asterisk Nohair (~jt@srscomp.demon.co.uk) |
| 19:00.15 | florz | Juggie: _changing_ latency |
| 19:00.16 | greg_work | unsynchronized clocks |
| 19:00.19 | Juggie | exactally. |
| 19:00.29 | Juggie | so my answer still stands, network latency causes problems |
| 19:00.32 | florz | Juggie: Nope |
| 19:00.32 | Delvar | Trenips! |
| 19:00.51 | florz | Juggie: It's not the latency that causes problems, just changes in the latency. |
| 19:01.15 | bjohnson | damn .. pstn dialing delay did it !! :) |
| 19:01.26 | Juggie | florz, changes in the latency shoudnt be a problem if your jitter buffer can handle it its when you have to drop a packet that it becomes an issue |
| 19:01.58 | bjohnson | Juggie: and the answer to the question "Can * terminate data calls?"? |
| 19:02.06 | Juggie | bjohnson, yes |
| 19:02.12 | `Sauron | How? |
| 19:02.12 | Juggie | you will get at most 28.8 though |
| 19:02.15 | bjohnson | how |
| 19:02.17 | Juggie | maybe 33.6 |
| 19:02.18 | Juggie | nothing higher |
| 19:02.20 | mikegrb | Juggie: it's the jitter |
| 19:02.22 | bjohnson | good enough |
| 19:02.39 | Juggie | bjohnson, thats on a LAN though. |
| 19:02.43 | *** join/#asterisk Yoda-BZH (~yoda-bzh@213.223.52.79) |
| 19:02.45 | greg_work | explaination of jitter/frame slip and faxes: http://opencall.org/faq/x47.html |
| 19:03.03 | Juggie | across the internet, i doubt you would get much satisfaction |
| 19:03.22 | greg_work | `Sauron: what are you running a pppd dialup server for anyways? |
| 19:03.30 | `Sauron | Juggie: _how_ do you terminate data calls through *? |
| 19:03.38 | bjohnson | Juggie: how can * terminate a dial in data call? |
| 19:03.38 | reseaux | some help on AGI? plz thz |
| 19:03.45 | `Sauron | Don't worry yourself with jitter, bandwidth, etc - that's my problem. |
| 19:04.04 | `Sauron | greg: Our users still use it |
| 19:04.06 | bjohnson | in my case would be coming straight in from pstn |
| 19:04.14 | terrapen | if i don't get some chapstick soon, i may die |
| 19:04.15 | Juggie | bjohnson, an ATA with your modem hoooked into it. |
| 19:04.16 | bjohnson | mine too |
| 19:04.22 | `Sauron | Argh. |
| 19:04.26 | `Sauron | That's a horseshit answer. |
| 19:04.33 | Nohair | any one using an iaxy ?? |
| 19:04.44 | BoRiS | * with data(modem)? lol!!!! good luck |
| 19:04.53 | `Sauron | So you're telling me I need 24 FXS devices, and 24 external modems? |
| 19:04.56 | `Sauron | Bwahaha |
| 19:05.09 | Juggie | BoRiS, across a lan it would manage 28.8/33.6 |
| 19:05.14 | greg_work | `Sauron: is it direct access to your network, or internet access? why not outsource it, use VPN? or even get a wholesale dialup provider account and you may be able to do the same thing |
| 19:05.15 | `Sauron | bjohnson: I think the real answer is "* can not terminate data calls" |
| 19:05.27 | Juggie | `Sauron, what are you trying to accomplish |
| 19:05.29 | `Sauron | greg: direct access |
| 19:05.41 | `Sauron | Juggie: read scrollback. |
| 19:05.45 | greg_work | `Sauron: whats wrong with putting a T1 directly to your modem box? |
| 19:05.46 | ManxPower | Asterisk can terminate ISDN 64K data calls using ZapRAS. Asterisk can terminate FAX calls using spandsp. Asterisk cannot terminate MODEM calls. Asterisk can pass them thru the system just like any other voice call. |
| 19:05.50 | greg_work | from * i mean |
| 19:05.54 | `Sauron | We have VPN, but there are people who can't run that |
| 19:05.55 | greg_work | configure it as fxs |
| 19:06.02 | `Sauron | greg: Because now you need 2 servers |
| 19:06.14 | Juggie | `Sauron, asterisk isnt a dialup server. |
| 19:06.15 | greg_work | `Sauron: tell them you don't support 5-year-old OS's anymore ;) |
| 19:06.16 | `Sauron | 1. the current PRI card is out of warranty, end of life, and end of service |
| 19:06.44 | greg_work | yeah, i'm really confused what * has to do with anything hten |
| 19:06.49 | `Sauron | 2. * can easily take over the fax server side |
| 19:06.56 | greg_work | mgetty+faxgetty should solve everything |
| 19:06.59 | `Sauron | 3. It would be nice if * could take over the dialup side as well |
| 19:06.59 | bjohnson | ManxPower: I think that's what I answered right at the beginning |
| 19:07.23 | greg_work | if anything, you should be looking into handling DID numbers in the mgetty+faxgetty package (if it doesn't handle it already) |
| 19:07.29 | `Sauron | Manx: I realise what you just said. Juggie kept claiming otherwise. |
| 19:07.37 | `Sauron | blah |
| 19:07.46 | `Sauron | don't y'all bother your brains |
| 19:08.02 | ManxPower | Juggie is wrong then. |
| 19:08.03 | *** join/#asterisk clive- (~pirch@rrba-146-118-73.telkomadsl.co.za) |
| 19:08.04 | Juggie | `Sauron, i said with an ATA you could pass a modem call through, not that * was a modem server. |
| 19:08.24 | bjohnson | in my case, I have a few employees that take advantage of free internet access by dialing into the office in the evenings .. I need a method to share the lines between fax/data/phone calls |
| 19:08.26 | `Sauron | Juggie: both me and bjohnson asked you "Can * terminate a data call?" |
| 19:08.28 | `Sauron | You said yes. |
| 19:08.38 | `Sauron | Quit backpedaling. |
| 19:08.50 | bjohnson | actually, I said you couldn't and he said you could |
| 19:08.57 | `Sauron | IT's a moot point |
| 19:09.05 | `Sauron | so don't you worry your little brain anymore |
| 19:09.13 | Juggie | `Sauron, i ment it can pass data to an ATA or whatever... if you are going to be an ass, just go. |
| 19:09.21 | bjohnson | `Sauron: end result .. still need modem hardware |
| 19:09.29 | Juggie | we dont have your modem hardware solution here. |
| 19:09.30 | terrapen | holy shite...im getting somewhere with the polycom... |
| 19:09.39 | terrapen | i'm gonna write this up on the wiki when i'm done |
| 19:09.40 | `Sauron | yawn |
| 19:09.53 | terrapen | nothing worse than documentation that is insufficient to get something working |
| 19:09.58 | bjohnson | hell .. I was excited that * could terminate data calls |
| 19:10.06 | bjohnson | what a let down |
| 19:10.12 | Juggie | bjohnson, a fax is a datacall. |
| 19:10.33 | Juggie | so it can terminate a data call, just not a modem call. |
| 19:10.37 | florz | ... and a ZapRAS connection is, too, isn't it? |
| 19:10.45 | Juggie | yes. |
| 19:10.47 | Juggie | thats ISDN. |
| 19:10.48 | terrapen | why would you want * to terminate a data (modem) call? |
| 19:10.58 | florz | Juggie: Fax is modem, isn't it? =:-) |
| 19:10.59 | terrapen | so many better solutions for that |
| 19:11.00 | Juggie | terrapen, thats a good question |
| 19:11.06 | Juggie | florz, yes it certainly is :) |
| 19:11.13 | bjohnson | terrapen: in my case, I have a few employees that take advantage of free internet access by dialing into the office in the evenings .. I need a method to share the lines between fax/data/phone calls |
| 19:11.13 | terrapen | use the best tools for the job |
| 19:11.27 | bjohnson | what ARE the best tools |
| 19:11.31 | Juggie | bjohnson, so you already have a dialup server? |
| 19:11.33 | terrapen | bj, you know what that reminds me of, bro... |
| 19:11.41 | florz | bjohnson: What kind of lines? |
| 19:11.41 | bjohnson | a small office? |
| 19:11.42 | terrapen | when i was 13, i ran a BBS |
| 19:11.50 | bjohnson | 3 analog lines |
| 19:11.51 | terrapen | (this is 1987) |
| 19:11.54 | bjohnson | 1 has a modem |
| 19:12.00 | terrapen | or maybe 1989...anyway |
| 19:12.08 | terrapen | i had a 286 with a 40Mb RLL disk |
| 19:12.18 | Juggie | terrapen, i ran one in like 93-95 |
| 19:12.21 | *** join/#asterisk mozrat (~mozrat@host81-130-140-197.in-addr.btopenworld.com) |
| 19:12.22 | terrapen | and for christmas, my dad got me a 340Mb western digital IDE drive |
| 19:12.35 | terrapen | and i tried for DAYS to make the RLL drive work alongside the IDE drive |
| 19:12.45 | `Sauron | greg, msgme your fwd number and I'll call you when I'm home for lunch and I can tell you what I'm trying to do. :p |
| 19:12.45 | terrapen | i even bought a special controller that claimed to make it work |
| 19:12.52 | terrapen | and spent hours on tech support |
| 19:12.53 | *** join/#asterisk DaLion2 (anon@Toronto-HSE-ppp3884464.sympatico.ca) |
| 19:13.04 | bjohnson | I suppose I should tell staff that I'm sorry to stop providing free internet access .. the super duper equipment I've been promoting .. just can't deal with that complexity |
| 19:13.05 | terrapen | in the end, i finally realized that is was a waste of time |
| 19:13.35 | Juggie | bj, with your 3 lines... after office hours, what exactally do you want to provide? |
| 19:13.36 | mozrat | Hello. Could some kind patient person help me get my n00b asterisk system working. I've set it up and defined a sip user and extension, and tried to connect using x-lite. I can see UDP sip datagrams sent from client to server but the server isn't playing |
| 19:13.38 | florz | bjohnson: What is your setup like right now? |
| 19:13.43 | redder86 | bjohnson: won't it work through TDM connections? |
| 19:13.48 | terrapen | bj: it's a bit extreme to expect a VoIP PBX to handle data calls, IMHO |
| 19:13.51 | bjohnson | tdm connections? |
| 19:13.53 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 19:14.11 | redder86 | terrapen: you can get an old AT/XT to run IDE. |
| 19:14.15 | florz | terrapen: After all, it does, no problem. |
| 19:14.15 | bjohnson | terrapen: umm .. I said it wasn't possible and was corrected. I'm still waiting for an answer |
| 19:14.29 | terrapen | moz, are you just trying to get a softphone working? if so, i have a better suggestion |
| 19:14.34 | Blackthorn | Could someone point me in the direction on how to setup voice menus? and how do you record a message into * that would be used as a menu? |
| 19:14.42 | Juggie | bjohnson, i specified you could do it with a modem and pass the signal through * not that * could act as a modem. |
| 19:14.46 | bjohnson | redder86: what'sTDM? |
| 19:14.54 | mozrat | terrapen: well the idea is to learn about Asterisk so I can use it properly in the future with hardware phones |
| 19:14.57 | redder86 | bjohnson: yeah, just run your "dialup DIDs" to TDM connections, not VoIP. |
| 19:15.05 | terrapen | redder, the machine ran the IDE fine...but it was just about impossible to get the RLL drive to work |
| 19:15.06 | redder86 | bjohnson: TDM card |
| 19:15.11 | bjohnson | redder86: I have 3 lines |
| 19:15.16 | terrapen | the point of the story is that sometimes, some wars aren't worth fighting |
| 19:15.41 | terrapen | i was pulling my hair out for days to save 40Mb! |
| 19:15.47 | Juggie | bjohnson, you have one dialup line now, what do you want to do, add another? |
| 19:15.49 | terrapen | (those were the days) |
| 19:15.53 | redder86 | bjohnson: how does the current system distinguish between data and voice calls? |
| 19:16.01 | terrapen | and now i have 200Gb free on my desktop computer |
| 19:16.05 | Juggie | you only have 3, you are going to have to keep one clear for emergency calls. |
| 19:16.12 | bjohnson | so the end of the story .. like at the beginning .. is that * CANNOT terminate calls via software .. it still requires modem hardware |
| 19:16.23 | bjohnson | redder86: a voice/fax/data switch |
| 19:16.31 | Juggie | bjohnson, correct.... at the moment * has no modem emulator. |
| 19:16.34 | bjohnson | Juggie: no. just keep it |
| 19:16.36 | redder86 | bjohnson: there is no softmodem in Asterisk yet |
| 19:16.53 | Mother_ | well the spandsp patch failed... |
| 19:16.55 | Juggie | you can pass modem calls, but you cant answer them. |
| 19:16.57 | terrapen | yeah, what if he put on of those voice/fax/data switches in front of his FXOs? |
| 19:16.58 | bjohnson | `Sauron wants to do the same thing with 23 lines |
| 19:17.04 | redder86 | bjohnson: what's wrong with just using that voice/fax/data switch still? |
| 19:17.09 | Mother_ | any pointers to adding the patch by hand? |
| 19:17.10 | Juggie | it would be ideal if you could answer them then pipe the output/input to PPP on a linux box. |
| 19:17.11 | Blackthorn | Could someone point me in the direction on how to setup voice menus? and how do you record a message into * that would be used as a menu? |
| 19:17.13 | Juggie | but we arnt there yet. |
| 19:17.24 | bjohnson | redder86: I've been posting for weeks for help with getting it to work with my hardware |
| 19:17.29 | Nohair | Any one using an Iaxy in the UK?? |
| 19:17.33 | terrapen | blackthorn, check out the Wiki |
| 19:17.52 | redder86 | bjohnson: what is it? a proprietary switch? |
| 19:17.58 | Blackthorn | terrapen: what is wiki |
| 19:18.05 | bjohnson | Bell supplied Ultraswitch 100 |
| 19:18.16 | redder86 | doesn't it just let voice calls go through one port, data thorugh another, fax through another? |
| 19:18.25 | terrapen | http://www.voip-info.org/tiki-index.php?page=Asterisk |
| 19:18.28 | terrapen | there. |
| 19:18.29 | bjohnson | redder86: one line in, 4 lines out (fax, modem, phone, and TAO) |
| 19:18.36 | Blackthorn | terra: thank you |
| 19:18.39 | bjohnson | redder86: yes |
| 19:19.02 | bjohnson | redder86: had a problem with callerid getting through to my SPA 3k without answering it first .. fixed that |
| 19:19.05 | redder86 | bjohnson: so plug your phone port on that into your Asterisk-connected FXO and be done with it |
| 19:19.12 | bjohnson | redder86: but I couldn't get the spa to dial out of it |
| 19:19.31 | bjohnson | redder86: replaced the spa with a x100p .. could dial out .. but couldn't get the callerid |
| 19:19.35 | bjohnson | see the circle |
| 19:20.05 | redder86 | bjohnson: does the x100p get caller*id without the switch in the way? |
| 19:20.14 | terrapen | Feb 3 19:15:57 WARNING[2379]: app_dial.c:1007 dial_exec: Had to drop call because I couldn't make SIP/poly1-e651 compatible with IAX2/NuFone/1 |
| 19:20.21 | terrapen | i guess i need to read up on transcoding |
| 19:20.22 | terrapen | or something |
| 19:20.24 | bjohnson | redder86: yes |
| 19:20.30 | DaLion2 | whois Dalion |
| 19:20.32 | *** join/#asterisk machinehd (~machinehd@storm.bcgroup.net) |
| 19:20.36 | DaLion2 | ~seen DaLion |
| 19:20.38 | jbot | dalion <anon@HSE-QuebecCity-ppp3497095.sympatico.ca> was last seen on IRC in channel #asterisk, 14h 58m 50s ago, saying: 'that would wotk'. |
| 19:20.38 | bjohnson | redder86: I think I have it working with the spa again .. now testing |
| 19:20.47 | DaLion2 | darn |
| 19:21.08 | redder86 | bjohnson: you shouldn't be surprised, though, you're probably the only Asterisk user trying to do this with that equipment in this way. |
| 19:21.27 | machinehd | getting a new server for * Dual Xeon or just a regular p4 ? |
| 19:21.35 | bjohnson | redder86: but you can maybe now understand my excitment when Juggie led BOTH `Sauron and I to believe that the modem hardware was not required (regardless of what info he intended to convey) |
| 19:21.39 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 19:21.45 | terrapen | machine: is that a question? |
| 19:21.55 | redder86 | bjohnson: I came into the conversation late. |
| 19:21.58 | bjohnson | redder86: according to the last few minutes .. I am not |
| 19:22.01 | machinehd | terrapen, yes in fact :) |
| 19:22.37 | bjohnson | redder86: at least not the only one with the goal of sharing a line for data/fax/phone calls |
| 19:22.39 | redder86 | bjohnson: there is some work by Steve Underwood to get Asterisk to do soft-modem emulation for fax modems (not data) |
| 19:23.09 | bjohnson | redder86: yes I know .. but if I need the external switch anyway for the modem calls .. the fax is not really the issue |
| 19:23.11 | Juggie | bjohnson, that was a misunderstanding, i simply ment asterisk COULD answer a data call, and then pass it off to an ATA or through another TDM port or the like.... i did not intend to imply that asterisk could ANSWER a modem call, all be it,. it can answer ISDN and FAX (both data calls) |
| 19:23.17 | redder86 | bjohnson: I share a line with voice/fax no problems, but putting fax through VoIP is not an ideal thing to do. |
| 19:23.47 | terrapen | what is the proper spelling of "homeez"? |
| 19:23.53 | terrapen | "homies"? |
| 19:24.09 | shido6 | are you writing a police report? |
| 19:24.17 | terrapen | hahah |
| 19:24.19 | bjohnson | check your spelling of that slang |
| 19:24.37 | terrapen | rofl shido |
| 19:24.39 | eKo1 | Its h0m33z |
| 19:24.46 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 19:24.57 | terrapen | h0meZZ if you are in East LA |
| 19:25.08 | Blackthorn | from the reading i just did it seems that in order to do menus you need to pre-record as a .gsm file. What program (say in windows) can record this type of file? and what directory do you drop your .gsm files into? |
| 19:25.20 | terrapen | get you Audacity |
| 19:25.25 | cypromis | make it 8hz wav |
| 19:25.25 | terrapen | record in WAV |
| 19:25.28 | *** part/#asterisk Tiranad (~tiranad@w034.z064000138.lax-ca.dsl.cnc.net) |
| 19:25.29 | cypromis | 8khz that is |
| 19:25.32 | terrapen | then use sox to re-encode it |
| 19:25.40 | shido6 | Blackthron |
| 19:25.42 | shido6 | good luck encoding gsm |
| 19:25.43 | terrapen | audacity.sf.net i think |
| 19:25.49 | shido6 | you can do it with Record,blah:gsm |
| 19:25.52 | shido6 | in the dialplan |
| 19:25.53 | zoa | :) |
| 19:25.56 | shido6 | and let asterisk record it |
| 19:25.56 | harryvv | Is there a comman that once the zap pstn is hanged up it will stop the ringing of the client? |
| 19:26.05 | shido6 | then use playback to listen to your silly prompt |
| 19:26.09 | zoa | :) |
| 19:26.12 | shido6 | unless you do this a lot youre gonna do it 3 or 4 times |
| 19:26.13 | zoa | then do it again |
| 19:26.15 | *** join/#asterisk justin_e (~justin_e@c-67-169-58-141.client.comcast.net) |
| 19:26.15 | shido6 | and have some ones u want to save |
| 19:26.19 | shido6 | cuz they sound funny as hell |
| 19:26.21 | terrapen | i had my sister record all of our voice prompts |
| 19:26.22 | bjohnson | shido6: had to switch around my hardware before getting a chance to try that dtmfmode=inband idea .. is there a way to do something similar with a X100P? I can't find a way to set dtmfmode in zapata.conf |
| 19:26.24 | zoa | yeah |
| 19:26.24 | terrapen | it took forever |
| 19:26.25 | cypromis | than calla studio and let them do a proper job |
| 19:26.28 | shido6 | drunk IVRs are best |
| 19:26.33 | zoa | because your little sister is crying in the background |
| 19:26.36 | shido6 | LOL |
| 19:26.38 | harryvv | terrapen, she has a good voice? |
| 19:26.38 | shido6 | right |
| 19:26.47 | zoa | because your dog now farted too close |
| 19:26.58 | terrapen | she has a pretty good voice but i should have used a studio-quality mic...too many bops |
| 19:27.04 | zoa | and you started coughing |
| 19:27.04 | `Sauron | bjohnson: I have an idea for how to do it, I'll have to do some poking around. |
| 19:27.17 | bjohnson | `Sauron: keep me updated |
| 19:27.20 | redder86 | I just record voice prompts from a phone. |
| 19:27.22 | shido6 | got a wet towel in the background? ( dont ask ) it helps with the sound |
| 19:27.23 | cypromis | and the train was pasing right by the window |
| 19:27.28 | cypromis | and a tsunami flooded the house |
| 19:27.31 | zoa | yeah |
| 19:27.42 | justin_e | is there a way to not answer an incomming call on my PRI? i.e. so that if they are calling my 800 # I won't be charged? |
| 19:27.47 | zoa | and then he accidentaly recorded over his first good recording |
| 19:28.05 | redder86 | justin_e: just don't answer it in your dialplan |
| 19:28.14 | terrapen | there used to be a website where you could get washed-up, former B-list celebrities to record messages for your friend's answering machine |
| 19:28.53 | cypromis | why are you calling me ? I don't have a telephone ... |
| 19:29.04 | Juggie | justin_e, are their certain numbers you dont want to answer? |
| 19:29.07 | justin_e | redder86 I have a catch all ext in my dial plan is the problem, can I specify a certain ext not to answer |
| 19:29.13 | Juggie | or when a certain number is called you dont want to answer |
| 19:29.20 | justin_e | yes Juggie |
| 19:29.25 | terrapen | you could get Gary Coleman to record your voice prompts |
| 19:29.27 | shido6 | if your number is say 8001234567 you could do ...... say |
| 19:29.30 | Juggie | (which one i said two) |
| 19:29.34 | harryvv | Alot of people get turned off from listening to IVRs and rather just press the 0 button. Im no exception. Having a really great female voice makes a difference :) |
| 19:29.34 | redder86 | justin_e: then catch the unanswerable number before the catchall. |
| 19:29.50 | Juggie | you can do exten=> 8001234567,1,Congestion |
| 19:29.58 | terrapen | since his bid for Governor of California didn't go so well, i'm sure he'd love the attention |
| 19:29.59 | shido6 | no |
| 19:30.00 | shido6 | no |
| 19:30.02 | Juggie | it will give them a this number is not in service. |
| 19:30.03 | shido6 | whats the number calling you |
| 19:30.16 | justin_e | ah, ok thanks I'll give that a try |
| 19:30.18 | harryvv | ohh wouldnt that be cool get a irv from Arnold |
| 19:30.19 | harryvv | :) |
| 19:30.23 | harryvv | ivr |
| 19:30.32 | Juggie | justin_e, do you want to block by number calling? or number being called? |
| 19:30.34 | shido6 | if the number calling you is 8105551212 and your 8xx number is 8001234567 then u could do |
| 19:30.44 | shido6 | exten => 8001234567/8105551212,1,Congestion |
| 19:30.53 | terrapen | people i'd love to have on my PBX prompts... |
| 19:30.54 | shido6 | or |
| 19:30.58 | shido6 | exten => 8001234567/8105551212,1,hangup |
| 19:31.00 | shido6 | hehee |
| 19:31.13 | justin_e | block by number being called |
| 19:31.19 | Juggie | ah |
| 19:31.20 | shido6 | exten => 8001234567/8105551212,1,Playback,fuckoff |
| 19:31.23 | zoa | im off |
| 19:31.24 | zoa | cheers |
| 19:31.27 | Juggie | then by number being called, do what i said justin. |
| 19:31.29 | redder86 | shido6: is that exten syntax new (calledexten/callerid) ? |
| 19:31.29 | modulus_ | anyone here u se voipjet? |
| 19:31.33 | modulus_ | their callerid is fubar |
| 19:31.36 | Juggie | assuming you get the 800 number from the pri when its called. |
| 19:31.40 | shido6 | this number is no longer in service |
| 19:31.54 | ManxPower | redder86, It's called the Ex-GF opton and has been documented in the handbook for at least 2 years. |
| 19:32.08 | harryvv | shido6, and the boss would use that number and you would get fired :) |
| 19:32.11 | bjohnson | modulus_: you just don't have any luck .. do you? |
| 19:32.17 | terrapen | wouldn't Congestion() answer the phone, though? |
| 19:32.18 | modulus_ | bjohnson, rarely |
| 19:32.24 | modulus_ | nufone callerid works fine |
| 19:32.26 | terrapen | and you would pay some call charges? |
| 19:32.28 | bjohnson | I use voipjet but haven't played with callerid at all yet |
| 19:32.28 | redder86 | ManxPower: nice, cool feature, although I've no use for it now. |
| 19:32.34 | modulus_ | voipjet only transmits last 4 digits of callerid |
| 19:32.42 | terrapen | callerID is so fun with * |
| 19:32.46 | ManxPower | redder86, I use it to send calls from my grandmother directly to my phone, bypassing the IVR. |
| 19:32.49 | modulus_ | not with voipjet |
| 19:32.52 | terrapen | i called an old girlfriend who would not answer my calls |
| 19:33.00 | modulus_ | psycho |
| 19:33.04 | modulus_ | creepy |
| 19:33.05 | terrapen | i used her phone number |
| 19:33.06 | terrapen | heh |
| 19:33.07 | redder86 | ManxPower: I ended up writing an AGI to do the same thing |
| 19:33.27 | terrapen | psycho would be using her mom's # |
| 19:33.34 | ManxPower | redder86, You also use a hammer to kill a fly, I assume. |
| 19:33.51 | redder86 | ManxPower: I have a "database" of numbers that I do that kind of thing for. I didn't want to modify the dialplan each time. |
| 19:34.17 | justin_e | it seems like Congestion does answer the call |
| 19:34.23 | redder86 | ManxPower: I had to adust Caller*ID, too, anyway. I hate when Caller*ID Name says "WIRELESS CALLER" |
| 19:34.28 | bjohnson | hammer's aren't really tha great for fly hunting |
| 19:34.42 | bjohnson | cars are better |
| 19:34.57 | justin_e | Accepting call from '5103xxxxxxx' to 'xxxxxxx' on channel 0/1, span 1 |
| 19:34.57 | justin_e | <PROTECTED> |
| 19:35.12 | justin_e | then show channels shows a call |
| 19:35.22 | justin_e | and later on I get -- Channel 0/1, span 1 got hangup |
| 19:35.33 | redder86 | very few wireless carriers provide anything useful in Caller*ID Name, as far as I've experienced. |
| 19:36.24 | ManxPower | justin_e, What are you trying to accomplish? |
| 19:36.25 | mikegrb | redder86: who do you use for inbound that you get CID Name? |
| 19:36.36 | redder86 | ManxPower: if my in-laws call, they go straight to voicemail. :-) Nobody here (inlcuding my wife) wants to talk with them. |
| 19:36.37 | ManxPower | ..er.... |
| 19:36.44 | ManxPower | justin_e, What are you trying to accomplish? |
| 19:36.52 | justin_e | I have some toll free numbers that are for a project that is currently not active |
| 19:37.02 | redder86 | mikegrb: I get Caller*ID Name on my Qwest line and on my NuFone DIDs. |
| 19:37.09 | redder86 | mikegrb: oh wait |
| 19:37.15 | justin_e | and we are getting tons of calls that are costing lots of money, because they are international toll free numbers |
| 19:37.16 | ManxPower | justin_e, leave the toll free number out of the dialplan. Asterisk will simply reject the call. |
| 19:37.18 | mikegrb | redder86: oh |
| 19:37.21 | redder86 | mikegrb: no, NuFone doesn't provide Caller*ID Name |
| 19:37.27 | mikegrb | redder86: right ;) |
| 19:37.30 | ManxPower | If it's a PRI the caller will usually give a "disconnected" message to the caller. |
| 19:37.56 | ManxPower | justin_e, That way you don't even have to answer the call |
| 19:38.09 | redder86 | mikegrg: that's why I keep a database of the Caller*ID Names that come in on my Qwest line. So if the call comes in on my NuFone DIDs, then it "looks the name up" from the database. |
| 19:38.17 | justin_e | ManxPower: The problem is I have a catch all at the bottom of the dialplan |
| 19:38.42 | justin_e | so I can't really "leave" it out, unless there is some way to say match every thing except these 20 numbers |
| 19:38.55 | ManxPower | justin_e, well that's a pretty stupid thing to do. Then you can't reject calls |
| 19:39.00 | redder86 | justin_e: if you don't answer those calls ever why don't you just disconnect that service? |
| 19:39.32 | justin_e | well it tooks several months to setup all the ITFN that we have, and we anticipate using them in the future |
| 19:39.52 | justin_e | just don't want to keep paying for $500/month of wrong numbers or people trying to mess with the system |
| 19:40.01 | ManxPower | It should be pretty easy to mopdify your catch-all to only catch the numbers you WANT. |
| 19:40.23 | redder86 | justin_e: and if you certainly shouldn't be receiving calls on random numbers. So you should only have a "catch all" for the block of numbers that you should be receiving. |
| 19:41.12 | justin_e | ok, I'll take a look at restructuring that. I just thought there might be an easy DontAnswer type solution |
| 19:41.28 | bjohnson | well .. just don't answer it |
| 19:41.42 | redder86 | mikegrb: for $0.45 per month Qwest forwards calls to my NuFone DID when my land-line is busy. |
| 19:42.01 | bjohnson | redder86: hey .. that's pretty good |
| 19:42.07 | *** join/#asterisk harryvetch (~noyb@S010600055d210201.vs.shawcable.net) |
| 19:42.12 | ManxPower | I still think that every time Asterisk encounters _. in the dialplan it should play "You're a lazy bastard!" over the sound card |
| 19:42.29 | harryvetch | or thay are on drugs |
| 19:42.37 | redder86 | yeah, that way I get a local number *and* I can keep a database of incoming Caller*ID Name/Number for later lookup if the call comes through NuFone. |
| 19:43.30 | harryvetch | one case two guys were so stoned thay called 911 and hungup. Next thing police knocked on the door and as one guy opened it the other was walking past him with a bad of marijane :) |
| 19:43.41 | harryvetch | BJ, who knows :) what is the ivr for? |
| 19:43.49 | ManxPower | harryvetch, Well stupid people SHOULD be arrested |
| 19:44.04 | redder86 | bjohnson: if only I could get my Qwest line for $0.02 per minute rather than $25.00 per month. |
| 19:44.14 | harryvetch | Manx I know. Usually thay are dumb :) Thay why thay cannot get a good job. |
| 19:44.24 | bjohnson | small business .. I'd like it to be entertaining .. but don't know if customers would share hurour |
| 19:44.27 | bjohnson | humour |
| 19:44.57 | harryvetch | What kind of biz is it? |
| 19:45.16 | bjohnson | redder86: I'm starting to track minute usage on pstn line to see if voip would be better |
| 19:45.16 | Beirdo | bjohnson: make the error message "What, are you a dumbass? That's an invalid choice!" |
| 19:45.40 | bjohnson | found a bunch of funny alison ivr messages aroung the net |
| 19:45.40 | harryvetch | fastest way to get fired |
| 19:45.41 | harryvetch | ;) |
| 19:45.49 | harryvetch | alison? |
| 19:45.54 | bjohnson | alison smith |
| 19:46.01 | redder86 | bjohnson: I had an ISP outage that lasted about 2 hours this past week. I was glad that I had the land line. |
| 19:46.02 | harryvetch | dont know who thatis |
| 19:46.12 | ManxPower | harryvetch, allison is the Voice of Asterisk |
| 19:46.17 | harryvetch | okay |
| 19:46.23 | bjohnson | something like ivrvoice.com |
| 19:47.22 | bjohnson | maybe we should collect some seasonal ivr stuff .. could get some cool halloween stuf I bet |
| 19:47.54 | harryvetch | yea, people dont like ivrs. might as well make them cheer up |
| 19:49.19 | _Brian | all call legs that originate from a sip phone to my asterisk system appear to be ulaw format, is there a way to force them to utilize something else (gsm). I have put in the appropriate disallow/allow entries in sip.conf, but it does not appear to effect these calls |
| 19:50.01 | bjohnson | how to you set dtmfmode=inband on a x100p? |
| 19:50.25 | ManxPower | bjohnson, PSTN calls are always inband |
| 19:50.28 | bjohnson | _Brian: reload? |
| 19:50.38 | _Brian | bjohnson: tried that :( |
| 19:50.40 | bjohnson | ManxPower: the x100p is connected to a Nortel ATA |
| 19:50.51 | *** join/#asterisk clive- (~pirch@rrba-146-124-148.telkomadsl.co.za) |
| 19:51.00 | _Brian | bjohnson: it does appear that the outbound call leg is using gsm, I want the inbound call leg to utilize it as well.. |
| 19:51.06 | bjohnson | ManxPower: no tones going through .. shido6 suggested trying inband |
| 19:51.16 | ManxPower | bjohnson, The Nortel ATA does not send DTMF. Sucks to be you. |
| 19:51.23 | bjohnson | yes it does |
| 19:51.50 | ManxPower | bjohnson, Call the nortel ATA from a Nortel digital phone, then dial from the ditigal set. you will not hear DTMF on the ATA |
| 19:51.55 | *** join/#asterisk herag (herag@ca-stmnca-cuda4-gen2c1-171.vnnyca.adelphia.net) |
| 19:52.14 | ManxPower | UNLESS you are using totally different Nortel ATAs than we tried using. |
| 19:52.18 | bjohnson | yes it does .. suck to be me |
| 19:52.56 | bjohnson | but .. maybe I AM trying a different ATA .. did you try the one at my office? |
| 19:56.09 | *** join/#asterisk Luhiwu (~marsosa@200.63.87.246) |
| 19:58.39 | *** join/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com) |
| 19:59.02 | FirstSword | hi, I just received Atcom AT-323^^ |
| 19:59.09 | *** part/#asterisk justin_e (~justin_e@c-67-169-58-141.client.comcast.net) |
| 19:59.14 | *** join/#asterisk buddah (~hnic@208.179.86.5) |
| 19:59.31 | buddah | anyone ever hear of a misconfigured polycom bringing down someones local network when used? |
| 20:01.18 | *** join/#asterisk charles___ (~charles@64.35.168.55) |
| 20:01.44 | denon | buddah: people claim that grandstreams do |
| 20:01.54 | ManxPower | bjohnson, But you have confirmed that when you plug an analog phone into the Nortel ATA you hear DTMF send from Nortel digital phones when you call it? |
| 20:01.58 | denon | or use to anyway |
| 20:02.05 | denon | used to |
| 20:02.07 | denon | whatever |
| 20:02.37 | ManxPower | bjohnson, I didn't know Nortel had more than one model of their ATA. The one we have looks like one of those silly fax/modem switch boxes (at least the case does) |
| 20:02.56 | bjohnson | this one is built into the CICS |
| 20:02.58 | bjohnson | one port |
| 20:03.30 | WildPikachu[BAR] | hrmmm |
| 20:03.38 | *** join/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 20:03.39 | WildPikachu[BAR] | does musiconhold allow any mp3? |
| 20:03.48 | Samoied | Hello all! |
| 20:04.05 | Samoied | Anyone have used k-1000 usb phone in linux? |
| 20:04.09 | ManxPower | bjohnson, Ah! not a stand alone device then? |
| 20:04.22 | Samoied | The sound function properly, but not the keys |
| 20:04.24 | sivana | ~seen normast |
| 20:04.26 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca> was last seen on IRC in channel #asterisk, 5h 26m 39s ago, saying: 'Nivex: I guess it's them same on most distro however..'. |
| 20:05.17 | modulus_ | voipjet isn't passing callerid correctly |
| 20:05.27 | bjohnson | ManxPower: no .. just one pair of the big cable that comes out of the cics |
| 20:05.50 | bjohnson | ManxPower: I head dtmf tones on the ata only if "Long tones" is enabled on the nortel handset |
| 20:06.11 | bjohnson | ManxPower: but using "long tones" seems to hang up the x100p |
| 20:06.27 | bjohnson | WildPikachu: yes .. even pron sounds |
| 20:07.00 | *** join/#asterisk r0d3nt|m (RatMan@64.60.114.35) |
| 20:07.28 | *** join/#asterisk empire667 (~user1@h71032.upc-h.chello.nl) |
| 20:07.46 | *** join/#asterisk jarrod (~jarrod@35256.ds.nac.net) |
| 20:07.56 | jarrod | is there any 'call manager software' for asterisk? |
| 20:08.02 | jarrod | either open source or commercial |
| 20:08.45 | ManxPower | bjohnson, turn off callprogress and busydetect |
| 20:10.16 | bjohnson | jarrod: yes .. check the wiki |
| 20:10.24 | tzafrir_home | Mother_, ? |
| 20:11.14 | bjohnson | ManxPower: both are commented out in my modified sample zapata.conf |
| 20:11.24 | bjohnson | SON !!!! |
| 20:14.33 | harryvetch | whats the switch or command to append two emails to one voicemail box? |
| 20:14.46 | harryvetch | is it a simple , ? |
| 20:15.33 | tzafrir_home | yes |
| 20:16.23 | harryvetch | Thanks |
| 20:16.24 | jaiger | is anyone familiar with the Adtran TA750 and FXO interfaces? |
| 20:18.27 | *** join/#asterisk jgaviria (~juan_manu@63.245.86.116) |
| 20:19.49 | jgaviria | i have an * box with a PRI connected, now i connected a TDM40 and when i pickup a phone connected to the TDM40, i got Unable to play dialtone on channel xx, somebody could helpme? |
| 20:20.17 | bjohnson | jaiger: I guess no-one here right now |
| 20:20.37 | jaiger | bjohnson: I guess so |
| 20:21.15 | blitzrage | ~zx81 |
| 20:21.16 | jbot | i guess zx81 is the creater of the Daily Asterisk News (see ~adn) |
| 20:21.25 | blitzrage | ~seen zx81 |
| 20:21.27 | jbot | zx81 <~ZX81@222-153-18-2.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 7h 34m 14s ago, saying: ':)'. |
| 20:21.36 | blitzrage | ~blitzrage |
| 20:21.37 | jbot | somebody said blitzrage was a super cool fellow |
| 20:21.40 | blitzrage | lol |
| 20:21.46 | fa | Feb 3 21:21:37 NOTICE[10636]: app_dial.c:762 dial_exec: Unable to create channel of type 'Zap' |
| 20:21.50 | fa | why? |
| 20:22.05 | blitzrage | hey, since ZX81 doesn't have his nickname registered, if someone sees him before me, tell him to contact me! |
| 20:22.19 | buddah | -- Got SIP response 503 "Service Unavailable" back from 67.110.252.10 |
| 20:22.27 | buddah | what would make something give that response? |
| 20:22.38 | buddah | its coming off a cisco vg224 |
| 20:23.16 | *** part/#asterisk Samoied (~samoied@popeye.opens.com.br) |
| 20:24.28 | *** join/#asterisk Mike (~mike@201.135.48.217) |
| 20:24.43 | *** join/#asterisk sung (~sung@fluorine.idge.net) |
| 20:24.46 | `Sauron | ~ads |
| 20:24.47 | jbot | please don't advertise in #debian... it's not effective anyhow. |
| 20:24.50 | `Sauron | ~adn |
| 20:24.51 | jbot | well, adn is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://www.sineapps.com/rssfeed.php for RSS |
| 20:24.53 | Mike | guey messages store for voicemail are stored with root permisions how can i change it to work with vmail.cgi? |
| 20:24.56 | Mike | -rwx------ 1 root root 4.0K 2005-02-03 14:54 msg0001.WAV* |
| 20:24.57 | Mike | calle69:/var/spool/asterisk/voicemail/asterisk/101/INBOX# |
| 20:26.53 | netsurfer | where does asterisk find the gsm files for saydigits ? i've replaced 0.gsm 1.gsm etc etc but its still the original audio even after restarting the server |
| 20:27.04 | greg_work | estimated wait time to reach sales rep: 44 minutes. time to hire more people .. |
| 20:27.07 | ManxPower | ~jbot google site:lists.digium.com vmail.cgi permission* |
| 20:27.36 | Beirdo | ManxPower: whose fine bot is that? |
| 20:27.44 | netsurfer | lol greg_work - for sure |
| 20:28.01 | *** part/#asterisk BozzaCamilleri (~connect@213.217.225.32) |
| 20:28.06 | `Sauron | greg: where's that? |
| 20:28.08 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 20:28.15 | PakiPenguin | can anyone please do a testcall for me |
| 20:28.24 | netsurfer | PakiPenguin - to where ? |
| 20:28.56 | *** join/#asterisk afrosheen (afrosheen@txprotoa8.august.net) |
| 20:29.15 | *** join/#asterisk Twister (~bob@pool-151-205-67-217.char.east.verizon.net) |
| 20:29.34 | fa | How can I disable that: received TEI check request for TEI = 87 |
| 20:29.35 | fa | Feb 3 21:29:08 WARNING[10838]: chan_zap.c:7411 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 |
| 20:30.29 | *** join/#asterisk ckruetze (~ckruetze@i3ED61EDF.versanet.de) |
| 20:30.40 | ManxPower | Beirdo, I don't know. I hate bots, but they are useful for people too lazy to use Google. |
| 20:30.51 | bjohnson | I still can't get my long distance pattern matches to leave my toll free calling alone :( |
| 20:30.57 | Beirdo | this one seems particularly useful |
| 20:31.12 | Beirdo | ~seen MajestiK |
| 20:31.13 | jbot | majestik is currently on #asterisk #debian |
| 20:31.28 | MajestiK | huh, what? did someone call me? |
| 20:31.31 | Beirdo | hehe |
| 20:31.42 | Beirdo | I was just using the bot |
| 20:31.44 | ManxPower | bjohnson, if you comment out your LD patters, do the toll free ones work? |
| 20:33.41 | PakiPenguin | exten => _77.,1,Dial(SIP/${EXTEN:2@domain.com}) <-- will this throw what ever dialed as 77XXXXX to domain.com? |
| 20:33.51 | *** join/#asterisk Beave (~beave@vistech.org) |
| 20:33.53 | Beave | hey all. |
| 20:34.02 | jarrod | it will throw XXXXX@domain.com |
| 20:34.14 | PakiPenguin | yes , i meant that |
| 20:34.34 | greg_work | isnt it Dial(SIP/${EXTEN:2}@domain.com) |
| 20:34.42 | Twister | hi all...i wanna get somethign straight...i have an office with 4 lines, Would i need to buy a TDM31B 3 FXS 1 FXO then add on another FXS? |
| 20:34.47 | Beave | I'm looking to test the max number of channels (SIP or IAX2) can handle at one time. Anyone know of a good way to stress test the voip links like this? |
| 20:34.47 | jarrod | ah yes |
| 20:34.57 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 20:35.22 | Twister | or does someone have a better solution |
| 20:36.17 | afrosheen | Beave: that's going to be matched to two things, bandwidth and codec |
| 20:36.34 | greg_work | twisted[work]: you have 4 CO lines now? |
| 20:36.42 | afrosheen | Beave: just calculate your available upstream and do some math on your codecs |
| 20:36.49 | Twister | yes |
| 20:36.51 | greg_work | twisted[work]: are you trying to keep all 4 lines or just share 1 CO between 3 phones? |
| 20:37.07 | Twister | trying to keep all 4 lines |
| 20:37.21 | Beave | afrosheen : thanks, and I understand that, but I have someone how actually wants to "see" this in action. Unfortunatly.. that's the only reason I ask. |
| 20:37.37 | greg_work | twisted[work]: you'd want to get a TDM04b then, which has 4 fxo ports. fxo connects to CO lines, fxs connects to phones |
| 20:37.47 | afrosheen | Beave: make some fake graphs in gimp |
| 20:37.54 | Beave | haha |
| 20:37.55 | Beave | nice. |
| 20:38.17 | afrosheen | 'see here boss? this is the max calls outbound at this plateau here with the flowers on it' |
| 20:38.18 | Twister | oh |
| 20:38.18 | Twister | ok |
| 20:38.24 | Twister | i was backwards then |
| 20:38.34 | *** part/#asterisk yogurt2ungue (~charlie@203-132-126-200.fibertel.com.ar) |
| 20:38.38 | Beave | I don't think that will impress them.. I can manually do calls across the link, but I thought someone might have been through these type of hoops before. |
| 20:38.49 | *** join/#asterisk Strom_C (~strom@66.159.243.60) |
| 20:38.49 | greg_work | twisted[work]: to confuse it more, you use fxs signalling with fxo ports, and fxo signalling on fxs ports ;) |
| 20:38.56 | afrosheen | Beave: with expensive software and a test lab it's very possible |
| 20:39.42 | Twister | /boggle |
| 20:39.47 | Twister | actually |
| 20:39.48 | Twister | no |
| 20:39.51 | Twister | that makes perfect sens |
| 20:39.51 | Twister | e |
| 20:40.01 | Beave | ? |
| 20:40.04 | *** join/#asterisk Slainte (Slainte@207.228.155.26) |
| 20:40.07 | bjohnson | ManxPower: damn .. a wayward '-' where it should have been a '_' |
| 20:40.11 | Twister | because you would need to use fxs signaling on the fxo to talk to the fxs and likewise ont he fxs |
| 20:40.22 | afrosheen | Beave:http://www.commweb.com/trends/54202074 |
| 20:40.47 | afrosheen | Beave: read up on the shunra storm wan thing |
| 20:40.57 | ManxPower | bjohnson, a simple "no" would have worked. *tease* |
| 20:41.01 | bjohnson | Twister: no .. there are many hardware configurations you could use |
| 20:41.19 | harryvetch | Im seeing this on my console wcfxo: Out of space to write register 05 with 08 anyone seen this before? |
| 20:41.20 | Twister | ah |
| 20:41.47 | Slainte | I have a T1 PRI and every so often even when no one is on the system I get a No one is available to answer at this time |
| 20:41.48 | greg_work | Twister: yeah, the tdm04b isn't the only way, but it's a good way (i have 4 lines, and that's what I got) |
| 20:42.05 | afrosheen | Slainte: sip phones? |
| 20:42.07 | Beave | Welp thanks. I'll just manually punch the number of calls they want throw. This doesnt have to be scientific, they just have to be able to hear it work. |
| 20:42.24 | PakiPenguin | http://pastebin.ca/5224 <-- should this in sip.conf mean , that every call coming from the ip in host , should goto the context defined? |
| 20:42.30 | afrosheen | Beave: try putting each phone in front of a different shoutcast stream |
| 20:42.38 | Beave | also, lastly, anyone ever play with ESI equipment? If possible, avoid it. |
| 20:42.43 | afrosheen | lol |
| 20:42.45 | jgaviria | i have an * box with a PRI connected, now i connected a TDM40 and when i pickup a phone connected to the TDM40, i got Unable to play dialtone on channel xx, somebody could helpme? |
| 20:43.08 | *** join/#asterisk ranliv (~ranliv@ipdial-246-155.info.com.ph) |
| 20:43.36 | Twister | thank you greg_work |
| 20:43.52 | ranliv | hello guys! i need help with my zap card |
| 20:44.03 | afrosheen | ranliv: well then! you're in the right place! |
| 20:44.06 | jaiger | harryvetch: I would grep through the driver source to see what causes that |
| 20:44.26 | PakiPenguin | anyone ? hello? |
| 20:44.33 | PakiPenguin | http://pastebin.ca/5224 |
| 20:44.48 | Slainte | äfro, yes |
| 20:44.59 | Slainte | Afro :, yes |
| 20:46.10 | ranliv | it seems asterisk could not create the zap channel |
| 20:46.25 | Twister | what are some other ways (just curious) |
| 20:46.37 | ManxPower | jgaviria, prolly shareing IRQs |
| 20:46.48 | ranliv | doing ztcfg -vv seems my card has been detected and install properly |
| 20:47.37 | ranliv | but doing show channel 1 @ * console = channel not known |
| 20:47.54 | ManxPower | ranliv, zap show channel 1 |
| 20:48.56 | ranliv | hehehe sorry... newbie mistake |
| 20:49.26 | Twister | so i would have to use fxs ports for the phones right (if im not doing ip phones) |
| 20:51.09 | jgaviria | ManxPower: i already check it, and t100p has 16 and wcfxs has 18 |
| 20:51.20 | mikegrb | my wife is learning to code! |
| 20:51.37 | ManxPower | mikegrb, be afraid. be very afraid. |
| 20:51.47 | bjohnson | Twister: yes .. for analog phones |
| 20:51.54 | mikegrb | http://thegrebs.com/~heidi/sshot.png |
| 20:52.13 | mikegrb | ManxPower: she even googled for builtin functions on her own! |
| 20:52.20 | ManxPower | mikegrb, eeewwwww!!!!!!! |
| 20:52.29 | bjohnson | 4 spa3000 at $100 ea would give you 4 fxo and 4 fxs .. I would suggest maybe a TDM40 with 2 and 2 and 2 of these |
| 20:52.33 | Twister | okiez, thanks :) |
| 20:52.43 | bjohnson | Twister: you haven't said how many phones total |
| 20:52.56 | mikegrb | ManxPower: would it make you feel better if I said she uses irssi and vim? |
| 20:53.01 | ManxPower | children are small, loud, disease carrying creatures that should be kept away from the civilized world. |
| 20:53.10 | bjohnson | for 4 lines + phones you might be better off with a channel bank |
| 20:53.17 | Twister | oh, theres 25 phones in my office, but im thinking of transfering the whole thing to ip phones |
| 20:53.49 | bjohnson | a T1 card from digium and a channel back could handle 23 of combined fxo and fxs |
| 20:53.58 | bjohnson | channel bank .. sorry |
| 20:54.29 | bjohnson | then use single fxs for other phones and/or add voip phones |
| 20:55.01 | Twister | didnt digium publish a book on asterisk? |
| 20:55.25 | bjohnson | I don't have a channel bank .. but I hear a used Adit600 with mixed fxo and fxs ports can be had for $200 on ebay |
| 20:55.59 | bjohnson | but then you need the digium T1 card and cabling from the chan bank to the phone |
| 20:56.02 | bjohnson | s |
| 20:56.04 | Twister | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=44993&item=5748619349&rd=1&ssPageName=WDVW |
| 20:57.17 | *** join/#asterisk PakiPenguin (~info@202.176.254.1) |
| 20:57.23 | Twister | thanks for all your help |
| 20:58.52 | PakiPenguin | how do i connect * with a B2BUA , i want to accept calls coming in from a specific ip and handle them in a context , i tried this < http://pastebin.ca/5224 > in sip.conf , but this doesnt work , when the call comes in from the ip mentioned in sip.conf , it takes it to the default incoming context , instead of the context i defined |
| 20:59.00 | tzanger | your coffee pot gurgles? |
| 20:59.19 | ManxPower | yup |
| 20:59.25 | tzanger | that's not good |
| 20:59.33 | _Brian | PakiPenguin: can you also pastebin the debugs from * |
| 20:59.48 | `Sauron | Has anyone connected * to cisco call manager through mgcp? |
| 20:59.57 | PakiPenguin | _Brian: yes sure |
| 21:00.53 | fa | <PROTECTED> |
| 21:00.53 | fa | <PROTECTED> |
| 21:00.54 | PakiPenguin | _Brian: http://pastebin.ca/5225 |
| 21:01.07 | fa | Why I have answered.. before in fact.. 0691761693 don't answer. |
| 21:01.11 | ManxPower | PakiPenguin, Your first step is to stop using silly acronyms like B2BUA |
| 21:01.16 | *** join/#asterisk Cresl1n (~matt@216.207.244.186) |
| 21:01.24 | *** join/#asterisk abbas_ (nidobas@203.81.194.242) |
| 21:02.38 | jaiger | I'm using an adtran channel bank + digium T1 card for my * installation |
| 21:02.46 | PakiPenguin | _Brian: any luck? |
| 21:03.26 | PakiPenguin | _Brian: it should goto incoming_newgateway , as my sip.conf entry says , but it still went to incoming |
| 21:03.40 | *** join/#asterisk file (~symlink@mctnnbsah25-142166093009.nb.aliant.net) |
| 21:03.41 | afrosheen | ManxPower: thought it was bubble, bubble |
| 21:03.53 | Slainte | <PROTECTED> |
| 21:04.02 | Slainte | Why do I keep getting this? |
| 21:04.15 | Slainte | Sip phones, T100p card, full PRI |
| 21:04.20 | Slainte | no one on the system. |
| 21:04.25 | afrosheen | Slainte: can you read this |
| 21:05.22 | ManxPower | PakiPenguin, you have a problem in sip.conf |
| 21:05.44 | Slainte | yes I can read this :) |
| 21:05.59 | ManxPower | Slainte, Without a pastebin of the console output around that message, nobody will be able to help you. |
| 21:06.09 | afrosheen | Slainte: ok good, I asked you much earlier, what kind of phones do you have |
| 21:06.53 | Slainte | I said Afro: yes responding to your sip question. Let me get the paste |
| 21:07.06 | afrosheen | Slainte: ok what brand are they and are they registering |
| 21:08.04 | PakiPenguin | ManxPower: pasting it , holdon |
| 21:08.04 | *** join/#asterisk sneak (~sneak@198.22.65.197) |
| 21:08.28 | *** join/#asterisk zeek (~zeekk@gw.dhivehinet.net.mv) |
| 21:08.50 | PakiPenguin | http://pastebin.ca/5229 <-- ManxPower sip.conf |
| 21:08.56 | Slainte | afrosheen: Polycom IP600 |
| 21:09.01 | zeek | anybody from spain here? |
| 21:09.13 | Slainte | afrosheen: getting logs for pastebin now |
| 21:09.29 | thieumS | what's the best calling card addon for asterisk ? |
| 21:09.35 | afrosheen | Slainte: our ip600 doesn't always take calls right away either, our 500's do though |
| 21:09.35 | ManxPower | PakiPenguin, You do not understand te difference between type=user and type=peer. |
| 21:09.55 | ManxPower | PakiPenguin, Once you understand the difference then you'll know how to fix the problem, Grasshopper. |
| 21:10.18 | _Brian | grasshopper...rofl |
| 21:10.22 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 21:10.28 | _Brian | wax on....wax off |
| 21:10.30 | Slainte | http://pastebin.ca/5230 When you say take calls you mean either incomming or outgoing? |
| 21:10.31 | afrosheen | when you can snatch the pebbles from my hand, you will then be ready |
| 21:10.34 | *** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 21:10.39 | afrosheen | Slainte: take = incoming |
| 21:11.16 | bjohnson | did someone say they needed phones? |
| 21:11.21 | bjohnson | big url |
| 21:11.23 | ManxPower | Slainte, SIP/2905 is tryin to make a call! |
| 21:11.23 | bjohnson | www.hbc.com/hbconline/catalog/product2.asp?langid=EN&imgthumb=True&productid=39917604&catid=d_681006&pcatid=d_1;d_681;d_681006&scatid=&Abstract=True&MSCSProfile=61E4CECF7275066FD87B9817DA5865CBE24730 256207A1D41B2943B1DE07E3E30A124987C33D31ED6A59E0E0 13DF8FB5188FB3C23E2D72DBAE8DA2BD85FEA293063C6043E3 F1F6E78F961D21625641C9F063BDB805DF2134DA86E615A93A 699215101D8061D0B6BFE6E2247AC21B3075EA78650EA8F4CF 7924DDE16EFB0DED081AF4DC7B36773B6B |
| 21:11.54 | ManxPower | Slainte, I don't see you handleing ANY disconnect codes! |
| 21:11.57 | Slainte | Slainte: Yes Sip/905 was tryingt o make an outside call but got a fast busy |
| 21:12.03 | _Brian | tinyurl.com is your friend |
| 21:12.15 | afrosheen | OW JESUS MY EYES |
| 21:12.26 | ManxPower | Slainte, right after the Dial line in your extensions.conf put a NoOp(CAUSECODE=${CAUSECODE}) |
| 21:12.31 | ManxPower | reload asterisk, try it again |
| 21:12.49 | Slainte | all my B Channels have reset, and now it works |
| 21:12.55 | ManxPower | Slainte, No! |
| 21:12.56 | Slainte | let me do the NoOp(CAUSECODE=${CAUSECODE}) thanks |
| 21:13.18 | ManxPower | resetting the bchannels is a normal part of Asterisk operation. you just don't understand how PRIs work and that is causing you problems. |
| 21:14.04 | zyke | any one got irish DIDs? |
| 21:14.45 | Slainte | Manx, so because I am not processing the disconnect codes, asterisk is keeping the channel open. When I get to 23, then it says all are unavailable? |
| 21:14.49 | Beirdo | bjohnson: nice of HBC to have such nasty URLs |
| 21:14.50 | ManxPower | Slainte, then pastebin the result. |
| 21:15.01 | `Sauron | bjohnson: use tinyurl or something, sheesh :p |
| 21:15.11 | ManxPower | Slainte, I did not see you state that before. |
| 21:16.19 | Slainte | Manx, I was asking as a theory. http://pastebin.ca/5231 |
| 21:17.04 | ManxPower | Slainte, call a busy or disconnected number |
| 21:17.10 | ManxPower | then pastebin the result. |
| 21:17.37 | ManxPower | Slainte, no, not handleing causecodes will just give you incorrect sounds when dialing non-answering number. |
| 21:17.46 | Slainte | ahhhh ok. |
| 21:17.53 | Slainte | let me try a disconnected number |
| 21:18.16 | ManxPower | Slainte, You want a cheat? try setting priindication=inband in zapata.conf. I have no idea if that will work or not. |
| 21:18.20 | *** join/#asterisk Alric (~nbowyer@masq.hyperusa.com) |
| 21:18.49 | ManxPower | Slainte, I THINK that will make your PRI act like an analog line. |
| 21:19.49 | ManxPower | Personally I think it's a horrible solution in the long term, but still, it is easy. |
| 21:19.56 | ManxPower | I'm never personally used priindication |
| 21:20.24 | Slainte | http://pastebin.ca/5232 |
| 21:20.28 | tzanger | should be outofband |
| 21:20.33 | tzanger | I never understood why mark left it inband |
| 21:20.34 | Slainte | still not doing what it should I dont think |
| 21:20.49 | tzanger | 'backward compatibility' makes ZERO sense, especially when he made the default of the dialplan fallthrough to 'true' in -HEAD |
| 21:20.58 | tzanger | which is totally assinine, IMO but I haven't heard his reasoning |
| 21:21.30 | ManxPower | WHOO!!! WHOO!!!! Sipura fixed a bug I was experiencing! |
| 21:21.43 | *** join/#asterisk atmel (~vlad@ruxi.dynamic.ucsd.edu) |
| 21:21.50 | ManxPower | Slainte, I don't see the NoOp |
| 21:22.18 | Slainte | either do I. Let me check my extensions.conf and makse sure I put it in properly |
| 21:23.07 | *** join/#asterisk fearnor (~alex@66.250.55.66) |
| 21:23.13 | gambolputty | what sipura bug is that? |
| 21:23.32 | Slainte | Manx, I am reading http://www.voip-info.org/wiki-Asterisk+variable+hangupcause one sec |
| 21:23.57 | fa | anyone use 3 cards od ISDN - hfc ? |
| 21:24.02 | ManxPower | gambolputty, If dial plan has a comma to invoke outside dialtone, input digits are |
| 21:24.02 | ManxPower | <PROTECTED> |
| 21:24.05 | blitzrage | so what is this I keep hearing about Asterisk's 100 simultaneous call limit, truth or FUD? |
| 21:24.20 | vaewyn | FUD |
| 21:26.02 | *** join/#asterisk TokyoJimu (~jimmy@198.51.175.64) |
| 21:26.25 | ManxPower | gambolputty, now I have to figure out how to increase the gain on the handset microphone and the speakerphone microphone. |
| 21:27.07 | afrosheen | blitzrage: what's the reasoning behind that |
| 21:27.13 | TokyoJimu | Anyone else notice that if you use the Google search box on the asterisk.org page to search the archives, it makes you enter your email address? What's with that? |
| 21:27.24 | blitzrage | vaewyn: thats what I was thinking, because I've never seen anyone who says there is a 100 call limit actually give me information as to what they were using, or why they could only get 100 calls |
| 21:27.31 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 21:27.32 | blitzrage | afrosheen: I have no idea, thats what I was wondering :) |
| 21:27.50 | ManxPower | Slainte, I created [macro-dial-result] a long time ago and put it on my site. good to see someone "borrowed" it and put on the wiki. |
| 21:28.00 | afrosheen | maybe on their p3 733 machine it can only handle 100 calls before it crumbles, who knows |
| 21:28.13 | vaewyn | I don't think I would want to put more than 100 on a single machine... but it can easily be done |
| 21:28.19 | ManxPower | TokyoJimu, That search box doesn't wrk well |
| 21:28.27 | afrosheen | plus saying '100 calls' is way too nebulous, there are a million variables that apply to calls |
| 21:28.40 | TokyoJimu | And even if you enter an address it just sends you to Digium's home page. |
| 21:28.47 | _Brian | ManxPower: i was just reading that macro earlier today...... |
| 21:29.04 | ManxPower | TokyoJimu, just go to www.google.com and add site:lists.digium.com to your search terms |
| 21:29.17 | ManxPower | _Brian, What did you think of it? |
| 21:29.42 | terrapen | anyone know what this means (wiki/google turns up next to nothing) |
| 21:29.43 | terrapen | Feb 3 21:24:26 WARNING[2390]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 512/4) |
| 21:29.52 | _Brian | ManxPower: interterestging....i might implement it at a later time |
| 21:29.53 | Delmar | greg_work areound? |
| 21:29.57 | Delmar | around*. |
| 21:30.11 | ManxPower | terrapen, "show codecs" |
| 21:30.46 | terrapen | come on, pastebin, load dammit |
| 21:30.56 | terrapen | surely there must be a better pastebin |
| 21:31.10 | ManxPower | pastebin.com and pastebin.ca |
| 21:31.19 | TokyoJimu | I'm trying to get an ATA-186 SIP box working. Without NAT it works fine, but behind a NAT box, calls to that number ring the phone but you can't answer. Outgoing calls ring the destination number but no audio in either direction. Any hints? |
| 21:31.37 | *** join/#asterisk BozzaCamilleri (~connect@213.217.225.32) |
| 21:31.51 | ManxPower | TokyoJimu, Is Asterisk behind nat too? |
| 21:32.02 | TokyoJimu | No, asterisk is not behind NAT. |
| 21:32.09 | Nohair | Hi any one know why I get Ouch... error while writing audio data: Broken pipe |
| 21:32.19 | ManxPower | TokyoJimu, The I have no idea. All of my stuff that's behind nat works just find. |
| 21:32.25 | ManxPower | Nohair, That's from mpg123 |
| 21:32.29 | *** join/#asterisk W1thdraw (~W1thdraw@ip68-5-125-44.oc.oc.cox.net) |
| 21:32.31 | gambolputty | manxpower: PSTN To SPA Gain |
| 21:32.41 | gambolputty | thats that setting I used on my sipura 3000 |
| 21:32.45 | ManxPower | gambolputty, that option does not exist on the SPA-841 |
| 21:32.52 | gambolputty | whoops |
| 21:32.55 | Nohair | Manxpower any idea why |
| 21:33.01 | gambolputty | wrong product I had in mind |
| 21:33.02 | ManxPower | Nohair, What version does mog123 -v show you? |
| 21:33.25 | TokyoJimu | See anything wrong with this sip.conf entry? |
| 21:33.27 | TokyoJimu | [168801] |
| 21:33.27 | TokyoJimu | type=friend |
| 21:33.27 | TokyoJimu | username=168801 |
| 21:33.27 | TokyoJimu | secret=blahblah |
| 21:33.27 | TokyoJimu | host=dynamic |
| 21:33.28 | TokyoJimu | ;This is the context (in extensions.conf) that handles internal calls |
| 21:33.30 | TokyoJimu | context=outtrunk-unlimited |
| 21:33.32 | TokyoJimu | canreinvite=no |
| 21:33.34 | TokyoJimu | dtmfmode=rfc2833 |
| 21:33.36 | TokyoJimu | nat=yes |
| 21:34.12 | ManxPower | TokyoJimu, use pastebin! |
| 21:34.26 | _Brian | O the flood!! O the flood!! make it stop |
| 21:34.31 | ManxPower | gambolputty, I have Audio Volume -- Ringer:Speaker:Handset:Headset: |
| 21:34.40 | ManxPower | gambolputty, but nothing for the microphones |
| 21:34.59 | Nohair | manxpower Ver 0.59q |
| 21:35.04 | ManxPower | TokyoJimu, where is your disallow=all and allow=ulaw. |
| 21:35.26 | ManxPower | Nohair, uninstall that mpg123 and in the asterisk source directory type "make mpg123" You have the wrong verison of mpg123 |
| 21:35.55 | TokyoJimu | I thought I'd start by allowing anything. No errors about unsupported formats, but I'll add it and see if it helps. |
| 21:35.59 | terrapen | Manx: http://www.pastebin.com/237306 |
| 21:36.08 | Nohair | Manxpower thanks whats the correct version |
| 21:36.09 | ManxPower | TokyoJimu, allowing all makes nothing work |
| 21:36.24 | ManxPower | Nohair, 0.59r, which is what "make mpg123" will download and install. |
| 21:36.24 | _Brian | Nohair: make mpg123 will download the correct version etc |
| 21:36.27 | TokyoJimu | Oh, OK. Thanks. I'll try this. |
| 21:36.39 | Delmar | TokyoJimu, whats the SIP client...that is using that ? |
| 21:36.58 | fa | Notice: Configuration file is /etc/zaptel.conf |
| 21:36.59 | fa | line 2: No such tone zone known: pl |
| 21:36.59 | fa | 1 error(s) detected |
| 21:36.59 | *** join/#asterisk ToyMan (~konversat@user-12lcqq2.cable.mindspring.com) |
| 21:37.08 | Nohair | manxpower how do i remove the wrong version |
| 21:37.23 | TokyoJimu | A Cisco ATA-186. |
| 21:37.30 | Delmar | ah right... ManxPower i didn't know that allow=all would break stuff.. thats handy to know. |
| 21:37.32 | terrapen | nohair, find t he binary |
| 21:37.34 | terrapen | and delete it |
| 21:37.49 | afrosheen | Nohair: slocate -u ; locate mpg123 |
| 21:38.04 | terrapen | % which mpg123 |
| 21:38.26 | terrapen | if its in your $PATH... |
| 21:38.30 | terrapen | which it probably is |
| 21:38.41 | fa | I have that |
| 21:38.41 | _Brian | ( cd / ; find . -name mpg123 -exec rm {} \; ) |
| 21:38.42 | Delmar | locate mpg123 |more |
| 21:38.47 | fa | Channel 01: Individual Clear channel (A-law) (Slaves: 01) |
| 21:38.47 | fa | Channel 02: Individual Clear channel (A-law) (Slaves: 02) |
| 21:38.47 | fa | Channel 03: D-channel (Default) (Slaves: 03) |
| 21:38.47 | fa | 3 channels configured. |
| 21:38.59 | Nohair | terrapen afrosheen found it |
| 21:39.02 | fa | but, when try to call i have Feb 3 22:38:51 NOTICE[11843]: app_dial.c:762 dial_exec: Unable to create channel of type 'Zap' |
| 21:39.05 | fa | why ? |
| 21:39.21 | TokyoJimu | But remember that it works fine w/o NAT, so it doesn't seem that formats is a problem. But I'll still try it. |
| 21:39.36 | Delmar | could be a few reasons for that fa. |
| 21:40.23 | ManxPower | TokyoJimu, no firewall involved? |
| 21:40.38 | terrapen | i wish there was a way to buy someone a soda or coffee over the internet |
| 21:40.39 | Delmar | fa, send your zapata.conf to pastebin and show me. |
| 21:40.48 | Delmar | fa, also your extensions.conf |
| 21:40.54 | terrapen | because i will buy someone a cappucino for helping me get this Polycom going |
| 21:40.56 | *** part/#asterisk BozzaCamilleri (~connect@213.217.225.32) |
| 21:40.58 | afrosheen | terrapen: I'm still looking for a 'remote beating' service |
| 21:41.05 | terrapen | heh, afro |
| 21:41.18 | terrapen | ok, i'll paypal $5 |
| 21:41.19 | TokyoJimu | MaxPower: Yes, a firewall on the asterisk side, allowing tcp and udp to the SIP port. |
| 21:41.19 | fa | Delmar zapata.conf http://pastebin.ca/5237 |
| 21:41.37 | fa | Delmar extensions.conf http://pastebin.ca/5238 |
| 21:41.45 | Delmar | afrosheen yeah. the threaten to beat with a big stick service (press 1) for the actual beating Press 2...if you would like the bondage version press 3. Muhaha |
| 21:42.01 | terrapen | $5 to whoever can help me fix this: http://www.pastebin.com/237307 |
| 21:42.02 | *** join/#asterisk xkev (kevin@orbit.xmission.com) |
| 21:42.03 | terrapen | :) |
| 21:42.03 | Nohair | maxpower have you used an Ixay |
| 21:42.10 | terrapen | oooo xmission |
| 21:42.11 | ManxPower | TokyoJimu, audio does not use udp port 5060 |
| 21:42.12 | Delmar | ok hang on fa. |
| 21:42.26 | terrapen | i miss living in UT and having xmission |
| 21:42.27 | ManxPower | Nohair, yes, but not in many months |
| 21:42.47 | TokyoJimu | Oh, what does it use? |
| 21:42.52 | xkev | is there an app that can check 'this file exists'? |
| 21:43.04 | xkev | so I can Exists(file.wav); Playback(file.wav); |
| 21:43.14 | xkev | erm not .wav on playback of course |
| 21:43.18 | ManxPower | TokyoJimu, a dynamically allocated set of ports. Asterisk defaults to UDP 10,000 - 20,000 and the Ciscos use 16,384 - 32,768 |
| 21:43.28 | terrapen | xkev: AGI? |
| 21:43.42 | TokyoJimu | Ahh, what a mess. But that definitely explains the problem. |
| 21:43.48 | Slainte | Manx, I have the NoOp working, but all I get for buys or bogus numbers is CAUSECODE=1 |
| 21:43.51 | xkev | terrapen that seems excessive :) |
| 21:43.52 | Slainte | Uunallocated |
| 21:43.54 | ManxPower | TokyoJimu, Welcome to the world of voIP |
| 21:43.56 | terrapen | yep :) |
| 21:44.00 | xkev | I could just do a Perl() then |
| 21:44.07 | ManxPower | Slainte, causecode 1??? what version of Asterisk are you using?? |
| 21:44.12 | xkev | seems like I should write an app_exists.c |
| 21:44.20 | Slainte | 1.05 |
| 21:44.40 | ManxPower | Slainte, Yes. Now you have to use a gotoif to play a "number disconnected" message when Causecode = 1 |
| 21:44.55 | ManxPower | You'll also have to handle BUSY, which I THINK is 16 or 17 CAUSECODE |
| 21:45.02 | nestAr | WHEEE |
| 21:45.04 | nestAr | i have PRI's |
| 21:45.16 | Slainte | Manx, why does the table say Unallocated? Are the numbers different now? |
| 21:45.20 | xkev | I have a causecode macro. want it? |
| 21:45.20 | afrosheen | I have a good iaxtrunk provider finally :) |
| 21:45.22 | TokyoJimu | MaxPower: Thanks for pointing me in the right direction. I've also ordered an IAXy to compare. |
| 21:45.26 | ManxPower | Slainte, Huh? |
| 21:45.29 | Slainte | xkev, please |
| 21:45.30 | terrapen | afro: who? |
| 21:45.35 | afrosheen | terrapen: txlink.net |
| 21:45.44 | terrapen | ah |
| 21:45.51 | afrosheen | I think we have a 5ms ping to their iax trunk box |
| 21:45.57 | terrapen | i thought about them...i forget why i didn't go that way |
| 21:45.58 | afrosheen | it's crazy |
| 21:45.59 | terrapen | wow |
| 21:46.02 | Slainte | http://www.voip-info.org/wiki-Asterisk+variable+hangupcause says causecode=1 is UNALLOCATED |
| 21:46.02 | terrapen | where are you? |
| 21:46.09 | afrosheen | dallas/richardson |
| 21:46.10 | terrapen | what's their box IP? |
| 21:46.13 | terrapen | ah |
| 21:46.15 | terrapen | i'm in SATX |
| 21:46.29 | jarrod | what kinda ip phones are you guys using |
| 21:46.32 | afrosheen | yeah they serve this area |
| 21:46.43 | ManxPower | Slainte, look at path/to/asterisk/include/asterisk/causes.h or something like that for the REAL numbers. |
| 21:46.46 | Slainte | Poly IP600 |
| 21:46.48 | terrapen | jarrod, i'm *trying* to use a Polycom...can't get it going yet.... |
| 21:46.50 | afrosheen | jarrod: we're 100% polycom |
| 21:47.00 | afrosheen | 300/500/600 |
| 21:47.03 | vaewyn | w0000!!!! I got money to hit up VON! schweeetttt! |
| 21:47.09 | terrapen | can I see somebody's iax.conf snippet for an IP600? |
| 21:47.12 | Slainte | Manx, coool thanks |
| 21:47.16 | `Sauron | What's the correct syntax for a SIP dial string? |
| 21:47.27 | `Sauron | I've seen 2 different versions, and neither is working in this case |
| 21:47.27 | terrapen | i'm getting translation issues from SIP to IAX2 |
| 21:47.28 | afrosheen | terrapen: we're having * handle iax, the phone itself is still sip |
| 21:47.30 | ManxPower | Slainte, unallocated is a number that uis not in service. |
| 21:47.30 | vaewyn | terrapen: you mean sip.conf correct? |
| 21:47.31 | jarrod | werd |
| 21:47.34 | jarrod | ive got some polycoms |
| 21:47.35 | jarrod | i like them |
| 21:47.36 | terrapen | err |
| 21:47.37 | Slainte | ahh makes sense now |
| 21:47.40 | terrapen | yes, sip.conf durrr |
| 21:47.44 | ManxPower | Slainte, looks like the wiki list is correct. |
| 21:47.46 | terrapen | my bad :) |
| 21:47.49 | afrosheen | terrapen: nat or no |
| 21:47.53 | terrapen | no NAT |
| 21:47.55 | Slainte | I have problems with the # on my polycoms |
| 21:48.06 | afrosheen | terrapen: ever try using AMP? |
| 21:48.08 | ManxPower | `Sauron, Dialing a phone or a SIP service provider? |
| 21:48.10 | Slainte | I cant use it to park a call |
| 21:48.16 | terrapen | AMP? never heard of it. |
| 21:48.18 | `Sauron | SIP gateway |
| 21:48.19 | afrosheen | a drooling retard can admin asterisk after amp is installed |
| 21:48.22 | xkev | slainte, sanitizing.. one moment |
| 21:48.27 | afrosheen | and we have one here |
| 21:48.31 | Slainte | xkev, thanks m8 |
| 21:48.39 | jarrod | wheres a good place to get some ip600's |
| 21:48.41 | afrosheen | amp.voxbox.ca |
| 21:48.41 | ManxPower | `Sauron, no idea. Traditionally it's SIP/username@sipconfentry/extension |
| 21:48.43 | _Brian | afrosheen: does a drooling retart come with AMP? |
| 21:48.44 | Slainte | AMP is the voxbox.ca app |
| 21:48.45 | jarrod | (not ebay?) |
| 21:48.47 | ManxPower | But I don't know if that would apply to SIP |
| 21:48.59 | afrosheen | _Brian: not unless you wanna fly me ^^^him out there |
| 21:49.03 | Slainte | Management front end |
| 21:49.08 | _Brian | :) |
| 21:49.24 | `Sauron | hum, ahha |
| 21:49.46 | `Sauron | Manx: Like this? |
| 21:49.47 | `Sauron | exten => s,1,Dial(SIP/auscm01/49914,30) |
| 21:49.50 | Slainte | my mp3 playback is garbbled. I read somewhere that the T100P can effect it. any ideas? |
| 21:50.33 | jaiger | does anyone use WAV or GSM for music on hold? |
| 21:50.36 | `Sauron | auscm01 is defined in sip.conf |
| 21:50.40 | terrapen | Feb 3 21:46:45 WARNING[2392]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/poly1-1be4(256) to IAX2/NuFone/1(512) |
| 21:50.41 | terrapen | fucker. |
| 21:50.50 | *** join/#asterisk Corydon-w (~tilghman@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 21:52.36 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 21:52.43 | ManxPower | terrapen, "show codecs" |
| 21:53.20 | terrapen | http://www.pastebin.com/237306 |
| 21:53.24 | terrapen | there you go, manx |
| 21:53.28 | ManxPower | You apparently are trying to use G729 without a license or you have allow=all or you have a bandwidth= |
| 21:53.37 | jarrod | where can i buy some polycom phones |
| 21:53.38 | `Sauron | Manx: When I use ,Dial(SIP/auscm01/49914,30), I get chan_sip.c:1714 create_addr: No such host: auscm01/49914 |
| 21:53.40 | `Sauron | Any ideas? |
| 21:53.48 | *** join/#asterisk mflorell (~mattf@rrcs-24-173-158-34.se.biz.rr.com) |
| 21:53.55 | Nohair | manxpower I have re installed mpg123 but it wont work now |
| 21:54.24 | ManxPower | sambal, do you have a [auscm01] set up as either a type=peer or type=friend. |
| 21:54.24 | terrapen | manx, reload that pastebin |
| 21:54.28 | *** join/#asterisk ChulJin (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net) |
| 21:54.31 | terrapen | put my sip.conf section in there |
| 21:54.39 | terrapen | shouldn't be using g729 at all |
| 21:54.49 | `Sauron | ManxPower: type=friend |
| 21:54.55 | ManxPower | terrapen, Well you are allowing it somwhere. |
| 21:55.04 | ManxPower | terrapen, I do not see your sip.conf |
| 21:55.32 | terrapen | at the bottom? lines 28-43 |
| 21:55.33 | ManxPower | `Sauron, does "sip show peers" show auscm01? |
| 21:55.36 | *** join/#asterisk JunK-Y (~grepmoo@65.39.228.5) |
| 21:55.40 | `Sauron | Yes. |
| 21:55.53 | ManxPower | terrapen, create a new one. I only get line 22 |
| 21:56.02 | JunK-Y | which recents to res_agi, GET VARIABLE should give the same value as ${UNIQUEID} ? |
| 21:56.07 | ManxPower | `Sauron, have you tried the format I gave you? |
| 21:56.13 | `Sauron | with OK (XX ms) |
| 21:56.31 | mikegrb | ManxPower: you have to click the comment where it says terrapen has added a comment |
| 21:56.38 | ManxPower | JunK-Y, historically you cannot get autocreated variables via AGI |
| 21:56.38 | Slainte | Nohair: When you say wont wok what do you mean? |
| 21:56.40 | `Sauron | Hum, let me try |
| 21:56.47 | xkev | Slainte, pri cause macros and such: http://orbit.xmission.com/~kevin/ext_dialing.conf (I #include this file) |
| 21:56.57 | terrapen | http://www.pastebin.com/237318 |
| 21:57.10 | xkev | you will also need http://orbit.xmission.com/~kevin/asterisk-sounds-cause.tar.gz (untar with cwd=/var/lib/asterisk/sounds) |
| 21:57.19 | xkev | they are all allison voices |
| 21:57.20 | JunK-Y | ManxPower: we can now with recent cvs. |
| 21:57.26 | Slainte | Thanks xkev |
| 21:57.37 | xkev | one of these days I'll wiki some of my trickery |
| 21:57.43 | JunK-Y | GET VARIABLE CALLERIDNUM works, like any other pre-defined vars. |
| 21:57.57 | ManxPower | terrapen, in [general] put context=INVALID |
| 21:58.05 | JunK-Y | it seems there's just a small problem with UNIQUEID |
| 21:58.20 | ManxPower | JunK-Y, They must have fixed that recently. post a bug report. |
| 21:58.23 | `Sauron | Hum. |
| 21:58.24 | `Sauron | Grr. |
| 21:58.32 | Yoda-BZH`ZzZ | une bonne nuit je vous souhaite ! / A good night I wish you |
| 21:58.32 | JunK-Y | will do. |
| 21:58.36 | terrapen | wooo |
| 21:58.36 | `Sauron | I got: -- Got SIP response 500 "Internal Server Error" back from 10.20.10.38 |
| 21:58.40 | terrapen | manx. |
| 21:58.46 | `Sauron | Now to find out why the hell I'm getting that. |
| 21:58.47 | terrapen | you da man. |
| 21:58.55 | ManxPower | `Sauron, what is 10.20.10.38 |
| 21:59.39 | `Sauron | Cisco Call Manager box |
| 21:59.55 | `Sauron | You don't even have to say anything... ;) |
| 22:00.18 | *** join/#asterisk yogurt2ungue (~yogurt2un@host15.201-252-158.telecom.net.ar) |
| 22:00.22 | ManxPower | ..er.. `Sauron |
| 22:00.50 | Juggie | does cisco even support SIP? |
| 22:00.51 | ManxPower | I think I'm going to start charging for support. |
| 22:00.56 | *** join/#asterisk hardwire (~hardwire@209.112.194.45) |
| 22:01.01 | hardwire | file: seen the Snom 360? |
| 22:01.05 | `Sauron | :) |
| 22:01.15 | xkev | 360? |
| 22:01.56 | xkev | ahhh hell, I left my iaxtel password in there |
| 22:02.12 | tzanger | :-) |
| 22:02.25 | xkev | too late :) |
| 22:02.31 | tzanger | no worries |
| 22:02.33 | tzanger | ttyl |
| 22:03.14 | ManxPower | desperate people will frequently actually pay real money to fix something. |
| 22:03.26 | `Sauron | hehe |
| 22:03.32 | `Sauron | That's how I make money :) |
| 22:03.46 | xkev | that reminds me, I have a guy who wants to pay me to fix his crap :) |
| 22:03.53 | hardwire | http://www.snom.com/snom360_voip_phone.html?&L=1 |
| 22:04.02 | hardwire | SIP(S) |
| 22:04.03 | hardwire | :) |
| 22:04.06 | afrosheen | when is polycom making new sip phones |
| 22:04.12 | xkev | ooh that's a pretty phone |
| 22:04.19 | ManxPower | `Sauron, The problem is that usually the people that are desperate enough to pay are assholes that don't listen to advice and instructions anyway. |
| 22:04.27 | xkev | afrosheen the ip600 works pretty damn good for me. It could use some presence lights though |
| 22:04.43 | terrapen | i'm an asshole who doesn't listen to advice and instructions? |
| 22:04.45 | afrosheen | xkev: ours just acts funny sometimes, like it forgets it's registered or something |
| 22:04.53 | xkev | they have bugs alright |
| 22:05.08 | afrosheen | xkev: another new sip.ld is rumored to be coming very soon |
| 22:05.12 | afrosheen | 1.3.4 is the newest |
| 22:05.20 | xkev | like I set my min rtp port to 65500 (since it doesn't have a max) and it rolls past 65535 and uses port 0 |
| 22:05.23 | ManxPower | terrapen, I was not SPECIFICALLY referring to you. |
| 22:05.27 | terrapen | ok :) |
| 22:05.30 | xkev | I need to track me down a 1.3.4, it fixes some bugs I have |
| 22:05.38 | ManxPower | terrapen, You gave me $5, that's not real money. That's the cost of a Latte. |
| 22:05.46 | terrapen | exactly. |
| 22:05.51 | afrosheen | xkev: we have it, 1.3.4 isn't supposed to introduce any new features at all |
| 22:06.01 | afrosheen | ManxPower: lol |
| 22:06.04 | terrapen | but was it not worth it? |
| 22:06.07 | afrosheen | a real latte |
| 22:06.10 | ManxPower | Anywhere else and I would have considered it an insult. But this is Asterisk. |
| 22:06.17 | xkev | afrosheen yeah, like that ringing bug I have seen a few times |
| 22:06.18 | terrapen | exactly :) |
| 22:06.19 | mflorell | http://www.freedomphones.net/polycom/files/ |
| 22:06.25 | mflorell | firmware 1.4.1 |
| 22:06.29 | terrapen | i answer people's unix questions all the time for free |
| 22:06.34 | xkev | oh it's on freedomphones. I haven't checked there since the day after it came out |
| 22:06.38 | ManxPower | terrapen, If you were a customer, and we spend 30 mins on this, you would have gotten a bill for $60 if I liked you, or $120 if I didn't like you. |
| 22:06.46 | afrosheen | mflorell: huh, what's the date on that |
| 22:06.48 | terrapen | because i've been doing unix for 13 years and i have to do diddly squat to answer the questiono |
| 22:06.58 | mflorell | not sure, let me check |
| 22:07.06 | *** join/#asterisk WifiFred (~wififred@apollo.bcwireless.net) |
| 22:07.06 | xkev | hrm 1.4.1 /me looks for release notes |
| 22:07.31 | vaewyn | So now many of you guys are gonna be at VON? |
| 22:07.33 | terrapen | sure, if a client called me and wanted unix support, i'd bill $100/hr but if someone asked a question on IRC and I answered it, i'd not expect payment |
| 22:07.34 | xkev | $120 an hour eh |
| 22:07.37 | vaewyn | how even |
| 22:07.40 | xkev | I'm only $100 :) |
| 22:07.43 | ManxPower | EGADS! Valentines Day is coming up soon. |
| 22:07.50 | sivana | ~seen normast |
| 22:07.52 | jbot | normast <HydraIRC@Ottawa-HSE-ppp4122264.sympatico.ca> was last seen on IRC in channel #asterisk, 7h 30m 5s ago, saying: 'Nivex: I guess it's them same on most distro however..'. |
| 22:07.55 | mflorell | build date of the sip image is 14-Dec-04 11:54 |
| 22:07.56 | terrapen | and if somebody sent me latte money, i'd be ecstatic |
| 22:08.03 | ManxPower | terrapen, Oh, I don't EXPECT payment. Which is good. |
| 22:08.08 | `Sauron | y'all are cheap, bastards. |
| 22:08.10 | afrosheen | mflorell: weird |
| 22:08.11 | terrapen | because i've answered a million questions on IRC and never gotten jack shit for it |
| 22:08.24 | ManxPower | terrapen, "it's the thought that counts" DOES count for something. |
| 22:08.32 | afrosheen | it's all about your rep in a channel |
| 22:08.44 | vaewyn | I've never gotten jack for answering questions.. but I have gotten shit so... |
| 22:08.45 | vaewyn | ;P |
| 22:08.47 | xkev | terrapen, you need a script to accept and verify credit card numbers and you'll dcc chat undistracted for as long as they want :) |
| 22:08.47 | `Sauron | afroshmen: That's a load of crap. |
| 22:08.59 | terrapen | i just get so frustrated because i'm very good a *nix but such a fucking * n00b that i annoy myself sometimes |
| 22:09.04 | terrapen | its not fun to be new at something again |
| 22:09.15 | `Sauron | Cuz there's always n00bs coming in who don't have a clue, and think they're all that - when in reality they don't know jack |
| 22:09.21 | `Sauron | they don't care if you have a rep or not |
| 22:09.37 | afrosheen | they learn soon enough |
| 22:09.47 | xkev | um looks like 1.4.1 requires bootrom 2.6.0, but 2.5.0 is on freedomphones |
| 22:09.47 | afrosheen | after everyone says 'ask sauron' |
| 22:09.48 | `Sauron | If they stay around long enough |
| 22:09.55 | ManxPower | I just changed my /away message from "I am not your Personal Asterisk Support Bitch!" to something a little nicer. |
| 22:10.04 | xkev | oh /me notes the differently-named 2.6.1 |
| 22:10.07 | jarrod | manx thnx |
| 22:10.09 | terrapen | hah |
| 22:10.21 | mflorell | http://www.freedomphones.net/polycom/files/bootrom_2_6_1.zip |
| 22:10.26 | xkev | yeah |
| 22:10.34 | xkev | lousy changing filenames |
| 22:10.41 | mflorell | tell me about it |
| 22:10.45 | xkev | huh? there's a 501 and a 301 too? |
| 22:10.49 | xkev | (models) |
| 22:10.49 | *** join/#asterisk freat[laptop] (~freat[lap@204.118.23.66) |
| 22:10.57 | ManxPower | There. updated. |
| 22:11.04 | terrapen | i answer unix-related questions on IRC for free, in gratitude for all of my questions that were kindly answered, back in the day |
| 22:11.13 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 22:11.14 | terrapen | everybody was a n00b once. |
| 22:11.18 | xkev | true dat |
| 22:11.23 | `Sauron | Manx: Is that paid in advance? |
| 22:11.49 | ManxPower | `Sauron, Of course. Do I LOOK gullible? |
| 22:11.52 | *** join/#asterisk harryvv (~noyb@S010600055d210201.vs.shawcable.net) |
| 22:11.54 | `Sauron | Like, cuz I won't pay you $40 to help me with CCM for 30 minutes, when you refuse to help :) |
| 22:11.55 | terrapen | mflorrel, be careful, the latest firmware from that site did not load on my IP600 |
| 22:12.00 | junky[work] | Thu Feb 3 17:10:33 2005 1107468633.120 GET FULL VARIABLE UNIQUEID |
| 22:12.01 | junky[work] | Thu Feb 3 17:10:33 2005 1107468633.120 res=Allow: INVITE, ACK,) |
| 22:12.01 | terrapen | i had to use the next-to-latest firmware |
| 22:12.03 | junky[work] | huh? |
| 22:12.20 | harryvv | Under no circumstances can the zapcard share a irq with another device like vidio card? |
| 22:12.40 | ManxPower | hardwire, that is correct if you want it to be even close to reliable. |
| 22:12.40 | harryvv | I am now getting errors and think its a irq sharing issue. |
| 22:12.41 | terrapen | why would you want it to share an IRQ? |
| 22:12.42 | xkev | "removed idle display microbrowser configuration" WHAT THE HELL? |
| 22:12.56 | mflorell | I believe some of the newer ip600s have more memory than older versions |
| 22:13.02 | ManxPower | hardwire, but most video cards can have the IRQ disabled in the motherboard bios |
| 22:13.06 | terrapen | mflorrell: yep, aparrently so |
| 22:13.08 | harryvv | terrapen, thats just it...I dont and my asterisk has worked really well untill now. |
| 22:13.08 | hardwire | are you crqzy |
| 22:13.14 | `Sauron | "Oh dear beloved ManxPower, I would like help writing wonderful love letters to my boyfriend. Enclosed, find $40 for half an hour's letter. Thanks, Bobby Jo" |
| 22:13.15 | hardwire | ManxPower: :) |
| 22:13.19 | hardwire | you are crazy. |
| 22:13.22 | terrapen | my IP600 did not have enough memory for the l8est firmware |
| 22:13.29 | afrosheen | terrapen: how can that be |
| 22:13.34 | xkev | I use that idle browser. anyone running 1.4.1 yet? am I reading this wrong or is it gone? |
| 22:13.36 | afrosheen | get a chop-shop model? |
| 22:13.49 | terrapen | afro: i just got a brand-fucking-new phone from voipsupply |
| 22:13.50 | afrosheen | xkev: the browser is still there, the idle mode is gone |
| 22:13.56 | xkev | that's stupid |
| 22:14.07 | ManxPower | Dear Bobby Jo, I wrote the requested letter and your boyfriend will be calling to dump you within the hour and live with me as my lovepuppet. |
| 22:14.16 | `Sauron | hehe |
| 22:14.30 | terrapen | i didn't save a boot log from the attempted upgrade |
| 22:14.36 | terrapen | but it fails for lack of disk space |
| 22:14.41 | Nohair | Any one using Asterisk on fedora core 3 |
| 22:14.52 | xkev | terrapen, then it reverts ok or becomes a doorstop? |
| 22:15.06 | terrapen | it just reboots |
| 22:15.07 | terrapen | over and over. |
| 22:15.12 | terrapen | (thank fucking god) |
| 22:15.20 | xkev | ..until you put the old sip.ld back I assume |
| 22:15.24 | afrosheen | yeah polycoms rarely become so hosed you can't fix them |
| 22:15.31 | terrapen | for that i am thankful |
| 22:15.35 | xkev | aight |
| 22:15.36 | afrosheen | I've messed the ones here up pretty good before |
| 22:15.46 | afrosheen | jesus look at the config file changes... |
| 22:15.52 | *** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 22:16.30 | afrosheen | oh, most of the new stuff is for the sp4000 |
| 22:16.49 | jarrod | have the ip600's been discontinued? |
| 22:16.51 | xkev | you guys ever figure out how they implemented the sip dialog event (like snom busy lamps subscribe to) that they list compliance with at teh end of 1.3.1 admin guide? |
| 22:17.23 | afrosheen | jarrod: I doubt it |
| 22:17.33 | afrosheen | jarrod: but I expect polycom to introduce new models today |
| 22:17.35 | xkev | ..at some point they even had a doc with samples of how it does certain subscribe/notify features, but they pulled it out |
| 22:17.37 | afrosheen | or uh this year |
| 22:17.45 | xkev | (at least there was a paste in the list archive) |
| 22:18.12 | harryvv | what is the default irq for zap? |
| 22:18.13 | xkev | "Receiving a check-sync can cause the file system to be reformatted on a SountPoint IP 500" cute |
| 22:18.46 | afrosheen | xkev: yeah lovely innnit |
| 22:19.00 | *** join/#asterisk niZon (ilt@S0106deadbeef6977.wp.shawcable.net) |
| 22:19.03 | afrosheen | harryvv: there isn't one, it varies with your motherboard and setup |
| 22:19.07 | xkev | I check-sync'd my IP500 a lot and it never did that |
| 22:19.37 | harryvv | afros okay |
| 22:20.15 | *** join/#asterisk florz (nobody@odnb-d9baa586.pool.mediaWays.net) |
| 22:20.41 | afrosheen | harryvv: if you have decent hardware, the zaptel card will (you hope) get an irq all to itself |
| 22:21.10 | florz | is there BTW anyone in here who has some irssi script that makes sure nickserv auth completes before joining channels? |
| 22:21.51 | florz | it's really annoying to have to join #asterisk manually after every reconnect :-) |
| 22:22.49 | *** join/#asterisk mh720 (~mike@nwcorp-fw.nationwide.net) |
| 22:22.59 | mh720 | how-D |
| 22:22.59 | harryvv | i afroshen, just a number9/3com/esoniq 128/zap |
| 22:23.01 | afrosheen | florz: I know, I have that issue sometimes |
| 22:23.03 | *** part/#asterisk jgaviria (~juan_manu@63.245.86.116) |
| 22:23.15 | *** join/#asterisk DrPete (~Pete@82-40-26-136.cable.ubr04.uddi.blueyonder.co.uk) |
| 22:23.17 | afrosheen | harryvv: as long as it's not sharing that irq you should be good |
| 22:23.52 | Nohair | Any one knwo where to get the lastest iaxy software |
| 22:24.17 | afrosheen | Nohair: doesn't it come with *? |
| 22:24.25 | Slainte | xkev: what is in the stream directory in your cause code macro? |
| 22:24.53 | Slainte | xkev: ast_openstream: File silence/1 does not exist in any format |
| 22:24.58 | afrosheen | holy crap, fox news bought al-jazeera tv today |
| 22:25.08 | mh720 | ?: does Monitor() only work with Zap channels? app_chanspy for SIP only? |
| 22:25.26 | Slainte | Does that mean they will make an Arabic version of The Simpsons? |
| 22:25.29 | afrosheen | mh720: you have barge with zap channels also |
| 22:25.44 | afrosheen | Slainte: who knows, it sounds like a nutty acquisition |
| 22:25.48 | florz | afrosheen: But no solution either? =:-) |
| 22:26.05 | Nohair | afrosheen yes but i understand there is a later version than the one i have |
| 22:26.21 | afrosheen | Nohair: then it will be in a newer asterisk right |
| 22:26.39 | afrosheen | cvs or head |
| 22:26.47 | mh720 | afro, thx, so Monitor only works on Zap technology channels? |
| 22:27.03 | afrosheen | mh720: no clue..I'm still a noob |
| 22:27.44 | DrPete | netsurfer: hi |
| 22:27.46 | mh720 | heh arent we all :) |
| 22:27.52 | netsurfer | hi DrPete |
| 22:28.04 | DrPete | netsurfer: card here yet? |
| 22:28.06 | mtqh | No Monitor works on all channels |
| 22:29.12 | netsurfer | DrPete - nope :( |
| 22:29.14 | *** join/#asterisk RoyK (~roy@83.80-203-29.nextgentel.com) |
| 22:31.05 | harryvv | since my zap for some od reason out of the blue is using the same irq as my vidio monitor irq 9 what is the fastest way to change it? |
| 22:31.13 | xkev | move slots |
| 22:31.18 | afrosheen | harryvv: yeah, move it over |
| 22:31.20 | mh720 | hmmmm.. ok, thanks mtqh, I read include/asterisk/monitor.h but only found reference to it in channel.c |
| 22:31.29 | xkev | disable video irq if you don't need it, or get an APIC board :) |
| 22:31.38 | xkev | apic is love |
| 22:31.56 | mh720 | I followed the wiki for Monitor() but no files get written, strange |
| 22:32.11 | mh720 | and I'm reasonably old-school with * |
| 22:32.33 | harryvv | kdev my asterisk been working for a week now reliably this this problem poped up this morning for no reason. |
| 22:32.39 | afrosheen | apic is love until you get a 2.6.x kernel that doesn't like it |
| 22:32.49 | cypromis | hehe |
| 22:33.25 | redder86 | off hand do you know how to embed a PDF into a browser window? |
| 22:33.33 | redder86 | does it need to take up the whole window? |
| 22:33.40 | terrapen | ok, i'm excited now...the IP600 is working now...finally, i feel like i'm getting somewhere |
| 22:33.43 | xkev | afro, I'm runnin 2.6.9-mm1 on a supermicro and it's superb |
| 22:33.52 | terrapen | now, i need to learn more about its config files |
| 22:33.57 | afrosheen | yeah I heard .9 was better with acpi |
| 22:33.58 | harryvv | kdev, whats a apic board? |
| 22:34.05 | mtqh | mh720: what are you trying to do |
| 22:34.06 | *** join/#asterisk Godot (~dr_schnac@pD9EC7E46.dip.t-dialin.net) |
| 22:34.11 | xkev | you have lots and lots and lots of interrupts, harry |
| 22:34.19 | xkev | XT-PIC versus APIC |
| 22:34.26 | mh720 | mtqh, just very basic record any outgoing call for starters |
| 22:34.36 | mtqh | show application monitor |
| 22:34.41 | afrosheen | terrapen: do most of the basic config through the phone's interface first |
| 22:34.52 | xkev | "formatting file system..." |
| 22:35.05 | mh720 | exten => 100,1,Wait,1 |
| 22:35.05 | mh720 | exten => 100,2,Playback(transfer,skip) |
| 22:35.05 | mh720 | exten => 100,4,Monitor(wav,mywavfile,mb) |
| 22:35.05 | mh720 | exten => 100,5,Dial(SIP/100,25) |
| 22:35.05 | mh720 | exten => 100,6,StopMonitor |
| 22:35.14 | mh720 | lol |
| 22:35.16 | mtqh | mb? |
| 22:35.17 | mh720 | where is 3? |
| 22:35.22 | afrosheen | profit! |
| 22:35.29 | mtqh | mh720 that would be an issue was well |
| 22:35.32 | *** join/#asterisk AgiNamu (~AgiNamu@200.12.43.74) |
| 22:35.57 | AgiNamu | Hey, numbers can't be longer than 3 + 15 right? |
| 22:36.02 | AgiNamu | countrycode + national destination? |
| 22:36.03 | mh720 | m - mux the recordings using soxmix after the call, b - don't record unless the call is bridged (ie. dont record if the call doesnt go through) |
| 22:36.03 | DrPete | netsurfer: :( |
| 22:36.08 | *** join/#asterisk WiFiGuy (WiFiGuy@CPE-69-76-99-187.wi.rr.com) |
| 22:36.13 | mtqh | mh720I know |
| 22:36.21 | mh720 | I've tried without these options |
| 22:36.23 | terrapen | afro, that's what i've been doing...does the phone's interface override the phones FTP'd config files? |
| 22:36.24 | AgiNamu | I want to make sure I can get away with a VarChar(21) |
| 22:36.28 | xkev | aginamu it's best not to assume on international calls |
| 22:36.49 | terrapen | afro, and is there a way to get it to dump config files that include the stuff which i set up using the phone's interface? |
| 22:36.49 | AgiNamu | xkev there's gotta be some limit |
| 22:37.21 | afrosheen | terrapen: yeah the phone usually keeps that stuff in flash and will dump it to the config files by itself |
| 22:37.34 | afrosheen | terrapen: you're using ftp to store the configs right? |
| 22:38.07 | xkev | seems like 3+15 would cover it |
| 22:38.16 | xkev | (3 being e.g. 011 in us) |
| 22:38.18 | terrapen | well, the phone is fetching boot/firmware images and dumping logs via FTP |
| 22:38.23 | AgiNamu | er, 3 being 1 in USA |
| 22:38.26 | terrapen | does that mean that it is storing configs, too? |
| 22:38.29 | AgiNamu | USA cc is 1 |
| 22:38.31 | afrosheen | terrapen: yes |
| 22:38.39 | xkev | terrapen when you reload the phone, it writes the config, unless the config is newer |
| 22:38.41 | afrosheen | terrapen: check the times on it's MAC.cfg |
| 22:38.45 | AgiNamu | 011 is their silly international prefix thing |
| 22:38.49 | AgiNamu | ..right? |
| 22:38.51 | afrosheen | yeah |
| 22:38.56 | terrapen | <napoleon dynamite> SWEET. </napoleon dynamite> |
| 22:39.00 | xkev | aginamu I thought that's what you meant |
| 22:39.10 | AgiNamu | no, 1-3 digit area code |
| 22:39.13 | AgiNamu | er, country code |
| 22:39.21 | AgiNamu | and 15 digit national destination |
| 22:39.21 | afrosheen | xkev: please tell me you know how to get custom ringtones working on these phones |
| 22:39.22 | xkev | 011446688008800 is about the longest UK number I can think of dialing from bushville |
| 22:39.39 | AgiNamu | 011 doesnt count though |
| 22:39.40 | `Sauron | I would guess that phone numbers will generally be less than 40 characters long, guaranteed. |
| 22:39.42 | xkev | right |
| 22:39.43 | AgiNamu | that's just for your switch. I drop it |
| 22:39.47 | xkev | 15 digits should cover it yeah |
| 22:40.00 | AgiNamu | Sauron, yea, well, im trying to make my DB as small as possible |
| 22:40.12 | AgiNamu | and by declaring varchar(30) I waste a lot of space |
| 22:40.15 | xkev | 3digit country + 15 national dial area (that's still a long national dial area, heh) |
| 22:40.20 | terrapen | the Polycom's UI gives me a woody |
| 22:40.25 | AgiNamu | xkev yea... that's what E.164 says |
| 22:40.26 | terrapen | one of the best i've seen on any device |
| 22:40.42 | Mother_ | terrapen: better than Cisco? :> |
| 22:40.43 | xkev | ok uh |
| 22:40.45 | AgiNamu | cc=1-3 digits maximum 15-cc digits [International public telecommunication number for geographic areas (maximum 15 digits)] |
| 22:40.47 | terrapen | i'm dreading trying to get the Cisco 7960 to work |
| 22:40.49 | xkev | 1.4.1 I still have my idle microbrowser |
| 22:40.56 | terrapen | i was thinking about sending it back, even... |
| 22:40.56 | AgiNamu | so... varchar(18) yey! |
| 22:41.06 | Mother_ | terrapen: worry not, it works just fine |
| 22:41.10 | afrosheen | xkev: update the bootrom to 2.6.1? |
| 22:41.13 | xkev | yes |
| 22:41.31 | ManxPower | Well I can't seem to make shared call apperaances to work with the SPA-841 |
| 22:41.32 | Mother_ | plus when I hear "polycom" I get this weird rash.... |
| 22:41.35 | afrosheen | terrapen: the polycom config is sorely lacking |
| 22:41.48 | xkev | system status -> general says "app. version: 1.4.1.0040' |
| 22:41.50 | Slainte | xkev: ast_openstream: File silence/1 does not exist in any format |
| 22:41.52 | afrosheen | terrapen: compare it to the ipmid.cfg file and you'll see what I mean |
| 22:41.59 | afrosheen | brb |
| 22:42.15 | afrosheen | meantime, someone explain how to load custom ringtones on polycoms |
| 22:42.15 | xkev | slainte, you may need asterisk-sounds |
| 22:42.49 | ManxPower | the silence files are in asterisk-sounds as xkev says |
| 22:42.53 | xkev | that's just a pause-maker to give audio time to fire up, since you are not answering the call and Answer/Wait routine is not appropriate |
| 22:43.31 | xkev | slainte, I'll change my tar to include a silence.gsm and adjust the dialplan accordingly |
| 22:44.11 | Slainte | xkev, good one. I do have the /usr/share/asterisk/sounds but there is no silence subdir |
| 22:44.42 | ManxPower | Slainte, cvs co asterisk-sounds |
| 22:44.49 | xkev | slerp the new tar.gz I just copied over if you don't want to get asterisk-sounds |
| 22:44.59 | xkev | and adjust all "silence/1" to "cause/silence" |
| 22:45.18 | terrapen | the only thing i dislike about Polycom is that my ex-girlfriend's name is Polly |
| 22:45.28 | xkev | you dated someone named POLLY? :) |
| 22:45.31 | terrapen | so every time i see the word, i get an uncomfortable feeling in my stomach |
| 22:45.33 | terrapen | yeah :) |
| 22:45.36 | Slainte | is the silence a new sound? |
| 22:45.38 | xkev | ..want a cracker? :) |
| 22:45.39 | terrapen | she was a tall, cute blonde :) |
| 22:46.06 | xkev | well I suppose that makes up for it |
| 22:46.09 | `Sauron | blonde? |
| 22:46.24 | hermie | terrapen: forget about her because, trust me, she's forgot about you. :) |
| 22:46.47 | ManxPower | it is the sound of silence |
| 22:47.03 | terrapen | haha |
| 22:47.07 | terrapen | probably so :) |
| 22:47.08 | *** join/#asterisk adrianhensler (~chatzilla@blk-222-123-93.eastlink.ca) |
| 22:47.25 | hermie | terrapen: that's from the Rejection Hotline :) |
| 22:48.19 | hermie | get a number from rejectionhotline.com and call it some time |
| 22:50.10 | Nohair | <PROTECTED> |
| 22:50.42 | Slainte | can someone tell me what the bluetooth daemon is for? |
| 22:51.21 | ManxPower | Nohair, bugs.digium.com but if it's a MoH bug forget about it. |
| 22:52.02 | Nohair | hi Manxpower its bug with the Iaxy and Euro flashtimings |
| 22:52.13 | `Sauron | Why forget about MoH bugs? |
| 22:52.18 | afrosheen | so nobody knows about custom ringtones huh |
| 22:52.26 | Mother_ | hmm can I detect for an incoming fax on a shared line by answering and providing a ringing tone? |
| 22:52.47 | `Sauron | Mother: You can do |
| 22:52.53 | Mother_ | so that if the caller is human, they will still hear a ring |
| 22:52.57 | `Sauron | exten => fax,1,Dial(alternatefaxextension) |
| 22:53.05 | terrapen | word...mountain bike ride tonight |
| 22:53.08 | terrapen | its gonna be COLD |
| 22:53.14 | `Sauron | Exten => s,1,Dial(regularextension) |
| 22:53.18 | ManxPower | `Sauron, Because MoH bugs are usually user error. |
| 22:53.22 | `Sauron | faxes will be sent to the fax place |
| 22:53.28 | `Sauron | and users are dealt with elsewhere |
| 22:53.42 | redder86 | hermie: you know, that rejectionhotline number is worth the call just to listen to it |
| 22:53.45 | `Sauron | Manx: Isee |
| 22:53.52 | Mother_ | Sauron: OK, will that happen before and withouth having to send an Answer command? |
| 22:54.07 | `Sauron | I think so |
| 22:54.15 | `Sauron | check the docs |
| 22:54.28 | Mother_ | great, I'll try it out, I've checked the wiki and lists but it wasn't all that clear |
| 22:54.30 | `Sauron | or pay Manx to answer you :) |
| 22:54.33 | Mother_ | lol! |
| 22:54.50 | hermie | redder86: yah, I got it from somebody who heard it on the radio, not from any real rejection ;) |
| 22:54.56 | Mother_ | I'll just have a play with it, but didn't want to dwell into futility |
| 22:55.21 | Mother_ | at least I managed to manually patch the CVS and compile OK |
| 22:55.40 | harryvv | xkev, I was looking though bios and found a selection that said Vidio search can be either pci/agp or agp/pci and selected the second option. That was what probebly worked. |
| 22:55.48 | harryvv | I also notices irqs changed |
| 22:58.31 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 22:59.04 | *** join/#asterisk mrproper_ (~mrproper_@61.95.55.242) |
| 22:59.56 | *** part/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 23:00.07 | *** join/#asterisk srt (~nobody@gw0-cgn.reucon.net) |
| 23:00.35 | mrproper_ | I have finished a complete guide for asterisk on fedora core 1, can anyone tell me where i can put up this guide for everyone else? |
| 23:00.58 | afrosheen | mrproper_: in the voip-info wiki somewhere |
| 23:01.42 | *** join/#asterisk kFuQ (~somedude@c-24-17-224-78.client.comcast.net) |
| 23:02.19 | ManxPower | mrproper_, voip-info.org AND asteriskdocs.org |
| 23:02.58 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 23:03.08 | mrproper_ | thanks guys |
| 23:04.58 | eKo1 | Crap. Asterisk is dying on me. |
| 23:06.13 | eKo1 | WTF?! When I do a 'sip show channels' there are and I quote: 96 active SIP channel(s) |
| 23:06.28 | xkev | 0 active SIP channel(s) |
| 23:06.32 | eKo1 | Most of them from the same peer. |
| 23:06.41 | xkev | ahhh, I've seen old cruft hang around before |
| 23:06.48 | xkev | are they guest peers? |
| 23:06.51 | eKo1 | With format set to unknow(d) |
| 23:07.00 | afrosheen | eKo1: they should have ip |
| 23:07.03 | eKo1 | It's a peer. |
| 23:07.05 | afrosheen | ip's attached right |
| 23:07.11 | eKo1 | Yeah, it has an IP. |
| 23:07.16 | afrosheen | it or they? |
| 23:07.17 | eKo1 | It's the IP of my provider. |
| 23:07.21 | afrosheen | oh ok |
| 23:07.35 | xkev | I had some guy calling my music on hold from iptel, and it would leave all his sessions on their with no rtp |
| 23:07.46 | afrosheen | lol |
| 23:07.54 | afrosheen | did you have a local shoutcast stream or smth |
| 23:08.03 | xkev | I called him after like 6 hours and said "are you listening to all of these?" |
| 23:08.04 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 23:08.07 | eKo1 | This doesn't make sense since 'show channels' on shows: 22 active channel(s) |
| 23:08.24 | *** join/#asterisk bkw_ (~brian@65.38.28.146) |
| 23:08.25 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 23:08.26 | bkw_ | yo |
| 23:08.27 | xkev | we have a huge collection of mp3 we randomize. mostly ambient, some cookie monster and drunken sailors too |
| 23:08.29 | eKo1 | xkev: That happens when the phone doesn't hang up properly. |
| 23:08.36 | bkw_ | hey |
| 23:08.40 | eKo1 | I've had that happen. |
| 23:08.54 | xkev | hola bkw |
| 23:09.05 | bkw_ | I need someone to fax me a 10,20,30 and 50 page document to 877-2787565 |
| 23:09.08 | bkw_ | please please |
| 23:09.10 | bkw_ | please |
| 23:09.12 | eKo1 | OK, it just grew to 101. |
| 23:09.12 | xkev | whaaa? |
| 23:09.22 | xkev | I can send you 50 pages of line feeds |
| 23:09.28 | afrosheen | lol |
| 23:09.39 | afrosheen | that won't help, he's gotta see something |
| 23:09.51 | xkev | there's a fax->email gateway I can use. /me looks for a pdf to send |
| 23:09.56 | xkev | erm email->fax, duh |
| 23:10.20 | xkev | eKo1, you're being dossed |
| 23:10.22 | xkev | :) |
| 23:10.44 | jarrod | whats a good pc softphone that can act like a 'call manager' that an operator could use |
| 23:10.46 | xkev | bkw got a ps or pdf or somethin I can send you? |
| 23:11.00 | xkev | jarrod, none? |
| 23:11.27 | *** join/#asterisk venix (~venix@209.5.255.68) |
| 23:11.30 | jarrod | lame |
| 23:11.44 | harryvv | This comes up when booting unable to open pid file /var/run/asterisk permission denied got three hits on google. Do not make sence of one of them. |
| 23:11.47 | xkev | jarrod, I'm workin on this unit: http://orbit.xmission.com/~kevin/op.png ..it could easily tie to a local install of asterisk and use the sound card for answering calls |
| 23:12.10 | xkev | (run it on a touchscreen lcd) |
| 23:12.19 | afrosheen | jarrod: uhmm flash operator panel? |
| 23:12.40 | afrosheen | jarrod: it comes with AMP |
| 23:12.46 | jarrod | well so my operator can point and click |
| 23:12.54 | afrosheen | yeah that's FOP then |
| 23:12.56 | jarrod | instead of having to answer and transfer with the phone |
| 23:13.07 | eKo1 | I did a 'stop now' and I'm still on the CLI. |
| 23:13.10 | jarrod | wow |
| 23:13.13 | jarrod | xkev i like that |
| 23:13.29 | xkev | I can cook it up for you to attach to an asterisk -r or something |
| 23:13.41 | xkev | no guarantees it won't eat your first born though |
| 23:14.16 | afrosheen | he can have another one |
| 23:14.29 | xkev | jarrod, if you know linux development in C++ and perl, you could redesign it to suit your needs |
| 23:14.40 | terrapen | oh fuck! i just realized that i bought an IP500., not an IP600 |
| 23:14.42 | terrapen | doh! |
| 23:14.47 | xkev | terrapen hawz |
| 23:14.55 | xkev | ip600 is so much nicer too |
| 23:14.57 | terrapen | not that this is a bad phone... |
| 23:15.01 | jarrod | yes |
| 23:15.01 | terrapen | is it really? |
| 23:15.06 | terrapen | what does it do better? |
| 23:15.23 | xkev | yeah, the screen is twice as nice, and the mute/speaker/registration/etc light up |
| 23:15.24 | afrosheen | more blinking lights |
| 23:15.26 | jarrod | actually the languages i know very very well are c++ and php |
| 23:15.33 | jarrod | i can do enough perl to figure it out :) |
| 23:15.44 | terrapen | mahtrfacker. |
| 23:15.50 | terrapen | that sucks! |
| 23:15.57 | terrapen | how the hell did i buy a 500.... |
| 23:16.01 | xkev | jarrod, this is built in Qt designer. the perl is just a matter of configuring what the buttons should be. it's a thin client of sorts |
| 23:16.08 | afrosheen | terrapen: how much did you pay for it |
| 23:16.30 | terrapen | <PROTECTED> |
| 23:16.47 | afrosheen | that's the right price for a 500 |
| 23:16.47 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 23:16.51 | *** join/#asterisk tessier_ (~treed@146.82.146.22) |
| 23:16.54 | afrosheen | maybe a little cheaper |
| 23:17.16 | afrosheen | how much are the 600's going for now |
| 23:17.24 | xkev | I"ve seen em as low as $255 |
| 23:17.36 | xkev | more commonly $350-ish |
| 23:18.05 | afrosheen | yeah so a 600 for 195 would be a bargain |
| 23:18.22 | terrapen | yeah, they sent me the quote for the 500 and that's what i bought |
| 23:18.28 | terrapen | it's ok...i'll just get another 600 |
| 23:18.31 | afrosheen | it sucked trying to buy these last year, everyone was running out |
| 23:18.32 | ManxPower | I have never seen a place with really cheap polycoms actually be able to SHIP them |
| 23:18.36 | terrapen | i wish i could return this Cisco 7960 |
| 23:18.42 | terrapen | maybe i should sell it on ebay |
| 23:18.50 | DaLion2 | terra how much u want for it |
| 23:18.52 | afrosheen | yeah it'll bring money |
| 23:18.58 | terrapen | whatever i paid for it...its brand new |
| 23:19.00 | terrapen | lemme see |
| 23:19.27 | terrapen | i paid 250.00, including power cube |
| 23:19.39 | DaLion2 | ah got them less |
| 23:19.39 | terrapen | (compatible power cube, not cisco power cube) |
| 23:19.41 | DaLion2 | why u selling it |
| 23:19.55 | xkev | manxpower that is true. the place I found the cheapest seemed to actually have a proper inventory linkage to their storefront, as 2 days later there were 12 more |
| 23:19.57 | *** join/#asterisk toddf (klnwlw3ax7@default.fries.net) |
| 23:20.03 | terrapen | well, i like the polycom and after all the trouble getting it going, im wondering if i have the patience to jack with the cisco |
| 23:20.24 | terrapen | i bought 1 IP500 and 1 7960 to test and compare |
| 23:20.24 | xkev | I have a 7960 here too |
| 23:20.33 | xkev | IP500 > 7960 |
| 23:20.38 | jarrod | polycom > cisco |
| 23:20.44 | xkev | polycom > * :) |
| 23:20.44 | jarrod | ip500 is only 3 lines also |
| 23:21.05 | jarrod | i traded in cisco for ip300 @ home |
| 23:21.17 | DaLion2 | http://www.btts.com/ds_polycom_ip500-600.htm |
| 23:21.19 | mrproper_ | New guide up there for anyone wanting to run Asterisk + AVM Fritz + CAPI on Fedora Core 1: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Fedora |
| 23:21.25 | xkev | 7960 screen is clearer than the ip500, but the rez is worse.. also always having to hit 'new call' on the 7960, instead of just dialing, is annoying |
| 23:21.50 | DaLion2 | how many lines can 7960 handle ? |
| 23:21.52 | jarrod | 6 |
| 23:21.55 | modulus_ | hrm using agi for prepaid solution is slow |
| 23:21.56 | jarrod | thats the way my ip300 is |
| 23:22.01 | jarrod | have to hit dial |
| 23:22.10 | DaLion2 | and cant u just deteck off hook ? im sure its a conf issue on cisco |
| 23:22.13 | modulus_ | jarrod that sucks |
| 23:22.17 | modulus_ | jarrod, just go eat at subways |
| 23:22.28 | jarrod | *ahem* thats jared |
| 23:22.29 | jarrod | :-P |
| 23:22.37 | modulus_ | homonym |
| 23:22.40 | modulus_ | :P |
| 23:22.43 | jarrod | haha |
| 23:22.47 | jarrod | props |
| 23:23.31 | *** part/#asterisk bkw_ (~brian@65.38.28.146) |
| 23:27.41 | *** part/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 23:29.19 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
| 23:29.19 | *** topic/#asterisk is Asterisk: The Open Source PBX || DUNDi: http://www.dundi.com #dundi || chan_zap in 1.0.4 had a bug, so 1.0.5 has been released |
| 23:31.11 | postel | xkev: you're not supposed to browse slashdot on it, supposed to be inches away from your face on a well lit desk during business hours, its an executive phone for crying out loud, if you dont want to hit New Call put the numbers on speed dials |
| 23:31.35 | *** join/#asterisk file[laptop] (~file_lapt@mctnnbsah25-142166093009.nb.aliant.net) |
| 23:31.36 | xkev | it's just been annoying since I'm used to the polycom |
| 23:31.59 | xkev | really minor and stupid nitpicking :) |
| 23:33.51 | mrproper_ | xkev: you should use radvision instead of polycom =P |
| 23:34.40 | *** join/#asterisk netsurfer (~netsurfer@dreambox.myvnc.com) |
| 23:35.08 | fa | how will go tomorrow for math exam for me? |
| 23:35.25 | fa | wowo it's not tomorrow. it's today |
| 23:35.36 | *** join/#asterisk jgaviria (~juan_manu@63.245.86.116) |
| 23:35.37 | eKo1 | What kind of maths.? |
| 23:35.46 | fa | analyz |
| 23:35.47 | afrosheen | geometry |
| 23:36.00 | fa | fuck! |
| 23:36.00 | eKo1 | That's childs play. |
| 23:36.06 | jgaviria | hi, anybody using a good softphone in linux with asterisk? |
| 23:36.08 | fa | yeash.. of course.. |
| 23:36.15 | afrosheen | jgaviria: there isn't one |
| 23:36.36 | fa | jgaviria no, we use only bad softphone in linux with astrisk. |
| 23:36.36 | eKo1 | That reminds me, I'm still stuck in graph theory. |
| 23:37.01 | jgaviria | afrosheen: have you probed sflphone? |
| 23:37.03 | afrosheen | kphone is unfinished/forgotten, gnomemeeting is kinda wack |
| 23:37.16 | afrosheen | jgaviria: yeah but my probe broke off in it |
| 23:37.33 | *** join/#asterisk angler_ (~angler@suid.digium.com) [NETSPLIT VICTIM] |
| 23:37.33 | *** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com) [NETSPLIT VICTIM] |
| 23:37.44 | fa | I am goign to bad. |
| 23:37.52 | fa | bye |
| 23:37.59 | afrosheen | have fun in bad |
| 23:38.05 | eKo1 | fa is going bad. haha |
| 23:38.22 | afrosheen | he could've said I'm going, too bad |
| 23:38.22 | jgaviria | afrosheen, i probed sflphone, and this doesnt register |
| 23:38.22 | *** join/#asterisk wwalker (~wwalker@wwalker.sustaining.supporter.pdpc) |
| 23:38.22 | *** join/#asterisk Makenshi (~makenshi@2001:630:1c0:2001:280:c8ff:fee2:921f) [NETSPLIT VICTIM] |
| 23:38.49 | eKo1 | afrosheen: Could be a she... |
| 23:39.24 | afrosheen | naw, fadi is a man's name |
| 23:39.33 | afrosheen | jgaviria: I may take a look at that, looks decent enough to probe |
| 23:40.13 | redder86 | punctuation is SO important |
| 23:40.13 | *** join/#asterisk chaoscon (stormy@chaoscon.user) [NETSPLIT VICTIM] |
| 23:40.13 | blitzrage | jgaviria: use X-Lite in WINE |
| 23:40.13 | blitzrage | when trying to load Asterisk with 6 CPS after about 34 seconds I get this error: http://pastebin.ca/5248 |
| 23:40.13 | blitzrage | netsplit! |
| 23:40.14 | redder86 | well, and spelling too, in this case |
| 23:41.05 | afrosheen | radvision looks like all video stuff |
| 23:41.06 | *** join/#asterisk ST-3 (ser@dipsy.tch.org) |
| 23:41.14 | xkev | aye it seems |
| 23:41.21 | afrosheen | hence the vision in their name |
| 23:41.43 | eKo1 | blitzrage: You ran out of pipes? |
| 23:42.08 | afrosheen | blitzrage: xlite works in wine? |
| 23:42.11 | blitzrage | incase my post didn't go through |
| 23:42.11 | blitzrage | when trying to load Asterisk with 6 CPS after about 34 seconds I get this error: http://pastebin.ca/5248 |
| 23:42.11 | blitzrage | any ideas? :) |
| 23:42.26 | blitzrage | afrosheen: yep |
| 23:42.31 | blitzrage | eKo1: uh... yah :) |
| 23:42.40 | blitzrage | eKo1: not sure why though... |
| 23:42.46 | blitzrage | eKo1: what causes something like that? |
| 23:42.47 | eKo1 | blitzrage: Power cyle. |
| 23:42.56 | blitzrage | eKo1: I'm sure that won't fix it |
| 23:43.04 | eKo1 | blitzrage: Try it. |
| 23:44.29 | *** join/#asterisk Cresl1n (~matt@216.207.245.23) |
| 23:44.30 | *** join/#asterisk GreyFoxx (greg@out.of.phaze.org) |
| 23:44.43 | blitzrage | Cresl1n: hey, maybe you can help me :) |
| 23:44.49 | Cresl1n | howdy :-) |
| 23:44.55 | blitzrage | Cresl1n: sorry, hi, how are you? |
| 23:44.55 | eKo1 | The CLI is choking my screen with these: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 23:45.05 | Cresl1n | pretty good |
| 23:45.06 | Cresl1n | what's up? |
| 23:45.17 | eKo1 | Dang it. |
| 23:45.17 | ManxPower | eKo1, turn off VAD on the client |
| 23:45.32 | eKo1 | There is no VAD option on the client. |
| 23:45.58 | blitzrage | Cresl1n: I'm trying to load test an Asterisk server with SIPP. We're doing about 6 CPS and it dies after about 34 seconds... and I'm not sure why this error would occur: http://pastebin.ca/5248 |
| 23:46.16 | Cresl1n | huh, that's weird |
| 23:46.20 | eKo1 | Could it be because it is using G.729B? |
| 23:46.20 | blitzrage | we're just doing an Answer() and Playback() of a 1 minute long wave file |
| 23:46.26 | afrosheen | what would cause it to run out of pipes |
| 23:46.26 | *** join/#asterisk mflorell (mflorell@171-5.202-68.tampabay.rr.com) |
| 23:46.28 | *** join/#asterisk BoRiS (~boris@S01060040ca1e5b54.wp.shawcable.net) |
| 23:46.38 | blitzrage | afrosheen: yah, thats what I'm not sure about... |
| 23:46.46 | *** part/#asterisk mflorell (mflorell@171-5.202-68.tampabay.rr.com) |
| 23:47.10 | Cresl1n | blitzrage: did you already look at ulimit? |
| 23:47.19 | blitzrage | Cresl1n: ulimit? :) |
| 23:47.35 | Cresl1n | yep ;-) `man ulimit` |
| 23:47.36 | Cresl1n | :-) |
| 23:47.42 | blitzrage | Cresl1n: aha! checking |
| 23:48.01 | eKo1 | ulimit -a |
| 23:48.15 | *** join/#asterisk t3t (~t3t@galley.pangalacticgargleblaster.com) |
| 23:49.05 | *** join/#asterisk bjohnson (~bjohnson@ip226-181.tor.istop.com) |
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| 23:52.49 | Chuji | In order to get cdr_odbc to work, I just need to edit the config right? |
| 23:53.01 | Chuji | I don't have to tell anything else to start using it? |
| 23:58.13 | terrapen | wierdness |
| 23:58.42 | terrapen | inbound calls to my NuFone 1-800 DID show up as "Toll-Free Call" on my IP500's caller ID display |
| 23:58.51 | jarrod | rtpp_test: support for RTP proxyhas been disabled temporarily |
| 23:58.53 | terrapen | but the callerID number does get recorded in the call history |
| 23:58.56 | jarrod | grr why does ser give me that :( |
| 23:59.01 | terrapen | its just that the phone doesn't display it |
